From ac9ef6cf9196107115930e9fc66207199ef395b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Jan 2012 12:08:44 +0100 Subject: ALSA: hda - Use bint for enable_msi option The new bint module option type suits well with this one. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fb35474c120..9cbde2fc7b1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -94,7 +94,7 @@ MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization."); module_param(single_cmd, bool, 0444); MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs " "(for debugging only)."); -module_param(enable_msi, int, 0444); +module_param(enable_msi, bint, 0444); MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); #ifdef CONFIG_SND_HDA_PATCH_LOADER module_param_array(patch, charp, NULL, 0444); -- cgit v1.2.3-18-g5258 From 0d6df67583bb40fdc365210740bcce0bd27420f7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:12:45 +0000 Subject: ASoC: Make WM8978 I2C usage unconditional The driver only supports I2C. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 85d514d63a4..0b1f7ada17b 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1001,7 +1001,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8978 = { .reg_cache_default = wm8978_reg, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1043,27 +1042,22 @@ static struct i2c_driver wm8978_i2c_driver = { .remove = __devexit_p(wm8978_i2c_remove), .id_table = wm8978_i2c_id, }; -#endif static int __init wm8978_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8978_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8978 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8978_modinit); static void __exit wm8978_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8978_i2c_driver); -#endif } module_exit(wm8978_exit); -- cgit v1.2.3-18-g5258 From ad6cdec507d877189c9813655dfa30579256a2fc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:13:15 +0000 Subject: ASoC: Remove unused control type from wm8978 driver Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 0b1f7ada17b..2ba8f8c88ba 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -50,7 +50,6 @@ static const u16 wm8978_reg[WM8978_CACHEREGNUM] = { /* codec private data */ struct wm8978_priv { - enum snd_soc_control_type control_type; unsigned int f_pllout; unsigned int f_mclk; unsigned int f_256fs; -- cgit v1.2.3-18-g5258 From 803b37885d355438192516d73ba3565e744a8b90 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:15:43 +0000 Subject: ASoC: Convert wm8978 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 25 ++++++++----------------- 1 file changed, 8 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 2ba8f8c88ba..36468f85f30 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -302,7 +302,7 @@ static const struct snd_soc_dapm_widget wm8978_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("RSPK"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8978_dapm_routes[] = { /* Output mixer */ {"Right Output Mixer", "PCM Playback Switch", "Right DAC"}, {"Right Output Mixer", "Aux Playback Switch", "RAUX"}, @@ -351,18 +351,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left Input Mixer", "MicP Switch", "LMICP"}, }; -static int wm8978_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8978_dapm_widgets, - ARRAY_SIZE(wm8978_dapm_widgets)); - /* set up the WM8978 audio map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* PLL divisors */ struct wm8978_pll_div { u32 k; @@ -975,10 +963,6 @@ static int wm8978_probe(struct snd_soc_codec *codec) wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8978_snd_controls, - ARRAY_SIZE(wm8978_snd_controls)); - wm8978_add_widgets(codec); - return 0; } @@ -998,6 +982,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8978 = { .reg_cache_size = ARRAY_SIZE(wm8978_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8978_reg, + + .controls = wm8978_snd_controls, + .num_controls = ARRAY_SIZE(wm8978_snd_controls), + .dapm_widgets = wm8978_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8978_dapm_widgets), + .dapm_routes = wm8978_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8978_dapm_routes), }; static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.3-18-g5258 From 623105dc97010f851f8fd22b7ce80b77d860b5f4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:16:53 +0000 Subject: ASoC: Convert wm8978 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 36468f85f30..051f5d0d37d 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -997,7 +997,8 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, struct wm8978_priv *wm8978; int ret; - wm8978 = kzalloc(sizeof(struct wm8978_priv), GFP_KERNEL); + wm8978 = devm_kzalloc(&i2c->dev, sizeof(struct wm8978_priv), + GFP_KERNEL); if (wm8978 == NULL) return -ENOMEM; @@ -1005,15 +1006,14 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8978, &wm8978_dai, 1); - if (ret < 0) - kfree(wm8978); + return ret; } static __devexit int wm8978_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + return 0; } -- cgit v1.2.3-18-g5258 From f98692ea6dda68c7eda6d53a3bc850702c3b8fde Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:32:09 +0000 Subject: ASoC: Use standard cache sync for WM8978 Saves a bit of code and supports further refactoring. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 051f5d0d37d..0ab339c034e 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -891,16 +891,9 @@ static int wm8978_suspend(struct snd_soc_codec *codec) static int wm8978_resume(struct snd_soc_codec *codec) { struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); - int i; - u16 *cache = codec->reg_cache; /* Sync reg_cache with the hardware */ - for (i = 0; i < ARRAY_SIZE(wm8978_reg); i++) { - if (i == WM8978_RESET) - continue; - if (cache[i] != wm8978_reg[i]) - snd_soc_write(codec, i, cache[i]); - } + snd_soc_cache_sync(codec); wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.2.3-18-g5258 From ee60d0155d653888de75b642182b0300c21ce07a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:39:39 +0000 Subject: ASoC: Convert wm8978 to direct regmap API usage Helps push the register cache code down out of ASoC and improves resume times by using the more efficient regmap cache sync code. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 116 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8978.h | 2 + 2 files changed, 96 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 0ab339c034e..5ff8734d5d2 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -29,27 +30,74 @@ #include "wm8978.h" -/* wm8978 register cache. Note that register 0 is not included in the cache. */ -static const u16 wm8978_reg[WM8978_CACHEREGNUM] = { - 0x0000, 0x0000, 0x0000, 0x0000, /* 0x00...0x03 */ - 0x0050, 0x0000, 0x0140, 0x0000, /* 0x04...0x07 */ - 0x0000, 0x0000, 0x0000, 0x00ff, /* 0x08...0x0b */ - 0x00ff, 0x0000, 0x0100, 0x00ff, /* 0x0c...0x0f */ - 0x00ff, 0x0000, 0x012c, 0x002c, /* 0x10...0x13 */ - 0x002c, 0x002c, 0x002c, 0x0000, /* 0x14...0x17 */ - 0x0032, 0x0000, 0x0000, 0x0000, /* 0x18...0x1b */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 0x1c...0x1f */ - 0x0038, 0x000b, 0x0032, 0x0000, /* 0x20...0x23 */ - 0x0008, 0x000c, 0x0093, 0x00e9, /* 0x24...0x27 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 0x28...0x2b */ - 0x0033, 0x0010, 0x0010, 0x0100, /* 0x2c...0x2f */ - 0x0100, 0x0002, 0x0001, 0x0001, /* 0x30...0x33 */ - 0x0039, 0x0039, 0x0039, 0x0039, /* 0x34...0x37 */ - 0x0001, 0x0001, /* 0x38...0x3b */ +static const struct reg_default wm8978_reg_defaults[] = { + { 1, 0x0000 }, + { 2, 0x0000 }, + { 3, 0x0000 }, + { 4, 0x0050 }, + { 5, 0x0000 }, + { 6, 0x0140 }, + { 7, 0x0000 }, + { 8, 0x0000 }, + { 9, 0x0000 }, + { 10, 0x0000 }, + { 11, 0x00ff }, + { 12, 0x00ff }, + { 13, 0x0000 }, + { 14, 0x0100 }, + { 15, 0x00ff }, + { 16, 0x00ff }, + { 17, 0x0000 }, + { 18, 0x012c }, + { 19, 0x002c }, + { 20, 0x002c }, + { 21, 0x002c }, + { 22, 0x002c }, + { 23, 0x0000 }, + { 24, 0x0032 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 28, 0x0000 }, + { 29, 0x0000 }, + { 30, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0038 }, + { 33, 0x000b }, + { 34, 0x0032 }, + { 35, 0x0000 }, + { 36, 0x0008 }, + { 37, 0x000c }, + { 38, 0x0093 }, + { 39, 0x00e9 }, + { 40, 0x0000 }, + { 41, 0x0000 }, + { 42, 0x0000 }, + { 43, 0x0000 }, + { 44, 0x0033 }, + { 45, 0x0010 }, + { 46, 0x0010 }, + { 47, 0x0100 }, + { 48, 0x0100 }, + { 49, 0x0002 }, + { 50, 0x0001 }, + { 51, 0x0001 }, + { 52, 0x0039 }, + { 53, 0x0039 }, + { 54, 0x0039 }, + { 55, 0x0039 }, + { 56, 0x0001 }, + { 57, 0x0001 }, }; +static bool wm8978_volatile(struct device *dev, unsigned int reg) +{ + return reg == WM8978_RESET; +} + /* codec private data */ struct wm8978_priv { + struct regmap *regmap; unsigned int f_pllout; unsigned int f_mclk; unsigned int f_256fs; @@ -881,10 +929,14 @@ static struct snd_soc_dai_driver wm8978_dai = { static int wm8978_suspend(struct snd_soc_codec *codec) { + struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); + wm8978_set_bias_level(codec, SND_SOC_BIAS_OFF); /* Also switch PLL off */ snd_soc_write(codec, WM8978_POWER_MANAGEMENT_1, 0); + regcache_mark_dirty(wm8978->regmap); + return 0; } @@ -893,7 +945,7 @@ static int wm8978_resume(struct snd_soc_codec *codec) struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); /* Sync reg_cache with the hardware */ - snd_soc_cache_sync(codec); + regcache_sync(wm8978->regmap); wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -933,7 +985,8 @@ static int wm8978_probe(struct snd_soc_codec *codec) * default hardware setting */ wm8978->sysclk = WM8978_PLL; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C); + codec->control_data = wm8978->regmap; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -972,9 +1025,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8978 = { .suspend = wm8978_suspend, .resume = wm8978_resume, .set_bias_level = wm8978_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8978_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8978_reg, .controls = wm8978_snd_controls, .num_controls = ARRAY_SIZE(wm8978_snd_controls), @@ -984,6 +1034,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8978 = { .num_dapm_routes = ARRAY_SIZE(wm8978_dapm_routes), }; +static const struct regmap_config wm8978_regmap_config = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8978_MAX_REGISTER, + .volatile_reg = wm8978_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8978_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8978_reg_defaults), +}; + static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -995,6 +1057,13 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, if (wm8978 == NULL) return -ENOMEM; + wm8978->regmap = regmap_init_i2c(i2c, &wm8978_regmap_config); + if (IS_ERR(wm8978->regmap)) { + ret = PTR_ERR(wm8978->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8978); ret = snd_soc_register_codec(&i2c->dev, @@ -1005,7 +1074,10 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, static __devexit int wm8978_i2c_remove(struct i2c_client *client) { + struct wm8978_priv *wm8978 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + regmap_exit(wm8978->regmap); return 0; } diff --git a/sound/soc/codecs/wm8978.h b/sound/soc/codecs/wm8978.h index c75525b7f15..6ae43495b7c 100644 --- a/sound/soc/codecs/wm8978.h +++ b/sound/soc/codecs/wm8978.h @@ -67,6 +67,8 @@ #define WM8978_OUT3_MIXER_CONTROL 0x38 #define WM8978_OUT4_MIXER_CONTROL 0x39 +#define WM8978_MAX_REGISTER 0x39 + #define WM8978_CACHEREGNUM 58 /* Clock divider Id's */ -- cgit v1.2.3-18-g5258 From 008f8d4f9955b5f20be06ed99434cc2f8b025e06 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:44:03 +0000 Subject: ASoC: Push wm8978 reset down into the I2C probe Ensures that we get control of the CODEC earlier and don't try to probe the card at all if register I/O isn't working. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 5ff8734d5d2..72d5fdcd3cc 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1000,13 +1000,6 @@ static int wm8978_probe(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(update_reg); i++) snd_soc_update_bits(codec, update_reg[i], 0x100, 0x100); - /* Reset the codec */ - ret = snd_soc_write(codec, WM8978_RESET, 0); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - wm8978_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -1066,9 +1059,24 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8978); + /* Reset the codec */ + ret = regmap_write(wm8978->regmap, WM8978_RESET, 0); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + goto err; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8978, &wm8978_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + goto err; + } + return 0; + +err: + regmap_exit(wm8978->regmap); return ret; } -- cgit v1.2.3-18-g5258 From ec2c0fec11f072222b63eb160da6be01773bfe65 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Dec 2011 21:20:59 +0800 Subject: ASoC: Convert WM9090 to use regmap directly Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 242 ++++++++++++++++++++++++---------------------- 1 file changed, 128 insertions(+), 114 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 41ebe0dce77..4be5551f06a 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -33,116 +34,51 @@ #include "wm9090.h" -static const u16 wm9090_reg_defaults[] = { - 0x9093, /* R0 - Software Reset */ - 0x0006, /* R1 - Power Management (1) */ - 0x6000, /* R2 - Power Management (2) */ - 0x0000, /* R3 - Power Management (3) */ - 0x0000, /* R4 */ - 0x0000, /* R5 */ - 0x01C0, /* R6 - Clocking 1 */ - 0x0000, /* R7 */ - 0x0000, /* R8 */ - 0x0000, /* R9 */ - 0x0000, /* R10 */ - 0x0000, /* R11 */ - 0x0000, /* R12 */ - 0x0000, /* R13 */ - 0x0000, /* R14 */ - 0x0000, /* R15 */ - 0x0000, /* R16 */ - 0x0000, /* R17 */ - 0x0000, /* R18 */ - 0x0000, /* R19 */ - 0x0000, /* R20 */ - 0x0000, /* R21 */ - 0x0003, /* R22 - IN1 Line Control */ - 0x0003, /* R23 - IN2 Line Control */ - 0x0083, /* R24 - IN1 Line Input A Volume */ - 0x0083, /* R25 - IN1 Line Input B Volume */ - 0x0083, /* R26 - IN2 Line Input A Volume */ - 0x0083, /* R27 - IN2 Line Input B Volume */ - 0x002D, /* R28 - Left Output Volume */ - 0x002D, /* R29 - Right Output Volume */ - 0x0000, /* R30 */ - 0x0000, /* R31 */ - 0x0000, /* R32 */ - 0x0000, /* R33 */ - 0x0100, /* R34 - SPKMIXL Attenuation */ - 0x0000, /* R35 */ - 0x0010, /* R36 - SPKOUT Mixers */ - 0x0140, /* R37 - ClassD3 */ - 0x0039, /* R38 - Speaker Volume Left */ - 0x0000, /* R39 */ - 0x0000, /* R40 */ - 0x0000, /* R41 */ - 0x0000, /* R42 */ - 0x0000, /* R43 */ - 0x0000, /* R44 */ - 0x0000, /* R45 - Output Mixer1 */ - 0x0000, /* R46 - Output Mixer2 */ - 0x0100, /* R47 - Output Mixer3 */ - 0x0100, /* R48 - Output Mixer4 */ - 0x0000, /* R49 */ - 0x0000, /* R50 */ - 0x0000, /* R51 */ - 0x0000, /* R52 */ - 0x0000, /* R53 */ - 0x0000, /* R54 - Speaker Mixer */ - 0x0000, /* R55 */ - 0x0000, /* R56 */ - 0x000D, /* R57 - AntiPOP2 */ - 0x0000, /* R58 */ - 0x0000, /* R59 */ - 0x0000, /* R60 */ - 0x0000, /* R61 */ - 0x0000, /* R62 */ - 0x0000, /* R63 */ - 0x0000, /* R64 */ - 0x0000, /* R65 */ - 0x0000, /* R66 */ - 0x0000, /* R67 */ - 0x0000, /* R68 */ - 0x0000, /* R69 */ - 0x0000, /* R70 - Write Sequencer 0 */ - 0x0000, /* R71 - Write Sequencer 1 */ - 0x0000, /* R72 - Write Sequencer 2 */ - 0x0000, /* R73 - Write Sequencer 3 */ - 0x0000, /* R74 - Write Sequencer 4 */ - 0x0000, /* R75 - Write Sequencer 5 */ - 0x1F25, /* R76 - Charge Pump 1 */ - 0x0000, /* R77 */ - 0x0000, /* R78 */ - 0x0000, /* R79 */ - 0x0000, /* R80 */ - 0x0000, /* R81 */ - 0x0000, /* R82 */ - 0x0000, /* R83 */ - 0x0000, /* R84 - DC Servo 0 */ - 0x054A, /* R85 - DC Servo 1 */ - 0x0000, /* R86 */ - 0x0000, /* R87 - DC Servo 3 */ - 0x0000, /* R88 - DC Servo Readback 0 */ - 0x0000, /* R89 - DC Servo Readback 1 */ - 0x0000, /* R90 - DC Servo Readback 2 */ - 0x0000, /* R91 */ - 0x0000, /* R92 */ - 0x0000, /* R93 */ - 0x0000, /* R94 */ - 0x0000, /* R95 */ - 0x0100, /* R96 - Analogue HP 0 */ - 0x0000, /* R97 */ - 0x8640, /* R98 - AGC Control 0 */ - 0xC000, /* R99 - AGC Control 1 */ - 0x0200, /* R100 - AGC Control 2 */ +static const struct reg_default wm9090_reg_defaults[] = { + { 1, 0x0006 }, /* R1 - Power Management (1) */ + { 2, 0x6000 }, /* R2 - Power Management (2) */ + { 3, 0x0000 }, /* R3 - Power Management (3) */ + { 6, 0x01C0 }, /* R6 - Clocking 1 */ + { 22, 0x0003 }, /* R22 - IN1 Line Control */ + { 23, 0x0003 }, /* R23 - IN2 Line Control */ + { 24, 0x0083 }, /* R24 - IN1 Line Input A Volume */ + { 25, 0x0083 }, /* R25 - IN1 Line Input B Volume */ + { 26, 0x0083 }, /* R26 - IN2 Line Input A Volume */ + { 27, 0x0083 }, /* R27 - IN2 Line Input B Volume */ + { 28, 0x002D }, /* R28 - Left Output Volume */ + { 29, 0x002D }, /* R29 - Right Output Volume */ + { 34, 0x0100 }, /* R34 - SPKMIXL Attenuation */ + { 35, 0x0010 }, /* R36 - SPKOUT Mixers */ + { 37, 0x0140 }, /* R37 - ClassD3 */ + { 38, 0x0039 }, /* R38 - Speaker Volume Left */ + { 45, 0x0000 }, /* R45 - Output Mixer1 */ + { 46, 0x0000 }, /* R46 - Output Mixer2 */ + { 47, 0x0100 }, /* R47 - Output Mixer3 */ + { 48, 0x0100 }, /* R48 - Output Mixer4 */ + { 54, 0x0000 }, /* R54 - Speaker Mixer */ + { 57, 0x000D }, /* R57 - AntiPOP2 */ + { 70, 0x0000 }, /* R70 - Write Sequencer 0 */ + { 71, 0x0000 }, /* R71 - Write Sequencer 1 */ + { 72, 0x0000 }, /* R72 - Write Sequencer 2 */ + { 73, 0x0000 }, /* R73 - Write Sequencer 3 */ + { 74, 0x0000 }, /* R74 - Write Sequencer 4 */ + { 75, 0x0000 }, /* R75 - Write Sequencer 5 */ + { 76, 0x1F25 }, /* R76 - Charge Pump 1 */ + { 85, 0x054A }, /* R85 - DC Servo 1 */ + { 87, 0x0000 }, /* R87 - DC Servo 3 */ + { 96, 0x0100 }, /* R96 - Analogue HP 0 */ + { 98, 0x8640 }, /* R98 - AGC Control 0 */ + { 99, 0xC000 }, /* R99 - AGC Control 1 */ + { 100, 0x0200 }, /* R100 - AGC Control 2 */ }; /* This struct is used to save the context */ struct wm9090_priv { struct wm9090_platform_data pdata; + struct regmap *regmap; }; -static int wm9090_volatile(struct snd_soc_codec *codec, unsigned int reg) +static bool wm9090_volatile(struct device *dev, unsigned int reg) { switch (reg) { case WM9090_SOFTWARE_RESET: @@ -150,10 +86,60 @@ static int wm9090_volatile(struct snd_soc_codec *codec, unsigned int reg) case WM9090_DC_SERVO_READBACK_0: case WM9090_DC_SERVO_READBACK_1: case WM9090_DC_SERVO_READBACK_2: - return 1; + return true; default: - return 0; + return false; + } +} + +static bool wm9090_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM9090_SOFTWARE_RESET: + case WM9090_POWER_MANAGEMENT_1: + case WM9090_POWER_MANAGEMENT_2: + case WM9090_POWER_MANAGEMENT_3: + case WM9090_CLOCKING_1: + case WM9090_IN1_LINE_CONTROL: + case WM9090_IN2_LINE_CONTROL: + case WM9090_IN1_LINE_INPUT_A_VOLUME: + case WM9090_IN1_LINE_INPUT_B_VOLUME: + case WM9090_IN2_LINE_INPUT_A_VOLUME: + case WM9090_IN2_LINE_INPUT_B_VOLUME: + case WM9090_LEFT_OUTPUT_VOLUME: + case WM9090_RIGHT_OUTPUT_VOLUME: + case WM9090_SPKMIXL_ATTENUATION: + case WM9090_SPKOUT_MIXERS: + case WM9090_CLASSD3: + case WM9090_SPEAKER_VOLUME_LEFT: + case WM9090_OUTPUT_MIXER1: + case WM9090_OUTPUT_MIXER2: + case WM9090_OUTPUT_MIXER3: + case WM9090_OUTPUT_MIXER4: + case WM9090_SPEAKER_MIXER: + case WM9090_ANTIPOP2: + case WM9090_WRITE_SEQUENCER_0: + case WM9090_WRITE_SEQUENCER_1: + case WM9090_WRITE_SEQUENCER_2: + case WM9090_WRITE_SEQUENCER_3: + case WM9090_WRITE_SEQUENCER_4: + case WM9090_WRITE_SEQUENCER_5: + case WM9090_CHARGE_PUMP_1: + case WM9090_DC_SERVO_0: + case WM9090_DC_SERVO_1: + case WM9090_DC_SERVO_3: + case WM9090_DC_SERVO_READBACK_0: + case WM9090_DC_SERVO_READBACK_1: + case WM9090_DC_SERVO_READBACK_2: + case WM9090_ANALOGUE_HP_0: + case WM9090_AGC_CONTROL_0: + case WM9090_AGC_CONTROL_1: + case WM9090_AGC_CONTROL_2: + return true; + + default: + return false; } } @@ -492,8 +478,7 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) static int wm9090_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 *reg_cache = codec->reg_cache; - int i, ret; + struct wm9090_priv *wm9090 = snd_soc_codec_get_drvdata(codec); switch (level) { case SND_SOC_BIAS_ON: @@ -513,7 +498,7 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Restore the register cache */ - snd_soc_cache_sync(codec); + regcache_sync(wm9090->regmap); } /* We keep VMID off during standby since the combination of @@ -537,9 +522,11 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { + struct wm9090_priv *wm9090 = dev_get_drvdata(codec->dev); int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + codec->control_data = wm9090->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -548,7 +535,7 @@ static int wm9090_probe(struct snd_soc_codec *codec) ret = snd_soc_read(codec, WM9090_SOFTWARE_RESET); if (ret < 0) return ret; - if (ret != wm9090_reg_defaults[WM9090_SOFTWARE_RESET]) { + if (ret != 0x9093) { dev_err(codec->dev, "Device is not a WM9090, ID=%x\n", ret); return -EINVAL; } @@ -624,12 +611,22 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9090 = { .suspend = wm9090_suspend, .resume = wm9090_resume, .set_bias_level = wm9090_set_bias_level, - .reg_cache_size = (WM9090_MAX_REGISTER + 1), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm9090_reg_defaults, - .volatile_register = wm9090_volatile, }; +static const struct regmap_config wm9090_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM9090_MAX_REGISTER, + .volatile_reg = wm9090_volatile, + .readable_reg = wm9090_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm9090_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm9090_reg_defaults), +}; + + static int wm9090_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -642,6 +639,13 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } + wm9090->regmap = regmap_init_i2c(i2c, &wm9090_regmap); + if (IS_ERR(wm9090->regmap)) { + ret = PTR_ERR(wm9090->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + if (i2c->dev.platform_data) memcpy(&wm9090->pdata, i2c->dev.platform_data, sizeof(wm9090->pdata)); @@ -650,6 +654,15 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9090, NULL, 0); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + goto err; + } + + return 0; + +err: + regmap_exit(wm9090->regmap); return ret; } @@ -658,6 +671,7 @@ static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); snd_soc_unregister_codec(&i2c->dev); + regmap_exit(wm9090->regmap); return 0; } -- cgit v1.2.3-18-g5258 From 391d9e4e5ce50bf14400ed04d13821e7b56e84f7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Dec 2011 21:43:01 +0800 Subject: ASoC: Move WM9090 device identification and reset to I2C probe Signed-off-by: Mark Brown --- sound/soc/codecs/wm9090.c | 26 ++++++++++++++------------ 1 file changed, 14 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4be5551f06a..a2b9208a08f 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -532,18 +532,6 @@ static int wm9090_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_read(codec, WM9090_SOFTWARE_RESET); - if (ret < 0) - return ret; - if (ret != 0x9093) { - dev_err(codec->dev, "Device is not a WM9090, ID=%x\n", ret); - return -EINVAL; - } - - ret = snd_soc_write(codec, WM9090_SOFTWARE_RESET, 0); - if (ret < 0) - return ret; - /* Configure some defaults; they will be written out when we * bring the bias up. */ @@ -631,6 +619,7 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm9090_priv *wm9090; + unsigned int reg; int ret; wm9090 = devm_kzalloc(&i2c->dev, sizeof(*wm9090), GFP_KERNEL); @@ -646,6 +635,19 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, return ret; } + ret = regmap_read(wm9090->regmap, WM9090_SOFTWARE_RESET, ®); + if (ret < 0) + goto err; + if (reg != 0x9093) { + dev_err(&i2c->dev, "Device is not a WM9090, ID=%x\n", ret); + ret = -ENODEV; + goto err; + } + + ret = regmap_write(wm9090->regmap, WM9090_SOFTWARE_RESET, 0); + if (ret < 0) + goto err; + if (i2c->dev.platform_data) memcpy(&wm9090->pdata, i2c->dev.platform_data, sizeof(wm9090->pdata)); -- cgit v1.2.3-18-g5258 From d0ad0af0432f7b4fe439a6a46e7a31f8dd5d3d55 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Dec 2011 11:53:06 +0800 Subject: ASoC: Convert wm8993 to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 420 ++++++++++++++++++++++++++++++---------------- 1 file changed, 271 insertions(+), 149 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 7c7fd925db8..53213020caf 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -40,134 +41,112 @@ static const char *wm8993_supply_names[WM8993_NUM_SUPPLIES] = { "SPKVDD", }; -static u16 wm8993_reg_defaults[WM8993_REGISTER_COUNT] = { - 0x8993, /* R0 - Software Reset */ - 0x0000, /* R1 - Power Management (1) */ - 0x6000, /* R2 - Power Management (2) */ - 0x0000, /* R3 - Power Management (3) */ - 0x4050, /* R4 - Audio Interface (1) */ - 0x4000, /* R5 - Audio Interface (2) */ - 0x01C8, /* R6 - Clocking 1 */ - 0x0000, /* R7 - Clocking 2 */ - 0x0000, /* R8 - Audio Interface (3) */ - 0x0040, /* R9 - Audio Interface (4) */ - 0x0004, /* R10 - DAC CTRL */ - 0x00C0, /* R11 - Left DAC Digital Volume */ - 0x00C0, /* R12 - Right DAC Digital Volume */ - 0x0000, /* R13 - Digital Side Tone */ - 0x0300, /* R14 - ADC CTRL */ - 0x00C0, /* R15 - Left ADC Digital Volume */ - 0x00C0, /* R16 - Right ADC Digital Volume */ - 0x0000, /* R17 */ - 0x0000, /* R18 - GPIO CTRL 1 */ - 0x0010, /* R19 - GPIO1 */ - 0x0000, /* R20 - IRQ_DEBOUNCE */ - 0x0000, /* R21 */ - 0x8000, /* R22 - GPIOCTRL 2 */ - 0x0800, /* R23 - GPIO_POL */ - 0x008B, /* R24 - Left Line Input 1&2 Volume */ - 0x008B, /* R25 - Left Line Input 3&4 Volume */ - 0x008B, /* R26 - Right Line Input 1&2 Volume */ - 0x008B, /* R27 - Right Line Input 3&4 Volume */ - 0x006D, /* R28 - Left Output Volume */ - 0x006D, /* R29 - Right Output Volume */ - 0x0066, /* R30 - Line Outputs Volume */ - 0x0020, /* R31 - HPOUT2 Volume */ - 0x0079, /* R32 - Left OPGA Volume */ - 0x0079, /* R33 - Right OPGA Volume */ - 0x0003, /* R34 - SPKMIXL Attenuation */ - 0x0003, /* R35 - SPKMIXR Attenuation */ - 0x0011, /* R36 - SPKOUT Mixers */ - 0x0100, /* R37 - SPKOUT Boost */ - 0x0079, /* R38 - Speaker Volume Left */ - 0x0079, /* R39 - Speaker Volume Right */ - 0x0000, /* R40 - Input Mixer2 */ - 0x0000, /* R41 - Input Mixer3 */ - 0x0000, /* R42 - Input Mixer4 */ - 0x0000, /* R43 - Input Mixer5 */ - 0x0000, /* R44 - Input Mixer6 */ - 0x0000, /* R45 - Output Mixer1 */ - 0x0000, /* R46 - Output Mixer2 */ - 0x0000, /* R47 - Output Mixer3 */ - 0x0000, /* R48 - Output Mixer4 */ - 0x0000, /* R49 - Output Mixer5 */ - 0x0000, /* R50 - Output Mixer6 */ - 0x0000, /* R51 - HPOUT2 Mixer */ - 0x0000, /* R52 - Line Mixer1 */ - 0x0000, /* R53 - Line Mixer2 */ - 0x0000, /* R54 - Speaker Mixer */ - 0x0000, /* R55 - Additional Control */ - 0x0000, /* R56 - AntiPOP1 */ - 0x0000, /* R57 - AntiPOP2 */ - 0x0000, /* R58 - MICBIAS */ - 0x0000, /* R59 */ - 0x0000, /* R60 - FLL Control 1 */ - 0x0000, /* R61 - FLL Control 2 */ - 0x0000, /* R62 - FLL Control 3 */ - 0x2EE0, /* R63 - FLL Control 4 */ - 0x0002, /* R64 - FLL Control 5 */ - 0x2287, /* R65 - Clocking 3 */ - 0x025F, /* R66 - Clocking 4 */ - 0x0000, /* R67 - MW Slave Control */ - 0x0000, /* R68 */ - 0x0002, /* R69 - Bus Control 1 */ - 0x0000, /* R70 - Write Sequencer 0 */ - 0x0000, /* R71 - Write Sequencer 1 */ - 0x0000, /* R72 - Write Sequencer 2 */ - 0x0000, /* R73 - Write Sequencer 3 */ - 0x0000, /* R74 - Write Sequencer 4 */ - 0x0000, /* R75 - Write Sequencer 5 */ - 0x1F25, /* R76 - Charge Pump 1 */ - 0x0000, /* R77 */ - 0x0000, /* R78 */ - 0x0000, /* R79 */ - 0x0000, /* R80 */ - 0x0000, /* R81 - Class W 0 */ - 0x0000, /* R82 */ - 0x0000, /* R83 */ - 0x0000, /* R84 - DC Servo 0 */ - 0x054A, /* R85 - DC Servo 1 */ - 0x0000, /* R86 */ - 0x0000, /* R87 - DC Servo 3 */ - 0x0000, /* R88 - DC Servo Readback 0 */ - 0x0000, /* R89 - DC Servo Readback 1 */ - 0x0000, /* R90 - DC Servo Readback 2 */ - 0x0000, /* R91 */ - 0x0000, /* R92 */ - 0x0000, /* R93 */ - 0x0000, /* R94 */ - 0x0000, /* R95 */ - 0x0100, /* R96 - Analogue HP 0 */ - 0x0000, /* R97 */ - 0x0000, /* R98 - EQ1 */ - 0x000C, /* R99 - EQ2 */ - 0x000C, /* R100 - EQ3 */ - 0x000C, /* R101 - EQ4 */ - 0x000C, /* R102 - EQ5 */ - 0x000C, /* R103 - EQ6 */ - 0x0FCA, /* R104 - EQ7 */ - 0x0400, /* R105 - EQ8 */ - 0x00D8, /* R106 - EQ9 */ - 0x1EB5, /* R107 - EQ10 */ - 0xF145, /* R108 - EQ11 */ - 0x0B75, /* R109 - EQ12 */ - 0x01C5, /* R110 - EQ13 */ - 0x1C58, /* R111 - EQ14 */ - 0xF373, /* R112 - EQ15 */ - 0x0A54, /* R113 - EQ16 */ - 0x0558, /* R114 - EQ17 */ - 0x168E, /* R115 - EQ18 */ - 0xF829, /* R116 - EQ19 */ - 0x07AD, /* R117 - EQ20 */ - 0x1103, /* R118 - EQ21 */ - 0x0564, /* R119 - EQ22 */ - 0x0559, /* R120 - EQ23 */ - 0x4000, /* R121 - EQ24 */ - 0x0000, /* R122 - Digital Pulls */ - 0x0F08, /* R123 - DRC Control 1 */ - 0x0000, /* R124 - DRC Control 2 */ - 0x0080, /* R125 - DRC Control 3 */ - 0x0000, /* R126 - DRC Control 4 */ +static struct reg_default wm8993_reg_defaults[] = { + { 1, 0x0000 }, /* R1 - Power Management (1) */ + { 2, 0x6000 }, /* R2 - Power Management (2) */ + { 3, 0x0000 }, /* R3 - Power Management (3) */ + { 4, 0x4050 }, /* R4 - Audio Interface (1) */ + { 5, 0x4000 }, /* R5 - Audio Interface (2) */ + { 6, 0x01C8 }, /* R6 - Clocking 1 */ + { 7, 0x0000 }, /* R7 - Clocking 2 */ + { 8, 0x0000 }, /* R8 - Audio Interface (3) */ + { 9, 0x0040 }, /* R9 - Audio Interface (4) */ + { 10, 0x0004 }, /* R10 - DAC CTRL */ + { 11, 0x00C0 }, /* R11 - Left DAC Digital Volume */ + { 12, 0x00C0 }, /* R12 - Right DAC Digital Volume */ + { 13, 0x0000 }, /* R13 - Digital Side Tone */ + { 14, 0x0300 }, /* R14 - ADC CTRL */ + { 15, 0x00C0 }, /* R15 - Left ADC Digital Volume */ + { 16, 0x00C0 }, /* R16 - Right ADC Digital Volume */ + { 18, 0x0000 }, /* R18 - GPIO CTRL 1 */ + { 19, 0x0010 }, /* R19 - GPIO1 */ + { 20, 0x0000 }, /* R20 - IRQ_DEBOUNCE */ + { 21, 0x8000 }, /* R22 - GPIOCTRL 2 */ + { 22, 0x0800 }, /* R23 - GPIO_POL */ + { 24, 0x008B }, /* R24 - Left Line Input 1&2 Volume */ + { 25, 0x008B }, /* R25 - Left Line Input 3&4 Volume */ + { 26, 0x008B }, /* R26 - Right Line Input 1&2 Volume */ + { 27, 0x008B }, /* R27 - Right Line Input 3&4 Volume */ + { 28, 0x006D }, /* R28 - Left Output Volume */ + { 29, 0x006D }, /* R29 - Right Output Volume */ + { 30, 0x0066 }, /* R30 - Line Outputs Volume */ + { 31, 0x0020 }, /* R31 - HPOUT2 Volume */ + { 32, 0x0079 }, /* R32 - Left OPGA Volume */ + { 33, 0x0079 }, /* R33 - Right OPGA Volume */ + { 34, 0x0003 }, /* R34 - SPKMIXL Attenuation */ + { 35, 0x0003 }, /* R35 - SPKMIXR Attenuation */ + { 36, 0x0011 }, /* R36 - SPKOUT Mixers */ + { 37, 0x0100 }, /* R37 - SPKOUT Boost */ + { 38, 0x0079 }, /* R38 - Speaker Volume Left */ + { 39, 0x0079 }, /* R39 - Speaker Volume Right */ + { 40, 0x0000 }, /* R40 - Input Mixer2 */ + { 41, 0x0000 }, /* R41 - Input Mixer3 */ + { 42, 0x0000 }, /* R42 - Input Mixer4 */ + { 43, 0x0000 }, /* R43 - Input Mixer5 */ + { 44, 0x0000 }, /* R44 - Input Mixer6 */ + { 45, 0x0000 }, /* R45 - Output Mixer1 */ + { 46, 0x0000 }, /* R46 - Output Mixer2 */ + { 47, 0x0000 }, /* R47 - Output Mixer3 */ + { 48, 0x0000 }, /* R48 - Output Mixer4 */ + { 49, 0x0000 }, /* R49 - Output Mixer5 */ + { 50, 0x0000 }, /* R50 - Output Mixer6 */ + { 51, 0x0000 }, /* R51 - HPOUT2 Mixer */ + { 52, 0x0000 }, /* R52 - Line Mixer1 */ + { 53, 0x0000 }, /* R53 - Line Mixer2 */ + { 54, 0x0000 }, /* R54 - Speaker Mixer */ + { 55, 0x0000 }, /* R55 - Additional Control */ + { 56, 0x0000 }, /* R56 - AntiPOP1 */ + { 57, 0x0000 }, /* R57 - AntiPOP2 */ + { 58, 0x0000 }, /* R58 - MICBIAS */ + { 60, 0x0000 }, /* R60 - FLL Control 1 */ + { 61, 0x0000 }, /* R61 - FLL Control 2 */ + { 62, 0x0000 }, /* R62 - FLL Control 3 */ + { 63, 0x2EE0 }, /* R63 - FLL Control 4 */ + { 64, 0x0002 }, /* R64 - FLL Control 5 */ + { 65, 0x2287 }, /* R65 - Clocking 3 */ + { 66, 0x025F }, /* R66 - Clocking 4 */ + { 67, 0x0000 }, /* R67 - MW Slave Control */ + { 69, 0x0002 }, /* R69 - Bus Control 1 */ + { 70, 0x0000 }, /* R70 - Write Sequencer 0 */ + { 71, 0x0000 }, /* R71 - Write Sequencer 1 */ + { 72, 0x0000 }, /* R72 - Write Sequencer 2 */ + { 73, 0x0000 }, /* R73 - Write Sequencer 3 */ + { 74, 0x0000 }, /* R74 - Write Sequencer 4 */ + { 75, 0x0000 }, /* R75 - Write Sequencer 5 */ + { 76, 0x1F25 }, /* R76 - Charge Pump 1 */ + { 81, 0x0000 }, /* R81 - Class W 0 */ + { 85, 0x054A }, /* R85 - DC Servo 1 */ + { 87, 0x0000 }, /* R87 - DC Servo 3 */ + { 96, 0x0100 }, /* R96 - Analogue HP 0 */ + { 98, 0x0000 }, /* R98 - EQ1 */ + { 99, 0x000C }, /* R99 - EQ2 */ + { 100, 0x000C }, /* R100 - EQ3 */ + { 101, 0x000C }, /* R101 - EQ4 */ + { 102, 0x000C }, /* R102 - EQ5 */ + { 103, 0x000C }, /* R103 - EQ6 */ + { 104, 0x0FCA }, /* R104 - EQ7 */ + { 105, 0x0400 }, /* R105 - EQ8 */ + { 106, 0x00D8 }, /* R106 - EQ9 */ + { 107, 0x1EB5 }, /* R107 - EQ10 */ + { 108, 0xF145 }, /* R108 - EQ11 */ + { 109, 0x0B75 }, /* R109 - EQ12 */ + { 110, 0x01C5 }, /* R110 - EQ13 */ + { 111, 0x1C58 }, /* R111 - EQ14 */ + { 112, 0xF373 }, /* R112 - EQ15 */ + { 113, 0x0A54 }, /* R113 - EQ16 */ + { 114, 0x0558 }, /* R114 - EQ17 */ + { 115, 0x168E }, /* R115 - EQ18 */ + { 116, 0xF829 }, /* R116 - EQ19 */ + { 117, 0x07AD }, /* R117 - EQ20 */ + { 118, 0x1103 }, /* R118 - EQ21 */ + { 119, 0x0564 }, /* R119 - EQ22 */ + { 120, 0x0559 }, /* R120 - EQ23 */ + { 121, 0x4000 }, /* R121 - EQ24 */ + { 122, 0x0000 }, /* R122 - Digital Pulls */ + { 123, 0x0F08 }, /* R123 - DRC Control 1 */ + { 124, 0x0000 }, /* R124 - DRC Control 2 */ + { 125, 0x0080 }, /* R125 - DRC Control 3 */ + { 126, 0x0000 }, /* R126 - DRC Control 4 */ }; static struct { @@ -225,9 +204,9 @@ static struct { struct wm8993_priv { struct wm_hubs_data hubs_data; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8993_NUM_SUPPLIES]; struct wm8993_platform_data pdata; - enum snd_soc_control_type control_type; int master; int sysclk_source; int tdm_slots; @@ -242,7 +221,7 @@ struct wm8993_priv { int fll_src; }; -static int wm8993_volatile(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8993_volatile(struct device *dev, unsigned int reg) { switch (reg) { case WM8993_SOFTWARE_RESET: @@ -250,9 +229,128 @@ static int wm8993_volatile(struct snd_soc_codec *codec, unsigned int reg) case WM8993_DC_SERVO_READBACK_0: case WM8993_DC_SERVO_READBACK_1: case WM8993_DC_SERVO_READBACK_2: - return 1; + return true; default: - return 0; + return false; + } +} + +static bool wm8993_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8993_SOFTWARE_RESET: + case WM8993_POWER_MANAGEMENT_1: + case WM8993_POWER_MANAGEMENT_2: + case WM8993_POWER_MANAGEMENT_3: + case WM8993_AUDIO_INTERFACE_1: + case WM8993_AUDIO_INTERFACE_2: + case WM8993_CLOCKING_1: + case WM8993_CLOCKING_2: + case WM8993_AUDIO_INTERFACE_3: + case WM8993_AUDIO_INTERFACE_4: + case WM8993_DAC_CTRL: + case WM8993_LEFT_DAC_DIGITAL_VOLUME: + case WM8993_RIGHT_DAC_DIGITAL_VOLUME: + case WM8993_DIGITAL_SIDE_TONE: + case WM8993_ADC_CTRL: + case WM8993_LEFT_ADC_DIGITAL_VOLUME: + case WM8993_RIGHT_ADC_DIGITAL_VOLUME: + case WM8993_GPIO_CTRL_1: + case WM8993_GPIO1: + case WM8993_IRQ_DEBOUNCE: + case WM8993_GPIOCTRL_2: + case WM8993_GPIO_POL: + case WM8993_LEFT_LINE_INPUT_1_2_VOLUME: + case WM8993_LEFT_LINE_INPUT_3_4_VOLUME: + case WM8993_RIGHT_LINE_INPUT_1_2_VOLUME: + case WM8993_RIGHT_LINE_INPUT_3_4_VOLUME: + case WM8993_LEFT_OUTPUT_VOLUME: + case WM8993_RIGHT_OUTPUT_VOLUME: + case WM8993_LINE_OUTPUTS_VOLUME: + case WM8993_HPOUT2_VOLUME: + case WM8993_LEFT_OPGA_VOLUME: + case WM8993_RIGHT_OPGA_VOLUME: + case WM8993_SPKMIXL_ATTENUATION: + case WM8993_SPKMIXR_ATTENUATION: + case WM8993_SPKOUT_MIXERS: + case WM8993_SPKOUT_BOOST: + case WM8993_SPEAKER_VOLUME_LEFT: + case WM8993_SPEAKER_VOLUME_RIGHT: + case WM8993_INPUT_MIXER2: + case WM8993_INPUT_MIXER3: + case WM8993_INPUT_MIXER4: + case WM8993_INPUT_MIXER5: + case WM8993_INPUT_MIXER6: + case WM8993_OUTPUT_MIXER1: + case WM8993_OUTPUT_MIXER2: + case WM8993_OUTPUT_MIXER3: + case WM8993_OUTPUT_MIXER4: + case WM8993_OUTPUT_MIXER5: + case WM8993_OUTPUT_MIXER6: + case WM8993_HPOUT2_MIXER: + case WM8993_LINE_MIXER1: + case WM8993_LINE_MIXER2: + case WM8993_SPEAKER_MIXER: + case WM8993_ADDITIONAL_CONTROL: + case WM8993_ANTIPOP1: + case WM8993_ANTIPOP2: + case WM8993_MICBIAS: + case WM8993_FLL_CONTROL_1: + case WM8993_FLL_CONTROL_2: + case WM8993_FLL_CONTROL_3: + case WM8993_FLL_CONTROL_4: + case WM8993_FLL_CONTROL_5: + case WM8993_CLOCKING_3: + case WM8993_CLOCKING_4: + case WM8993_MW_SLAVE_CONTROL: + case WM8993_BUS_CONTROL_1: + case WM8993_WRITE_SEQUENCER_0: + case WM8993_WRITE_SEQUENCER_1: + case WM8993_WRITE_SEQUENCER_2: + case WM8993_WRITE_SEQUENCER_3: + case WM8993_WRITE_SEQUENCER_4: + case WM8993_WRITE_SEQUENCER_5: + case WM8993_CHARGE_PUMP_1: + case WM8993_CLASS_W_0: + case WM8993_DC_SERVO_0: + case WM8993_DC_SERVO_1: + case WM8993_DC_SERVO_3: + case WM8993_DC_SERVO_READBACK_0: + case WM8993_DC_SERVO_READBACK_1: + case WM8993_DC_SERVO_READBACK_2: + case WM8993_ANALOGUE_HP_0: + case WM8993_EQ1: + case WM8993_EQ2: + case WM8993_EQ3: + case WM8993_EQ4: + case WM8993_EQ5: + case WM8993_EQ6: + case WM8993_EQ7: + case WM8993_EQ8: + case WM8993_EQ9: + case WM8993_EQ10: + case WM8993_EQ11: + case WM8993_EQ12: + case WM8993_EQ13: + case WM8993_EQ14: + case WM8993_EQ15: + case WM8993_EQ16: + case WM8993_EQ17: + case WM8993_EQ18: + case WM8993_EQ19: + case WM8993_EQ20: + case WM8993_EQ21: + case WM8993_EQ22: + case WM8993_EQ23: + case WM8993_EQ24: + case WM8993_DIGITAL_PULLS: + case WM8993_DRC_CONTROL_1: + case WM8993_DRC_CONTROL_2: + case WM8993_DRC_CONTROL_3: + case WM8993_DRC_CONTROL_4: + return true; + default: + return false; } } @@ -963,7 +1061,8 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, if (ret != 0) return ret; - snd_soc_cache_sync(codec); + regcache_cache_only(wm8993->regmap, false); + regcache_sync(wm8993->regmap); /* Tune DC servo configuration */ snd_soc_write(codec, 0x44, 3); @@ -1024,14 +1123,8 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, WM8993_VMID_RAMP_MASK | WM8993_BIAS_SRC, 0); -#ifdef CONFIG_REGULATOR - /* Post 2.6.34 we will be able to get a callback when - * the regulators are disabled which we can use but - * for now just assume that the power will be cut if - * the regulator API is in use. - */ - codec->cache_sync = 1; -#endif + regcache_cache_only(wm8993->regmap, true); + regcache_mark_dirty(wm8993->regmap); regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); @@ -1425,7 +1518,8 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->hubs_data.dcs_codes_r = -2; wm8993->hubs_data.series_startup = 1; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + codec->control_data = wm8993->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1449,7 +1543,7 @@ static int wm8993_probe(struct snd_soc_codec *codec) } val = snd_soc_read(codec, WM8993_SOFTWARE_RESET); - if (val != wm8993_reg_defaults[WM8993_SOFTWARE_RESET]) { + if (val != 0x8993) { dev_err(codec->dev, "Invalid ID register value %x\n", val); ret = -EINVAL; goto err_enable; @@ -1459,7 +1553,7 @@ static int wm8993_probe(struct snd_soc_codec *codec) if (ret != 0) goto err_enable; - codec->cache_only = 1; + regcache_cache_only(wm8993->regmap, true); /* By default we're using the output mixers */ wm8993->class_w_users = 2; @@ -1578,16 +1672,25 @@ static int wm8993_resume(struct snd_soc_codec *codec) #define wm8993_resume NULL #endif +static const struct regmap_config wm8993_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM8993_MAX_REGISTER, + .volatile_reg = wm8993_volatile, + .readable_reg = wm8993_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8993_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8993_reg_defaults), +}; + static struct snd_soc_codec_driver soc_codec_dev_wm8993 = { .probe = wm8993_probe, .remove = wm8993_remove, .suspend = wm8993_suspend, .resume = wm8993_resume, .set_bias_level = wm8993_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8993_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8993_reg_defaults, - .volatile_register = wm8993_volatile, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) @@ -1602,17 +1705,36 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, if (wm8993 == NULL) return -ENOMEM; + wm8993->regmap = regmap_init_i2c(i2c, &wm8993_regmap); + if (IS_ERR(wm8993->regmap)) { + ret = PTR_ERR(wm8993->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, wm8993); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8993, &wm8993_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + goto err; + } + + return ret; + +err: + regmap_exit(wm8993->regmap); return ret; } static __devexit int wm8993_i2c_remove(struct i2c_client *client) { + struct wm8993_priv *wm8993 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(wm8993->regmap); + return 0; } -- cgit v1.2.3-18-g5258 From bfea3abb804f5ab97c5da3da7c6386664531d698 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Dec 2011 12:31:14 +0800 Subject: ASoC: Move WM8993 resource acquisition and device reset to bus probe Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 89 ++++++++++++++++++++++++++--------------------- 1 file changed, 49 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 53213020caf..e7ae9fda3f5 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1511,7 +1511,7 @@ static int wm8993_probe(struct snd_soc_codec *codec) { struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret, i, val; + int ret; wm8993->hubs_data.hp_startup_mode = 1; wm8993->hubs_data.dcs_codes_l = -2; @@ -1525,36 +1525,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8993->supplies); i++) - wm8993->supplies[i].supply = wm8993_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8993->supplies), - wm8993->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies), - wm8993->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - - val = snd_soc_read(codec, WM8993_SOFTWARE_RESET); - if (val != 0x8993) { - dev_err(codec->dev, "Invalid ID register value %x\n", val); - ret = -EINVAL; - goto err_enable; - } - - ret = snd_soc_write(codec, WM8993_SOFTWARE_RESET, 0xffff); - if (ret != 0) - goto err_enable; - - regcache_cache_only(wm8993->regmap, true); - /* By default we're using the output mixers */ wm8993->class_w_users = 2; @@ -1583,7 +1553,7 @@ static int wm8993_probe(struct snd_soc_codec *codec) ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret != 0) - goto err_enable; + return ret; snd_soc_add_controls(codec, wm8993_snd_controls, ARRAY_SIZE(wm8993_snd_controls)); @@ -1605,11 +1575,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) return 0; -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); - return ret; } static int wm8993_remove(struct snd_soc_codec *codec) @@ -1698,7 +1663,8 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8993_priv *wm8993; - int ret; + unsigned int reg; + int ret, i; wm8993 = devm_kzalloc(&i2c->dev, sizeof(struct wm8993_priv), GFP_KERNEL); @@ -1714,15 +1680,56 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8993); + for (i = 0; i < ARRAY_SIZE(wm8993->supplies); i++) + wm8993->supplies[i].supply = wm8993_supply_names[i]; + + ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8993->supplies), + wm8993->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + goto err_get; + } + + ret = regmap_read(wm8993->regmap, WM8993_SOFTWARE_RESET, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); + goto err_enable; + } + + if (reg != 0x8993) { + dev_err(&i2c->dev, "Invalid ID register value %x\n", reg); + ret = -EINVAL; + goto err_enable; + } + + ret = regmap_write(wm8993->regmap, WM8993_SOFTWARE_RESET, 0xffff); + if (ret != 0) + goto err_enable; + + regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); + + regcache_cache_only(wm8993->regmap, true); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8993, &wm8993_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - goto err; + goto err_enable; } - return ret; + return 0; +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); +err_get: + regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); err: regmap_exit(wm8993->regmap); return ret; @@ -1734,6 +1741,8 @@ static __devexit int wm8993_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(wm8993->regmap); + regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); return 0; } -- cgit v1.2.3-18-g5258 From d3398ff05907167f463e119421b053ce043741d1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Nov 2011 16:32:03 +0000 Subject: ASoC: Convert WM8753 to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 186 ++++++++++++++++++++++++++++++++++------------ 1 file changed, 139 insertions(+), 47 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index b114c19f530..21ed75de41f 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include #include @@ -65,28 +66,86 @@ static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec, * We can't read the WM8753 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8753_reg[] = { - 0x0000, 0x0008, 0x0000, 0x000a, - 0x000a, 0x0033, 0x0000, 0x0007, - 0x00ff, 0x00ff, 0x000f, 0x000f, - 0x007b, 0x0000, 0x0032, 0x0000, - 0x00c3, 0x00c3, 0x00c0, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0000, 0x0000, 0x0000, - 0x0055, 0x0005, 0x0050, 0x0055, - 0x0050, 0x0055, 0x0050, 0x0055, - 0x0079, 0x0079, 0x0079, 0x0079, - 0x0079, 0x0000, 0x0000, 0x0000, - 0x0000, 0x0097, 0x0097, 0x0000, - 0x0004, 0x0000, 0x0083, 0x0024, - 0x01ba, 0x0000, 0x0083, 0x0024, - 0x01ba, 0x0000, 0x0000, 0x0000 +static const struct reg_default wm8753_reg_defaults[] = { + { 0x00, 0x0000 }, + { 0x01, 0x0008 }, + { 0x02, 0x0000 }, + { 0x03, 0x000a }, + { 0x04, 0x000a }, + { 0x05, 0x0033 }, + { 0x06, 0x0000 }, + { 0x07, 0x0007 }, + { 0x08, 0x00ff }, + { 0x09, 0x00ff }, + { 0x0a, 0x000f }, + { 0x0b, 0x000f }, + { 0x0c, 0x007b }, + { 0x0d, 0x0000 }, + { 0x0e, 0x0032 }, + { 0x0f, 0x0000 }, + { 0x10, 0x00c3 }, + { 0x11, 0x00c3 }, + { 0x12, 0x00c0 }, + { 0x13, 0x0000 }, + { 0x14, 0x0000 }, + { 0x15, 0x0000 }, + { 0x16, 0x0000 }, + { 0x17, 0x0000 }, + { 0x18, 0x0000 }, + { 0x19, 0x0000 }, + { 0x1a, 0x0000 }, + { 0x1b, 0x0000 }, + { 0x1c, 0x0000 }, + { 0x1d, 0x0000 }, + { 0x1e, 0x0000 }, + { 0x1f, 0x0000 }, + { 0x20, 0x0055 }, + { 0x21, 0x0005 }, + { 0x22, 0x0050 }, + { 0x23, 0x0055 }, + { 0x24, 0x0050 }, + { 0x25, 0x0055 }, + { 0x26, 0x0050 }, + { 0x27, 0x0055 }, + { 0x28, 0x0079 }, + { 0x29, 0x0079 }, + { 0x2a, 0x0079 }, + { 0x2b, 0x0079 }, + { 0x2c, 0x0079 }, + { 0x2d, 0x0000 }, + { 0x2e, 0x0000 }, + { 0x2f, 0x0000 }, + { 0x30, 0x0000 }, + { 0x31, 0x0097 }, + { 0x32, 0x0097 }, + { 0x33, 0x0000 }, + { 0x34, 0x0004 }, + { 0x35, 0x0000 }, + { 0x36, 0x0083 }, + { 0x37, 0x0024 }, + { 0x38, 0x01ba }, + { 0x39, 0x0000 }, + { 0x3a, 0x0083 }, + { 0x3b, 0x0024 }, + { 0x3c, 0x01ba }, + { 0x3d, 0x0000 }, + { 0x3e, 0x0000 }, + { 0x3f, 0x0000 }, }; +static bool wm8753_volatile(struct device *dev, unsigned int reg) +{ + return reg == WM8753_RESET; +} + +static bool wm8753_writeable(struct device *dev, unsigned int reg) +{ + return reg <= WM8753_ADCTL2; +} + /* codec private data */ struct wm8753_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; unsigned int sysclk; unsigned int pcmclk; @@ -1383,25 +1442,15 @@ static void wm8753_work(struct work_struct *work) static int wm8753_suspend(struct snd_soc_codec *codec) { wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); + codec->cache_sync = 1; return 0; } static int wm8753_resume(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; - int i; - - /* Sync reg_cache with the hardware */ - for (i = 1; i < ARRAY_SIZE(wm8753_reg); i++) { - if (i == WM8753_RESET) - continue; - - /* No point in writing hardware default values back */ - if (reg_cache[i] == wm8753_reg[i]) - continue; + struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec); - snd_soc_write(codec, i, reg_cache[i]); - } + regcache_sync(wm8753->regmap); wm8753_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1423,7 +1472,8 @@ static int wm8753_probe(struct snd_soc_codec *codec) INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8753->control_type); + codec->control_data = wm8753->regmap; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -1473,9 +1523,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8753 = { .suspend = wm8753_suspend, .resume = wm8753_resume, .set_bias_level = wm8753_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8753_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8753_reg, .controls = wm8753_snd_controls, .num_controls = ARRAY_SIZE(wm8753_snd_controls), @@ -1491,6 +1538,19 @@ static const struct of_device_id wm8753_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8753_of_match); +static const struct regmap_config wm8753_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8753_ADCTL2, + .writeable_reg = wm8753_writeable, + .volatile_reg = wm8753_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8753_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8753_reg_defaults), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8753_spi_probe(struct spi_device *spi) { @@ -1501,20 +1561,36 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) if (wm8753 == NULL) return -ENOMEM; - wm8753->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8753); - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_wm8753, wm8753_dai, ARRAY_SIZE(wm8753_dai)); - if (ret < 0) - kfree(wm8753); + wm8753->regmap = regmap_init_spi(spi, &wm8753_regmap); + if (IS_ERR(wm8753->regmap)) { + ret = PTR_ERR(wm8753->regmap); + dev_err(&spi->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8753, + wm8753_dai, ARRAY_SIZE(wm8753_dai)); + if (ret != 0) { + dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret); + goto err_regmap; + } +err_regmap: + regmap_exit(wm8753->regmap); +err: + kfree(wm8753); return ret; } static int __devexit wm8753_spi_remove(struct spi_device *spi) { + struct wm8753_priv *wm8753 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + regmap_exit(wm8753->regmap); + kfree(wm8753); return 0; } @@ -1541,19 +1617,35 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8753); - wm8753->control_type = SND_SOC_I2C; - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_wm8753, wm8753_dai, ARRAY_SIZE(wm8753_dai)); - if (ret < 0) - kfree(wm8753); + wm8753->regmap = regmap_init_i2c(i2c, &wm8753_regmap); + if (IS_ERR(wm8753->regmap)) { + ret = PTR_ERR(wm8753->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8753, + wm8753_dai, ARRAY_SIZE(wm8753_dai)); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + goto err_regmap; + } +err_regmap: + regmap_exit(wm8753->regmap); +err: + kfree(wm8753); return ret; } static __devexit int wm8753_i2c_remove(struct i2c_client *client) { + struct wm8753_priv *wm8753 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(wm8753->regmap); + kfree(wm8753); return 0; } -- cgit v1.2.3-18-g5258 From 542cc361de509798b311999ada07ebf2bd5673cf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 10:59:39 +0000 Subject: ASoC: Make WM8971 I2C usage unconditional The driver only supports I2C so no need to worry about SPI only systems. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 4af893601f0..a1db1509dcf 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -688,7 +688,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8971 = { .reg_cache_default = wm8971_reg, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8971_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -731,27 +730,22 @@ static struct i2c_driver wm8971_i2c_driver = { .remove = __devexit_p(wm8971_i2c_remove), .id_table = wm8971_i2c_id, }; -#endif static int __init wm8971_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8971_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8971 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8971_modinit); static void __exit wm8971_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8971_i2c_driver); -#endif } module_exit(wm8971_exit); -- cgit v1.2.3-18-g5258 From c4850644ceaeb4fb6be3bd3305afc995a0f46a6c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:03:08 +0000 Subject: ASoC: Convert wm8971 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index a1db1509dcf..3a5d67c59a2 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -252,7 +252,7 @@ static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8971_dapm_routes[] = { /* left mixer */ {"Left Mixer", "Playback Switch", "Left DAC"}, {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, @@ -329,17 +329,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Right ADC", NULL, "Right ADC Mux"}, }; -static int wm8971_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8971_dapm_widgets, - ARRAY_SIZE(wm8971_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - struct _coeff_div { u32 mclk; u32 rate; @@ -659,10 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8971_LINVOL, 0x0100, 0x0100); snd_soc_update_bits(codec, WM8971_RINVOL, 0x0100, 0x0100); - snd_soc_add_controls(codec, wm8971_snd_controls, - ARRAY_SIZE(wm8971_snd_controls)); - wm8971_add_widgets(codec); - return ret; } @@ -686,6 +671,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8971 = { .reg_cache_size = ARRAY_SIZE(wm8971_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8971_reg, + + .controls = wm8971_snd_controls, + .num_controls = ARRAY_SIZE(wm8971_snd_controls), + .dapm_widgets = wm8971_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8971_dapm_widgets), + .dapm_routes = wm8971_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8971_dapm_routes), }; static __devinit int wm8971_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.3-18-g5258 From 028b0a0a92daa36bd9e0f6d01954fe67c969a095 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:04:04 +0000 Subject: ASoC: Convert wm8971 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8971.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 3a5d67c59a2..28fe59e3ce0 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -686,7 +686,8 @@ static __devinit int wm8971_i2c_probe(struct i2c_client *i2c, struct wm8971_priv *wm8971; int ret; - wm8971 = kzalloc(sizeof(struct wm8971_priv), GFP_KERNEL); + wm8971 = devm_kzalloc(&i2c->dev, sizeof(struct wm8971_priv), + GFP_KERNEL); if (wm8971 == NULL) return -ENOMEM; @@ -695,15 +696,13 @@ static __devinit int wm8971_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8971, &wm8971_dai, 1); - if (ret < 0) - kfree(wm8971); + return ret; } static __devexit int wm8971_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3-18-g5258 From 7a389651bd88f884b46ddcada78730d294ccf1eb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:08:21 +0000 Subject: ASoC: Make wm8974 I2C usage unconditional The driver only supports I2C at present. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 4a6a7b5a61b..80c264e3ef8 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -636,7 +636,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8974 = { .reg_cache_default = wm8974_reg, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8974_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -678,27 +677,22 @@ static struct i2c_driver wm8974_i2c_driver = { .remove = __devexit_p(wm8974_i2c_remove), .id_table = wm8974_i2c_id, }; -#endif static int __init wm8974_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8974_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8974 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8974_modinit); static void __exit wm8974_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8974_i2c_driver); -#endif } module_exit(wm8974_exit); -- cgit v1.2.3-18-g5258 From a2bd691c64383ab290732d771a7404e26c0b9d53 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:10:27 +0000 Subject: ASoC: Convert wm8974 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 23 ++++++++--------------- 1 file changed, 8 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 80c264e3ef8..1e7a87a5616 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -235,7 +235,7 @@ SND_SOC_DAPM_OUTPUT("SPKOUTP"), SND_SOC_DAPM_OUTPUT("SPKOUTN"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8974_dapm_routes[] = { /* Mono output mixer */ {"Mono Mixer", "PCM Playback Switch", "DAC"}, {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, @@ -269,17 +269,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Aux Input", NULL, "AUX"}, }; -static int wm8974_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8974_dapm_widgets, - ARRAY_SIZE(wm8974_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - struct pll_ { unsigned int pre_div:1; unsigned int n:4; @@ -611,9 +600,6 @@ static int wm8974_probe(struct snd_soc_codec *codec) } wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8974_snd_controls, - ARRAY_SIZE(wm8974_snd_controls)); - wm8974_add_widgets(codec); return ret; } @@ -634,6 +620,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8974 = { .reg_cache_size = ARRAY_SIZE(wm8974_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8974_reg, + + .controls = wm8974_snd_controls, + .num_controls = ARRAY_SIZE(wm8974_snd_controls), + .dapm_widgets = wm8974_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8974_dapm_widgets), + .dapm_routes = wm8974_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8974_dapm_routes), }; static __devinit int wm8974_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.3-18-g5258 From c2562a8e3b5f871ad0b73caf98bb7541e8724efc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 11:11:25 +0000 Subject: ASoC: Remove wm8974 private data It's only ever referenced when being allocated and freed. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 1e7a87a5616..d93c03f820c 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -48,10 +48,6 @@ static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { #define WM8974_POWER1_BIASEN 0x08 #define WM8974_POWER1_BUFIOEN 0x04 -struct wm8974_priv { - enum snd_soc_control_type control_type; -}; - #define wm8974_reset(c) snd_soc_write(c, WM8974_RESET, 0) static const char *wm8974_companding[] = {"Off", "NC", "u-law", "A-law" }; @@ -632,26 +628,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8974 = { static __devinit int wm8974_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct wm8974_priv *wm8974; int ret; - wm8974 = kzalloc(sizeof(struct wm8974_priv), GFP_KERNEL); - if (wm8974 == NULL) - return -ENOMEM; - - i2c_set_clientdata(i2c, wm8974); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8974, &wm8974_dai, 1); - if (ret < 0) - kfree(wm8974); + return ret; } static __devexit int wm8974_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + return 0; } -- cgit v1.2.3-18-g5258 From e055cd67fda76dd8efdcdd4038f5adfe0f8e85a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 19:13:37 +0000 Subject: ASoC: Use standard cache sync for wm8804 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 20 +------------------- 1 file changed, 1 insertion(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index d54a3ca5e19..4d79cefe85d 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -482,24 +482,6 @@ static int wm8804_set_clkdiv(struct snd_soc_dai *dai, return 0; } -static void wm8804_sync_cache(struct snd_soc_codec *codec) -{ - short i; - u8 *cache; - - if (!codec->cache_sync) - return; - - codec->cache_only = 0; - cache = codec->reg_cache; - for (i = 0; i < codec->driver->reg_cache_size; i++) { - if (i == WM8804_RST_DEVID1 || cache[i] == wm8804_reg_defs[i]) - continue; - snd_soc_write(codec, i, cache[i]); - } - codec->cache_sync = 0; -} - static int wm8804_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -524,7 +506,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, ret); return ret; } - wm8804_sync_cache(codec); + snd_soc_cache_sync(codec); } /* power down the OSC and the PLL */ snd_soc_update_bits(codec, WM8804_PWRDN, 0x9, 0x9); -- cgit v1.2.3-18-g5258 From f649f1a8aadeb9ba146359daf51cc5ed137b394e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 19:19:18 +0000 Subject: ASoC: Convert wm8804 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 4d79cefe85d..a9f1eb334f7 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -713,7 +713,7 @@ static int __devinit wm8804_spi_probe(struct spi_device *spi) struct wm8804_priv *wm8804; int ret; - wm8804 = kzalloc(sizeof *wm8804, GFP_KERNEL); + wm8804 = devm_kzalloc(&spi->dev, sizeof *wm8804, GFP_KERNEL); if (!wm8804) return -ENOMEM; @@ -722,15 +722,13 @@ static int __devinit wm8804_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8804, &wm8804_dai, 1); - if (ret < 0) - kfree(wm8804); + return ret; } static int __devexit wm8804_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -752,7 +750,7 @@ static __devinit int wm8804_i2c_probe(struct i2c_client *i2c, struct wm8804_priv *wm8804; int ret; - wm8804 = kzalloc(sizeof *wm8804, GFP_KERNEL); + wm8804 = devm_kzalloc(&i2c->dev, sizeof *wm8804, GFP_KERNEL); if (!wm8804) return -ENOMEM; @@ -761,15 +759,14 @@ static __devinit int wm8804_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8804, &wm8804_dai, 1); - if (ret < 0) - kfree(wm8804); + return ret; } static __devexit int wm8804_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + return 0; } -- cgit v1.2.3-18-g5258 From 891271c28f06881332c6131158ea13f328401aa7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 19:58:06 +0000 Subject: ASoC: Convert wm8804 to direct regmap API usage The register map for this device is actually fairly sparse so the rbtree should be beneficial. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 115 +++++++++++++++++++++++++--------------------- 1 file changed, 62 insertions(+), 53 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index a9f1eb334f7..8abe3757a97 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -35,45 +36,33 @@ static const char *wm8804_supply_names[WM8804_NUM_SUPPLIES] = { "DVDD" }; -static const u8 wm8804_reg_defs[] = { - 0x05, /* R0 - RST/DEVID1 */ - 0x88, /* R1 - DEVID2 */ - 0x04, /* R2 - DEVREV */ - 0x21, /* R3 - PLL1 */ - 0xFD, /* R4 - PLL2 */ - 0x36, /* R5 - PLL3 */ - 0x07, /* R6 - PLL4 */ - 0x16, /* R7 - PLL5 */ - 0x18, /* R8 - PLL6 */ - 0xFF, /* R9 - SPDMODE */ - 0x00, /* R10 - INTMASK */ - 0x00, /* R11 - INTSTAT */ - 0x00, /* R12 - SPDSTAT */ - 0x00, /* R13 - RXCHAN1 */ - 0x00, /* R14 - RXCHAN2 */ - 0x00, /* R15 - RXCHAN3 */ - 0x00, /* R16 - RXCHAN4 */ - 0x00, /* R17 - RXCHAN5 */ - 0x00, /* R18 - SPDTX1 */ - 0x00, /* R19 - SPDTX2 */ - 0x00, /* R20 - SPDTX3 */ - 0x71, /* R21 - SPDTX4 */ - 0x0B, /* R22 - SPDTX5 */ - 0x70, /* R23 - GPO0 */ - 0x57, /* R24 - GPO1 */ - 0x00, /* R25 */ - 0x42, /* R26 - GPO2 */ - 0x06, /* R27 - AIFTX */ - 0x06, /* R28 - AIFRX */ - 0x80, /* R29 - SPDRX1 */ - 0x07, /* R30 - PWRDN */ +static const struct reg_default wm8804_reg_defaults[] = { + { 3, 0x21 }, /* R3 - PLL1 */ + { 4, 0xFD }, /* R4 - PLL2 */ + { 5, 0x36 }, /* R5 - PLL3 */ + { 6, 0x07 }, /* R6 - PLL4 */ + { 7, 0x16 }, /* R7 - PLL5 */ + { 8, 0x18 }, /* R8 - PLL6 */ + { 9, 0xFF }, /* R9 - SPDMODE */ + { 10, 0x00 }, /* R10 - INTMASK */ + { 18, 0x00 }, /* R18 - SPDTX1 */ + { 19, 0x00 }, /* R19 - SPDTX2 */ + { 20, 0x00 }, /* R20 - SPDTX3 */ + { 21, 0x71 }, /* R21 - SPDTX4 */ + { 22, 0x0B }, /* R22 - SPDTX5 */ + { 23, 0x70 }, /* R23 - GPO0 */ + { 24, 0x57 }, /* R24 - GPO1 */ + { 26, 0x42 }, /* R26 - GPO2 */ + { 27, 0x06 }, /* R27 - AIFTX */ + { 28, 0x06 }, /* R28 - AIFRX */ + { 29, 0x80 }, /* R29 - SPDRX1 */ + { 30, 0x07 }, /* R30 - PWRDN */ }; struct wm8804_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8804_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8804_NUM_SUPPLIES]; - struct snd_soc_codec *codec; }; static int txsrc_get(struct snd_kcontrol *kcontrol, @@ -94,7 +83,7 @@ static int wm8804_regulator_event_##n(struct notifier_block *nb, \ struct wm8804_priv *wm8804 = container_of(nb, struct wm8804_priv, \ disable_nb[n]); \ if (event & REGULATOR_EVENT_DISABLE) { \ - wm8804->codec->cache_sync = 1; \ + regcache_mark_dirty(wm8804->regmap); \ } \ return 0; \ } @@ -176,7 +165,7 @@ static int txsrc_put(struct snd_kcontrol *kcontrol, return 0; } -static int wm8804_volatile(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8804_volatile(struct device *dev, unsigned int reg) { switch (reg) { case WM8804_RST_DEVID1: @@ -189,12 +178,10 @@ static int wm8804_volatile(struct snd_soc_codec *codec, unsigned int reg) case WM8804_RXCHAN3: case WM8804_RXCHAN4: case WM8804_RXCHAN5: - return 1; + return true; default: - break; + return false; } - - return 0; } static int wm8804_reset(struct snd_soc_codec *codec) @@ -506,7 +493,7 @@ static int wm8804_set_bias_level(struct snd_soc_codec *codec, ret); return ret; } - snd_soc_cache_sync(codec); + regcache_sync(wm8804->regmap); } /* power down the OSC and the PLL */ snd_soc_update_bits(codec, WM8804_PWRDN, 0x9, 0x9); @@ -561,11 +548,11 @@ static int wm8804_probe(struct snd_soc_codec *codec) int i, id1, id2, ret; wm8804 = snd_soc_codec_get_drvdata(codec); - wm8804->codec = codec; codec->dapm.idle_bias_off = 1; + codec->control_data = wm8804->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, wm8804->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; @@ -618,8 +605,7 @@ static int wm8804_probe(struct snd_soc_codec *codec) id2 = (id2 << 8) | id1; - if (id2 != ((wm8804_reg_defs[WM8804_DEVID2] << 8) - | wm8804_reg_defs[WM8804_RST_DEVID1])) { + if (id2 != 0x8805) { dev_err(codec->dev, "Invalid device ID: %#x\n", id2); ret = -EINVAL; goto err_reg_enable; @@ -692,10 +678,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .suspend = wm8804_suspend, .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8804_reg_defs), - .reg_word_size = sizeof(u8), - .reg_cache_default = wm8804_reg_defs, - .volatile_register = wm8804_volatile, .controls = wm8804_snd_controls, .num_controls = ARRAY_SIZE(wm8804_snd_controls), @@ -707,6 +689,18 @@ static const struct of_device_id wm8804_of_match[] = { }; MODULE_DEVICE_TABLE(of, wm8804_of_match); +static struct regmap_config wm8804_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = WM8804_MAX_REGISTER, + .volatile_reg = wm8804_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8804_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8804_reg_defaults), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8804_spi_probe(struct spi_device *spi) { @@ -717,7 +711,12 @@ static int __devinit wm8804_spi_probe(struct spi_device *spi) if (!wm8804) return -ENOMEM; - wm8804->control_type = SND_SOC_SPI; + wm8804->regmap = regmap_init_spi(spi, &wm8804_regmap_config); + if (IS_ERR(wm8804->regmap)) { + ret = PTR_ERR(wm8804->regmap); + return ret; + } + spi_set_drvdata(spi, wm8804); ret = snd_soc_register_codec(&spi->dev, @@ -728,7 +727,9 @@ static int __devinit wm8804_spi_probe(struct spi_device *spi) static int __devexit wm8804_spi_remove(struct spi_device *spi) { + struct wm8804_priv *wm8804 = spi_get_drvdata(spi); snd_soc_unregister_codec(&spi->dev); + regmap_exit(wm8804->regmap); return 0; } @@ -754,18 +755,26 @@ static __devinit int wm8804_i2c_probe(struct i2c_client *i2c, if (!wm8804) return -ENOMEM; - wm8804->control_type = SND_SOC_I2C; i2c_set_clientdata(i2c, wm8804); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8804, &wm8804_dai, 1); + if (ret != 0) + goto err; + return 0; + +err: + regmap_exit(wm8804->regmap); return ret; } -static __devexit int wm8804_i2c_remove(struct i2c_client *client) +static __devexit int wm8804_i2c_remove(struct i2c_client *i2c) { - snd_soc_unregister_codec(&client->dev); + struct wm8804_priv *wm8804 = i2c_get_clientdata(i2c); + + snd_soc_unregister_codec(&i2c->dev); + regmap_exit(wm8804->regmap); return 0; } -- cgit v1.2.3-18-g5258 From 429440c947e72b6f6f317fa4346ddebf9d5e5413 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 20:03:23 +0000 Subject: ASoC: Make WM8904 I2C usage unconditional Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f31c754c886..1ad93737f0f 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2525,7 +2525,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8904 = { .volatile_register = wm8904_volatile_register, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2571,27 +2570,22 @@ static struct i2c_driver wm8904_i2c_driver = { .remove = __devexit_p(wm8904_i2c_remove), .id_table = wm8904_i2c_id, }; -#endif static int __init wm8904_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8904_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8904 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8904_modinit); static void __exit wm8904_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8904_i2c_driver); -#endif } module_exit(wm8904_exit); -- cgit v1.2.3-18-g5258 From 93e26d4e44e65ef803ea35cfd0fe14fd8af49cb0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 20:05:00 +0000 Subject: ASoC: Convert wm8904 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 1ad93737f0f..98d4f815b4a 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2531,7 +2531,8 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, struct wm8904_priv *wm8904; int ret; - wm8904 = kzalloc(sizeof(struct wm8904_priv), GFP_KERNEL); + wm8904 = devm_kzalloc(&i2c->dev, sizeof(struct wm8904_priv), + GFP_KERNEL); if (wm8904 == NULL) return -ENOMEM; @@ -2541,15 +2542,13 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8904, &wm8904_dai, 1); - if (ret < 0) - kfree(wm8904); + return ret; } static __devexit int wm8904_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3-18-g5258 From 274eb8f9d8780903ccd40e007928f26c3a8c6e15 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 21:07:04 +0000 Subject: ASoC: Use standard cache sync for WM8904 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 28 +--------------------------- 1 file changed, 1 insertion(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 98d4f815b4a..673a2fe585b 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2088,32 +2088,6 @@ static int wm8904_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static void wm8904_sync_cache(struct snd_soc_codec *codec) -{ - u16 *reg_cache = codec->reg_cache; - int i; - - if (!codec->cache_sync) - return; - - codec->cache_only = 0; - - /* Sync back cached values if they're different from the - * hardware default. - */ - for (i = 1; i < codec->driver->reg_cache_size; i++) { - if (!wm8904_access[i].writable) - continue; - - if (reg_cache[i] == wm8904_reg[i]) - continue; - - snd_soc_write(codec, i, reg_cache[i]); - } - - codec->cache_sync = 0; -} - static int wm8904_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2146,7 +2120,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, return ret; } - wm8904_sync_cache(codec); + snd_soc_cache_sync(codec); /* Enable bias */ snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, -- cgit v1.2.3-18-g5258 From 84d0d83180a8db564483f5e2ff58a7661d78858b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 21:12:51 +0000 Subject: ASoC: Convert WM8904 to direct regmap API usage The device has a very sparse register map so should benefit from using a rbtree cache. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 768 +++++++++++++++------------------------------- 1 file changed, 254 insertions(+), 514 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 673a2fe585b..f2a740de3ad 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -17,6 +17,7 @@ #include #include #include +#include #include #include #include @@ -47,6 +48,7 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = { /* codec private data */ struct wm8904_priv { + struct regmap *regmap; enum wm8904_type devtype; @@ -86,517 +88,229 @@ struct wm8904_priv { int dcs_state[WM8904_NUM_DCS_CHANNELS]; }; -static const u16 wm8904_reg[WM8904_MAX_REGISTER + 1] = { - 0x8904, /* R0 - SW Reset and ID */ - 0x0000, /* R1 - Revision */ - 0x0000, /* R2 */ - 0x0000, /* R3 */ - 0x0018, /* R4 - Bias Control 0 */ - 0x0000, /* R5 - VMID Control 0 */ - 0x0000, /* R6 - Mic Bias Control 0 */ - 0x0000, /* R7 - Mic Bias Control 1 */ - 0x0001, /* R8 - Analogue DAC 0 */ - 0x9696, /* R9 - mic Filter Control */ - 0x0001, /* R10 - Analogue ADC 0 */ - 0x0000, /* R11 */ - 0x0000, /* R12 - Power Management 0 */ - 0x0000, /* R13 */ - 0x0000, /* R14 - Power Management 2 */ - 0x0000, /* R15 - Power Management 3 */ - 0x0000, /* R16 */ - 0x0000, /* R17 */ - 0x0000, /* R18 - Power Management 6 */ - 0x0000, /* R19 */ - 0x945E, /* R20 - Clock Rates 0 */ - 0x0C05, /* R21 - Clock Rates 1 */ - 0x0006, /* R22 - Clock Rates 2 */ - 0x0000, /* R23 */ - 0x0050, /* R24 - Audio Interface 0 */ - 0x000A, /* R25 - Audio Interface 1 */ - 0x00E4, /* R26 - Audio Interface 2 */ - 0x0040, /* R27 - Audio Interface 3 */ - 0x0000, /* R28 */ - 0x0000, /* R29 */ - 0x00C0, /* R30 - DAC Digital Volume Left */ - 0x00C0, /* R31 - DAC Digital Volume Right */ - 0x0000, /* R32 - DAC Digital 0 */ - 0x0008, /* R33 - DAC Digital 1 */ - 0x0000, /* R34 */ - 0x0000, /* R35 */ - 0x00C0, /* R36 - ADC Digital Volume Left */ - 0x00C0, /* R37 - ADC Digital Volume Right */ - 0x0010, /* R38 - ADC Digital 0 */ - 0x0000, /* R39 - Digital Microphone 0 */ - 0x01AF, /* R40 - DRC 0 */ - 0x3248, /* R41 - DRC 1 */ - 0x0000, /* R42 - DRC 2 */ - 0x0000, /* R43 - DRC 3 */ - 0x0085, /* R44 - Analogue Left Input 0 */ - 0x0085, /* R45 - Analogue Right Input 0 */ - 0x0044, /* R46 - Analogue Left Input 1 */ - 0x0044, /* R47 - Analogue Right Input 1 */ - 0x0000, /* R48 */ - 0x0000, /* R49 */ - 0x0000, /* R50 */ - 0x0000, /* R51 */ - 0x0000, /* R52 */ - 0x0000, /* R53 */ - 0x0000, /* R54 */ - 0x0000, /* R55 */ - 0x0000, /* R56 */ - 0x002D, /* R57 - Analogue OUT1 Left */ - 0x002D, /* R58 - Analogue OUT1 Right */ - 0x0039, /* R59 - Analogue OUT2 Left */ - 0x0039, /* R60 - Analogue OUT2 Right */ - 0x0000, /* R61 - Analogue OUT12 ZC */ - 0x0000, /* R62 */ - 0x0000, /* R63 */ - 0x0000, /* R64 */ - 0x0000, /* R65 */ - 0x0000, /* R66 */ - 0x0000, /* R67 - DC Servo 0 */ - 0x0000, /* R68 - DC Servo 1 */ - 0xAAAA, /* R69 - DC Servo 2 */ - 0x0000, /* R70 */ - 0xAAAA, /* R71 - DC Servo 4 */ - 0xAAAA, /* R72 - DC Servo 5 */ - 0x0000, /* R73 - DC Servo 6 */ - 0x0000, /* R74 - DC Servo 7 */ - 0x0000, /* R75 - DC Servo 8 */ - 0x0000, /* R76 - DC Servo 9 */ - 0x0000, /* R77 - DC Servo Readback 0 */ - 0x0000, /* R78 */ - 0x0000, /* R79 */ - 0x0000, /* R80 */ - 0x0000, /* R81 */ - 0x0000, /* R82 */ - 0x0000, /* R83 */ - 0x0000, /* R84 */ - 0x0000, /* R85 */ - 0x0000, /* R86 */ - 0x0000, /* R87 */ - 0x0000, /* R88 */ - 0x0000, /* R89 */ - 0x0000, /* R90 - Analogue HP 0 */ - 0x0000, /* R91 */ - 0x0000, /* R92 */ - 0x0000, /* R93 */ - 0x0000, /* R94 - Analogue Lineout 0 */ - 0x0000, /* R95 */ - 0x0000, /* R96 */ - 0x0000, /* R97 */ - 0x0000, /* R98 - Charge Pump 0 */ - 0x0000, /* R99 */ - 0x0000, /* R100 */ - 0x0000, /* R101 */ - 0x0000, /* R102 */ - 0x0000, /* R103 */ - 0x0004, /* R104 - Class W 0 */ - 0x0000, /* R105 */ - 0x0000, /* R106 */ - 0x0000, /* R107 */ - 0x0000, /* R108 - Write Sequencer 0 */ - 0x0000, /* R109 - Write Sequencer 1 */ - 0x0000, /* R110 - Write Sequencer 2 */ - 0x0000, /* R111 - Write Sequencer 3 */ - 0x0000, /* R112 - Write Sequencer 4 */ - 0x0000, /* R113 */ - 0x0000, /* R114 */ - 0x0000, /* R115 */ - 0x0000, /* R116 - FLL Control 1 */ - 0x0007, /* R117 - FLL Control 2 */ - 0x0000, /* R118 - FLL Control 3 */ - 0x2EE0, /* R119 - FLL Control 4 */ - 0x0004, /* R120 - FLL Control 5 */ - 0x0014, /* R121 - GPIO Control 1 */ - 0x0010, /* R122 - GPIO Control 2 */ - 0x0010, /* R123 - GPIO Control 3 */ - 0x0000, /* R124 - GPIO Control 4 */ - 0x0000, /* R125 */ - 0x0000, /* R126 - Digital Pulls */ - 0x0000, /* R127 - Interrupt Status */ - 0xFFFF, /* R128 - Interrupt Status Mask */ - 0x0000, /* R129 - Interrupt Polarity */ - 0x0000, /* R130 - Interrupt Debounce */ - 0x0000, /* R131 */ - 0x0000, /* R132 */ - 0x0000, /* R133 */ - 0x0000, /* R134 - EQ1 */ - 0x000C, /* R135 - EQ2 */ - 0x000C, /* R136 - EQ3 */ - 0x000C, /* R137 - EQ4 */ - 0x000C, /* R138 - EQ5 */ - 0x000C, /* R139 - EQ6 */ - 0x0FCA, /* R140 - EQ7 */ - 0x0400, /* R141 - EQ8 */ - 0x00D8, /* R142 - EQ9 */ - 0x1EB5, /* R143 - EQ10 */ - 0xF145, /* R144 - EQ11 */ - 0x0B75, /* R145 - EQ12 */ - 0x01C5, /* R146 - EQ13 */ - 0x1C58, /* R147 - EQ14 */ - 0xF373, /* R148 - EQ15 */ - 0x0A54, /* R149 - EQ16 */ - 0x0558, /* R150 - EQ17 */ - 0x168E, /* R151 - EQ18 */ - 0xF829, /* R152 - EQ19 */ - 0x07AD, /* R153 - EQ20 */ - 0x1103, /* R154 - EQ21 */ - 0x0564, /* R155 - EQ22 */ - 0x0559, /* R156 - EQ23 */ - 0x4000, /* R157 - EQ24 */ - 0x0000, /* R158 */ - 0x0000, /* R159 */ - 0x0000, /* R160 */ - 0x0000, /* R161 - Control Interface Test 1 */ - 0x0000, /* R162 */ - 0x0000, /* R163 */ - 0x0000, /* R164 */ - 0x0000, /* R165 */ - 0x0000, /* R166 */ - 0x0000, /* R167 */ - 0x0000, /* R168 */ - 0x0000, /* R169 */ - 0x0000, /* R170 */ - 0x0000, /* R171 */ - 0x0000, /* R172 */ - 0x0000, /* R173 */ - 0x0000, /* R174 */ - 0x0000, /* R175 */ - 0x0000, /* R176 */ - 0x0000, /* R177 */ - 0x0000, /* R178 */ - 0x0000, /* R179 */ - 0x0000, /* R180 */ - 0x0000, /* R181 */ - 0x0000, /* R182 */ - 0x0000, /* R183 */ - 0x0000, /* R184 */ - 0x0000, /* R185 */ - 0x0000, /* R186 */ - 0x0000, /* R187 */ - 0x0000, /* R188 */ - 0x0000, /* R189 */ - 0x0000, /* R190 */ - 0x0000, /* R191 */ - 0x0000, /* R192 */ - 0x0000, /* R193 */ - 0x0000, /* R194 */ - 0x0000, /* R195 */ - 0x0000, /* R196 */ - 0x0000, /* R197 */ - 0x0000, /* R198 */ - 0x0000, /* R199 */ - 0x0000, /* R200 */ - 0x0000, /* R201 */ - 0x0000, /* R202 */ - 0x0000, /* R203 */ - 0x0000, /* R204 - Analogue Output Bias 0 */ - 0x0000, /* R205 */ - 0x0000, /* R206 */ - 0x0000, /* R207 */ - 0x0000, /* R208 */ - 0x0000, /* R209 */ - 0x0000, /* R210 */ - 0x0000, /* R211 */ - 0x0000, /* R212 */ - 0x0000, /* R213 */ - 0x0000, /* R214 */ - 0x0000, /* R215 */ - 0x0000, /* R216 */ - 0x0000, /* R217 */ - 0x0000, /* R218 */ - 0x0000, /* R219 */ - 0x0000, /* R220 */ - 0x0000, /* R221 */ - 0x0000, /* R222 */ - 0x0000, /* R223 */ - 0x0000, /* R224 */ - 0x0000, /* R225 */ - 0x0000, /* R226 */ - 0x0000, /* R227 */ - 0x0000, /* R228 */ - 0x0000, /* R229 */ - 0x0000, /* R230 */ - 0x0000, /* R231 */ - 0x0000, /* R232 */ - 0x0000, /* R233 */ - 0x0000, /* R234 */ - 0x0000, /* R235 */ - 0x0000, /* R236 */ - 0x0000, /* R237 */ - 0x0000, /* R238 */ - 0x0000, /* R239 */ - 0x0000, /* R240 */ - 0x0000, /* R241 */ - 0x0000, /* R242 */ - 0x0000, /* R243 */ - 0x0000, /* R244 */ - 0x0000, /* R245 */ - 0x0000, /* R246 */ - 0x0000, /* R247 - FLL NCO Test 0 */ - 0x0019, /* R248 - FLL NCO Test 1 */ +static const struct reg_default wm8904_reg_defaults[] = { + { 4, 0x0018 }, /* R4 - Bias Control 0 */ + { 5, 0x0000 }, /* R5 - VMID Control 0 */ + { 6, 0x0000 }, /* R6 - Mic Bias Control 0 */ + { 7, 0x0000 }, /* R7 - Mic Bias Control 1 */ + { 8, 0x0001 }, /* R8 - Analogue DAC 0 */ + { 9, 0x9696 }, /* R9 - mic Filter Control */ + { 10, 0x0001 }, /* R10 - Analogue ADC 0 */ + { 12, 0x0000 }, /* R12 - Power Management 0 */ + { 14, 0x0000 }, /* R14 - Power Management 2 */ + { 15, 0x0000 }, /* R15 - Power Management 3 */ + { 18, 0x0000 }, /* R18 - Power Management 6 */ + { 19, 0x945E }, /* R20 - Clock Rates 0 */ + { 21, 0x0C05 }, /* R21 - Clock Rates 1 */ + { 22, 0x0006 }, /* R22 - Clock Rates 2 */ + { 24, 0x0050 }, /* R24 - Audio Interface 0 */ + { 25, 0x000A }, /* R25 - Audio Interface 1 */ + { 26, 0x00E4 }, /* R26 - Audio Interface 2 */ + { 27, 0x0040 }, /* R27 - Audio Interface 3 */ + { 30, 0x00C0 }, /* R30 - DAC Digital Volume Left */ + { 31, 0x00C0 }, /* R31 - DAC Digital Volume Right */ + { 32, 0x0000 }, /* R32 - DAC Digital 0 */ + { 33, 0x0008 }, /* R33 - DAC Digital 1 */ + { 36, 0x00C0 }, /* R36 - ADC Digital Volume Left */ + { 37, 0x00C0 }, /* R37 - ADC Digital Volume Right */ + { 38, 0x0010 }, /* R38 - ADC Digital 0 */ + { 39, 0x0000 }, /* R39 - Digital Microphone 0 */ + { 40, 0x01AF }, /* R40 - DRC 0 */ + { 41, 0x3248 }, /* R41 - DRC 1 */ + { 42, 0x0000 }, /* R42 - DRC 2 */ + { 43, 0x0000 }, /* R43 - DRC 3 */ + { 44, 0x0085 }, /* R44 - Analogue Left Input 0 */ + { 45, 0x0085 }, /* R45 - Analogue Right Input 0 */ + { 46, 0x0044 }, /* R46 - Analogue Left Input 1 */ + { 47, 0x0044 }, /* R47 - Analogue Right Input 1 */ + { 57, 0x002D }, /* R57 - Analogue OUT1 Left */ + { 58, 0x002D }, /* R58 - Analogue OUT1 Right */ + { 59, 0x0039 }, /* R59 - Analogue OUT2 Left */ + { 60, 0x0039 }, /* R60 - Analogue OUT2 Right */ + { 61, 0x0000 }, /* R61 - Analogue OUT12 ZC */ + { 67, 0x0000 }, /* R67 - DC Servo 0 */ + { 69, 0xAAAA }, /* R69 - DC Servo 2 */ + { 71, 0xAAAA }, /* R71 - DC Servo 4 */ + { 72, 0xAAAA }, /* R72 - DC Servo 5 */ + { 90, 0x0000 }, /* R90 - Analogue HP 0 */ + { 94, 0x0000 }, /* R94 - Analogue Lineout 0 */ + { 98, 0x0000 }, /* R98 - Charge Pump 0 */ + { 104, 0x0004 }, /* R104 - Class W 0 */ + { 108, 0x0000 }, /* R108 - Write Sequencer 0 */ + { 109, 0x0000 }, /* R109 - Write Sequencer 1 */ + { 110, 0x0000 }, /* R110 - Write Sequencer 2 */ + { 111, 0x0000 }, /* R111 - Write Sequencer 3 */ + { 112, 0x0000 }, /* R112 - Write Sequencer 4 */ + { 116, 0x0000 }, /* R116 - FLL Control 1 */ + { 117, 0x0007 }, /* R117 - FLL Control 2 */ + { 118, 0x0000 }, /* R118 - FLL Control 3 */ + { 119, 0x2EE0 }, /* R119 - FLL Control 4 */ + { 120, 0x0004 }, /* R120 - FLL Control 5 */ + { 121, 0x0014 }, /* R121 - GPIO Control 1 */ + { 122, 0x0010 }, /* R122 - GPIO Control 2 */ + { 123, 0x0010 }, /* R123 - GPIO Control 3 */ + { 124, 0x0000 }, /* R124 - GPIO Control 4 */ + { 126, 0x0000 }, /* R126 - Digital Pulls */ + { 128, 0xFFFF }, /* R128 - Interrupt Status Mask */ + { 129, 0x0000 }, /* R129 - Interrupt Polarity */ + { 130, 0x0000 }, /* R130 - Interrupt Debounce */ + { 134, 0x0000 }, /* R134 - EQ1 */ + { 135, 0x000C }, /* R135 - EQ2 */ + { 136, 0x000C }, /* R136 - EQ3 */ + { 137, 0x000C }, /* R137 - EQ4 */ + { 138, 0x000C }, /* R138 - EQ5 */ + { 139, 0x000C }, /* R139 - EQ6 */ + { 140, 0x0FCA }, /* R140 - EQ7 */ + { 141, 0x0400 }, /* R141 - EQ8 */ + { 142, 0x00D8 }, /* R142 - EQ9 */ + { 143, 0x1EB5 }, /* R143 - EQ10 */ + { 144, 0xF145 }, /* R144 - EQ11 */ + { 145, 0x0B75 }, /* R145 - EQ12 */ + { 146, 0x01C5 }, /* R146 - EQ13 */ + { 147, 0x1C58 }, /* R147 - EQ14 */ + { 148, 0xF373 }, /* R148 - EQ15 */ + { 149, 0x0A54 }, /* R149 - EQ16 */ + { 150, 0x0558 }, /* R150 - EQ17 */ + { 151, 0x168E }, /* R151 - EQ18 */ + { 152, 0xF829 }, /* R152 - EQ19 */ + { 153, 0x07AD }, /* R153 - EQ20 */ + { 154, 0x1103 }, /* R154 - EQ21 */ + { 155, 0x0564 }, /* R155 - EQ22 */ + { 156, 0x0559 }, /* R156 - EQ23 */ + { 157, 0x4000 }, /* R157 - EQ24 */ + { 161, 0x0000 }, /* R161 - Control Interface Test 1 */ + { 204, 0x0000 }, /* R204 - Analogue Output Bias 0 */ + { 247, 0x0000 }, /* R247 - FLL NCO Test 0 */ + { 248, 0x0019 }, /* R248 - FLL NCO Test 1 */ }; -static struct { - int readable; - int writable; - int vol; -} wm8904_access[] = { - { 0xFFFF, 0xFFFF, 1 }, /* R0 - SW Reset and ID */ - { 0x0000, 0x0000, 0 }, /* R1 - Revision */ - { 0x0000, 0x0000, 0 }, /* R2 */ - { 0x0000, 0x0000, 0 }, /* R3 */ - { 0x001F, 0x001F, 0 }, /* R4 - Bias Control 0 */ - { 0x0047, 0x0047, 0 }, /* R5 - VMID Control 0 */ - { 0x007F, 0x007F, 0 }, /* R6 - Mic Bias Control 0 */ - { 0xC007, 0xC007, 0 }, /* R7 - Mic Bias Control 1 */ - { 0x001E, 0x001E, 0 }, /* R8 - Analogue DAC 0 */ - { 0xFFFF, 0xFFFF, 0 }, /* R9 - mic Filter Control */ - { 0x0001, 0x0001, 0 }, /* R10 - Analogue ADC 0 */ - { 0x0000, 0x0000, 0 }, /* R11 */ - { 0x0003, 0x0003, 0 }, /* R12 - Power Management 0 */ - { 0x0000, 0x0000, 0 }, /* R13 */ - { 0x0003, 0x0003, 0 }, /* R14 - Power Management 2 */ - { 0x0003, 0x0003, 0 }, /* R15 - Power Management 3 */ - { 0x0000, 0x0000, 0 }, /* R16 */ - { 0x0000, 0x0000, 0 }, /* R17 */ - { 0x000F, 0x000F, 0 }, /* R18 - Power Management 6 */ - { 0x0000, 0x0000, 0 }, /* R19 */ - { 0x7001, 0x7001, 0 }, /* R20 - Clock Rates 0 */ - { 0x3C07, 0x3C07, 0 }, /* R21 - Clock Rates 1 */ - { 0xD00F, 0xD00F, 0 }, /* R22 - Clock Rates 2 */ - { 0x0000, 0x0000, 0 }, /* R23 */ - { 0x1FFF, 0x1FFF, 0 }, /* R24 - Audio Interface 0 */ - { 0x3DDF, 0x3DDF, 0 }, /* R25 - Audio Interface 1 */ - { 0x0F1F, 0x0F1F, 0 }, /* R26 - Audio Interface 2 */ - { 0x0FFF, 0x0FFF, 0 }, /* R27 - Audio Interface 3 */ - { 0x0000, 0x0000, 0 }, /* R28 */ - { 0x0000, 0x0000, 0 }, /* R29 */ - { 0x00FF, 0x01FF, 0 }, /* R30 - DAC Digital Volume Left */ - { 0x00FF, 0x01FF, 0 }, /* R31 - DAC Digital Volume Right */ - { 0x0FFF, 0x0FFF, 0 }, /* R32 - DAC Digital 0 */ - { 0x1E4E, 0x1E4E, 0 }, /* R33 - DAC Digital 1 */ - { 0x0000, 0x0000, 0 }, /* R34 */ - { 0x0000, 0x0000, 0 }, /* R35 */ - { 0x00FF, 0x01FF, 0 }, /* R36 - ADC Digital Volume Left */ - { 0x00FF, 0x01FF, 0 }, /* R37 - ADC Digital Volume Right */ - { 0x0073, 0x0073, 0 }, /* R38 - ADC Digital 0 */ - { 0x1800, 0x1800, 0 }, /* R39 - Digital Microphone 0 */ - { 0xDFEF, 0xDFEF, 0 }, /* R40 - DRC 0 */ - { 0xFFFF, 0xFFFF, 0 }, /* R41 - DRC 1 */ - { 0x003F, 0x003F, 0 }, /* R42 - DRC 2 */ - { 0x07FF, 0x07FF, 0 }, /* R43 - DRC 3 */ - { 0x009F, 0x009F, 0 }, /* R44 - Analogue Left Input 0 */ - { 0x009F, 0x009F, 0 }, /* R45 - Analogue Right Input 0 */ - { 0x007F, 0x007F, 0 }, /* R46 - Analogue Left Input 1 */ - { 0x007F, 0x007F, 0 }, /* R47 - Analogue Right Input 1 */ - { 0x0000, 0x0000, 0 }, /* R48 */ - { 0x0000, 0x0000, 0 }, /* R49 */ - { 0x0000, 0x0000, 0 }, /* R50 */ - { 0x0000, 0x0000, 0 }, /* R51 */ - { 0x0000, 0x0000, 0 }, /* R52 */ - { 0x0000, 0x0000, 0 }, /* R53 */ - { 0x0000, 0x0000, 0 }, /* R54 */ - { 0x0000, 0x0000, 0 }, /* R55 */ - { 0x0000, 0x0000, 0 }, /* R56 */ - { 0x017F, 0x01FF, 0 }, /* R57 - Analogue OUT1 Left */ - { 0x017F, 0x01FF, 0 }, /* R58 - Analogue OUT1 Right */ - { 0x017F, 0x01FF, 0 }, /* R59 - Analogue OUT2 Left */ - { 0x017F, 0x01FF, 0 }, /* R60 - Analogue OUT2 Right */ - { 0x000F, 0x000F, 0 }, /* R61 - Analogue OUT12 ZC */ - { 0x0000, 0x0000, 0 }, /* R62 */ - { 0x0000, 0x0000, 0 }, /* R63 */ - { 0x0000, 0x0000, 0 }, /* R64 */ - { 0x0000, 0x0000, 0 }, /* R65 */ - { 0x0000, 0x0000, 0 }, /* R66 */ - { 0x000F, 0x000F, 0 }, /* R67 - DC Servo 0 */ - { 0xFFFF, 0xFFFF, 1 }, /* R68 - DC Servo 1 */ - { 0x0F0F, 0x0F0F, 0 }, /* R69 - DC Servo 2 */ - { 0x0000, 0x0000, 0 }, /* R70 */ - { 0x007F, 0x007F, 0 }, /* R71 - DC Servo 4 */ - { 0x007F, 0x007F, 0 }, /* R72 - DC Servo 5 */ - { 0x00FF, 0x00FF, 1 }, /* R73 - DC Servo 6 */ - { 0x00FF, 0x00FF, 1 }, /* R74 - DC Servo 7 */ - { 0x00FF, 0x00FF, 1 }, /* R75 - DC Servo 8 */ - { 0x00FF, 0x00FF, 1 }, /* R76 - DC Servo 9 */ - { 0x0FFF, 0x0000, 1 }, /* R77 - DC Servo Readback 0 */ - { 0x0000, 0x0000, 0 }, /* R78 */ - { 0x0000, 0x0000, 0 }, /* R79 */ - { 0x0000, 0x0000, 0 }, /* R80 */ - { 0x0000, 0x0000, 0 }, /* R81 */ - { 0x0000, 0x0000, 0 }, /* R82 */ - { 0x0000, 0x0000, 0 }, /* R83 */ - { 0x0000, 0x0000, 0 }, /* R84 */ - { 0x0000, 0x0000, 0 }, /* R85 */ - { 0x0000, 0x0000, 0 }, /* R86 */ - { 0x0000, 0x0000, 0 }, /* R87 */ - { 0x0000, 0x0000, 0 }, /* R88 */ - { 0x0000, 0x0000, 0 }, /* R89 */ - { 0x00FF, 0x00FF, 0 }, /* R90 - Analogue HP 0 */ - { 0x0000, 0x0000, 0 }, /* R91 */ - { 0x0000, 0x0000, 0 }, /* R92 */ - { 0x0000, 0x0000, 0 }, /* R93 */ - { 0x00FF, 0x00FF, 0 }, /* R94 - Analogue Lineout 0 */ - { 0x0000, 0x0000, 0 }, /* R95 */ - { 0x0000, 0x0000, 0 }, /* R96 */ - { 0x0000, 0x0000, 0 }, /* R97 */ - { 0x0001, 0x0001, 0 }, /* R98 - Charge Pump 0 */ - { 0x0000, 0x0000, 0 }, /* R99 */ - { 0x0000, 0x0000, 0 }, /* R100 */ - { 0x0000, 0x0000, 0 }, /* R101 */ - { 0x0000, 0x0000, 0 }, /* R102 */ - { 0x0000, 0x0000, 0 }, /* R103 */ - { 0x0001, 0x0001, 0 }, /* R104 - Class W 0 */ - { 0x0000, 0x0000, 0 }, /* R105 */ - { 0x0000, 0x0000, 0 }, /* R106 */ - { 0x0000, 0x0000, 0 }, /* R107 */ - { 0x011F, 0x011F, 0 }, /* R108 - Write Sequencer 0 */ - { 0x7FFF, 0x7FFF, 0 }, /* R109 - Write Sequencer 1 */ - { 0x4FFF, 0x4FFF, 0 }, /* R110 - Write Sequencer 2 */ - { 0x003F, 0x033F, 0 }, /* R111 - Write Sequencer 3 */ - { 0x03F1, 0x0000, 0 }, /* R112 - Write Sequencer 4 */ - { 0x0000, 0x0000, 0 }, /* R113 */ - { 0x0000, 0x0000, 0 }, /* R114 */ - { 0x0000, 0x0000, 0 }, /* R115 */ - { 0x0007, 0x0007, 0 }, /* R116 - FLL Control 1 */ - { 0x3F77, 0x3F77, 0 }, /* R117 - FLL Control 2 */ - { 0xFFFF, 0xFFFF, 0 }, /* R118 - FLL Control 3 */ - { 0x7FEF, 0x7FEF, 0 }, /* R119 - FLL Control 4 */ - { 0x001B, 0x001B, 0 }, /* R120 - FLL Control 5 */ - { 0x003F, 0x003F, 0 }, /* R121 - GPIO Control 1 */ - { 0x003F, 0x003F, 0 }, /* R122 - GPIO Control 2 */ - { 0x003F, 0x003F, 0 }, /* R123 - GPIO Control 3 */ - { 0x038F, 0x038F, 0 }, /* R124 - GPIO Control 4 */ - { 0x0000, 0x0000, 0 }, /* R125 */ - { 0x00FF, 0x00FF, 0 }, /* R126 - Digital Pulls */ - { 0x07FF, 0x03FF, 1 }, /* R127 - Interrupt Status */ - { 0x03FF, 0x03FF, 0 }, /* R128 - Interrupt Status Mask */ - { 0x03FF, 0x03FF, 0 }, /* R129 - Interrupt Polarity */ - { 0x03FF, 0x03FF, 0 }, /* R130 - Interrupt Debounce */ - { 0x0000, 0x0000, 0 }, /* R131 */ - { 0x0000, 0x0000, 0 }, /* R132 */ - { 0x0000, 0x0000, 0 }, /* R133 */ - { 0x0001, 0x0001, 0 }, /* R134 - EQ1 */ - { 0x001F, 0x001F, 0 }, /* R135 - EQ2 */ - { 0x001F, 0x001F, 0 }, /* R136 - EQ3 */ - { 0x001F, 0x001F, 0 }, /* R137 - EQ4 */ - { 0x001F, 0x001F, 0 }, /* R138 - EQ5 */ - { 0x001F, 0x001F, 0 }, /* R139 - EQ6 */ - { 0xFFFF, 0xFFFF, 0 }, /* R140 - EQ7 */ - { 0xFFFF, 0xFFFF, 0 }, /* R141 - EQ8 */ - { 0xFFFF, 0xFFFF, 0 }, /* R142 - EQ9 */ - { 0xFFFF, 0xFFFF, 0 }, /* R143 - EQ10 */ - { 0xFFFF, 0xFFFF, 0 }, /* R144 - EQ11 */ - { 0xFFFF, 0xFFFF, 0 }, /* R145 - EQ12 */ - { 0xFFFF, 0xFFFF, 0 }, /* R146 - EQ13 */ - { 0xFFFF, 0xFFFF, 0 }, /* R147 - EQ14 */ - { 0xFFFF, 0xFFFF, 0 }, /* R148 - EQ15 */ - { 0xFFFF, 0xFFFF, 0 }, /* R149 - EQ16 */ - { 0xFFFF, 0xFFFF, 0 }, /* R150 - EQ17 */ - { 0xFFFF, 0xFFFF, 0 }, /* R151wm8523_dai - EQ18 */ - { 0xFFFF, 0xFFFF, 0 }, /* R152 - EQ19 */ - { 0xFFFF, 0xFFFF, 0 }, /* R153 - EQ20 */ - { 0xFFFF, 0xFFFF, 0 }, /* R154 - EQ21 */ - { 0xFFFF, 0xFFFF, 0 }, /* R155 - EQ22 */ - { 0xFFFF, 0xFFFF, 0 }, /* R156 - EQ23 */ - { 0xFFFF, 0xFFFF, 0 }, /* R157 - EQ24 */ - { 0x0000, 0x0000, 0 }, /* R158 */ - { 0x0000, 0x0000, 0 }, /* R159 */ - { 0x0000, 0x0000, 0 }, /* R160 */ - { 0x0002, 0x0002, 0 }, /* R161 - Control Interface Test 1 */ - { 0x0000, 0x0000, 0 }, /* R162 */ - { 0x0000, 0x0000, 0 }, /* R163 */ - { 0x0000, 0x0000, 0 }, /* R164 */ - { 0x0000, 0x0000, 0 }, /* R165 */ - { 0x0000, 0x0000, 0 }, /* R166 */ - { 0x0000, 0x0000, 0 }, /* R167 */ - { 0x0000, 0x0000, 0 }, /* R168 */ - { 0x0000, 0x0000, 0 }, /* R169 */ - { 0x0000, 0x0000, 0 }, /* R170 */ - { 0x0000, 0x0000, 0 }, /* R171 */ - { 0x0000, 0x0000, 0 }, /* R172 */ - { 0x0000, 0x0000, 0 }, /* R173 */ - { 0x0000, 0x0000, 0 }, /* R174 */ - { 0x0000, 0x0000, 0 }, /* R175 */ - { 0x0000, 0x0000, 0 }, /* R176 */ - { 0x0000, 0x0000, 0 }, /* R177 */ - { 0x0000, 0x0000, 0 }, /* R178 */ - { 0x0000, 0x0000, 0 }, /* R179 */ - { 0x0000, 0x0000, 0 }, /* R180 */ - { 0x0000, 0x0000, 0 }, /* R181 */ - { 0x0000, 0x0000, 0 }, /* R182 */ - { 0x0000, 0x0000, 0 }, /* R183 */ - { 0x0000, 0x0000, 0 }, /* R184 */ - { 0x0000, 0x0000, 0 }, /* R185 */ - { 0x0000, 0x0000, 0 }, /* R186 */ - { 0x0000, 0x0000, 0 }, /* R187 */ - { 0x0000, 0x0000, 0 }, /* R188 */ - { 0x0000, 0x0000, 0 }, /* R189 */ - { 0x0000, 0x0000, 0 }, /* R190 */ - { 0x0000, 0x0000, 0 }, /* R191 */ - { 0x0000, 0x0000, 0 }, /* R192 */ - { 0x0000, 0x0000, 0 }, /* R193 */ - { 0x0000, 0x0000, 0 }, /* R194 */ - { 0x0000, 0x0000, 0 }, /* R195 */ - { 0x0000, 0x0000, 0 }, /* R196 */ - { 0x0000, 0x0000, 0 }, /* R197 */ - { 0x0000, 0x0000, 0 }, /* R198 */ - { 0x0000, 0x0000, 0 }, /* R199 */ - { 0x0000, 0x0000, 0 }, /* R200 */ - { 0x0000, 0x0000, 0 }, /* R201 */ - { 0x0000, 0x0000, 0 }, /* R202 */ - { 0x0000, 0x0000, 0 }, /* R203 */ - { 0x0070, 0x0070, 0 }, /* R204 - Analogue Output Bias 0 */ - { 0x0000, 0x0000, 0 }, /* R205 */ - { 0x0000, 0x0000, 0 }, /* R206 */ - { 0x0000, 0x0000, 0 }, /* R207 */ - { 0x0000, 0x0000, 0 }, /* R208 */ - { 0x0000, 0x0000, 0 }, /* R209 */ - { 0x0000, 0x0000, 0 }, /* R210 */ - { 0x0000, 0x0000, 0 }, /* R211 */ - { 0x0000, 0x0000, 0 }, /* R212 */ - { 0x0000, 0x0000, 0 }, /* R213 */ - { 0x0000, 0x0000, 0 }, /* R214 */ - { 0x0000, 0x0000, 0 }, /* R215 */ - { 0x0000, 0x0000, 0 }, /* R216 */ - { 0x0000, 0x0000, 0 }, /* R217 */ - { 0x0000, 0x0000, 0 }, /* R218 */ - { 0x0000, 0x0000, 0 }, /* R219 */ - { 0x0000, 0x0000, 0 }, /* R220 */ - { 0x0000, 0x0000, 0 }, /* R221 */ - { 0x0000, 0x0000, 0 }, /* R222 */ - { 0x0000, 0x0000, 0 }, /* R223 */ - { 0x0000, 0x0000, 0 }, /* R224 */ - { 0x0000, 0x0000, 0 }, /* R225 */ - { 0x0000, 0x0000, 0 }, /* R226 */ - { 0x0000, 0x0000, 0 }, /* R227 */ - { 0x0000, 0x0000, 0 }, /* R228 */ - { 0x0000, 0x0000, 0 }, /* R229 */ - { 0x0000, 0x0000, 0 }, /* R230 */ - { 0x0000, 0x0000, 0 }, /* R231 */ - { 0x0000, 0x0000, 0 }, /* R232 */ - { 0x0000, 0x0000, 0 }, /* R233 */ - { 0x0000, 0x0000, 0 }, /* R234 */ - { 0x0000, 0x0000, 0 }, /* R235 */ - { 0x0000, 0x0000, 0 }, /* R236 */ - { 0x0000, 0x0000, 0 }, /* R237 */ - { 0x0000, 0x0000, 0 }, /* R238 */ - { 0x0000, 0x0000, 0 }, /* R239 */ - { 0x0000, 0x0000, 0 }, /* R240 */ - { 0x0000, 0x0000, 0 }, /* R241 */ - { 0x0000, 0x0000, 0 }, /* R242 */ - { 0x0000, 0x0000, 0 }, /* R243 */ - { 0x0000, 0x0000, 0 }, /* R244 */ - { 0x0000, 0x0000, 0 }, /* R245 */ - { 0x0000, 0x0000, 0 }, /* R246 */ - { 0x0001, 0x0001, 0 }, /* R247 - FLL NCO Test 0 */ - { 0x003F, 0x003F, 0 }, /* R248 - FLL NCO Test 1 */ -}; +static bool wm8904_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8904_SW_RESET_AND_ID: + case WM8904_REVISION: + case WM8904_DC_SERVO_1: + case WM8904_DC_SERVO_6: + case WM8904_DC_SERVO_7: + case WM8904_DC_SERVO_8: + case WM8904_DC_SERVO_9: + case WM8904_DC_SERVO_READBACK_0: + case WM8904_INTERRUPT_STATUS: + return true; + default: + return false; + } +} -static int wm8904_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm8904_readable_register(struct device *dev, unsigned int reg) { - return wm8904_access[reg].vol; + switch (reg) { + case WM8904_SW_RESET_AND_ID: + case WM8904_REVISION: + case WM8904_BIAS_CONTROL_0: + case WM8904_VMID_CONTROL_0: + case WM8904_MIC_BIAS_CONTROL_0: + case WM8904_MIC_BIAS_CONTROL_1: + case WM8904_ANALOGUE_DAC_0: + case WM8904_MIC_FILTER_CONTROL: + case WM8904_ANALOGUE_ADC_0: + case WM8904_POWER_MANAGEMENT_0: + case WM8904_POWER_MANAGEMENT_2: + case WM8904_POWER_MANAGEMENT_3: + case WM8904_POWER_MANAGEMENT_6: + case WM8904_CLOCK_RATES_0: + case WM8904_CLOCK_RATES_1: + case WM8904_CLOCK_RATES_2: + case WM8904_AUDIO_INTERFACE_0: + case WM8904_AUDIO_INTERFACE_1: + case WM8904_AUDIO_INTERFACE_2: + case WM8904_AUDIO_INTERFACE_3: + case WM8904_DAC_DIGITAL_VOLUME_LEFT: + case WM8904_DAC_DIGITAL_VOLUME_RIGHT: + case WM8904_DAC_DIGITAL_0: + case WM8904_DAC_DIGITAL_1: + case WM8904_ADC_DIGITAL_VOLUME_LEFT: + case WM8904_ADC_DIGITAL_VOLUME_RIGHT: + case WM8904_ADC_DIGITAL_0: + case WM8904_DIGITAL_MICROPHONE_0: + case WM8904_DRC_0: + case WM8904_DRC_1: + case WM8904_DRC_2: + case WM8904_DRC_3: + case WM8904_ANALOGUE_LEFT_INPUT_0: + case WM8904_ANALOGUE_RIGHT_INPUT_0: + case WM8904_ANALOGUE_LEFT_INPUT_1: + case WM8904_ANALOGUE_RIGHT_INPUT_1: + case WM8904_ANALOGUE_OUT1_LEFT: + case WM8904_ANALOGUE_OUT1_RIGHT: + case WM8904_ANALOGUE_OUT2_LEFT: + case WM8904_ANALOGUE_OUT2_RIGHT: + case WM8904_ANALOGUE_OUT12_ZC: + case WM8904_DC_SERVO_0: + case WM8904_DC_SERVO_1: + case WM8904_DC_SERVO_2: + case WM8904_DC_SERVO_4: + case WM8904_DC_SERVO_5: + case WM8904_DC_SERVO_6: + case WM8904_DC_SERVO_7: + case WM8904_DC_SERVO_8: + case WM8904_DC_SERVO_9: + case WM8904_DC_SERVO_READBACK_0: + case WM8904_ANALOGUE_HP_0: + case WM8904_ANALOGUE_LINEOUT_0: + case WM8904_CHARGE_PUMP_0: + case WM8904_CLASS_W_0: + case WM8904_WRITE_SEQUENCER_0: + case WM8904_WRITE_SEQUENCER_1: + case WM8904_WRITE_SEQUENCER_2: + case WM8904_WRITE_SEQUENCER_3: + case WM8904_WRITE_SEQUENCER_4: + case WM8904_FLL_CONTROL_1: + case WM8904_FLL_CONTROL_2: + case WM8904_FLL_CONTROL_3: + case WM8904_FLL_CONTROL_4: + case WM8904_FLL_CONTROL_5: + case WM8904_GPIO_CONTROL_1: + case WM8904_GPIO_CONTROL_2: + case WM8904_GPIO_CONTROL_3: + case WM8904_GPIO_CONTROL_4: + case WM8904_DIGITAL_PULLS: + case WM8904_INTERRUPT_STATUS: + case WM8904_INTERRUPT_STATUS_MASK: + case WM8904_INTERRUPT_POLARITY: + case WM8904_INTERRUPT_DEBOUNCE: + case WM8904_EQ1: + case WM8904_EQ2: + case WM8904_EQ3: + case WM8904_EQ4: + case WM8904_EQ5: + case WM8904_EQ6: + case WM8904_EQ7: + case WM8904_EQ8: + case WM8904_EQ9: + case WM8904_EQ10: + case WM8904_EQ11: + case WM8904_EQ12: + case WM8904_EQ13: + case WM8904_EQ14: + case WM8904_EQ15: + case WM8904_EQ16: + case WM8904_EQ17: + case WM8904_EQ18: + case WM8904_EQ19: + case WM8904_EQ20: + case WM8904_EQ21: + case WM8904_EQ22: + case WM8904_EQ23: + case WM8904_EQ24: + case WM8904_CONTROL_INTERFACE_TEST_1: + case WM8904_ANALOGUE_OUTPUT_BIAS_0: + case WM8904_FLL_NCO_TEST_0: + case WM8904_FLL_NCO_TEST_1: + return true; + default: + return true; + } } static int wm8904_reset(struct snd_soc_codec *codec) @@ -2120,7 +1834,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, return ret; } - snd_soc_cache_sync(codec); + regcache_sync(wm8904->regmap); /* Enable bias */ snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, @@ -2346,6 +2060,7 @@ static int wm8904_probe(struct snd_soc_codec *codec) codec->cache_sync = 1; codec->dapm.idle_bias_off = 1; + codec->control_data = wm8904->regmap; switch (wm8904->devtype) { case WM8904: @@ -2359,7 +2074,7 @@ static int wm8904_probe(struct snd_soc_codec *codec) return -EINVAL; } - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -2387,7 +2102,7 @@ static int wm8904_probe(struct snd_soc_codec *codec) dev_err(codec->dev, "Failed to read ID register\n"); goto err_enable; } - if (ret != wm8904_reg[WM8904_SW_RESET_AND_ID]) { + if (ret != 0x8904) { dev_err(codec->dev, "Device is not a WM8904, ID is %x\n", ret); ret = -EINVAL; goto err_enable; @@ -2493,10 +2208,19 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8904 = { .suspend = wm8904_suspend, .resume = wm8904_resume, .set_bias_level = wm8904_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8904_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8904_reg, - .volatile_register = wm8904_volatile_register, +}; + +static const struct regmap_config wm8904_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM8904_MAX_REGISTER, + .volatile_reg = wm8904_volatile_register, + .readable_reg = wm8904_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8904_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8904_reg_defaults), }; static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, @@ -2510,19 +2234,35 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, if (wm8904 == NULL) return -ENOMEM; + wm8904->regmap = regmap_init_i2c(i2c, &wm8904_regmap); + if (IS_ERR(wm8904->regmap)) { + ret = PTR_ERR(wm8904->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + wm8904->devtype = id->driver_data; i2c_set_clientdata(i2c, wm8904); wm8904->pdata = i2c->dev.platform_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8904, &wm8904_dai, 1); + if (ret != 0) + goto err; + + return 0; +err: + regmap_exit(wm8904->regmap); return ret; } static __devexit int wm8904_i2c_remove(struct i2c_client *client) { + struct wm8904_priv *wm8904 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); + regmap_exit(wm8904->regmap); return 0; } -- cgit v1.2.3-18-g5258 From b5531205f51c6f5f073fd0c63cfa9659fa4d912b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 21:15:48 +0000 Subject: ASoC: Make I2C usage unconditional in WM8940 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 14039ea2f3e..ffc123bc943 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -743,7 +743,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = { .volatile_register = wm8940_volatile_register, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8940_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -786,27 +785,22 @@ static struct i2c_driver wm8940_i2c_driver = { .remove = __devexit_p(wm8940_i2c_remove), .id_table = wm8940_i2c_id, }; -#endif static int __init wm8940_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8940_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register wm8940 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8940_modinit); static void __exit wm8940_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8940_i2c_driver); -#endif } module_exit(wm8940_exit); -- cgit v1.2.3-18-g5258 From 42dad0d84a318d5245b8b311644388dae5f521c0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 21:16:59 +0000 Subject: ASoC: Convert WM8940 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index ffc123bc943..ae1933ed3e0 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -749,7 +749,8 @@ static __devinit int wm8940_i2c_probe(struct i2c_client *i2c, struct wm8940_priv *wm8940; int ret; - wm8940 = kzalloc(sizeof(struct wm8940_priv), GFP_KERNEL); + wm8940 = devm_kzalloc(&i2c->dev, sizeof(struct wm8940_priv), + GFP_KERNEL); if (wm8940 == NULL) return -ENOMEM; @@ -758,15 +759,14 @@ static __devinit int wm8940_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8940, &wm8940_dai, 1); - if (ret < 0) - kfree(wm8940); + return ret; } static __devexit int wm8940_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + return 0; } -- cgit v1.2.3-18-g5258 From 1e9c898df0ef659dacbc9ee037f825cc380854cf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 21:21:49 +0000 Subject: ASoC: Make I2C usage unconditional in WM8955 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 924548182d5..adcfdcaa9fb 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -1001,7 +1001,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8955 = { .reg_cache_default = wm8955_reg, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8955_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1044,27 +1043,22 @@ static struct i2c_driver wm8955_i2c_driver = { .remove = __devexit_p(wm8955_i2c_remove), .id_table = wm8955_i2c_id, }; -#endif static int __init wm8955_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8955_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8955 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8955_modinit); static void __exit wm8955_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8955_i2c_driver); -#endif } module_exit(wm8955_exit); -- cgit v1.2.3-18-g5258 From ba5c88d02de255b51d399001115384f8847cb0df Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 21:23:04 +0000 Subject: ASoC: Convert WM8955 to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index adcfdcaa9fb..cc6f6692bf5 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -1007,7 +1007,8 @@ static __devinit int wm8955_i2c_probe(struct i2c_client *i2c, struct wm8955_priv *wm8955; int ret; - wm8955 = kzalloc(sizeof(struct wm8955_priv), GFP_KERNEL); + wm8955 = devm_kzalloc(&i2c->dev, sizeof(struct wm8955_priv), + GFP_KERNEL); if (wm8955 == NULL) return -ENOMEM; @@ -1016,15 +1017,13 @@ static __devinit int wm8955_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8955, &wm8955_dai, 1); - if (ret < 0) - kfree(wm8955); + return ret; } static __devexit int wm8955_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3-18-g5258 From 9887cb9e651da91c5bad2578d71e7ff8410e14b7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 21:39:44 +0000 Subject: ASoC: Use standard register cache sync for WM8955 Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index cc6f6692bf5..559c96b656a 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -795,18 +795,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec, return ret; } - /* Sync back cached values if they're - * different from the hardware default. - */ - for (i = 0; i < codec->driver->reg_cache_size; i++) { - if (i == WM8955_RESET) - continue; - - if (reg_cache[i] == wm8955_reg[i]) - continue; - - snd_soc_write(codec, i, reg_cache[i]); - } + snd_soc_cache_sync(codec); /* Enable VREF and VMID */ snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, -- cgit v1.2.3-18-g5258 From 95860fdf0f565f96fc37561b6794177604f89097 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 21:42:36 +0000 Subject: ASoC: Convert WM8955 to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 198 ++++++++++++++++++++++++++++------------------ 1 file changed, 123 insertions(+), 75 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 559c96b656a..11fb2dd40c5 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -38,7 +39,7 @@ static const char *wm8955_supply_names[WM8955_NUM_SUPPLIES] = { /* codec private data */ struct wm8955_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; unsigned int mclk_rate; @@ -48,69 +49,85 @@ struct wm8955_priv { struct regulator_bulk_data supplies[WM8955_NUM_SUPPLIES]; }; -static const u16 wm8955_reg[WM8955_MAX_REGISTER + 1] = { - 0x0000, /* R0 */ - 0x0000, /* R1 */ - 0x0079, /* R2 - LOUT1 volume */ - 0x0079, /* R3 - ROUT1 volume */ - 0x0000, /* R4 */ - 0x0008, /* R5 - DAC Control */ - 0x0000, /* R6 */ - 0x000A, /* R7 - Audio Interface */ - 0x0000, /* R8 - Sample Rate */ - 0x0000, /* R9 */ - 0x00FF, /* R10 - Left DAC volume */ - 0x00FF, /* R11 - Right DAC volume */ - 0x000F, /* R12 - Bass control */ - 0x000F, /* R13 - Treble control */ - 0x0000, /* R14 */ - 0x0000, /* R15 - Reset */ - 0x0000, /* R16 */ - 0x0000, /* R17 */ - 0x0000, /* R18 */ - 0x0000, /* R19 */ - 0x0000, /* R20 */ - 0x0000, /* R21 */ - 0x0000, /* R22 */ - 0x00C1, /* R23 - Additional control (1) */ - 0x0000, /* R24 - Additional control (2) */ - 0x0000, /* R25 - Power Management (1) */ - 0x0000, /* R26 - Power Management (2) */ - 0x0000, /* R27 - Additional Control (3) */ - 0x0000, /* R28 */ - 0x0000, /* R29 */ - 0x0000, /* R30 */ - 0x0000, /* R31 */ - 0x0000, /* R32 */ - 0x0000, /* R33 */ - 0x0050, /* R34 - Left out Mix (1) */ - 0x0050, /* R35 - Left out Mix (2) */ - 0x0050, /* R36 - Right out Mix (1) */ - 0x0050, /* R37 - Right Out Mix (2) */ - 0x0050, /* R38 - Mono out Mix (1) */ - 0x0050, /* R39 - Mono out Mix (2) */ - 0x0079, /* R40 - LOUT2 volume */ - 0x0079, /* R41 - ROUT2 volume */ - 0x0079, /* R42 - MONOOUT volume */ - 0x0000, /* R43 - Clocking / PLL */ - 0x0103, /* R44 - PLL Control 1 */ - 0x0024, /* R45 - PLL Control 2 */ - 0x01BA, /* R46 - PLL Control 3 */ - 0x0000, /* R47 */ - 0x0000, /* R48 */ - 0x0000, /* R49 */ - 0x0000, /* R50 */ - 0x0000, /* R51 */ - 0x0000, /* R52 */ - 0x0000, /* R53 */ - 0x0000, /* R54 */ - 0x0000, /* R55 */ - 0x0000, /* R56 */ - 0x0000, /* R57 */ - 0x0000, /* R58 */ - 0x0000, /* R59 - PLL Control 4 */ +static const struct reg_default wm8955_reg_defaults[] = { + { 2, 0x0079 }, /* R2 - LOUT1 volume */ + { 3, 0x0079 }, /* R3 - ROUT1 volume */ + { 5, 0x0008 }, /* R5 - DAC Control */ + { 7, 0x000A }, /* R7 - Audio Interface */ + { 8, 0x0000 }, /* R8 - Sample Rate */ + { 10, 0x00FF }, /* R10 - Left DAC volume */ + { 11, 0x00FF }, /* R11 - Right DAC volume */ + { 12, 0x000F }, /* R12 - Bass control */ + { 13, 0x000F }, /* R13 - Treble control */ + { 23, 0x00C1 }, /* R23 - Additional control (1) */ + { 24, 0x0000 }, /* R24 - Additional control (2) */ + { 25, 0x0000 }, /* R25 - Power Management (1) */ + { 26, 0x0000 }, /* R26 - Power Management (2) */ + { 27, 0x0000 }, /* R27 - Additional Control (3) */ + { 34, 0x0050 }, /* R34 - Left out Mix (1) */ + { 35, 0x0050 }, /* R35 - Left out Mix (2) */ + { 36, 0x0050 }, /* R36 - Right out Mix (1) */ + { 37, 0x0050 }, /* R37 - Right Out Mix (2) */ + { 38, 0x0050 }, /* R38 - Mono out Mix (1) */ + { 39, 0x0050 }, /* R39 - Mono out Mix (2) */ + { 40, 0x0079 }, /* R40 - LOUT2 volume */ + { 41, 0x0079 }, /* R41 - ROUT2 volume */ + { 42, 0x0079 }, /* R42 - MONOOUT volume */ + { 43, 0x0000 }, /* R43 - Clocking / PLL */ + { 44, 0x0103 }, /* R44 - PLL Control 1 */ + { 45, 0x0024 }, /* R45 - PLL Control 2 */ + { 46, 0x01BA }, /* R46 - PLL Control 3 */ + { 59, 0x0000 }, /* R59 - PLL Control 4 */ }; +static bool wm8955_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8955_LOUT1_VOLUME: + case WM8955_ROUT1_VOLUME: + case WM8955_DAC_CONTROL: + case WM8955_AUDIO_INTERFACE: + case WM8955_SAMPLE_RATE: + case WM8955_LEFT_DAC_VOLUME: + case WM8955_RIGHT_DAC_VOLUME: + case WM8955_BASS_CONTROL: + case WM8955_TREBLE_CONTROL: + case WM8955_RESET: + case WM8955_ADDITIONAL_CONTROL_1: + case WM8955_ADDITIONAL_CONTROL_2: + case WM8955_POWER_MANAGEMENT_1: + case WM8955_POWER_MANAGEMENT_2: + case WM8955_ADDITIONAL_CONTROL_3: + case WM8955_LEFT_OUT_MIX_1: + case WM8955_LEFT_OUT_MIX_2: + case WM8955_RIGHT_OUT_MIX_1: + case WM8955_RIGHT_OUT_MIX_2: + case WM8955_MONO_OUT_MIX_1: + case WM8955_MONO_OUT_MIX_2: + case WM8955_LOUT2_VOLUME: + case WM8955_ROUT2_VOLUME: + case WM8955_MONOOUT_VOLUME: + case WM8955_CLOCKING_PLL: + case WM8955_PLL_CONTROL_1: + case WM8955_PLL_CONTROL_2: + case WM8955_PLL_CONTROL_3: + case WM8955_PLL_CONTROL_4: + return true; + default: + return false; + } +} + +static bool wm8955_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8955_RESET: + return true; + default: + return false; + } +} + static int wm8955_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, WM8955_RESET, 0); @@ -765,8 +782,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - u16 *reg_cache = codec->reg_cache; - int ret, i; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -795,7 +811,7 @@ static int wm8955_set_bias_level(struct snd_soc_codec *codec, return ret; } - snd_soc_cache_sync(codec); + regcache_sync(wm8955->regmap); /* Enable VREF and VMID */ snd_soc_update_bits(codec, WM8955_POWER_MANAGEMENT_1, @@ -869,8 +885,12 @@ static struct snd_soc_dai_driver wm8955_dai = { #ifdef CONFIG_PM static int wm8955_suspend(struct snd_soc_codec *codec) { + struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); + wm8955_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_mark_dirty(wm8955->regmap); + return 0; } @@ -889,10 +909,11 @@ static int wm8955_probe(struct snd_soc_codec *codec) { struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); struct wm8955_pdata *pdata = dev_get_platdata(codec->dev); - u16 *reg_cache = codec->reg_cache; int ret, i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8955->control_type); + codec->control_data = wm8955->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -947,12 +968,12 @@ static int wm8955_probe(struct snd_soc_codec *codec) /* Set platform data values */ if (pdata) { if (pdata->out2_speaker) - reg_cache[WM8955_ADDITIONAL_CONTROL_2] - |= WM8955_ROUT2INV; + snd_soc_update_bits(codec, WM8955_ADDITIONAL_CONTROL_2, + WM8955_ROUT2INV, WM8955_ROUT2INV); if (pdata->monoin_diff) - reg_cache[WM8955_MONO_OUT_MIX_1] - |= WM8955_DMEN; + snd_soc_update_bits(codec, WM8955_MONO_OUT_MIX_1, + WM8955_DMEN, WM8955_DMEN); } wm8955_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -985,9 +1006,19 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8955 = { .suspend = wm8955_suspend, .resume = wm8955_resume, .set_bias_level = wm8955_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8955_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8955_reg, +}; + +static const struct regmap_config wm8955_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8955_MAX_REGISTER, + .volatile_reg = wm8955_volatile, + .writeable_reg = wm8955_writeable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8955_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8955_reg_defaults), }; static __devinit int wm8955_i2c_probe(struct i2c_client *i2c, @@ -1001,18 +1032,35 @@ static __devinit int wm8955_i2c_probe(struct i2c_client *i2c, if (wm8955 == NULL) return -ENOMEM; + wm8955->regmap = regmap_init_i2c(i2c, &wm8955_regmap); + if (IS_ERR(wm8955->regmap)) { + ret = PTR_ERR(wm8955->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + i2c_set_clientdata(i2c, wm8955); - wm8955->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8955, &wm8955_dai, 1); + if (ret != 0) + goto err; + + return ret; +err: + regmap_exit(wm8955->regmap); return ret; } static __devexit int wm8955_i2c_remove(struct i2c_client *client) { + struct wm8955_priv *wm8955 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + regmap_exit(wm8955->regmap); + return 0; } -- cgit v1.2.3-18-g5258 From 3294c4c603a1c4ce00e5b8495e99dd3ba076f1e3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 29 Dec 2011 21:45:27 +0000 Subject: ASoC: Convert WM8955 to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 25 ++++++++----------------- 1 file changed, 8 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 11fb2dd40c5..61fe97433e7 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -544,7 +544,7 @@ SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("OUT3"), }; -static const struct snd_soc_dapm_route wm8955_intercon[] = { +static const struct snd_soc_dapm_route wm8955_dapm_routes[] = { { "DACL", NULL, "SYSCLK" }, { "DACR", NULL, "SYSCLK" }, @@ -589,21 +589,6 @@ static const struct snd_soc_dapm_route wm8955_intercon[] = { { "OUT3", NULL, "OUT3 PGA" }, }; -static int wm8955_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_add_controls(codec, wm8955_snd_controls, - ARRAY_SIZE(wm8955_snd_controls)); - - snd_soc_dapm_new_controls(dapm, wm8955_dapm_widgets, - ARRAY_SIZE(wm8955_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, wm8955_intercon, - ARRAY_SIZE(wm8955_intercon)); - - return 0; -} - static int wm8955_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -981,7 +966,6 @@ static int wm8955_probe(struct snd_soc_codec *codec) /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(wm8955->supplies), wm8955->supplies); - wm8955_add_widgets(codec); return 0; err_enable: @@ -1006,6 +990,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8955 = { .suspend = wm8955_suspend, .resume = wm8955_resume, .set_bias_level = wm8955_set_bias_level, + + .controls = wm8955_snd_controls, + .num_controls = ARRAY_SIZE(wm8955_snd_controls), + .dapm_widgets = wm8955_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8955_dapm_widgets), + .dapm_routes = wm8955_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8955_dapm_routes), }; static const struct regmap_config wm8955_regmap = { -- cgit v1.2.3-18-g5258 From 6296914ccefe6efefee811436dd7cfad6545f2eb Mon Sep 17 00:00:00 2001 From: Joachim Eastwood Date: Sun, 1 Jan 2012 02:14:24 +0100 Subject: ASoC: use proper defines for stream directions in pcm engines Signed-off-by: Joachim Eastwood Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 10 +++++----- sound/soc/fsl/mpc5200_dma.c | 12 ++++++------ sound/soc/s6000/s6000-pcm.c | 5 +++-- 3 files changed, 14 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 4f59bbaba48..96bb92dd174 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -311,23 +311,23 @@ static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) * should allocate a DMA buffer only for the streams that are valid. */ - if (pcm->streams[0].substream) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[0].substream->dma_buffer); + &pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer); if (ret) { dev_err(card->dev, "can't alloc playback dma buffer\n"); return ret; } } - if (pcm->streams[1].substream) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[1].substream->dma_buffer); + &pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->dma_buffer); if (ret) { dev_err(card->dev, "can't alloc capture dma buffer\n"); - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + snd_dma_free_pages(&pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer); return ret; } } diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index e7803d34c42..2112e224a4a 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -316,16 +316,16 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; - if (pcm->streams[0].substream) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, - size, &pcm->streams[0].substream->dma_buffer); + size, &pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer); if (rc) goto playback_alloc_err; } - if (pcm->streams[1].substream) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, - size, &pcm->streams[1].substream->dma_buffer); + size, &pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->dma_buffer); if (rc) goto capture_alloc_err; } @@ -336,8 +336,8 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) return 0; capture_alloc_err: - if (pcm->streams[0].substream) - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) + snd_dma_free_pages(&pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->dma_buffer); playback_alloc_err: dev_err(card->dev, "Cannot allocate buffer(s)\n"); diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 43c014f362f..716da861c62 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -435,7 +435,8 @@ static void s6000_pcm_free(struct snd_pcm *pcm) { struct snd_soc_pcm_runtime *runtime = pcm->private_data; struct s6000_pcm_dma_params *params = - snd_soc_dai_get_dma_data(runtime->cpu_dai, pcm->streams[0].substream); + snd_soc_dai_get_dma_data(runtime->cpu_dai, + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); free_irq(params->irq, pcm); snd_pcm_lib_preallocate_free_for_all(pcm); @@ -451,7 +452,7 @@ static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) int res; params = snd_soc_dai_get_dma_data(runtime->cpu_dai, - pcm->streams[0].substream); + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); if (!card->dev->dma_mask) card->dev->dma_mask = &s6000_pcm_dmamask; -- cgit v1.2.3-18-g5258 From 350e16d5293b54e2ef105ebd777f43dbe5a15ffa Mon Sep 17 00:00:00 2001 From: Joachim Eastwood Date: Sun, 1 Jan 2012 02:43:03 +0100 Subject: ASoC: replace 0xffffffff with DMA_BIT_MASK macro Signed-off-by: Joachim Eastwood Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm.c | 4 ++-- sound/soc/davinci/davinci-pcm.c | 4 ++-- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- sound/soc/fsl/mpc5200_dma.c | 4 ++-- sound/soc/kirkwood/kirkwood-dma.c | 4 ++-- sound/soc/samsung/dma.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- 7 files changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index a21ff459e5d..9b84f985770 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -362,7 +362,7 @@ static struct snd_pcm_ops atmel_pcm_ops = { /*--------------------------------------------------------------------------*\ * ASoC platform driver \*--------------------------------------------------------------------------*/ -static u64 atmel_pcm_dmamask = 0xffffffff; +static u64 atmel_pcm_dmamask = DMA_BIT_MASK(32); static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) { @@ -373,7 +373,7 @@ static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->dma_mask) card->dev->dma_mask = &atmel_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = atmel_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index b26401f87b8..97d77b29896 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -826,7 +826,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) } } -static u64 davinci_pcm_dmamask = 0xffffffff; +static u64 davinci_pcm_dmamask = DMA_BIT_MASK(32); static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) { @@ -837,7 +837,7 @@ static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->dma_mask) card->dev->dma_mask = &davinci_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = davinci_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index de839044987..32adca38b48 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -281,7 +281,7 @@ static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static u64 ep93xx_pcm_dmamask = 0xffffffff; +static u64 ep93xx_pcm_dmamask = DMA_BIT_MASK(32); static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) { @@ -292,7 +292,7 @@ static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->dma_mask) card->dev->dma_mask = &ep93xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = ep93xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 2112e224a4a..33adbf1e40d 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -298,7 +298,7 @@ static struct snd_pcm_ops psc_dma_ops = { .hw_params = psc_dma_hw_params, }; -static u64 psc_dma_dmamask = 0xffffffff; +static u64 psc_dma_dmamask = DMA_BIT_MASK(32); static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) { struct snd_card *card = rtd->card->snd_card; @@ -314,7 +314,7 @@ static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->dma_mask) card->dev->dma_mask = &psc_dma_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { rc = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, pcm->card->dev, diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index d0385402712..b9f16598324 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -55,7 +55,7 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = { .fifo_size = 0, }; -static u64 kirkwood_dma_dmamask = 0xFFFFFFFFUL; +static u64 kirkwood_dma_dmamask = DMA_BIT_MASK(32); static irqreturn_t kirkwood_dma_irq(int irq, void *dev_id) { @@ -324,7 +324,7 @@ static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->dma_mask) card->dev->dma_mask = &kirkwood_dma_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = kirkwood_dma_preallocate_dma_buffer(pcm, diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index e4ba17ce6b3..ddc6cde14e2 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -411,7 +411,7 @@ static int dma_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->dma_mask) card->dev->dma_mask = &dma_mask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = preallocate_dma_buffer(pcm, diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index c22431516ab..8b4457137c7 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -336,7 +336,7 @@ static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!card->dev->dma_mask) card->dev->dma_mask = &tegra_dma_mask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = 0xffffffff; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = tegra_pcm_preallocate_dma_buffer(pcm, -- cgit v1.2.3-18-g5258 From 3b09bb820dda209d5c81e329420ccf36dd30c8eb Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 9 Jan 2012 12:09:29 +0000 Subject: ASoC: core - Improve card registration error messaging for large DAI links. Print out the offending DAI link entry when a naming error occurs. Makes thing easier to debug for machines with a large number of DAI links. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b5ecf6d2321..41c8e45a23e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2864,7 +2864,8 @@ int snd_soc_register_card(struct snd_soc_card *card) */ if (!!link->codec_name == !!link->codec_of_node) { dev_err(card->dev, - "Neither/both codec name/of_node are set\n"); + "Neither/both codec name/of_node are set for %s\n", + link->name); return -EINVAL; } @@ -2874,7 +2875,7 @@ int snd_soc_register_card(struct snd_soc_card *card) */ if (link->platform_name && link->platform_of_node) { dev_err(card->dev, - "Both platform name/of_node are set\n"); + "Both platform name/of_node are set for %s\n", link->name); return -EINVAL; } @@ -2884,7 +2885,8 @@ int snd_soc_register_card(struct snd_soc_card *card) */ if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { dev_err(card->dev, - "Neither/both cpu_dai name/of_node are set\n"); + "Neither/both cpu_dai name/of_node are set for %s\n", + link->name); return -EINVAL; } } -- cgit v1.2.3-18-g5258 From 9b85fc90634972634a229aaa1c94f8c9a50fddbc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 7 Jan 2012 18:02:57 -0800 Subject: ASoC: Optimise performance of WM8904 ADC 128fs OSR mode Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 31 ++++++++++++++++++++++++++++++- sound/soc/codecs/wm8904.h | 11 +++++++++++ 2 files changed, 41 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f2a740de3ad..14afc119334 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -304,6 +304,7 @@ static bool wm8904_readable_register(struct device *dev, unsigned int reg) case WM8904_EQ23: case WM8904_EQ24: case WM8904_CONTROL_INTERFACE_TEST_1: + case WM8904_ADC_TEST_0: case WM8904_ANALOGUE_OUTPUT_BIAS_0: case WM8904_FLL_NCO_TEST_0: case WM8904_FLL_NCO_TEST_1: @@ -569,6 +570,29 @@ static const char *hpf_mode_text[] = { static const struct soc_enum hpf_mode = SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text); +static int wm8904_adc_osr_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int val; + int ret; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + if (ucontrol->value.integer.value[0]) + val = 0; + else + val = WM8904_ADC_128_OSR_TST_MODE | WM8904_ADC_BIASX1P5; + + snd_soc_update_bits(codec, WM8904_ADC_TEST_0, + WM8904_ADC_128_OSR_TST_MODE | WM8904_ADC_BIASX1P5, + val); + + return ret; +} + static const struct snd_kcontrol_new wm8904_adc_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8904_ADC_DIGITAL_VOLUME_LEFT, WM8904_ADC_DIGITAL_VOLUME_RIGHT, 1, 119, 0, digital_tlv), @@ -585,7 +609,12 @@ SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0, SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0), SOC_ENUM("High Pass Filter Mode", hpf_mode), -SOC_SINGLE("ADC 128x OSR Switch", WM8904_ANALOGUE_ADC_0, 0, 1, 0), +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC 128x OSR Switch", + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, + .put = wm8904_adc_osr_put, + .private_value = SOC_SINGLE_VALUE(WM8904_ANALOGUE_ADC_0, 0, 1, 0), +}, }; static const char *drc_path_text[] = { diff --git a/sound/soc/codecs/wm8904.h b/sound/soc/codecs/wm8904.h index 9e8c84188ba..c29a0e8131c 100644 --- a/sound/soc/codecs/wm8904.h +++ b/sound/soc/codecs/wm8904.h @@ -123,6 +123,7 @@ #define WM8904_EQ23 0x9C #define WM8904_EQ24 0x9D #define WM8904_CONTROL_INTERFACE_TEST_1 0xA1 +#define WM8904_ADC_TEST_0 0xC6 #define WM8904_ANALOGUE_OUTPUT_BIAS_0 0xCC #define WM8904_FLL_NCO_TEST_0 0xF7 #define WM8904_FLL_NCO_TEST_1 0xF8 @@ -1556,6 +1557,16 @@ #define WM8904_USER_KEY_SHIFT 1 /* USER_KEY */ #define WM8904_USER_KEY_WIDTH 1 /* USER_KEY */ +/* + * R198 (0xC6) - ADC Test 0 + */ +#define WM8904_ADC_128_OSR_TST_MODE 0x0004 /* ADC_128_OSR_TST_MODE */ +#define WM8904_ADC_128_OSR_TST_MODE_SHIFT 2 /* ADC_128_OSR_TST_MODE */ +#define WM8904_ADC_128_OSR_TST_MODE_WIDTH 1 /* ADC_128_OSR_TST_MODE */ +#define WM8904_ADC_BIASX1P5 0x0001 /* ADC_BIASX1P5 */ +#define WM8904_ADC_BIASX1P5_SHIFT 0 /* ADC_BIASX1P5 */ +#define WM8904_ADC_BIASX1P5_WIDTH 1 /* ADC_BIASX1P5 */ + /* * R204 (0xCC) - Analogue Output Bias 0 */ -- cgit v1.2.3-18-g5258 From 08656910bb80882aaad739faea6dac3a0818f71c Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 9 Jan 2012 12:10:16 +0000 Subject: ASoC: twl6040 - add method to query HS DC offset step size in mV Provide a method for mach drivers to query the HS DC offset step size in mV. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 13 +++++++++++++ sound/soc/codecs/twl6040.h | 1 + 2 files changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 5b9c79b6f65..284dd2e9997 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1052,6 +1052,19 @@ int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim) } EXPORT_SYMBOL_GPL(twl6040_get_trim_value); +int twl6040_get_hs_step_size(struct snd_soc_codec *codec) +{ + struct twl6040 *twl6040 = codec->control_data; + + if (twl6040_get_revid(twl6040) < TWL6040_REV_ES1_2) + /* For ES under ES_1.3 HS step is 2 mV */ + return 2; + else + /* For ES_1.3 HS step is 1 mV */ + return 1; +} +EXPORT_SYMBOL_GPL(twl6040_get_hs_step_size); + static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* Capture gains */ SOC_DOUBLE_TLV("Capture Preamplifier Volume", diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index ef273f1fac2..0611406ca7c 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -39,5 +39,6 @@ void twl6040_hs_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, int report); int twl6040_get_clk_id(struct snd_soc_codec *codec); int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim); +int twl6040_get_hs_step_size(struct snd_soc_codec *codec); #endif /* End of __TWL6040_H__ */ -- cgit v1.2.3-18-g5258 From 7aca69f9fe8f04ca37a01e2540960c53b24e3223 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 9 Jan 2012 12:36:24 +0000 Subject: ASoC: utils - Add support for a dummy codec driver. This is useful to create dummy codec devices where we need to have some DAI links without a real Codec. e.g. could be used to represent dumb FM, MODEM, etc This is also used by dynamic PCM for DAI links that have no codec. Signed-off-by: Liam Girdwood [Fixed the indentation -- broonie] Signed-off-by: Mark Brown --- sound/soc/soc-utils.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 4220bb0f273..60053709e41 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -89,14 +89,32 @@ static struct snd_soc_platform_driver dummy_platform = { .ops = &dummy_dma_ops, }; +static struct snd_soc_codec_driver dummy_codec; +static struct snd_soc_dai_driver dummy_dai = { + .name = "snd-soc-dummy-dai", +}; + static __devinit int snd_soc_dummy_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &dummy_platform); + int ret; + + ret = snd_soc_register_codec(&pdev->dev, &dummy_codec, &dummy_dai, 1); + if (ret < 0) + return ret; + + ret = snd_soc_register_platform(&pdev->dev, &dummy_platform); + if (ret < 0) { + snd_soc_unregister_codec(&pdev->dev); + return ret; + } + + return ret; } static __devexit int snd_soc_dummy_remove(struct platform_device *pdev) { snd_soc_unregister_platform(&pdev->dev); + snd_soc_unregister_codec(&pdev->dev); return 0; } -- cgit v1.2.3-18-g5258 From 2e932f29409ca6129578b10fa61b7cb9937b9f54 Mon Sep 17 00:00:00 2001 From: Manjunath Hadli Date: Tue, 10 Jan 2012 16:57:43 +0530 Subject: ASoC: CQ93VC: remove machine specific header file inclusion from codec driver remove unnecessary inclusion of machine specific header file mach/dm365.h from cq93vc.c voice codec driver which comes in the way of platform code consolidation. Signed-off-by: Manjunath Hadli Signed-off-by: Mark Brown --- sound/soc/codecs/cq93vc.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 4854b472d5f..06d2ea18a54 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -38,8 +38,6 @@ #include #include -#include - static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, unsigned int reg) { -- cgit v1.2.3-18-g5258 From 291d64be3137b9bb709ce3bc72dbe2ca6b647466 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 10 Jan 2012 10:53:49 -0800 Subject: ASoC: Make WM8962 I2C usage unconditional We only support I2C so no need to ifdef. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 296de4e30d2..63e908ccbf8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -4155,7 +4155,6 @@ static const struct regmap_config wm8962_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -4284,27 +4283,16 @@ static struct i2c_driver wm8962_i2c_driver = { .remove = __devexit_p(wm8962_i2c_remove), .id_table = wm8962_i2c_id, }; -#endif static int __init wm8962_modinit(void) { - int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&wm8962_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8962 I2C driver: %d\n", - ret); - } -#endif - return 0; + return i2c_add_driver(&wm8962_i2c_driver); } module_init(wm8962_modinit); static void __exit wm8962_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8962_i2c_driver); -#endif } module_exit(wm8962_exit); -- cgit v1.2.3-18-g5258 From 44fb864b8f696d3e8328b294e6515ad45aecfeb0 Mon Sep 17 00:00:00 2001 From: Ryan Mallon Date: Wed, 11 Jan 2012 14:14:31 +1100 Subject: ep93xx: Don't use system controller defines in audio drivers Both the Snapper CL15 and EDB93xx audio drivers set the same audio configuration in ep93xx_i2s_acquire. Remove the arguments to ep93xx_i2s_acquire so that the audio drivers no longer need the EP93XX_SYSCON defines exported. Cc: Hartley Sweeten Cc: Mika Westerberg Cc: Liam Girdwood Cc: Mark Brown Signed-off-by: Ryan Mallon Signed-off-by: Mark Brown --- sound/soc/ep93xx/edb93xx.c | 4 +--- sound/soc/ep93xx/snappercl15.c | 4 +--- 2 files changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index bae5cbbbd2b..e01cb02abd3 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -85,9 +85,7 @@ static int __devinit edb93xx_probe(struct platform_device *pdev) struct snd_soc_card *card = &snd_soc_edb93xx; int ret; - ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, - EP93XX_SYSCON_I2SCLKDIV_ORIDE | - EP93XX_SYSCON_I2SCLKDIV_SPOL); + ret = ep93xx_i2s_acquire(); if (ret) return ret; diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index ccae34a3f28..a193cea3cf3 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -103,9 +103,7 @@ static int __devinit snappercl15_probe(struct platform_device *pdev) struct snd_soc_card *card = &snd_soc_snappercl15; int ret; - ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, - EP93XX_SYSCON_I2SCLKDIV_ORIDE | - EP93XX_SYSCON_I2SCLKDIV_SPOL); + ret = ep93xx_i2s_acquire(); if (ret) return ret; -- cgit v1.2.3-18-g5258 From a6b44f1636f244c97eacb43720414ff356e17c6e Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Wed, 11 Jan 2012 13:21:05 +0100 Subject: ASoC: Route Mic Bias in Visstrim_M10 board. Visstrim_M10 board uses an external microphone that can be enabled/disabled by the user Signed-off-by: Javier Martin Signed-off-by: Mark Brown --- sound/soc/imx/mx27vis-aic32x4.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 3c2eed9094d..d37e23cfc94 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -74,6 +74,24 @@ static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { .hw_params = mx27vis_aic32x4_hw_params, }; +static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = { + SOC_DAPM_PIN_SWITCH("External Mic"), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_MIC("External Mic", NULL), +}; + +static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { + {"Mic Bias", NULL, "External Mic"}, + {"IN1_R", NULL, "Mic Bias"}, + {"IN2_R", NULL, "Mic Bias"}, + {"IN3_R", NULL, "Mic Bias"}, + {"IN1_L", NULL, "Mic Bias"}, + {"IN2_L", NULL, "Mic Bias"}, + {"IN3_L", NULL, "Mic Bias"}, +}; + static struct snd_soc_dai_link mx27vis_aic32x4_dai = { .name = "tlv320aic32x4", .stream_name = "TLV320AIC32X4", @@ -89,6 +107,12 @@ static struct snd_soc_card mx27vis_aic32x4 = { .owner = THIS_MODULE, .dai_link = &mx27vis_aic32x4_dai, .num_links = 1, + .controls = mx27vis_aic32x4_controls, + .num_controls = ARRAY_SIZE(mx27vis_aic32x4_controls), + .dapm_widgets = aic32x4_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), + .dapm_routes = aic32x4_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; static struct platform_device *mx27vis_aic32x4_snd_device; -- cgit v1.2.3-18-g5258 From cef6d1d450ba217dc173a83a50d12de9aaa32bb6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jan 2012 20:13:19 -0800 Subject: ASoC: Convert WM8962 register access map to modern style Much more compact, both in terms of source and especially in terms of RAM used at runtime. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 1803 ++++++++++++++++----------------------------- 1 file changed, 648 insertions(+), 1155 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 63e908ccbf8..a20e2b7ab26 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -797,1167 +797,660 @@ static struct reg_default wm8962_reg[] = { { 21139, 0x8580 }, /* R21139 - VSS_XTS32_0 */ }; -static const struct wm8962_reg_access { - u16 read; - u16 write; - u16 vol; -} wm8962_reg_access[WM8962_MAX_REGISTER + 1] = { - [0] = { 0x00FF, 0x01FF, 0x0000 }, /* R0 - Left Input volume */ - [1] = { 0xFEFF, 0x01FF, 0x0000 }, /* R1 - Right Input volume */ - [2] = { 0x00FF, 0x01FF, 0x0000 }, /* R2 - HPOUTL volume */ - [3] = { 0x00FF, 0x01FF, 0x0000 }, /* R3 - HPOUTR volume */ - [4] = { 0x07FE, 0x07FE, 0xFFFF }, /* R4 - Clocking1 */ - [5] = { 0x007F, 0x007F, 0x0000 }, /* R5 - ADC & DAC Control 1 */ - [6] = { 0x37ED, 0x37ED, 0x0000 }, /* R6 - ADC & DAC Control 2 */ - [7] = { 0x1FFF, 0x1FFF, 0x0000 }, /* R7 - Audio Interface 0 */ - [8] = { 0x0FEF, 0x0FEF, 0xFFFF }, /* R8 - Clocking2 */ - [9] = { 0x0B9F, 0x039F, 0x0000 }, /* R9 - Audio Interface 1 */ - [10] = { 0x00FF, 0x01FF, 0x0000 }, /* R10 - Left DAC volume */ - [11] = { 0x00FF, 0x01FF, 0x0000 }, /* R11 - Right DAC volume */ - [14] = { 0x07FF, 0x07FF, 0x0000 }, /* R14 - Audio Interface 2 */ - [15] = { 0xFFFF, 0xFFFF, 0xFFFF }, /* R15 - Software Reset */ - [17] = { 0x07FF, 0x07FF, 0x0000 }, /* R17 - ALC1 */ - [18] = { 0xF8FF, 0x00FF, 0xFFFF }, /* R18 - ALC2 */ - [19] = { 0x1DFF, 0x1DFF, 0x0000 }, /* R19 - ALC3 */ - [20] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20 - Noise Gate */ - [21] = { 0x00FF, 0x01FF, 0x0000 }, /* R21 - Left ADC volume */ - [22] = { 0x00FF, 0x01FF, 0x0000 }, /* R22 - Right ADC volume */ - [23] = { 0x0161, 0x0161, 0x0000 }, /* R23 - Additional control(1) */ - [24] = { 0x0008, 0x0008, 0x0000 }, /* R24 - Additional control(2) */ - [25] = { 0x07FE, 0x07FE, 0x0000 }, /* R25 - Pwr Mgmt (1) */ - [26] = { 0x01FB, 0x01FB, 0x0000 }, /* R26 - Pwr Mgmt (2) */ - [27] = { 0x0017, 0x0017, 0x0000 }, /* R27 - Additional Control (3) */ - [28] = { 0x001C, 0x001C, 0x0000 }, /* R28 - Anti-pop */ - - [30] = { 0xFFFE, 0xFFFE, 0x0000 }, /* R30 - Clocking 3 */ - [31] = { 0x000F, 0x000F, 0x0000 }, /* R31 - Input mixer control (1) */ - [32] = { 0x01FF, 0x01FF, 0x0000 }, /* R32 - Left input mixer volume */ - [33] = { 0x01FF, 0x01FF, 0x0000 }, /* R33 - Right input mixer volume */ - [34] = { 0x003F, 0x003F, 0x0000 }, /* R34 - Input mixer control (2) */ - [35] = { 0x003F, 0x003F, 0x0000 }, /* R35 - Input bias control */ - [37] = { 0x001F, 0x001F, 0x0000 }, /* R37 - Left input PGA control */ - [38] = { 0x001F, 0x001F, 0x0000 }, /* R38 - Right input PGA control */ - [40] = { 0x00FF, 0x01FF, 0x0000 }, /* R40 - SPKOUTL volume */ - [41] = { 0x00FF, 0x01FF, 0x0000 }, /* R41 - SPKOUTR volume */ - - [47] = { 0x000F, 0x0000, 0xFFFF }, /* R47 - Thermal Shutdown Status */ - [48] = { 0x7EC7, 0x7E07, 0xFFFF }, /* R48 - Additional Control (4) */ - [49] = { 0x00D3, 0x00D7, 0xFFFF }, /* R49 - Class D Control 1 */ - [51] = { 0x0047, 0x0047, 0x0000 }, /* R51 - Class D Control 2 */ - [56] = { 0x001E, 0x001E, 0x0000 }, /* R56 - Clocking 4 */ - [57] = { 0x02FC, 0x02FC, 0x0000 }, /* R57 - DAC DSP Mixing (1) */ - [58] = { 0x00FC, 0x00FC, 0x0000 }, /* R58 - DAC DSP Mixing (2) */ - [60] = { 0x00CC, 0x00CC, 0x0000 }, /* R60 - DC Servo 0 */ - [61] = { 0x00DD, 0x00DD, 0x0000 }, /* R61 - DC Servo 1 */ - [64] = { 0x3F80, 0x3F80, 0x0000 }, /* R64 - DC Servo 4 */ - [66] = { 0x0780, 0x0000, 0xFFFF }, /* R66 - DC Servo 6 */ - [68] = { 0x0007, 0x0007, 0x0000 }, /* R68 - Analogue PGA Bias */ - [69] = { 0x00FF, 0x00FF, 0x0000 }, /* R69 - Analogue HP 0 */ - [71] = { 0x01FF, 0x01FF, 0x0000 }, /* R71 - Analogue HP 2 */ - [72] = { 0x0001, 0x0001, 0x0000 }, /* R72 - Charge Pump 1 */ - [82] = { 0x0001, 0x0001, 0x0000 }, /* R82 - Charge Pump B */ - [87] = { 0x00A0, 0x00A0, 0x0000 }, /* R87 - Write Sequencer Control 1 */ - [90] = { 0x007F, 0x01FF, 0x0000 }, /* R90 - Write Sequencer Control 2 */ - [93] = { 0x03F9, 0x0000, 0x0000 }, /* R93 - Write Sequencer Control 3 */ - [94] = { 0x0070, 0x0070, 0x0000 }, /* R94 - Control Interface */ - [99] = { 0x000F, 0x000F, 0x0000 }, /* R99 - Mixer Enables */ - [100] = { 0x00BF, 0x00BF, 0x0000 }, /* R100 - Headphone Mixer (1) */ - [101] = { 0x00BF, 0x00BF, 0x0000 }, /* R101 - Headphone Mixer (2) */ - [102] = { 0x01FF, 0x01FF, 0x0000 }, /* R102 - Headphone Mixer (3) */ - [103] = { 0x01FF, 0x01FF, 0x0000 }, /* R103 - Headphone Mixer (4) */ - [105] = { 0x00BF, 0x00BF, 0x0000 }, /* R105 - Speaker Mixer (1) */ - [106] = { 0x00BF, 0x00BF, 0x0000 }, /* R106 - Speaker Mixer (2) */ - [107] = { 0x01FF, 0x01FF, 0x0000 }, /* R107 - Speaker Mixer (3) */ - [108] = { 0x01FF, 0x01FF, 0x0000 }, /* R108 - Speaker Mixer (4) */ - [109] = { 0x00F0, 0x00F0, 0x0000 }, /* R109 - Speaker Mixer (5) */ - [110] = { 0x00F7, 0x00F7, 0x0000 }, /* R110 - Beep Generator (1) */ - [115] = { 0x001F, 0x001F, 0x0000 }, /* R115 - Oscillator Trim (3) */ - [116] = { 0x001F, 0x001F, 0x0000 }, /* R116 - Oscillator Trim (4) */ - [119] = { 0x00FF, 0x00FF, 0x0000 }, /* R119 - Oscillator Trim (7) */ - [124] = { 0x0079, 0x0079, 0x0000 }, /* R124 - Analogue Clocking1 */ - [125] = { 0x00DF, 0x00DF, 0x0000 }, /* R125 - Analogue Clocking2 */ - [126] = { 0x000D, 0x000D, 0x0000 }, /* R126 - Analogue Clocking3 */ - [127] = { 0x0000, 0xFFFF, 0x0000 }, /* R127 - PLL Software Reset */ - [129] = { 0x00B0, 0x00B0, 0x0000 }, /* R129 - PLL2 */ - [131] = { 0x0003, 0x0003, 0x0000 }, /* R131 - PLL 4 */ - [136] = { 0x005F, 0x005F, 0x0000 }, /* R136 - PLL 9 */ - [137] = { 0x00FF, 0x00FF, 0x0000 }, /* R137 - PLL 10 */ - [138] = { 0x00FF, 0x00FF, 0x0000 }, /* R138 - PLL 11 */ - [139] = { 0x00FF, 0x00FF, 0x0000 }, /* R139 - PLL 12 */ - [140] = { 0x005F, 0x005F, 0x0000 }, /* R140 - PLL 13 */ - [141] = { 0x00FF, 0x00FF, 0x0000 }, /* R141 - PLL 14 */ - [142] = { 0x00FF, 0x00FF, 0x0000 }, /* R142 - PLL 15 */ - [143] = { 0x00FF, 0x00FF, 0x0000 }, /* R143 - PLL 16 */ - [155] = { 0x0067, 0x0067, 0x0000 }, /* R155 - FLL Control (1) */ - [156] = { 0x01FB, 0x01FB, 0x0000 }, /* R156 - FLL Control (2) */ - [157] = { 0x0007, 0x0007, 0x0000 }, /* R157 - FLL Control (3) */ - [159] = { 0x007F, 0x007F, 0x0000 }, /* R159 - FLL Control (5) */ - [160] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R160 - FLL Control (6) */ - [161] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R161 - FLL Control (7) */ - [162] = { 0x03FF, 0x03FF, 0x0000 }, /* R162 - FLL Control (8) */ - [252] = { 0x0005, 0x0005, 0x0000 }, /* R252 - General test 1 */ - [256] = { 0x000F, 0x000F, 0x0000 }, /* R256 - DF1 */ - [257] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R257 - DF2 */ - [258] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R258 - DF3 */ - [259] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R259 - DF4 */ - [260] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R260 - DF5 */ - [261] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R261 - DF6 */ - [262] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R262 - DF7 */ - [264] = { 0x0003, 0x0003, 0x0000 }, /* R264 - LHPF1 */ - [265] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R265 - LHPF2 */ - [268] = { 0x0077, 0x0077, 0x0000 }, /* R268 - THREED1 */ - [269] = { 0xFFFC, 0xFFFC, 0x0000 }, /* R269 - THREED2 */ - [270] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R270 - THREED3 */ - [271] = { 0xFFFC, 0xFFFC, 0x0000 }, /* R271 - THREED4 */ - [276] = { 0x7FFF, 0x7FFF, 0x0000 }, /* R276 - DRC 1 */ - [277] = { 0x1FFF, 0x1FFF, 0x0000 }, /* R277 - DRC 2 */ - [278] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R278 - DRC 3 */ - [279] = { 0x07FF, 0x07FF, 0x0000 }, /* R279 - DRC 4 */ - [280] = { 0x03FF, 0x03FF, 0x0000 }, /* R280 - DRC 5 */ - [285] = { 0x0003, 0x0003, 0x0000 }, /* R285 - Tloopback */ - [335] = { 0x0007, 0x0007, 0x0000 }, /* R335 - EQ1 */ - [336] = { 0xFFFE, 0xFFFE, 0x0000 }, /* R336 - EQ2 */ - [337] = { 0xFFC0, 0xFFC0, 0x0000 }, /* R337 - EQ3 */ - [338] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R338 - EQ4 */ - [339] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R339 - EQ5 */ - [340] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R340 - EQ6 */ - [341] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R341 - EQ7 */ - [342] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R342 - EQ8 */ - [343] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R343 - EQ9 */ - [344] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R344 - EQ10 */ - [345] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R345 - EQ11 */ - [346] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R346 - EQ12 */ - [347] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R347 - EQ13 */ - [348] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R348 - EQ14 */ - [349] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R349 - EQ15 */ - [350] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R350 - EQ16 */ - [351] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R351 - EQ17 */ - [352] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R352 - EQ18 */ - [353] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R353 - EQ19 */ - [354] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R354 - EQ20 */ - [355] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R355 - EQ21 */ - [356] = { 0xFFFE, 0xFFFE, 0x0000 }, /* R356 - EQ22 */ - [357] = { 0xFFC0, 0xFFC0, 0x0000 }, /* R357 - EQ23 */ - [358] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R358 - EQ24 */ - [359] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R359 - EQ25 */ - [360] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R360 - EQ26 */ - [361] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R361 - EQ27 */ - [362] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R362 - EQ28 */ - [363] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R363 - EQ29 */ - [364] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R364 - EQ30 */ - [365] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R365 - EQ31 */ - [366] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R366 - EQ32 */ - [367] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R367 - EQ33 */ - [368] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R368 - EQ34 */ - [369] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R369 - EQ35 */ - [370] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R370 - EQ36 */ - [371] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R371 - EQ37 */ - [372] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R372 - EQ38 */ - [373] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R373 - EQ39 */ - [374] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R374 - EQ40 */ - [375] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R375 - EQ41 */ - [513] = { 0x045F, 0x045F, 0x0000 }, /* R513 - GPIO 2 */ - [514] = { 0x045F, 0x045F, 0x0000 }, /* R514 - GPIO 3 */ - [516] = { 0xE75F, 0xE75F, 0x0000 }, /* R516 - GPIO 5 */ - [517] = { 0xE75F, 0xE75F, 0x0000 }, /* R517 - GPIO 6 */ - [560] = { 0x0030, 0x0030, 0xFFFF }, /* R560 - Interrupt Status 1 */ - [561] = { 0xFFED, 0xFFED, 0xFFFF }, /* R561 - Interrupt Status 2 */ - [568] = { 0x0030, 0x0030, 0x0000 }, /* R568 - Interrupt Status 1 Mask */ - [569] = { 0xFFED, 0xFFED, 0x0000 }, /* R569 - Interrupt Status 2 Mask */ - [576] = { 0x0001, 0x0001, 0x0000 }, /* R576 - Interrupt Control */ - [584] = { 0x002D, 0x002D, 0x0000 }, /* R584 - IRQ Debounce */ - [586] = { 0xC000, 0xC000, 0x0000 }, /* R586 - MICINT Source Pol */ - [768] = { 0x0001, 0x0001, 0x0000 }, /* R768 - DSP2 Power Management */ - [1037] = { 0x0000, 0x003F, 0xFFFF }, /* R1037 - DSP2_ExecControl */ - [4096] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4096 - Write Sequencer 0 */ - [4097] = { 0x00FF, 0x00FF, 0x0000 }, /* R4097 - Write Sequencer 1 */ - [4098] = { 0x070F, 0x070F, 0x0000 }, /* R4098 - Write Sequencer 2 */ - [4099] = { 0x010F, 0x010F, 0x0000 }, /* R4099 - Write Sequencer 3 */ - [4100] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4100 - Write Sequencer 4 */ - [4101] = { 0x00FF, 0x00FF, 0x0000 }, /* R4101 - Write Sequencer 5 */ - [4102] = { 0x070F, 0x070F, 0x0000 }, /* R4102 - Write Sequencer 6 */ - [4103] = { 0x010F, 0x010F, 0x0000 }, /* R4103 - Write Sequencer 7 */ - [4104] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4104 - Write Sequencer 8 */ - [4105] = { 0x00FF, 0x00FF, 0x0000 }, /* R4105 - Write Sequencer 9 */ - [4106] = { 0x070F, 0x070F, 0x0000 }, /* R4106 - Write Sequencer 10 */ - [4107] = { 0x010F, 0x010F, 0x0000 }, /* R4107 - Write Sequencer 11 */ - [4108] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4108 - Write Sequencer 12 */ - [4109] = { 0x00FF, 0x00FF, 0x0000 }, /* R4109 - Write Sequencer 13 */ - [4110] = { 0x070F, 0x070F, 0x0000 }, /* R4110 - Write Sequencer 14 */ - [4111] = { 0x010F, 0x010F, 0x0000 }, /* R4111 - Write Sequencer 15 */ - [4112] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4112 - Write Sequencer 16 */ - [4113] = { 0x00FF, 0x00FF, 0x0000 }, /* R4113 - Write Sequencer 17 */ - [4114] = { 0x070F, 0x070F, 0x0000 }, /* R4114 - Write Sequencer 18 */ - [4115] = { 0x010F, 0x010F, 0x0000 }, /* R4115 - Write Sequencer 19 */ - [4116] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4116 - Write Sequencer 20 */ - [4117] = { 0x00FF, 0x00FF, 0x0000 }, /* R4117 - Write Sequencer 21 */ - [4118] = { 0x070F, 0x070F, 0x0000 }, /* R4118 - Write Sequencer 22 */ - [4119] = { 0x010F, 0x010F, 0x0000 }, /* R4119 - Write Sequencer 23 */ - [4120] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4120 - Write Sequencer 24 */ - [4121] = { 0x00FF, 0x00FF, 0x0000 }, /* R4121 - Write Sequencer 25 */ - [4122] = { 0x070F, 0x070F, 0x0000 }, /* R4122 - Write Sequencer 26 */ - [4123] = { 0x010F, 0x010F, 0x0000 }, /* R4123 - Write Sequencer 27 */ - [4124] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4124 - Write Sequencer 28 */ - [4125] = { 0x00FF, 0x00FF, 0x0000 }, /* R4125 - Write Sequencer 29 */ - [4126] = { 0x070F, 0x070F, 0x0000 }, /* R4126 - Write Sequencer 30 */ - [4127] = { 0x010F, 0x010F, 0x0000 }, /* R4127 - Write Sequencer 31 */ - [4128] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4128 - Write Sequencer 32 */ - [4129] = { 0x00FF, 0x00FF, 0x0000 }, /* R4129 - Write Sequencer 33 */ - [4130] = { 0x070F, 0x070F, 0x0000 }, /* R4130 - Write Sequencer 34 */ - [4131] = { 0x010F, 0x010F, 0x0000 }, /* R4131 - Write Sequencer 35 */ - [4132] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4132 - Write Sequencer 36 */ - [4133] = { 0x00FF, 0x00FF, 0x0000 }, /* R4133 - Write Sequencer 37 */ - [4134] = { 0x070F, 0x070F, 0x0000 }, /* R4134 - Write Sequencer 38 */ - [4135] = { 0x010F, 0x010F, 0x0000 }, /* R4135 - Write Sequencer 39 */ - [4136] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4136 - Write Sequencer 40 */ - [4137] = { 0x00FF, 0x00FF, 0x0000 }, /* R4137 - Write Sequencer 41 */ - [4138] = { 0x070F, 0x070F, 0x0000 }, /* R4138 - Write Sequencer 42 */ - [4139] = { 0x010F, 0x010F, 0x0000 }, /* R4139 - Write Sequencer 43 */ - [4140] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4140 - Write Sequencer 44 */ - [4141] = { 0x00FF, 0x00FF, 0x0000 }, /* R4141 - Write Sequencer 45 */ - [4142] = { 0x070F, 0x070F, 0x0000 }, /* R4142 - Write Sequencer 46 */ - [4143] = { 0x010F, 0x010F, 0x0000 }, /* R4143 - Write Sequencer 47 */ - [4144] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4144 - Write Sequencer 48 */ - [4145] = { 0x00FF, 0x00FF, 0x0000 }, /* R4145 - Write Sequencer 49 */ - [4146] = { 0x070F, 0x070F, 0x0000 }, /* R4146 - Write Sequencer 50 */ - [4147] = { 0x010F, 0x010F, 0x0000 }, /* R4147 - Write Sequencer 51 */ - [4148] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4148 - Write Sequencer 52 */ - [4149] = { 0x00FF, 0x00FF, 0x0000 }, /* R4149 - Write Sequencer 53 */ - [4150] = { 0x070F, 0x070F, 0x0000 }, /* R4150 - Write Sequencer 54 */ - [4151] = { 0x010F, 0x010F, 0x0000 }, /* R4151 - Write Sequencer 55 */ - [4152] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4152 - Write Sequencer 56 */ - [4153] = { 0x00FF, 0x00FF, 0x0000 }, /* R4153 - Write Sequencer 57 */ - [4154] = { 0x070F, 0x070F, 0x0000 }, /* R4154 - Write Sequencer 58 */ - [4155] = { 0x010F, 0x010F, 0x0000 }, /* R4155 - Write Sequencer 59 */ - [4156] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4156 - Write Sequencer 60 */ - [4157] = { 0x00FF, 0x00FF, 0x0000 }, /* R4157 - Write Sequencer 61 */ - [4158] = { 0x070F, 0x070F, 0x0000 }, /* R4158 - Write Sequencer 62 */ - [4159] = { 0x010F, 0x010F, 0x0000 }, /* R4159 - Write Sequencer 63 */ - [4160] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4160 - Write Sequencer 64 */ - [4161] = { 0x00FF, 0x00FF, 0x0000 }, /* R4161 - Write Sequencer 65 */ - [4162] = { 0x070F, 0x070F, 0x0000 }, /* R4162 - Write Sequencer 66 */ - [4163] = { 0x010F, 0x010F, 0x0000 }, /* R4163 - Write Sequencer 67 */ - [4164] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4164 - Write Sequencer 68 */ - [4165] = { 0x00FF, 0x00FF, 0x0000 }, /* R4165 - Write Sequencer 69 */ - [4166] = { 0x070F, 0x070F, 0x0000 }, /* R4166 - Write Sequencer 70 */ - [4167] = { 0x010F, 0x010F, 0x0000 }, /* R4167 - Write Sequencer 71 */ - [4168] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4168 - Write Sequencer 72 */ - [4169] = { 0x00FF, 0x00FF, 0x0000 }, /* R4169 - Write Sequencer 73 */ - [4170] = { 0x070F, 0x070F, 0x0000 }, /* R4170 - Write Sequencer 74 */ - [4171] = { 0x010F, 0x010F, 0x0000 }, /* R4171 - Write Sequencer 75 */ - [4172] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4172 - Write Sequencer 76 */ - [4173] = { 0x00FF, 0x00FF, 0x0000 }, /* R4173 - Write Sequencer 77 */ - [4174] = { 0x070F, 0x070F, 0x0000 }, /* R4174 - Write Sequencer 78 */ - [4175] = { 0x010F, 0x010F, 0x0000 }, /* R4175 - Write Sequencer 79 */ - [4176] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4176 - Write Sequencer 80 */ - [4177] = { 0x00FF, 0x00FF, 0x0000 }, /* R4177 - Write Sequencer 81 */ - [4178] = { 0x070F, 0x070F, 0x0000 }, /* R4178 - Write Sequencer 82 */ - [4179] = { 0x010F, 0x010F, 0x0000 }, /* R4179 - Write Sequencer 83 */ - [4180] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4180 - Write Sequencer 84 */ - [4181] = { 0x00FF, 0x00FF, 0x0000 }, /* R4181 - Write Sequencer 85 */ - [4182] = { 0x070F, 0x070F, 0x0000 }, /* R4182 - Write Sequencer 86 */ - [4183] = { 0x010F, 0x010F, 0x0000 }, /* R4183 - Write Sequencer 87 */ - [4184] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4184 - Write Sequencer 88 */ - [4185] = { 0x00FF, 0x00FF, 0x0000 }, /* R4185 - Write Sequencer 89 */ - [4186] = { 0x070F, 0x070F, 0x0000 }, /* R4186 - Write Sequencer 90 */ - [4187] = { 0x010F, 0x010F, 0x0000 }, /* R4187 - Write Sequencer 91 */ - [4188] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4188 - Write Sequencer 92 */ - [4189] = { 0x00FF, 0x00FF, 0x0000 }, /* R4189 - Write Sequencer 93 */ - [4190] = { 0x070F, 0x070F, 0x0000 }, /* R4190 - Write Sequencer 94 */ - [4191] = { 0x010F, 0x010F, 0x0000 }, /* R4191 - Write Sequencer 95 */ - [4192] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4192 - Write Sequencer 96 */ - [4193] = { 0x00FF, 0x00FF, 0x0000 }, /* R4193 - Write Sequencer 97 */ - [4194] = { 0x070F, 0x070F, 0x0000 }, /* R4194 - Write Sequencer 98 */ - [4195] = { 0x010F, 0x010F, 0x0000 }, /* R4195 - Write Sequencer 99 */ - [4196] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4196 - Write Sequencer 100 */ - [4197] = { 0x00FF, 0x00FF, 0x0000 }, /* R4197 - Write Sequencer 101 */ - [4198] = { 0x070F, 0x070F, 0x0000 }, /* R4198 - Write Sequencer 102 */ - [4199] = { 0x010F, 0x010F, 0x0000 }, /* R4199 - Write Sequencer 103 */ - [4200] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4200 - Write Sequencer 104 */ - [4201] = { 0x00FF, 0x00FF, 0x0000 }, /* R4201 - Write Sequencer 105 */ - [4202] = { 0x070F, 0x070F, 0x0000 }, /* R4202 - Write Sequencer 106 */ - [4203] = { 0x010F, 0x010F, 0x0000 }, /* R4203 - Write Sequencer 107 */ - [4204] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4204 - Write Sequencer 108 */ - [4205] = { 0x00FF, 0x00FF, 0x0000 }, /* R4205 - Write Sequencer 109 */ - [4206] = { 0x070F, 0x070F, 0x0000 }, /* R4206 - Write Sequencer 110 */ - [4207] = { 0x010F, 0x010F, 0x0000 }, /* R4207 - Write Sequencer 111 */ - [4208] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4208 - Write Sequencer 112 */ - [4209] = { 0x00FF, 0x00FF, 0x0000 }, /* R4209 - Write Sequencer 113 */ - [4210] = { 0x070F, 0x070F, 0x0000 }, /* R4210 - Write Sequencer 114 */ - [4211] = { 0x010F, 0x010F, 0x0000 }, /* R4211 - Write Sequencer 115 */ - [4212] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4212 - Write Sequencer 116 */ - [4213] = { 0x00FF, 0x00FF, 0x0000 }, /* R4213 - Write Sequencer 117 */ - [4214] = { 0x070F, 0x070F, 0x0000 }, /* R4214 - Write Sequencer 118 */ - [4215] = { 0x010F, 0x010F, 0x0000 }, /* R4215 - Write Sequencer 119 */ - [4216] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4216 - Write Sequencer 120 */ - [4217] = { 0x00FF, 0x00FF, 0x0000 }, /* R4217 - Write Sequencer 121 */ - [4218] = { 0x070F, 0x070F, 0x0000 }, /* R4218 - Write Sequencer 122 */ - [4219] = { 0x010F, 0x010F, 0x0000 }, /* R4219 - Write Sequencer 123 */ - [4220] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4220 - Write Sequencer 124 */ - [4221] = { 0x00FF, 0x00FF, 0x0000 }, /* R4221 - Write Sequencer 125 */ - [4222] = { 0x070F, 0x070F, 0x0000 }, /* R4222 - Write Sequencer 126 */ - [4223] = { 0x010F, 0x010F, 0x0000 }, /* R4223 - Write Sequencer 127 */ - [4224] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4224 - Write Sequencer 128 */ - [4225] = { 0x00FF, 0x00FF, 0x0000 }, /* R4225 - Write Sequencer 129 */ - [4226] = { 0x070F, 0x070F, 0x0000 }, /* R4226 - Write Sequencer 130 */ - [4227] = { 0x010F, 0x010F, 0x0000 }, /* R4227 - Write Sequencer 131 */ - [4228] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4228 - Write Sequencer 132 */ - [4229] = { 0x00FF, 0x00FF, 0x0000 }, /* R4229 - Write Sequencer 133 */ - [4230] = { 0x070F, 0x070F, 0x0000 }, /* R4230 - Write Sequencer 134 */ - [4231] = { 0x010F, 0x010F, 0x0000 }, /* R4231 - Write Sequencer 135 */ - [4232] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4232 - Write Sequencer 136 */ - [4233] = { 0x00FF, 0x00FF, 0x0000 }, /* R4233 - Write Sequencer 137 */ - [4234] = { 0x070F, 0x070F, 0x0000 }, /* R4234 - Write Sequencer 138 */ - [4235] = { 0x010F, 0x010F, 0x0000 }, /* R4235 - Write Sequencer 139 */ - [4236] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4236 - Write Sequencer 140 */ - [4237] = { 0x00FF, 0x00FF, 0x0000 }, /* R4237 - Write Sequencer 141 */ - [4238] = { 0x070F, 0x070F, 0x0000 }, /* R4238 - Write Sequencer 142 */ - [4239] = { 0x010F, 0x010F, 0x0000 }, /* R4239 - Write Sequencer 143 */ - [4240] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4240 - Write Sequencer 144 */ - [4241] = { 0x00FF, 0x00FF, 0x0000 }, /* R4241 - Write Sequencer 145 */ - [4242] = { 0x070F, 0x070F, 0x0000 }, /* R4242 - Write Sequencer 146 */ - [4243] = { 0x010F, 0x010F, 0x0000 }, /* R4243 - Write Sequencer 147 */ - [4244] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4244 - Write Sequencer 148 */ - [4245] = { 0x00FF, 0x00FF, 0x0000 }, /* R4245 - Write Sequencer 149 */ - [4246] = { 0x070F, 0x070F, 0x0000 }, /* R4246 - Write Sequencer 150 */ - [4247] = { 0x010F, 0x010F, 0x0000 }, /* R4247 - Write Sequencer 151 */ - [4248] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4248 - Write Sequencer 152 */ - [4249] = { 0x00FF, 0x00FF, 0x0000 }, /* R4249 - Write Sequencer 153 */ - [4250] = { 0x070F, 0x070F, 0x0000 }, /* R4250 - Write Sequencer 154 */ - [4251] = { 0x010F, 0x010F, 0x0000 }, /* R4251 - Write Sequencer 155 */ - [4252] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4252 - Write Sequencer 156 */ - [4253] = { 0x00FF, 0x00FF, 0x0000 }, /* R4253 - Write Sequencer 157 */ - [4254] = { 0x070F, 0x070F, 0x0000 }, /* R4254 - Write Sequencer 158 */ - [4255] = { 0x010F, 0x010F, 0x0000 }, /* R4255 - Write Sequencer 159 */ - [4256] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4256 - Write Sequencer 160 */ - [4257] = { 0x00FF, 0x00FF, 0x0000 }, /* R4257 - Write Sequencer 161 */ - [4258] = { 0x070F, 0x070F, 0x0000 }, /* R4258 - Write Sequencer 162 */ - [4259] = { 0x010F, 0x010F, 0x0000 }, /* R4259 - Write Sequencer 163 */ - [4260] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4260 - Write Sequencer 164 */ - [4261] = { 0x00FF, 0x00FF, 0x0000 }, /* R4261 - Write Sequencer 165 */ - [4262] = { 0x070F, 0x070F, 0x0000 }, /* R4262 - Write Sequencer 166 */ - [4263] = { 0x010F, 0x010F, 0x0000 }, /* R4263 - Write Sequencer 167 */ - [4264] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4264 - Write Sequencer 168 */ - [4265] = { 0x00FF, 0x00FF, 0x0000 }, /* R4265 - Write Sequencer 169 */ - [4266] = { 0x070F, 0x070F, 0x0000 }, /* R4266 - Write Sequencer 170 */ - [4267] = { 0x010F, 0x010F, 0x0000 }, /* R4267 - Write Sequencer 171 */ - [4268] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4268 - Write Sequencer 172 */ - [4269] = { 0x00FF, 0x00FF, 0x0000 }, /* R4269 - Write Sequencer 173 */ - [4270] = { 0x070F, 0x070F, 0x0000 }, /* R4270 - Write Sequencer 174 */ - [4271] = { 0x010F, 0x010F, 0x0000 }, /* R4271 - Write Sequencer 175 */ - [4272] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4272 - Write Sequencer 176 */ - [4273] = { 0x00FF, 0x00FF, 0x0000 }, /* R4273 - Write Sequencer 177 */ - [4274] = { 0x070F, 0x070F, 0x0000 }, /* R4274 - Write Sequencer 178 */ - [4275] = { 0x010F, 0x010F, 0x0000 }, /* R4275 - Write Sequencer 179 */ - [4276] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4276 - Write Sequencer 180 */ - [4277] = { 0x00FF, 0x00FF, 0x0000 }, /* R4277 - Write Sequencer 181 */ - [4278] = { 0x070F, 0x070F, 0x0000 }, /* R4278 - Write Sequencer 182 */ - [4279] = { 0x010F, 0x010F, 0x0000 }, /* R4279 - Write Sequencer 183 */ - [4280] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4280 - Write Sequencer 184 */ - [4281] = { 0x00FF, 0x00FF, 0x0000 }, /* R4281 - Write Sequencer 185 */ - [4282] = { 0x070F, 0x070F, 0x0000 }, /* R4282 - Write Sequencer 186 */ - [4283] = { 0x010F, 0x010F, 0x0000 }, /* R4283 - Write Sequencer 187 */ - [4284] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4284 - Write Sequencer 188 */ - [4285] = { 0x00FF, 0x00FF, 0x0000 }, /* R4285 - Write Sequencer 189 */ - [4286] = { 0x070F, 0x070F, 0x0000 }, /* R4286 - Write Sequencer 190 */ - [4287] = { 0x010F, 0x010F, 0x0000 }, /* R4287 - Write Sequencer 191 */ - [4288] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4288 - Write Sequencer 192 */ - [4289] = { 0x00FF, 0x00FF, 0x0000 }, /* R4289 - Write Sequencer 193 */ - [4290] = { 0x070F, 0x070F, 0x0000 }, /* R4290 - Write Sequencer 194 */ - [4291] = { 0x010F, 0x010F, 0x0000 }, /* R4291 - Write Sequencer 195 */ - [4292] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4292 - Write Sequencer 196 */ - [4293] = { 0x00FF, 0x00FF, 0x0000 }, /* R4293 - Write Sequencer 197 */ - [4294] = { 0x070F, 0x070F, 0x0000 }, /* R4294 - Write Sequencer 198 */ - [4295] = { 0x010F, 0x010F, 0x0000 }, /* R4295 - Write Sequencer 199 */ - [4296] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4296 - Write Sequencer 200 */ - [4297] = { 0x00FF, 0x00FF, 0x0000 }, /* R4297 - Write Sequencer 201 */ - [4298] = { 0x070F, 0x070F, 0x0000 }, /* R4298 - Write Sequencer 202 */ - [4299] = { 0x010F, 0x010F, 0x0000 }, /* R4299 - Write Sequencer 203 */ - [4300] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4300 - Write Sequencer 204 */ - [4301] = { 0x00FF, 0x00FF, 0x0000 }, /* R4301 - Write Sequencer 205 */ - [4302] = { 0x070F, 0x070F, 0x0000 }, /* R4302 - Write Sequencer 206 */ - [4303] = { 0x010F, 0x010F, 0x0000 }, /* R4303 - Write Sequencer 207 */ - [4304] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4304 - Write Sequencer 208 */ - [4305] = { 0x00FF, 0x00FF, 0x0000 }, /* R4305 - Write Sequencer 209 */ - [4306] = { 0x070F, 0x070F, 0x0000 }, /* R4306 - Write Sequencer 210 */ - [4307] = { 0x010F, 0x010F, 0x0000 }, /* R4307 - Write Sequencer 211 */ - [4308] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4308 - Write Sequencer 212 */ - [4309] = { 0x00FF, 0x00FF, 0x0000 }, /* R4309 - Write Sequencer 213 */ - [4310] = { 0x070F, 0x070F, 0x0000 }, /* R4310 - Write Sequencer 214 */ - [4311] = { 0x010F, 0x010F, 0x0000 }, /* R4311 - Write Sequencer 215 */ - [4312] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4312 - Write Sequencer 216 */ - [4313] = { 0x00FF, 0x00FF, 0x0000 }, /* R4313 - Write Sequencer 217 */ - [4314] = { 0x070F, 0x070F, 0x0000 }, /* R4314 - Write Sequencer 218 */ - [4315] = { 0x010F, 0x010F, 0x0000 }, /* R4315 - Write Sequencer 219 */ - [4316] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4316 - Write Sequencer 220 */ - [4317] = { 0x00FF, 0x00FF, 0x0000 }, /* R4317 - Write Sequencer 221 */ - [4318] = { 0x070F, 0x070F, 0x0000 }, /* R4318 - Write Sequencer 222 */ - [4319] = { 0x010F, 0x010F, 0x0000 }, /* R4319 - Write Sequencer 223 */ - [4320] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4320 - Write Sequencer 224 */ - [4321] = { 0x00FF, 0x00FF, 0x0000 }, /* R4321 - Write Sequencer 225 */ - [4322] = { 0x070F, 0x070F, 0x0000 }, /* R4322 - Write Sequencer 226 */ - [4323] = { 0x010F, 0x010F, 0x0000 }, /* R4323 - Write Sequencer 227 */ - [4324] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4324 - Write Sequencer 228 */ - [4325] = { 0x00FF, 0x00FF, 0x0000 }, /* R4325 - Write Sequencer 229 */ - [4326] = { 0x070F, 0x070F, 0x0000 }, /* R4326 - Write Sequencer 230 */ - [4327] = { 0x010F, 0x010F, 0x0000 }, /* R4327 - Write Sequencer 231 */ - [4328] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4328 - Write Sequencer 232 */ - [4329] = { 0x00FF, 0x00FF, 0x0000 }, /* R4329 - Write Sequencer 233 */ - [4330] = { 0x070F, 0x070F, 0x0000 }, /* R4330 - Write Sequencer 234 */ - [4331] = { 0x010F, 0x010F, 0x0000 }, /* R4331 - Write Sequencer 235 */ - [4332] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4332 - Write Sequencer 236 */ - [4333] = { 0x00FF, 0x00FF, 0x0000 }, /* R4333 - Write Sequencer 237 */ - [4334] = { 0x070F, 0x070F, 0x0000 }, /* R4334 - Write Sequencer 238 */ - [4335] = { 0x010F, 0x010F, 0x0000 }, /* R4335 - Write Sequencer 239 */ - [4336] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4336 - Write Sequencer 240 */ - [4337] = { 0x00FF, 0x00FF, 0x0000 }, /* R4337 - Write Sequencer 241 */ - [4338] = { 0x070F, 0x070F, 0x0000 }, /* R4338 - Write Sequencer 242 */ - [4339] = { 0x010F, 0x010F, 0x0000 }, /* R4339 - Write Sequencer 243 */ - [4340] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4340 - Write Sequencer 244 */ - [4341] = { 0x00FF, 0x00FF, 0x0000 }, /* R4341 - Write Sequencer 245 */ - [4342] = { 0x070F, 0x070F, 0x0000 }, /* R4342 - Write Sequencer 246 */ - [4343] = { 0x010F, 0x010F, 0x0000 }, /* R4343 - Write Sequencer 247 */ - [4344] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4344 - Write Sequencer 248 */ - [4345] = { 0x00FF, 0x00FF, 0x0000 }, /* R4345 - Write Sequencer 249 */ - [4346] = { 0x070F, 0x070F, 0x0000 }, /* R4346 - Write Sequencer 250 */ - [4347] = { 0x010F, 0x010F, 0x0000 }, /* R4347 - Write Sequencer 251 */ - [4348] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4348 - Write Sequencer 252 */ - [4349] = { 0x00FF, 0x00FF, 0x0000 }, /* R4349 - Write Sequencer 253 */ - [4350] = { 0x070F, 0x070F, 0x0000 }, /* R4350 - Write Sequencer 254 */ - [4351] = { 0x010F, 0x010F, 0x0000 }, /* R4351 - Write Sequencer 255 */ - [4352] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4352 - Write Sequencer 256 */ - [4353] = { 0x00FF, 0x00FF, 0x0000 }, /* R4353 - Write Sequencer 257 */ - [4354] = { 0x070F, 0x070F, 0x0000 }, /* R4354 - Write Sequencer 258 */ - [4355] = { 0x010F, 0x010F, 0x0000 }, /* R4355 - Write Sequencer 259 */ - [4356] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4356 - Write Sequencer 260 */ - [4357] = { 0x00FF, 0x00FF, 0x0000 }, /* R4357 - Write Sequencer 261 */ - [4358] = { 0x070F, 0x070F, 0x0000 }, /* R4358 - Write Sequencer 262 */ - [4359] = { 0x010F, 0x010F, 0x0000 }, /* R4359 - Write Sequencer 263 */ - [4360] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4360 - Write Sequencer 264 */ - [4361] = { 0x00FF, 0x00FF, 0x0000 }, /* R4361 - Write Sequencer 265 */ - [4362] = { 0x070F, 0x070F, 0x0000 }, /* R4362 - Write Sequencer 266 */ - [4363] = { 0x010F, 0x010F, 0x0000 }, /* R4363 - Write Sequencer 267 */ - [4364] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4364 - Write Sequencer 268 */ - [4365] = { 0x00FF, 0x00FF, 0x0000 }, /* R4365 - Write Sequencer 269 */ - [4366] = { 0x070F, 0x070F, 0x0000 }, /* R4366 - Write Sequencer 270 */ - [4367] = { 0x010F, 0x010F, 0x0000 }, /* R4367 - Write Sequencer 271 */ - [4368] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4368 - Write Sequencer 272 */ - [4369] = { 0x00FF, 0x00FF, 0x0000 }, /* R4369 - Write Sequencer 273 */ - [4370] = { 0x070F, 0x070F, 0x0000 }, /* R4370 - Write Sequencer 274 */ - [4371] = { 0x010F, 0x010F, 0x0000 }, /* R4371 - Write Sequencer 275 */ - [4372] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4372 - Write Sequencer 276 */ - [4373] = { 0x00FF, 0x00FF, 0x0000 }, /* R4373 - Write Sequencer 277 */ - [4374] = { 0x070F, 0x070F, 0x0000 }, /* R4374 - Write Sequencer 278 */ - [4375] = { 0x010F, 0x010F, 0x0000 }, /* R4375 - Write Sequencer 279 */ - [4376] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4376 - Write Sequencer 280 */ - [4377] = { 0x00FF, 0x00FF, 0x0000 }, /* R4377 - Write Sequencer 281 */ - [4378] = { 0x070F, 0x070F, 0x0000 }, /* R4378 - Write Sequencer 282 */ - [4379] = { 0x010F, 0x010F, 0x0000 }, /* R4379 - Write Sequencer 283 */ - [4380] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4380 - Write Sequencer 284 */ - [4381] = { 0x00FF, 0x00FF, 0x0000 }, /* R4381 - Write Sequencer 285 */ - [4382] = { 0x070F, 0x070F, 0x0000 }, /* R4382 - Write Sequencer 286 */ - [4383] = { 0x010F, 0x010F, 0x0000 }, /* R4383 - Write Sequencer 287 */ - [4384] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4384 - Write Sequencer 288 */ - [4385] = { 0x00FF, 0x00FF, 0x0000 }, /* R4385 - Write Sequencer 289 */ - [4386] = { 0x070F, 0x070F, 0x0000 }, /* R4386 - Write Sequencer 290 */ - [4387] = { 0x010F, 0x010F, 0x0000 }, /* R4387 - Write Sequencer 291 */ - [4388] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4388 - Write Sequencer 292 */ - [4389] = { 0x00FF, 0x00FF, 0x0000 }, /* R4389 - Write Sequencer 293 */ - [4390] = { 0x070F, 0x070F, 0x0000 }, /* R4390 - Write Sequencer 294 */ - [4391] = { 0x010F, 0x010F, 0x0000 }, /* R4391 - Write Sequencer 295 */ - [4392] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4392 - Write Sequencer 296 */ - [4393] = { 0x00FF, 0x00FF, 0x0000 }, /* R4393 - Write Sequencer 297 */ - [4394] = { 0x070F, 0x070F, 0x0000 }, /* R4394 - Write Sequencer 298 */ - [4395] = { 0x010F, 0x010F, 0x0000 }, /* R4395 - Write Sequencer 299 */ - [4396] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4396 - Write Sequencer 300 */ - [4397] = { 0x00FF, 0x00FF, 0x0000 }, /* R4397 - Write Sequencer 301 */ - [4398] = { 0x070F, 0x070F, 0x0000 }, /* R4398 - Write Sequencer 302 */ - [4399] = { 0x010F, 0x010F, 0x0000 }, /* R4399 - Write Sequencer 303 */ - [4400] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4400 - Write Sequencer 304 */ - [4401] = { 0x00FF, 0x00FF, 0x0000 }, /* R4401 - Write Sequencer 305 */ - [4402] = { 0x070F, 0x070F, 0x0000 }, /* R4402 - Write Sequencer 306 */ - [4403] = { 0x010F, 0x010F, 0x0000 }, /* R4403 - Write Sequencer 307 */ - [4404] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4404 - Write Sequencer 308 */ - [4405] = { 0x00FF, 0x00FF, 0x0000 }, /* R4405 - Write Sequencer 309 */ - [4406] = { 0x070F, 0x070F, 0x0000 }, /* R4406 - Write Sequencer 310 */ - [4407] = { 0x010F, 0x010F, 0x0000 }, /* R4407 - Write Sequencer 311 */ - [4408] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4408 - Write Sequencer 312 */ - [4409] = { 0x00FF, 0x00FF, 0x0000 }, /* R4409 - Write Sequencer 313 */ - [4410] = { 0x070F, 0x070F, 0x0000 }, /* R4410 - Write Sequencer 314 */ - [4411] = { 0x010F, 0x010F, 0x0000 }, /* R4411 - Write Sequencer 315 */ - [4412] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4412 - Write Sequencer 316 */ - [4413] = { 0x00FF, 0x00FF, 0x0000 }, /* R4413 - Write Sequencer 317 */ - [4414] = { 0x070F, 0x070F, 0x0000 }, /* R4414 - Write Sequencer 318 */ - [4415] = { 0x010F, 0x010F, 0x0000 }, /* R4415 - Write Sequencer 319 */ - [4416] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4416 - Write Sequencer 320 */ - [4417] = { 0x00FF, 0x00FF, 0x0000 }, /* R4417 - Write Sequencer 321 */ - [4418] = { 0x070F, 0x070F, 0x0000 }, /* R4418 - Write Sequencer 322 */ - [4419] = { 0x010F, 0x010F, 0x0000 }, /* R4419 - Write Sequencer 323 */ - [4420] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4420 - Write Sequencer 324 */ - [4421] = { 0x00FF, 0x00FF, 0x0000 }, /* R4421 - Write Sequencer 325 */ - [4422] = { 0x070F, 0x070F, 0x0000 }, /* R4422 - Write Sequencer 326 */ - [4423] = { 0x010F, 0x010F, 0x0000 }, /* R4423 - Write Sequencer 327 */ - [4424] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4424 - Write Sequencer 328 */ - [4425] = { 0x00FF, 0x00FF, 0x0000 }, /* R4425 - Write Sequencer 329 */ - [4426] = { 0x070F, 0x070F, 0x0000 }, /* R4426 - Write Sequencer 330 */ - [4427] = { 0x010F, 0x010F, 0x0000 }, /* R4427 - Write Sequencer 331 */ - [4428] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4428 - Write Sequencer 332 */ - [4429] = { 0x00FF, 0x00FF, 0x0000 }, /* R4429 - Write Sequencer 333 */ - [4430] = { 0x070F, 0x070F, 0x0000 }, /* R4430 - Write Sequencer 334 */ - [4431] = { 0x010F, 0x010F, 0x0000 }, /* R4431 - Write Sequencer 335 */ - [4432] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4432 - Write Sequencer 336 */ - [4433] = { 0x00FF, 0x00FF, 0x0000 }, /* R4433 - Write Sequencer 337 */ - [4434] = { 0x070F, 0x070F, 0x0000 }, /* R4434 - Write Sequencer 338 */ - [4435] = { 0x010F, 0x010F, 0x0000 }, /* R4435 - Write Sequencer 339 */ - [4436] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4436 - Write Sequencer 340 */ - [4437] = { 0x00FF, 0x00FF, 0x0000 }, /* R4437 - Write Sequencer 341 */ - [4438] = { 0x070F, 0x070F, 0x0000 }, /* R4438 - Write Sequencer 342 */ - [4439] = { 0x010F, 0x010F, 0x0000 }, /* R4439 - Write Sequencer 343 */ - [4440] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4440 - Write Sequencer 344 */ - [4441] = { 0x00FF, 0x00FF, 0x0000 }, /* R4441 - Write Sequencer 345 */ - [4442] = { 0x070F, 0x070F, 0x0000 }, /* R4442 - Write Sequencer 346 */ - [4443] = { 0x010F, 0x010F, 0x0000 }, /* R4443 - Write Sequencer 347 */ - [4444] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4444 - Write Sequencer 348 */ - [4445] = { 0x00FF, 0x00FF, 0x0000 }, /* R4445 - Write Sequencer 349 */ - [4446] = { 0x070F, 0x070F, 0x0000 }, /* R4446 - Write Sequencer 350 */ - [4447] = { 0x010F, 0x010F, 0x0000 }, /* R4447 - Write Sequencer 351 */ - [4448] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4448 - Write Sequencer 352 */ - [4449] = { 0x00FF, 0x00FF, 0x0000 }, /* R4449 - Write Sequencer 353 */ - [4450] = { 0x070F, 0x070F, 0x0000 }, /* R4450 - Write Sequencer 354 */ - [4451] = { 0x010F, 0x010F, 0x0000 }, /* R4451 - Write Sequencer 355 */ - [4452] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4452 - Write Sequencer 356 */ - [4453] = { 0x00FF, 0x00FF, 0x0000 }, /* R4453 - Write Sequencer 357 */ - [4454] = { 0x070F, 0x070F, 0x0000 }, /* R4454 - Write Sequencer 358 */ - [4455] = { 0x010F, 0x010F, 0x0000 }, /* R4455 - Write Sequencer 359 */ - [4456] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4456 - Write Sequencer 360 */ - [4457] = { 0x00FF, 0x00FF, 0x0000 }, /* R4457 - Write Sequencer 361 */ - [4458] = { 0x070F, 0x070F, 0x0000 }, /* R4458 - Write Sequencer 362 */ - [4459] = { 0x010F, 0x010F, 0x0000 }, /* R4459 - Write Sequencer 363 */ - [4460] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4460 - Write Sequencer 364 */ - [4461] = { 0x00FF, 0x00FF, 0x0000 }, /* R4461 - Write Sequencer 365 */ - [4462] = { 0x070F, 0x070F, 0x0000 }, /* R4462 - Write Sequencer 366 */ - [4463] = { 0x010F, 0x010F, 0x0000 }, /* R4463 - Write Sequencer 367 */ - [4464] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4464 - Write Sequencer 368 */ - [4465] = { 0x00FF, 0x00FF, 0x0000 }, /* R4465 - Write Sequencer 369 */ - [4466] = { 0x070F, 0x070F, 0x0000 }, /* R4466 - Write Sequencer 370 */ - [4467] = { 0x010F, 0x010F, 0x0000 }, /* R4467 - Write Sequencer 371 */ - [4468] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4468 - Write Sequencer 372 */ - [4469] = { 0x00FF, 0x00FF, 0x0000 }, /* R4469 - Write Sequencer 373 */ - [4470] = { 0x070F, 0x070F, 0x0000 }, /* R4470 - Write Sequencer 374 */ - [4471] = { 0x010F, 0x010F, 0x0000 }, /* R4471 - Write Sequencer 375 */ - [4472] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4472 - Write Sequencer 376 */ - [4473] = { 0x00FF, 0x00FF, 0x0000 }, /* R4473 - Write Sequencer 377 */ - [4474] = { 0x070F, 0x070F, 0x0000 }, /* R4474 - Write Sequencer 378 */ - [4475] = { 0x010F, 0x010F, 0x0000 }, /* R4475 - Write Sequencer 379 */ - [4476] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4476 - Write Sequencer 380 */ - [4477] = { 0x00FF, 0x00FF, 0x0000 }, /* R4477 - Write Sequencer 381 */ - [4478] = { 0x070F, 0x070F, 0x0000 }, /* R4478 - Write Sequencer 382 */ - [4479] = { 0x010F, 0x010F, 0x0000 }, /* R4479 - Write Sequencer 383 */ - [4480] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4480 - Write Sequencer 384 */ - [4481] = { 0x00FF, 0x00FF, 0x0000 }, /* R4481 - Write Sequencer 385 */ - [4482] = { 0x070F, 0x070F, 0x0000 }, /* R4482 - Write Sequencer 386 */ - [4483] = { 0x010F, 0x010F, 0x0000 }, /* R4483 - Write Sequencer 387 */ - [4484] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4484 - Write Sequencer 388 */ - [4485] = { 0x00FF, 0x00FF, 0x0000 }, /* R4485 - Write Sequencer 389 */ - [4486] = { 0x070F, 0x070F, 0x0000 }, /* R4486 - Write Sequencer 390 */ - [4487] = { 0x010F, 0x010F, 0x0000 }, /* R4487 - Write Sequencer 391 */ - [4488] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4488 - Write Sequencer 392 */ - [4489] = { 0x00FF, 0x00FF, 0x0000 }, /* R4489 - Write Sequencer 393 */ - [4490] = { 0x070F, 0x070F, 0x0000 }, /* R4490 - Write Sequencer 394 */ - [4491] = { 0x010F, 0x010F, 0x0000 }, /* R4491 - Write Sequencer 395 */ - [4492] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4492 - Write Sequencer 396 */ - [4493] = { 0x00FF, 0x00FF, 0x0000 }, /* R4493 - Write Sequencer 397 */ - [4494] = { 0x070F, 0x070F, 0x0000 }, /* R4494 - Write Sequencer 398 */ - [4495] = { 0x010F, 0x010F, 0x0000 }, /* R4495 - Write Sequencer 399 */ - [4496] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4496 - Write Sequencer 400 */ - [4497] = { 0x00FF, 0x00FF, 0x0000 }, /* R4497 - Write Sequencer 401 */ - [4498] = { 0x070F, 0x070F, 0x0000 }, /* R4498 - Write Sequencer 402 */ - [4499] = { 0x010F, 0x010F, 0x0000 }, /* R4499 - Write Sequencer 403 */ - [4500] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4500 - Write Sequencer 404 */ - [4501] = { 0x00FF, 0x00FF, 0x0000 }, /* R4501 - Write Sequencer 405 */ - [4502] = { 0x070F, 0x070F, 0x0000 }, /* R4502 - Write Sequencer 406 */ - [4503] = { 0x010F, 0x010F, 0x0000 }, /* R4503 - Write Sequencer 407 */ - [4504] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4504 - Write Sequencer 408 */ - [4505] = { 0x00FF, 0x00FF, 0x0000 }, /* R4505 - Write Sequencer 409 */ - [4506] = { 0x070F, 0x070F, 0x0000 }, /* R4506 - Write Sequencer 410 */ - [4507] = { 0x010F, 0x010F, 0x0000 }, /* R4507 - Write Sequencer 411 */ - [4508] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4508 - Write Sequencer 412 */ - [4509] = { 0x00FF, 0x00FF, 0x0000 }, /* R4509 - Write Sequencer 413 */ - [4510] = { 0x070F, 0x070F, 0x0000 }, /* R4510 - Write Sequencer 414 */ - [4511] = { 0x010F, 0x010F, 0x0000 }, /* R4511 - Write Sequencer 415 */ - [4512] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4512 - Write Sequencer 416 */ - [4513] = { 0x00FF, 0x00FF, 0x0000 }, /* R4513 - Write Sequencer 417 */ - [4514] = { 0x070F, 0x070F, 0x0000 }, /* R4514 - Write Sequencer 418 */ - [4515] = { 0x010F, 0x010F, 0x0000 }, /* R4515 - Write Sequencer 419 */ - [4516] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4516 - Write Sequencer 420 */ - [4517] = { 0x00FF, 0x00FF, 0x0000 }, /* R4517 - Write Sequencer 421 */ - [4518] = { 0x070F, 0x070F, 0x0000 }, /* R4518 - Write Sequencer 422 */ - [4519] = { 0x010F, 0x010F, 0x0000 }, /* R4519 - Write Sequencer 423 */ - [4520] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4520 - Write Sequencer 424 */ - [4521] = { 0x00FF, 0x00FF, 0x0000 }, /* R4521 - Write Sequencer 425 */ - [4522] = { 0x070F, 0x070F, 0x0000 }, /* R4522 - Write Sequencer 426 */ - [4523] = { 0x010F, 0x010F, 0x0000 }, /* R4523 - Write Sequencer 427 */ - [4524] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4524 - Write Sequencer 428 */ - [4525] = { 0x00FF, 0x00FF, 0x0000 }, /* R4525 - Write Sequencer 429 */ - [4526] = { 0x070F, 0x070F, 0x0000 }, /* R4526 - Write Sequencer 430 */ - [4527] = { 0x010F, 0x010F, 0x0000 }, /* R4527 - Write Sequencer 431 */ - [4528] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4528 - Write Sequencer 432 */ - [4529] = { 0x00FF, 0x00FF, 0x0000 }, /* R4529 - Write Sequencer 433 */ - [4530] = { 0x070F, 0x070F, 0x0000 }, /* R4530 - Write Sequencer 434 */ - [4531] = { 0x010F, 0x010F, 0x0000 }, /* R4531 - Write Sequencer 435 */ - [4532] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4532 - Write Sequencer 436 */ - [4533] = { 0x00FF, 0x00FF, 0x0000 }, /* R4533 - Write Sequencer 437 */ - [4534] = { 0x070F, 0x070F, 0x0000 }, /* R4534 - Write Sequencer 438 */ - [4535] = { 0x010F, 0x010F, 0x0000 }, /* R4535 - Write Sequencer 439 */ - [4536] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4536 - Write Sequencer 440 */ - [4537] = { 0x00FF, 0x00FF, 0x0000 }, /* R4537 - Write Sequencer 441 */ - [4538] = { 0x070F, 0x070F, 0x0000 }, /* R4538 - Write Sequencer 442 */ - [4539] = { 0x010F, 0x010F, 0x0000 }, /* R4539 - Write Sequencer 443 */ - [4540] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4540 - Write Sequencer 444 */ - [4541] = { 0x00FF, 0x00FF, 0x0000 }, /* R4541 - Write Sequencer 445 */ - [4542] = { 0x070F, 0x070F, 0x0000 }, /* R4542 - Write Sequencer 446 */ - [4543] = { 0x010F, 0x010F, 0x0000 }, /* R4543 - Write Sequencer 447 */ - [4544] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4544 - Write Sequencer 448 */ - [4545] = { 0x00FF, 0x00FF, 0x0000 }, /* R4545 - Write Sequencer 449 */ - [4546] = { 0x070F, 0x070F, 0x0000 }, /* R4546 - Write Sequencer 450 */ - [4547] = { 0x010F, 0x010F, 0x0000 }, /* R4547 - Write Sequencer 451 */ - [4548] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4548 - Write Sequencer 452 */ - [4549] = { 0x00FF, 0x00FF, 0x0000 }, /* R4549 - Write Sequencer 453 */ - [4550] = { 0x070F, 0x070F, 0x0000 }, /* R4550 - Write Sequencer 454 */ - [4551] = { 0x010F, 0x010F, 0x0000 }, /* R4551 - Write Sequencer 455 */ - [4552] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4552 - Write Sequencer 456 */ - [4553] = { 0x00FF, 0x00FF, 0x0000 }, /* R4553 - Write Sequencer 457 */ - [4554] = { 0x070F, 0x070F, 0x0000 }, /* R4554 - Write Sequencer 458 */ - [4555] = { 0x010F, 0x010F, 0x0000 }, /* R4555 - Write Sequencer 459 */ - [4556] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4556 - Write Sequencer 460 */ - [4557] = { 0x00FF, 0x00FF, 0x0000 }, /* R4557 - Write Sequencer 461 */ - [4558] = { 0x070F, 0x070F, 0x0000 }, /* R4558 - Write Sequencer 462 */ - [4559] = { 0x010F, 0x010F, 0x0000 }, /* R4559 - Write Sequencer 463 */ - [4560] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4560 - Write Sequencer 464 */ - [4561] = { 0x00FF, 0x00FF, 0x0000 }, /* R4561 - Write Sequencer 465 */ - [4562] = { 0x070F, 0x070F, 0x0000 }, /* R4562 - Write Sequencer 466 */ - [4563] = { 0x010F, 0x010F, 0x0000 }, /* R4563 - Write Sequencer 467 */ - [4564] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4564 - Write Sequencer 468 */ - [4565] = { 0x00FF, 0x00FF, 0x0000 }, /* R4565 - Write Sequencer 469 */ - [4566] = { 0x070F, 0x070F, 0x0000 }, /* R4566 - Write Sequencer 470 */ - [4567] = { 0x010F, 0x010F, 0x0000 }, /* R4567 - Write Sequencer 471 */ - [4568] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4568 - Write Sequencer 472 */ - [4569] = { 0x00FF, 0x00FF, 0x0000 }, /* R4569 - Write Sequencer 473 */ - [4570] = { 0x070F, 0x070F, 0x0000 }, /* R4570 - Write Sequencer 474 */ - [4571] = { 0x010F, 0x010F, 0x0000 }, /* R4571 - Write Sequencer 475 */ - [4572] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4572 - Write Sequencer 476 */ - [4573] = { 0x00FF, 0x00FF, 0x0000 }, /* R4573 - Write Sequencer 477 */ - [4574] = { 0x070F, 0x070F, 0x0000 }, /* R4574 - Write Sequencer 478 */ - [4575] = { 0x010F, 0x010F, 0x0000 }, /* R4575 - Write Sequencer 479 */ - [4576] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4576 - Write Sequencer 480 */ - [4577] = { 0x00FF, 0x00FF, 0x0000 }, /* R4577 - Write Sequencer 481 */ - [4578] = { 0x070F, 0x070F, 0x0000 }, /* R4578 - Write Sequencer 482 */ - [4579] = { 0x010F, 0x010F, 0x0000 }, /* R4579 - Write Sequencer 483 */ - [4580] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4580 - Write Sequencer 484 */ - [4581] = { 0x00FF, 0x00FF, 0x0000 }, /* R4581 - Write Sequencer 485 */ - [4582] = { 0x070F, 0x070F, 0x0000 }, /* R4582 - Write Sequencer 486 */ - [4583] = { 0x010F, 0x010F, 0x0000 }, /* R4583 - Write Sequencer 487 */ - [4584] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4584 - Write Sequencer 488 */ - [4585] = { 0x00FF, 0x00FF, 0x0000 }, /* R4585 - Write Sequencer 489 */ - [4586] = { 0x070F, 0x070F, 0x0000 }, /* R4586 - Write Sequencer 490 */ - [4587] = { 0x010F, 0x010F, 0x0000 }, /* R4587 - Write Sequencer 491 */ - [4588] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4588 - Write Sequencer 492 */ - [4589] = { 0x00FF, 0x00FF, 0x0000 }, /* R4589 - Write Sequencer 493 */ - [4590] = { 0x070F, 0x070F, 0x0000 }, /* R4590 - Write Sequencer 494 */ - [4591] = { 0x010F, 0x010F, 0x0000 }, /* R4591 - Write Sequencer 495 */ - [4592] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4592 - Write Sequencer 496 */ - [4593] = { 0x00FF, 0x00FF, 0x0000 }, /* R4593 - Write Sequencer 497 */ - [4594] = { 0x070F, 0x070F, 0x0000 }, /* R4594 - Write Sequencer 498 */ - [4595] = { 0x010F, 0x010F, 0x0000 }, /* R4595 - Write Sequencer 499 */ - [4596] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4596 - Write Sequencer 500 */ - [4597] = { 0x00FF, 0x00FF, 0x0000 }, /* R4597 - Write Sequencer 501 */ - [4598] = { 0x070F, 0x070F, 0x0000 }, /* R4598 - Write Sequencer 502 */ - [4599] = { 0x010F, 0x010F, 0x0000 }, /* R4599 - Write Sequencer 503 */ - [4600] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4600 - Write Sequencer 504 */ - [4601] = { 0x00FF, 0x00FF, 0x0000 }, /* R4601 - Write Sequencer 505 */ - [4602] = { 0x070F, 0x070F, 0x0000 }, /* R4602 - Write Sequencer 506 */ - [4603] = { 0x010F, 0x010F, 0x0000 }, /* R4603 - Write Sequencer 507 */ - [4604] = { 0x3FFF, 0x3FFF, 0x0000 }, /* R4604 - Write Sequencer 508 */ - [4605] = { 0x00FF, 0x00FF, 0x0000 }, /* R4605 - Write Sequencer 509 */ - [4606] = { 0x070F, 0x070F, 0x0000 }, /* R4606 - Write Sequencer 510 */ - [4607] = { 0x010F, 0x010F, 0x0000 }, /* R4607 - Write Sequencer 511 */ - [8192] = { 0x03FF, 0x03FF, 0x0000 }, /* R8192 - DSP2 Instruction RAM 0 */ - [9216] = { 0x003F, 0x003F, 0x0000 }, /* R9216 - DSP2 Address RAM 2 */ - [9217] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R9217 - DSP2 Address RAM 1 */ - [9218] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R9218 - DSP2 Address RAM 0 */ - [12288] = { 0x00FF, 0x00FF, 0x0000 }, /* R12288 - DSP2 Data1 RAM 1 */ - [12289] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R12289 - DSP2 Data1 RAM 0 */ - [13312] = { 0x00FF, 0x00FF, 0x0000 }, /* R13312 - DSP2 Data2 RAM 1 */ - [13313] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R13313 - DSP2 Data2 RAM 0 */ - [14336] = { 0x00FF, 0x00FF, 0x0000 }, /* R14336 - DSP2 Data3 RAM 1 */ - [14337] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R14337 - DSP2 Data3 RAM 0 */ - [15360] = { 0x07FF, 0x07FF, 0x0000 }, /* R15360 - DSP2 Coeff RAM 0 */ - [16384] = { 0x00FF, 0x00FF, 0x0000 }, /* R16384 - RETUNEADC_SHARED_COEFF_1 */ - [16385] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16385 - RETUNEADC_SHARED_COEFF_0 */ - [16386] = { 0x00FF, 0x00FF, 0x0000 }, /* R16386 - RETUNEDAC_SHARED_COEFF_1 */ - [16387] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16387 - RETUNEDAC_SHARED_COEFF_0 */ - [16388] = { 0x00FF, 0x00FF, 0x0000 }, /* R16388 - SOUNDSTAGE_ENABLES_1 */ - [16389] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16389 - SOUNDSTAGE_ENABLES_0 */ - [16896] = { 0x00FF, 0x00FF, 0x0000 }, /* R16896 - HDBASS_AI_1 */ - [16897] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16897 - HDBASS_AI_0 */ - [16898] = { 0x00FF, 0x00FF, 0x0000 }, /* R16898 - HDBASS_AR_1 */ - [16899] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16899 - HDBASS_AR_0 */ - [16900] = { 0x00FF, 0x00FF, 0x0000 }, /* R16900 - HDBASS_B_1 */ - [16901] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16901 - HDBASS_B_0 */ - [16902] = { 0x00FF, 0x00FF, 0x0000 }, /* R16902 - HDBASS_K_1 */ - [16903] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16903 - HDBASS_K_0 */ - [16904] = { 0x00FF, 0x00FF, 0x0000 }, /* R16904 - HDBASS_N1_1 */ - [16905] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16905 - HDBASS_N1_0 */ - [16906] = { 0x00FF, 0x00FF, 0x0000 }, /* R16906 - HDBASS_N2_1 */ - [16907] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16907 - HDBASS_N2_0 */ - [16908] = { 0x00FF, 0x00FF, 0x0000 }, /* R16908 - HDBASS_N3_1 */ - [16909] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16909 - HDBASS_N3_0 */ - [16910] = { 0x00FF, 0x00FF, 0x0000 }, /* R16910 - HDBASS_N4_1 */ - [16911] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16911 - HDBASS_N4_0 */ - [16912] = { 0x00FF, 0x00FF, 0x0000 }, /* R16912 - HDBASS_N5_1 */ - [16913] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16913 - HDBASS_N5_0 */ - [16914] = { 0x00FF, 0x00FF, 0x0000 }, /* R16914 - HDBASS_X1_1 */ - [16915] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16915 - HDBASS_X1_0 */ - [16916] = { 0x00FF, 0x00FF, 0x0000 }, /* R16916 - HDBASS_X2_1 */ - [16917] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16917 - HDBASS_X2_0 */ - [16918] = { 0x00FF, 0x00FF, 0x0000 }, /* R16918 - HDBASS_X3_1 */ - [16919] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16919 - HDBASS_X3_0 */ - [16920] = { 0x00FF, 0x00FF, 0x0000 }, /* R16920 - HDBASS_ATK_1 */ - [16921] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16921 - HDBASS_ATK_0 */ - [16922] = { 0x00FF, 0x00FF, 0x0000 }, /* R16922 - HDBASS_DCY_1 */ - [16923] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16923 - HDBASS_DCY_0 */ - [16924] = { 0x00FF, 0x00FF, 0x0000 }, /* R16924 - HDBASS_PG_1 */ - [16925] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R16925 - HDBASS_PG_0 */ - [17408] = { 0x00FF, 0x00FF, 0x0000 }, /* R17408 - HPF_C_1 */ - [17409] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17409 - HPF_C_0 */ - [17920] = { 0x00FF, 0x00FF, 0x0000 }, /* R17920 - ADCL_RETUNE_C1_1 */ - [17921] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17921 - ADCL_RETUNE_C1_0 */ - [17922] = { 0x00FF, 0x00FF, 0x0000 }, /* R17922 - ADCL_RETUNE_C2_1 */ - [17923] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17923 - ADCL_RETUNE_C2_0 */ - [17924] = { 0x00FF, 0x00FF, 0x0000 }, /* R17924 - ADCL_RETUNE_C3_1 */ - [17925] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17925 - ADCL_RETUNE_C3_0 */ - [17926] = { 0x00FF, 0x00FF, 0x0000 }, /* R17926 - ADCL_RETUNE_C4_1 */ - [17927] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17927 - ADCL_RETUNE_C4_0 */ - [17928] = { 0x00FF, 0x00FF, 0x0000 }, /* R17928 - ADCL_RETUNE_C5_1 */ - [17929] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17929 - ADCL_RETUNE_C5_0 */ - [17930] = { 0x00FF, 0x00FF, 0x0000 }, /* R17930 - ADCL_RETUNE_C6_1 */ - [17931] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17931 - ADCL_RETUNE_C6_0 */ - [17932] = { 0x00FF, 0x00FF, 0x0000 }, /* R17932 - ADCL_RETUNE_C7_1 */ - [17933] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17933 - ADCL_RETUNE_C7_0 */ - [17934] = { 0x00FF, 0x00FF, 0x0000 }, /* R17934 - ADCL_RETUNE_C8_1 */ - [17935] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17935 - ADCL_RETUNE_C8_0 */ - [17936] = { 0x00FF, 0x00FF, 0x0000 }, /* R17936 - ADCL_RETUNE_C9_1 */ - [17937] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17937 - ADCL_RETUNE_C9_0 */ - [17938] = { 0x00FF, 0x00FF, 0x0000 }, /* R17938 - ADCL_RETUNE_C10_1 */ - [17939] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17939 - ADCL_RETUNE_C10_0 */ - [17940] = { 0x00FF, 0x00FF, 0x0000 }, /* R17940 - ADCL_RETUNE_C11_1 */ - [17941] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17941 - ADCL_RETUNE_C11_0 */ - [17942] = { 0x00FF, 0x00FF, 0x0000 }, /* R17942 - ADCL_RETUNE_C12_1 */ - [17943] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17943 - ADCL_RETUNE_C12_0 */ - [17944] = { 0x00FF, 0x00FF, 0x0000 }, /* R17944 - ADCL_RETUNE_C13_1 */ - [17945] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17945 - ADCL_RETUNE_C13_0 */ - [17946] = { 0x00FF, 0x00FF, 0x0000 }, /* R17946 - ADCL_RETUNE_C14_1 */ - [17947] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17947 - ADCL_RETUNE_C14_0 */ - [17948] = { 0x00FF, 0x00FF, 0x0000 }, /* R17948 - ADCL_RETUNE_C15_1 */ - [17949] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17949 - ADCL_RETUNE_C15_0 */ - [17950] = { 0x00FF, 0x00FF, 0x0000 }, /* R17950 - ADCL_RETUNE_C16_1 */ - [17951] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17951 - ADCL_RETUNE_C16_0 */ - [17952] = { 0x00FF, 0x00FF, 0x0000 }, /* R17952 - ADCL_RETUNE_C17_1 */ - [17953] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17953 - ADCL_RETUNE_C17_0 */ - [17954] = { 0x00FF, 0x00FF, 0x0000 }, /* R17954 - ADCL_RETUNE_C18_1 */ - [17955] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17955 - ADCL_RETUNE_C18_0 */ - [17956] = { 0x00FF, 0x00FF, 0x0000 }, /* R17956 - ADCL_RETUNE_C19_1 */ - [17957] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17957 - ADCL_RETUNE_C19_0 */ - [17958] = { 0x00FF, 0x00FF, 0x0000 }, /* R17958 - ADCL_RETUNE_C20_1 */ - [17959] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17959 - ADCL_RETUNE_C20_0 */ - [17960] = { 0x00FF, 0x00FF, 0x0000 }, /* R17960 - ADCL_RETUNE_C21_1 */ - [17961] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17961 - ADCL_RETUNE_C21_0 */ - [17962] = { 0x00FF, 0x00FF, 0x0000 }, /* R17962 - ADCL_RETUNE_C22_1 */ - [17963] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17963 - ADCL_RETUNE_C22_0 */ - [17964] = { 0x00FF, 0x00FF, 0x0000 }, /* R17964 - ADCL_RETUNE_C23_1 */ - [17965] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17965 - ADCL_RETUNE_C23_0 */ - [17966] = { 0x00FF, 0x00FF, 0x0000 }, /* R17966 - ADCL_RETUNE_C24_1 */ - [17967] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17967 - ADCL_RETUNE_C24_0 */ - [17968] = { 0x00FF, 0x00FF, 0x0000 }, /* R17968 - ADCL_RETUNE_C25_1 */ - [17969] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17969 - ADCL_RETUNE_C25_0 */ - [17970] = { 0x00FF, 0x00FF, 0x0000 }, /* R17970 - ADCL_RETUNE_C26_1 */ - [17971] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17971 - ADCL_RETUNE_C26_0 */ - [17972] = { 0x00FF, 0x00FF, 0x0000 }, /* R17972 - ADCL_RETUNE_C27_1 */ - [17973] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17973 - ADCL_RETUNE_C27_0 */ - [17974] = { 0x00FF, 0x00FF, 0x0000 }, /* R17974 - ADCL_RETUNE_C28_1 */ - [17975] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17975 - ADCL_RETUNE_C28_0 */ - [17976] = { 0x00FF, 0x00FF, 0x0000 }, /* R17976 - ADCL_RETUNE_C29_1 */ - [17977] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17977 - ADCL_RETUNE_C29_0 */ - [17978] = { 0x00FF, 0x00FF, 0x0000 }, /* R17978 - ADCL_RETUNE_C30_1 */ - [17979] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17979 - ADCL_RETUNE_C30_0 */ - [17980] = { 0x00FF, 0x00FF, 0x0000 }, /* R17980 - ADCL_RETUNE_C31_1 */ - [17981] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17981 - ADCL_RETUNE_C31_0 */ - [17982] = { 0x00FF, 0x00FF, 0x0000 }, /* R17982 - ADCL_RETUNE_C32_1 */ - [17983] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R17983 - ADCL_RETUNE_C32_0 */ - [18432] = { 0x00FF, 0x00FF, 0x0000 }, /* R18432 - RETUNEADC_PG2_1 */ - [18433] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18433 - RETUNEADC_PG2_0 */ - [18434] = { 0x00FF, 0x00FF, 0x0000 }, /* R18434 - RETUNEADC_PG_1 */ - [18435] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18435 - RETUNEADC_PG_0 */ - [18944] = { 0x00FF, 0x00FF, 0x0000 }, /* R18944 - ADCR_RETUNE_C1_1 */ - [18945] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18945 - ADCR_RETUNE_C1_0 */ - [18946] = { 0x00FF, 0x00FF, 0x0000 }, /* R18946 - ADCR_RETUNE_C2_1 */ - [18947] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18947 - ADCR_RETUNE_C2_0 */ - [18948] = { 0x00FF, 0x00FF, 0x0000 }, /* R18948 - ADCR_RETUNE_C3_1 */ - [18949] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18949 - ADCR_RETUNE_C3_0 */ - [18950] = { 0x00FF, 0x00FF, 0x0000 }, /* R18950 - ADCR_RETUNE_C4_1 */ - [18951] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18951 - ADCR_RETUNE_C4_0 */ - [18952] = { 0x00FF, 0x00FF, 0x0000 }, /* R18952 - ADCR_RETUNE_C5_1 */ - [18953] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18953 - ADCR_RETUNE_C5_0 */ - [18954] = { 0x00FF, 0x00FF, 0x0000 }, /* R18954 - ADCR_RETUNE_C6_1 */ - [18955] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18955 - ADCR_RETUNE_C6_0 */ - [18956] = { 0x00FF, 0x00FF, 0x0000 }, /* R18956 - ADCR_RETUNE_C7_1 */ - [18957] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18957 - ADCR_RETUNE_C7_0 */ - [18958] = { 0x00FF, 0x00FF, 0x0000 }, /* R18958 - ADCR_RETUNE_C8_1 */ - [18959] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18959 - ADCR_RETUNE_C8_0 */ - [18960] = { 0x00FF, 0x00FF, 0x0000 }, /* R18960 - ADCR_RETUNE_C9_1 */ - [18961] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18961 - ADCR_RETUNE_C9_0 */ - [18962] = { 0x00FF, 0x00FF, 0x0000 }, /* R18962 - ADCR_RETUNE_C10_1 */ - [18963] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18963 - ADCR_RETUNE_C10_0 */ - [18964] = { 0x00FF, 0x00FF, 0x0000 }, /* R18964 - ADCR_RETUNE_C11_1 */ - [18965] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18965 - ADCR_RETUNE_C11_0 */ - [18966] = { 0x00FF, 0x00FF, 0x0000 }, /* R18966 - ADCR_RETUNE_C12_1 */ - [18967] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18967 - ADCR_RETUNE_C12_0 */ - [18968] = { 0x00FF, 0x00FF, 0x0000 }, /* R18968 - ADCR_RETUNE_C13_1 */ - [18969] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18969 - ADCR_RETUNE_C13_0 */ - [18970] = { 0x00FF, 0x00FF, 0x0000 }, /* R18970 - ADCR_RETUNE_C14_1 */ - [18971] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18971 - ADCR_RETUNE_C14_0 */ - [18972] = { 0x00FF, 0x00FF, 0x0000 }, /* R18972 - ADCR_RETUNE_C15_1 */ - [18973] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18973 - ADCR_RETUNE_C15_0 */ - [18974] = { 0x00FF, 0x00FF, 0x0000 }, /* R18974 - ADCR_RETUNE_C16_1 */ - [18975] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18975 - ADCR_RETUNE_C16_0 */ - [18976] = { 0x00FF, 0x00FF, 0x0000 }, /* R18976 - ADCR_RETUNE_C17_1 */ - [18977] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18977 - ADCR_RETUNE_C17_0 */ - [18978] = { 0x00FF, 0x00FF, 0x0000 }, /* R18978 - ADCR_RETUNE_C18_1 */ - [18979] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18979 - ADCR_RETUNE_C18_0 */ - [18980] = { 0x00FF, 0x00FF, 0x0000 }, /* R18980 - ADCR_RETUNE_C19_1 */ - [18981] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18981 - ADCR_RETUNE_C19_0 */ - [18982] = { 0x00FF, 0x00FF, 0x0000 }, /* R18982 - ADCR_RETUNE_C20_1 */ - [18983] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18983 - ADCR_RETUNE_C20_0 */ - [18984] = { 0x00FF, 0x00FF, 0x0000 }, /* R18984 - ADCR_RETUNE_C21_1 */ - [18985] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18985 - ADCR_RETUNE_C21_0 */ - [18986] = { 0x00FF, 0x00FF, 0x0000 }, /* R18986 - ADCR_RETUNE_C22_1 */ - [18987] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18987 - ADCR_RETUNE_C22_0 */ - [18988] = { 0x00FF, 0x00FF, 0x0000 }, /* R18988 - ADCR_RETUNE_C23_1 */ - [18989] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18989 - ADCR_RETUNE_C23_0 */ - [18990] = { 0x00FF, 0x00FF, 0x0000 }, /* R18990 - ADCR_RETUNE_C24_1 */ - [18991] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18991 - ADCR_RETUNE_C24_0 */ - [18992] = { 0x00FF, 0x00FF, 0x0000 }, /* R18992 - ADCR_RETUNE_C25_1 */ - [18993] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18993 - ADCR_RETUNE_C25_0 */ - [18994] = { 0x00FF, 0x00FF, 0x0000 }, /* R18994 - ADCR_RETUNE_C26_1 */ - [18995] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18995 - ADCR_RETUNE_C26_0 */ - [18996] = { 0x00FF, 0x00FF, 0x0000 }, /* R18996 - ADCR_RETUNE_C27_1 */ - [18997] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18997 - ADCR_RETUNE_C27_0 */ - [18998] = { 0x00FF, 0x00FF, 0x0000 }, /* R18998 - ADCR_RETUNE_C28_1 */ - [18999] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R18999 - ADCR_RETUNE_C28_0 */ - [19000] = { 0x00FF, 0x00FF, 0x0000 }, /* R19000 - ADCR_RETUNE_C29_1 */ - [19001] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19001 - ADCR_RETUNE_C29_0 */ - [19002] = { 0x00FF, 0x00FF, 0x0000 }, /* R19002 - ADCR_RETUNE_C30_1 */ - [19003] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19003 - ADCR_RETUNE_C30_0 */ - [19004] = { 0x00FF, 0x00FF, 0x0000 }, /* R19004 - ADCR_RETUNE_C31_1 */ - [19005] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19005 - ADCR_RETUNE_C31_0 */ - [19006] = { 0x00FF, 0x00FF, 0x0000 }, /* R19006 - ADCR_RETUNE_C32_1 */ - [19007] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19007 - ADCR_RETUNE_C32_0 */ - [19456] = { 0x00FF, 0x00FF, 0x0000 }, /* R19456 - DACL_RETUNE_C1_1 */ - [19457] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19457 - DACL_RETUNE_C1_0 */ - [19458] = { 0x00FF, 0x00FF, 0x0000 }, /* R19458 - DACL_RETUNE_C2_1 */ - [19459] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19459 - DACL_RETUNE_C2_0 */ - [19460] = { 0x00FF, 0x00FF, 0x0000 }, /* R19460 - DACL_RETUNE_C3_1 */ - [19461] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19461 - DACL_RETUNE_C3_0 */ - [19462] = { 0x00FF, 0x00FF, 0x0000 }, /* R19462 - DACL_RETUNE_C4_1 */ - [19463] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19463 - DACL_RETUNE_C4_0 */ - [19464] = { 0x00FF, 0x00FF, 0x0000 }, /* R19464 - DACL_RETUNE_C5_1 */ - [19465] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19465 - DACL_RETUNE_C5_0 */ - [19466] = { 0x00FF, 0x00FF, 0x0000 }, /* R19466 - DACL_RETUNE_C6_1 */ - [19467] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19467 - DACL_RETUNE_C6_0 */ - [19468] = { 0x00FF, 0x00FF, 0x0000 }, /* R19468 - DACL_RETUNE_C7_1 */ - [19469] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19469 - DACL_RETUNE_C7_0 */ - [19470] = { 0x00FF, 0x00FF, 0x0000 }, /* R19470 - DACL_RETUNE_C8_1 */ - [19471] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19471 - DACL_RETUNE_C8_0 */ - [19472] = { 0x00FF, 0x00FF, 0x0000 }, /* R19472 - DACL_RETUNE_C9_1 */ - [19473] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19473 - DACL_RETUNE_C9_0 */ - [19474] = { 0x00FF, 0x00FF, 0x0000 }, /* R19474 - DACL_RETUNE_C10_1 */ - [19475] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19475 - DACL_RETUNE_C10_0 */ - [19476] = { 0x00FF, 0x00FF, 0x0000 }, /* R19476 - DACL_RETUNE_C11_1 */ - [19477] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19477 - DACL_RETUNE_C11_0 */ - [19478] = { 0x00FF, 0x00FF, 0x0000 }, /* R19478 - DACL_RETUNE_C12_1 */ - [19479] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19479 - DACL_RETUNE_C12_0 */ - [19480] = { 0x00FF, 0x00FF, 0x0000 }, /* R19480 - DACL_RETUNE_C13_1 */ - [19481] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19481 - DACL_RETUNE_C13_0 */ - [19482] = { 0x00FF, 0x00FF, 0x0000 }, /* R19482 - DACL_RETUNE_C14_1 */ - [19483] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19483 - DACL_RETUNE_C14_0 */ - [19484] = { 0x00FF, 0x00FF, 0x0000 }, /* R19484 - DACL_RETUNE_C15_1 */ - [19485] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19485 - DACL_RETUNE_C15_0 */ - [19486] = { 0x00FF, 0x00FF, 0x0000 }, /* R19486 - DACL_RETUNE_C16_1 */ - [19487] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19487 - DACL_RETUNE_C16_0 */ - [19488] = { 0x00FF, 0x00FF, 0x0000 }, /* R19488 - DACL_RETUNE_C17_1 */ - [19489] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19489 - DACL_RETUNE_C17_0 */ - [19490] = { 0x00FF, 0x00FF, 0x0000 }, /* R19490 - DACL_RETUNE_C18_1 */ - [19491] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19491 - DACL_RETUNE_C18_0 */ - [19492] = { 0x00FF, 0x00FF, 0x0000 }, /* R19492 - DACL_RETUNE_C19_1 */ - [19493] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19493 - DACL_RETUNE_C19_0 */ - [19494] = { 0x00FF, 0x00FF, 0x0000 }, /* R19494 - DACL_RETUNE_C20_1 */ - [19495] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19495 - DACL_RETUNE_C20_0 */ - [19496] = { 0x00FF, 0x00FF, 0x0000 }, /* R19496 - DACL_RETUNE_C21_1 */ - [19497] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19497 - DACL_RETUNE_C21_0 */ - [19498] = { 0x00FF, 0x00FF, 0x0000 }, /* R19498 - DACL_RETUNE_C22_1 */ - [19499] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19499 - DACL_RETUNE_C22_0 */ - [19500] = { 0x00FF, 0x00FF, 0x0000 }, /* R19500 - DACL_RETUNE_C23_1 */ - [19501] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19501 - DACL_RETUNE_C23_0 */ - [19502] = { 0x00FF, 0x00FF, 0x0000 }, /* R19502 - DACL_RETUNE_C24_1 */ - [19503] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19503 - DACL_RETUNE_C24_0 */ - [19504] = { 0x00FF, 0x00FF, 0x0000 }, /* R19504 - DACL_RETUNE_C25_1 */ - [19505] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19505 - DACL_RETUNE_C25_0 */ - [19506] = { 0x00FF, 0x00FF, 0x0000 }, /* R19506 - DACL_RETUNE_C26_1 */ - [19507] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19507 - DACL_RETUNE_C26_0 */ - [19508] = { 0x00FF, 0x00FF, 0x0000 }, /* R19508 - DACL_RETUNE_C27_1 */ - [19509] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19509 - DACL_RETUNE_C27_0 */ - [19510] = { 0x00FF, 0x00FF, 0x0000 }, /* R19510 - DACL_RETUNE_C28_1 */ - [19511] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19511 - DACL_RETUNE_C28_0 */ - [19512] = { 0x00FF, 0x00FF, 0x0000 }, /* R19512 - DACL_RETUNE_C29_1 */ - [19513] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19513 - DACL_RETUNE_C29_0 */ - [19514] = { 0x00FF, 0x00FF, 0x0000 }, /* R19514 - DACL_RETUNE_C30_1 */ - [19515] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19515 - DACL_RETUNE_C30_0 */ - [19516] = { 0x00FF, 0x00FF, 0x0000 }, /* R19516 - DACL_RETUNE_C31_1 */ - [19517] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19517 - DACL_RETUNE_C31_0 */ - [19518] = { 0x00FF, 0x00FF, 0x0000 }, /* R19518 - DACL_RETUNE_C32_1 */ - [19519] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19519 - DACL_RETUNE_C32_0 */ - [19968] = { 0x00FF, 0x00FF, 0x0000 }, /* R19968 - RETUNEDAC_PG2_1 */ - [19969] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19969 - RETUNEDAC_PG2_0 */ - [19970] = { 0x00FF, 0x00FF, 0x0000 }, /* R19970 - RETUNEDAC_PG_1 */ - [19971] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R19971 - RETUNEDAC_PG_0 */ - [20480] = { 0x00FF, 0x00FF, 0x0000 }, /* R20480 - DACR_RETUNE_C1_1 */ - [20481] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20481 - DACR_RETUNE_C1_0 */ - [20482] = { 0x00FF, 0x00FF, 0x0000 }, /* R20482 - DACR_RETUNE_C2_1 */ - [20483] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20483 - DACR_RETUNE_C2_0 */ - [20484] = { 0x00FF, 0x00FF, 0x0000 }, /* R20484 - DACR_RETUNE_C3_1 */ - [20485] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20485 - DACR_RETUNE_C3_0 */ - [20486] = { 0x00FF, 0x00FF, 0x0000 }, /* R20486 - DACR_RETUNE_C4_1 */ - [20487] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20487 - DACR_RETUNE_C4_0 */ - [20488] = { 0x00FF, 0x00FF, 0x0000 }, /* R20488 - DACR_RETUNE_C5_1 */ - [20489] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20489 - DACR_RETUNE_C5_0 */ - [20490] = { 0x00FF, 0x00FF, 0x0000 }, /* R20490 - DACR_RETUNE_C6_1 */ - [20491] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20491 - DACR_RETUNE_C6_0 */ - [20492] = { 0x00FF, 0x00FF, 0x0000 }, /* R20492 - DACR_RETUNE_C7_1 */ - [20493] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20493 - DACR_RETUNE_C7_0 */ - [20494] = { 0x00FF, 0x00FF, 0x0000 }, /* R20494 - DACR_RETUNE_C8_1 */ - [20495] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20495 - DACR_RETUNE_C8_0 */ - [20496] = { 0x00FF, 0x00FF, 0x0000 }, /* R20496 - DACR_RETUNE_C9_1 */ - [20497] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20497 - DACR_RETUNE_C9_0 */ - [20498] = { 0x00FF, 0x00FF, 0x0000 }, /* R20498 - DACR_RETUNE_C10_1 */ - [20499] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20499 - DACR_RETUNE_C10_0 */ - [20500] = { 0x00FF, 0x00FF, 0x0000 }, /* R20500 - DACR_RETUNE_C11_1 */ - [20501] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20501 - DACR_RETUNE_C11_0 */ - [20502] = { 0x00FF, 0x00FF, 0x0000 }, /* R20502 - DACR_RETUNE_C12_1 */ - [20503] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20503 - DACR_RETUNE_C12_0 */ - [20504] = { 0x00FF, 0x00FF, 0x0000 }, /* R20504 - DACR_RETUNE_C13_1 */ - [20505] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20505 - DACR_RETUNE_C13_0 */ - [20506] = { 0x00FF, 0x00FF, 0x0000 }, /* R20506 - DACR_RETUNE_C14_1 */ - [20507] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20507 - DACR_RETUNE_C14_0 */ - [20508] = { 0x00FF, 0x00FF, 0x0000 }, /* R20508 - DACR_RETUNE_C15_1 */ - [20509] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20509 - DACR_RETUNE_C15_0 */ - [20510] = { 0x00FF, 0x00FF, 0x0000 }, /* R20510 - DACR_RETUNE_C16_1 */ - [20511] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20511 - DACR_RETUNE_C16_0 */ - [20512] = { 0x00FF, 0x00FF, 0x0000 }, /* R20512 - DACR_RETUNE_C17_1 */ - [20513] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20513 - DACR_RETUNE_C17_0 */ - [20514] = { 0x00FF, 0x00FF, 0x0000 }, /* R20514 - DACR_RETUNE_C18_1 */ - [20515] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20515 - DACR_RETUNE_C18_0 */ - [20516] = { 0x00FF, 0x00FF, 0x0000 }, /* R20516 - DACR_RETUNE_C19_1 */ - [20517] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20517 - DACR_RETUNE_C19_0 */ - [20518] = { 0x00FF, 0x00FF, 0x0000 }, /* R20518 - DACR_RETUNE_C20_1 */ - [20519] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20519 - DACR_RETUNE_C20_0 */ - [20520] = { 0x00FF, 0x00FF, 0x0000 }, /* R20520 - DACR_RETUNE_C21_1 */ - [20521] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20521 - DACR_RETUNE_C21_0 */ - [20522] = { 0x00FF, 0x00FF, 0x0000 }, /* R20522 - DACR_RETUNE_C22_1 */ - [20523] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20523 - DACR_RETUNE_C22_0 */ - [20524] = { 0x00FF, 0x00FF, 0x0000 }, /* R20524 - DACR_RETUNE_C23_1 */ - [20525] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20525 - DACR_RETUNE_C23_0 */ - [20526] = { 0x00FF, 0x00FF, 0x0000 }, /* R20526 - DACR_RETUNE_C24_1 */ - [20527] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20527 - DACR_RETUNE_C24_0 */ - [20528] = { 0x00FF, 0x00FF, 0x0000 }, /* R20528 - DACR_RETUNE_C25_1 */ - [20529] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20529 - DACR_RETUNE_C25_0 */ - [20530] = { 0x00FF, 0x00FF, 0x0000 }, /* R20530 - DACR_RETUNE_C26_1 */ - [20531] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20531 - DACR_RETUNE_C26_0 */ - [20532] = { 0x00FF, 0x00FF, 0x0000 }, /* R20532 - DACR_RETUNE_C27_1 */ - [20533] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20533 - DACR_RETUNE_C27_0 */ - [20534] = { 0x00FF, 0x00FF, 0x0000 }, /* R20534 - DACR_RETUNE_C28_1 */ - [20535] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20535 - DACR_RETUNE_C28_0 */ - [20536] = { 0x00FF, 0x00FF, 0x0000 }, /* R20536 - DACR_RETUNE_C29_1 */ - [20537] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20537 - DACR_RETUNE_C29_0 */ - [20538] = { 0x00FF, 0x00FF, 0x0000 }, /* R20538 - DACR_RETUNE_C30_1 */ - [20539] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20539 - DACR_RETUNE_C30_0 */ - [20540] = { 0x00FF, 0x00FF, 0x0000 }, /* R20540 - DACR_RETUNE_C31_1 */ - [20541] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20541 - DACR_RETUNE_C31_0 */ - [20542] = { 0x00FF, 0x00FF, 0x0000 }, /* R20542 - DACR_RETUNE_C32_1 */ - [20543] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20543 - DACR_RETUNE_C32_0 */ - [20992] = { 0x00FF, 0x00FF, 0x0000 }, /* R20992 - VSS_XHD2_1 */ - [20993] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20993 - VSS_XHD2_0 */ - [20994] = { 0x00FF, 0x00FF, 0x0000 }, /* R20994 - VSS_XHD3_1 */ - [20995] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20995 - VSS_XHD3_0 */ - [20996] = { 0x00FF, 0x00FF, 0x0000 }, /* R20996 - VSS_XHN1_1 */ - [20997] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20997 - VSS_XHN1_0 */ - [20998] = { 0x00FF, 0x00FF, 0x0000 }, /* R20998 - VSS_XHN2_1 */ - [20999] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R20999 - VSS_XHN2_0 */ - [21000] = { 0x00FF, 0x00FF, 0x0000 }, /* R21000 - VSS_XHN3_1 */ - [21001] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21001 - VSS_XHN3_0 */ - [21002] = { 0x00FF, 0x00FF, 0x0000 }, /* R21002 - VSS_XLA_1 */ - [21003] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21003 - VSS_XLA_0 */ - [21004] = { 0x00FF, 0x00FF, 0x0000 }, /* R21004 - VSS_XLB_1 */ - [21005] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21005 - VSS_XLB_0 */ - [21006] = { 0x00FF, 0x00FF, 0x0000 }, /* R21006 - VSS_XLG_1 */ - [21007] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21007 - VSS_XLG_0 */ - [21008] = { 0x00FF, 0x00FF, 0x0000 }, /* R21008 - VSS_PG2_1 */ - [21009] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21009 - VSS_PG2_0 */ - [21010] = { 0x00FF, 0x00FF, 0x0000 }, /* R21010 - VSS_PG_1 */ - [21011] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21011 - VSS_PG_0 */ - [21012] = { 0x00FF, 0x00FF, 0x0000 }, /* R21012 - VSS_XTD1_1 */ - [21013] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21013 - VSS_XTD1_0 */ - [21014] = { 0x00FF, 0x00FF, 0x0000 }, /* R21014 - VSS_XTD2_1 */ - [21015] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21015 - VSS_XTD2_0 */ - [21016] = { 0x00FF, 0x00FF, 0x0000 }, /* R21016 - VSS_XTD3_1 */ - [21017] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21017 - VSS_XTD3_0 */ - [21018] = { 0x00FF, 0x00FF, 0x0000 }, /* R21018 - VSS_XTD4_1 */ - [21019] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21019 - VSS_XTD4_0 */ - [21020] = { 0x00FF, 0x00FF, 0x0000 }, /* R21020 - VSS_XTD5_1 */ - [21021] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21021 - VSS_XTD5_0 */ - [21022] = { 0x00FF, 0x00FF, 0x0000 }, /* R21022 - VSS_XTD6_1 */ - [21023] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21023 - VSS_XTD6_0 */ - [21024] = { 0x00FF, 0x00FF, 0x0000 }, /* R21024 - VSS_XTD7_1 */ - [21025] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21025 - VSS_XTD7_0 */ - [21026] = { 0x00FF, 0x00FF, 0x0000 }, /* R21026 - VSS_XTD8_1 */ - [21027] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21027 - VSS_XTD8_0 */ - [21028] = { 0x00FF, 0x00FF, 0x0000 }, /* R21028 - VSS_XTD9_1 */ - [21029] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21029 - VSS_XTD9_0 */ - [21030] = { 0x00FF, 0x00FF, 0x0000 }, /* R21030 - VSS_XTD10_1 */ - [21031] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21031 - VSS_XTD10_0 */ - [21032] = { 0x00FF, 0x00FF, 0x0000 }, /* R21032 - VSS_XTD11_1 */ - [21033] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21033 - VSS_XTD11_0 */ - [21034] = { 0x00FF, 0x00FF, 0x0000 }, /* R21034 - VSS_XTD12_1 */ - [21035] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21035 - VSS_XTD12_0 */ - [21036] = { 0x00FF, 0x00FF, 0x0000 }, /* R21036 - VSS_XTD13_1 */ - [21037] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21037 - VSS_XTD13_0 */ - [21038] = { 0x00FF, 0x00FF, 0x0000 }, /* R21038 - VSS_XTD14_1 */ - [21039] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21039 - VSS_XTD14_0 */ - [21040] = { 0x00FF, 0x00FF, 0x0000 }, /* R21040 - VSS_XTD15_1 */ - [21041] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21041 - VSS_XTD15_0 */ - [21042] = { 0x00FF, 0x00FF, 0x0000 }, /* R21042 - VSS_XTD16_1 */ - [21043] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21043 - VSS_XTD16_0 */ - [21044] = { 0x00FF, 0x00FF, 0x0000 }, /* R21044 - VSS_XTD17_1 */ - [21045] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21045 - VSS_XTD17_0 */ - [21046] = { 0x00FF, 0x00FF, 0x0000 }, /* R21046 - VSS_XTD18_1 */ - [21047] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21047 - VSS_XTD18_0 */ - [21048] = { 0x00FF, 0x00FF, 0x0000 }, /* R21048 - VSS_XTD19_1 */ - [21049] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21049 - VSS_XTD19_0 */ - [21050] = { 0x00FF, 0x00FF, 0x0000 }, /* R21050 - VSS_XTD20_1 */ - [21051] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21051 - VSS_XTD20_0 */ - [21052] = { 0x00FF, 0x00FF, 0x0000 }, /* R21052 - VSS_XTD21_1 */ - [21053] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21053 - VSS_XTD21_0 */ - [21054] = { 0x00FF, 0x00FF, 0x0000 }, /* R21054 - VSS_XTD22_1 */ - [21055] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21055 - VSS_XTD22_0 */ - [21056] = { 0x00FF, 0x00FF, 0x0000 }, /* R21056 - VSS_XTD23_1 */ - [21057] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21057 - VSS_XTD23_0 */ - [21058] = { 0x00FF, 0x00FF, 0x0000 }, /* R21058 - VSS_XTD24_1 */ - [21059] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21059 - VSS_XTD24_0 */ - [21060] = { 0x00FF, 0x00FF, 0x0000 }, /* R21060 - VSS_XTD25_1 */ - [21061] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21061 - VSS_XTD25_0 */ - [21062] = { 0x00FF, 0x00FF, 0x0000 }, /* R21062 - VSS_XTD26_1 */ - [21063] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21063 - VSS_XTD26_0 */ - [21064] = { 0x00FF, 0x00FF, 0x0000 }, /* R21064 - VSS_XTD27_1 */ - [21065] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21065 - VSS_XTD27_0 */ - [21066] = { 0x00FF, 0x00FF, 0x0000 }, /* R21066 - VSS_XTD28_1 */ - [21067] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21067 - VSS_XTD28_0 */ - [21068] = { 0x00FF, 0x00FF, 0x0000 }, /* R21068 - VSS_XTD29_1 */ - [21069] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21069 - VSS_XTD29_0 */ - [21070] = { 0x00FF, 0x00FF, 0x0000 }, /* R21070 - VSS_XTD30_1 */ - [21071] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21071 - VSS_XTD30_0 */ - [21072] = { 0x00FF, 0x00FF, 0x0000 }, /* R21072 - VSS_XTD31_1 */ - [21073] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21073 - VSS_XTD31_0 */ - [21074] = { 0x00FF, 0x00FF, 0x0000 }, /* R21074 - VSS_XTD32_1 */ - [21075] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21075 - VSS_XTD32_0 */ - [21076] = { 0x00FF, 0x00FF, 0x0000 }, /* R21076 - VSS_XTS1_1 */ - [21077] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21077 - VSS_XTS1_0 */ - [21078] = { 0x00FF, 0x00FF, 0x0000 }, /* R21078 - VSS_XTS2_1 */ - [21079] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21079 - VSS_XTS2_0 */ - [21080] = { 0x00FF, 0x00FF, 0x0000 }, /* R21080 - VSS_XTS3_1 */ - [21081] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21081 - VSS_XTS3_0 */ - [21082] = { 0x00FF, 0x00FF, 0x0000 }, /* R21082 - VSS_XTS4_1 */ - [21083] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21083 - VSS_XTS4_0 */ - [21084] = { 0x00FF, 0x00FF, 0x0000 }, /* R21084 - VSS_XTS5_1 */ - [21085] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21085 - VSS_XTS5_0 */ - [21086] = { 0x00FF, 0x00FF, 0x0000 }, /* R21086 - VSS_XTS6_1 */ - [21087] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21087 - VSS_XTS6_0 */ - [21088] = { 0x00FF, 0x00FF, 0x0000 }, /* R21088 - VSS_XTS7_1 */ - [21089] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21089 - VSS_XTS7_0 */ - [21090] = { 0x00FF, 0x00FF, 0x0000 }, /* R21090 - VSS_XTS8_1 */ - [21091] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21091 - VSS_XTS8_0 */ - [21092] = { 0x00FF, 0x00FF, 0x0000 }, /* R21092 - VSS_XTS9_1 */ - [21093] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21093 - VSS_XTS9_0 */ - [21094] = { 0x00FF, 0x00FF, 0x0000 }, /* R21094 - VSS_XTS10_1 */ - [21095] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21095 - VSS_XTS10_0 */ - [21096] = { 0x00FF, 0x00FF, 0x0000 }, /* R21096 - VSS_XTS11_1 */ - [21097] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21097 - VSS_XTS11_0 */ - [21098] = { 0x00FF, 0x00FF, 0x0000 }, /* R21098 - VSS_XTS12_1 */ - [21099] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21099 - VSS_XTS12_0 */ - [21100] = { 0x00FF, 0x00FF, 0x0000 }, /* R21100 - VSS_XTS13_1 */ - [21101] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21101 - VSS_XTS13_0 */ - [21102] = { 0x00FF, 0x00FF, 0x0000 }, /* R21102 - VSS_XTS14_1 */ - [21103] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21103 - VSS_XTS14_0 */ - [21104] = { 0x00FF, 0x00FF, 0x0000 }, /* R21104 - VSS_XTS15_1 */ - [21105] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21105 - VSS_XTS15_0 */ - [21106] = { 0x00FF, 0x00FF, 0x0000 }, /* R21106 - VSS_XTS16_1 */ - [21107] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21107 - VSS_XTS16_0 */ - [21108] = { 0x00FF, 0x00FF, 0x0000 }, /* R21108 - VSS_XTS17_1 */ - [21109] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21109 - VSS_XTS17_0 */ - [21110] = { 0x00FF, 0x00FF, 0x0000 }, /* R21110 - VSS_XTS18_1 */ - [21111] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21111 - VSS_XTS18_0 */ - [21112] = { 0x00FF, 0x00FF, 0x0000 }, /* R21112 - VSS_XTS19_1 */ - [21113] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21113 - VSS_XTS19_0 */ - [21114] = { 0x00FF, 0x00FF, 0x0000 }, /* R21114 - VSS_XTS20_1 */ - [21115] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21115 - VSS_XTS20_0 */ - [21116] = { 0x00FF, 0x00FF, 0x0000 }, /* R21116 - VSS_XTS21_1 */ - [21117] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21117 - VSS_XTS21_0 */ - [21118] = { 0x00FF, 0x00FF, 0x0000 }, /* R21118 - VSS_XTS22_1 */ - [21119] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21119 - VSS_XTS22_0 */ - [21120] = { 0x00FF, 0x00FF, 0x0000 }, /* R21120 - VSS_XTS23_1 */ - [21121] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21121 - VSS_XTS23_0 */ - [21122] = { 0x00FF, 0x00FF, 0x0000 }, /* R21122 - VSS_XTS24_1 */ - [21123] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21123 - VSS_XTS24_0 */ - [21124] = { 0x00FF, 0x00FF, 0x0000 }, /* R21124 - VSS_XTS25_1 */ - [21125] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21125 - VSS_XTS25_0 */ - [21126] = { 0x00FF, 0x00FF, 0x0000 }, /* R21126 - VSS_XTS26_1 */ - [21127] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21127 - VSS_XTS26_0 */ - [21128] = { 0x00FF, 0x00FF, 0x0000 }, /* R21128 - VSS_XTS27_1 */ - [21129] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21129 - VSS_XTS27_0 */ - [21130] = { 0x00FF, 0x00FF, 0x0000 }, /* R21130 - VSS_XTS28_1 */ - [21131] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21131 - VSS_XTS28_0 */ - [21132] = { 0x00FF, 0x00FF, 0x0000 }, /* R21132 - VSS_XTS29_1 */ - [21133] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21133 - VSS_XTS29_0 */ - [21134] = { 0x00FF, 0x00FF, 0x0000 }, /* R21134 - VSS_XTS30_1 */ - [21135] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21135 - VSS_XTS30_0 */ - [21136] = { 0x00FF, 0x00FF, 0x0000 }, /* R21136 - VSS_XTS31_1 */ - [21137] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21137 - VSS_XTS31_0 */ - [21138] = { 0x00FF, 0x00FF, 0x0000 }, /* R21138 - VSS_XTS32_1 */ - [21139] = { 0xFFFF, 0xFFFF, 0x0000 }, /* R21139 - VSS_XTS32_0 */ -}; - static bool wm8962_volatile_register(struct device *dev, unsigned int reg) { - if (wm8962_reg_access[reg].vol) - return 1; - else - return 0; + switch (reg) { + case WM8962_CLOCKING1: + case WM8962_CLOCKING2: + case WM8962_SOFTWARE_RESET: + case WM8962_ALC2: + case WM8962_THERMAL_SHUTDOWN_STATUS: + case WM8962_ADDITIONAL_CONTROL_4: + case WM8962_CLASS_D_CONTROL_1: + case WM8962_DC_SERVO_6: + case WM8962_INTERRUPT_STATUS_1: + case WM8962_INTERRUPT_STATUS_2: + case WM8962_DSP2_EXECCONTROL: + return true; + default: + return false; + } } static bool wm8962_readable_register(struct device *dev, unsigned int reg) { - if (wm8962_reg_access[reg].read) - return 1; - else - return 0; + switch (reg) { + case WM8962_LEFT_INPUT_VOLUME: + case WM8962_RIGHT_INPUT_VOLUME: + case WM8962_HPOUTL_VOLUME: + case WM8962_HPOUTR_VOLUME: + case WM8962_CLOCKING1: + case WM8962_ADC_DAC_CONTROL_1: + case WM8962_ADC_DAC_CONTROL_2: + case WM8962_AUDIO_INTERFACE_0: + case WM8962_CLOCKING2: + case WM8962_AUDIO_INTERFACE_1: + case WM8962_LEFT_DAC_VOLUME: + case WM8962_RIGHT_DAC_VOLUME: + case WM8962_AUDIO_INTERFACE_2: + case WM8962_SOFTWARE_RESET: + case WM8962_ALC1: + case WM8962_ALC2: + case WM8962_ALC3: + case WM8962_NOISE_GATE: + case WM8962_LEFT_ADC_VOLUME: + case WM8962_RIGHT_ADC_VOLUME: + case WM8962_ADDITIONAL_CONTROL_1: + case WM8962_ADDITIONAL_CONTROL_2: + case WM8962_PWR_MGMT_1: + case WM8962_PWR_MGMT_2: + case WM8962_ADDITIONAL_CONTROL_3: + case WM8962_ANTI_POP: + case WM8962_CLOCKING_3: + case WM8962_INPUT_MIXER_CONTROL_1: + case WM8962_LEFT_INPUT_MIXER_VOLUME: + case WM8962_RIGHT_INPUT_MIXER_VOLUME: + case WM8962_INPUT_MIXER_CONTROL_2: + case WM8962_INPUT_BIAS_CONTROL: + case WM8962_LEFT_INPUT_PGA_CONTROL: + case WM8962_RIGHT_INPUT_PGA_CONTROL: + case WM8962_SPKOUTL_VOLUME: + case WM8962_SPKOUTR_VOLUME: + case WM8962_THERMAL_SHUTDOWN_STATUS: + case WM8962_ADDITIONAL_CONTROL_4: + case WM8962_CLASS_D_CONTROL_1: + case WM8962_CLASS_D_CONTROL_2: + case WM8962_CLOCKING_4: + case WM8962_DAC_DSP_MIXING_1: + case WM8962_DAC_DSP_MIXING_2: + case WM8962_DC_SERVO_0: + case WM8962_DC_SERVO_1: + case WM8962_DC_SERVO_4: + case WM8962_DC_SERVO_6: + case WM8962_ANALOGUE_PGA_BIAS: + case WM8962_ANALOGUE_HP_0: + case WM8962_ANALOGUE_HP_2: + case WM8962_CHARGE_PUMP_1: + case WM8962_CHARGE_PUMP_B: + case WM8962_WRITE_SEQUENCER_CONTROL_1: + case WM8962_WRITE_SEQUENCER_CONTROL_2: + case WM8962_WRITE_SEQUENCER_CONTROL_3: + case WM8962_CONTROL_INTERFACE: + case WM8962_MIXER_ENABLES: + case WM8962_HEADPHONE_MIXER_1: + case WM8962_HEADPHONE_MIXER_2: + case WM8962_HEADPHONE_MIXER_3: + case WM8962_HEADPHONE_MIXER_4: + case WM8962_SPEAKER_MIXER_1: + case WM8962_SPEAKER_MIXER_2: + case WM8962_SPEAKER_MIXER_3: + case WM8962_SPEAKER_MIXER_4: + case WM8962_SPEAKER_MIXER_5: + case WM8962_BEEP_GENERATOR_1: + case WM8962_OSCILLATOR_TRIM_3: + case WM8962_OSCILLATOR_TRIM_4: + case WM8962_OSCILLATOR_TRIM_7: + case WM8962_ANALOGUE_CLOCKING1: + case WM8962_ANALOGUE_CLOCKING2: + case WM8962_ANALOGUE_CLOCKING3: + case WM8962_PLL_SOFTWARE_RESET: + case WM8962_PLL2: + case WM8962_PLL_4: + case WM8962_PLL_9: + case WM8962_PLL_10: + case WM8962_PLL_11: + case WM8962_PLL_12: + case WM8962_PLL_13: + case WM8962_PLL_14: + case WM8962_PLL_15: + case WM8962_PLL_16: + case WM8962_FLL_CONTROL_1: + case WM8962_FLL_CONTROL_2: + case WM8962_FLL_CONTROL_3: + case WM8962_FLL_CONTROL_5: + case WM8962_FLL_CONTROL_6: + case WM8962_FLL_CONTROL_7: + case WM8962_FLL_CONTROL_8: + case WM8962_GENERAL_TEST_1: + case WM8962_DF1: + case WM8962_DF2: + case WM8962_DF3: + case WM8962_DF4: + case WM8962_DF5: + case WM8962_DF6: + case WM8962_DF7: + case WM8962_LHPF1: + case WM8962_LHPF2: + case WM8962_THREED1: + case WM8962_THREED2: + case WM8962_THREED3: + case WM8962_THREED4: + case WM8962_DRC_1: + case WM8962_DRC_2: + case WM8962_DRC_3: + case WM8962_DRC_4: + case WM8962_DRC_5: + case WM8962_TLOOPBACK: + case WM8962_EQ1: + case WM8962_EQ2: + case WM8962_EQ3: + case WM8962_EQ4: + case WM8962_EQ5: + case WM8962_EQ6: + case WM8962_EQ7: + case WM8962_EQ8: + case WM8962_EQ9: + case WM8962_EQ10: + case WM8962_EQ11: + case WM8962_EQ12: + case WM8962_EQ13: + case WM8962_EQ14: + case WM8962_EQ15: + case WM8962_EQ16: + case WM8962_EQ17: + case WM8962_EQ18: + case WM8962_EQ19: + case WM8962_EQ20: + case WM8962_EQ21: + case WM8962_EQ22: + case WM8962_EQ23: + case WM8962_EQ24: + case WM8962_EQ25: + case WM8962_EQ26: + case WM8962_EQ27: + case WM8962_EQ28: + case WM8962_EQ29: + case WM8962_EQ30: + case WM8962_EQ31: + case WM8962_EQ32: + case WM8962_EQ33: + case WM8962_EQ34: + case WM8962_EQ35: + case WM8962_EQ36: + case WM8962_EQ37: + case WM8962_EQ38: + case WM8962_EQ39: + case WM8962_EQ40: + case WM8962_EQ41: + case WM8962_GPIO_BASE: + case WM8962_GPIO_2: + case WM8962_GPIO_3: + case WM8962_GPIO_5: + case WM8962_GPIO_6: + case WM8962_INTERRUPT_STATUS_1: + case WM8962_INTERRUPT_STATUS_2: + case WM8962_INTERRUPT_STATUS_1_MASK: + case WM8962_INTERRUPT_STATUS_2_MASK: + case WM8962_INTERRUPT_CONTROL: + case WM8962_IRQ_DEBOUNCE: + case WM8962_MICINT_SOURCE_POL: + case WM8962_DSP2_POWER_MANAGEMENT: + case WM8962_DSP2_EXECCONTROL: + case WM8962_DSP2_INSTRUCTION_RAM_0: + case WM8962_DSP2_ADDRESS_RAM_2: + case WM8962_DSP2_ADDRESS_RAM_1: + case WM8962_DSP2_ADDRESS_RAM_0: + case WM8962_DSP2_DATA1_RAM_1: + case WM8962_DSP2_DATA1_RAM_0: + case WM8962_DSP2_DATA2_RAM_1: + case WM8962_DSP2_DATA2_RAM_0: + case WM8962_DSP2_DATA3_RAM_1: + case WM8962_DSP2_DATA3_RAM_0: + case WM8962_DSP2_COEFF_RAM_0: + case WM8962_RETUNEADC_SHARED_COEFF_1: + case WM8962_RETUNEADC_SHARED_COEFF_0: + case WM8962_RETUNEDAC_SHARED_COEFF_1: + case WM8962_RETUNEDAC_SHARED_COEFF_0: + case WM8962_SOUNDSTAGE_ENABLES_1: + case WM8962_SOUNDSTAGE_ENABLES_0: + case WM8962_HDBASS_AI_1: + case WM8962_HDBASS_AI_0: + case WM8962_HDBASS_AR_1: + case WM8962_HDBASS_AR_0: + case WM8962_HDBASS_B_1: + case WM8962_HDBASS_B_0: + case WM8962_HDBASS_K_1: + case WM8962_HDBASS_K_0: + case WM8962_HDBASS_N1_1: + case WM8962_HDBASS_N1_0: + case WM8962_HDBASS_N2_1: + case WM8962_HDBASS_N2_0: + case WM8962_HDBASS_N3_1: + case WM8962_HDBASS_N3_0: + case WM8962_HDBASS_N4_1: + case WM8962_HDBASS_N4_0: + case WM8962_HDBASS_N5_1: + case WM8962_HDBASS_N5_0: + case WM8962_HDBASS_X1_1: + case WM8962_HDBASS_X1_0: + case WM8962_HDBASS_X2_1: + case WM8962_HDBASS_X2_0: + case WM8962_HDBASS_X3_1: + case WM8962_HDBASS_X3_0: + case WM8962_HDBASS_ATK_1: + case WM8962_HDBASS_ATK_0: + case WM8962_HDBASS_DCY_1: + case WM8962_HDBASS_DCY_0: + case WM8962_HDBASS_PG_1: + case WM8962_HDBASS_PG_0: + case WM8962_HPF_C_1: + case WM8962_HPF_C_0: + case WM8962_ADCL_RETUNE_C1_1: + case WM8962_ADCL_RETUNE_C1_0: + case WM8962_ADCL_RETUNE_C2_1: + case WM8962_ADCL_RETUNE_C2_0: + case WM8962_ADCL_RETUNE_C3_1: + case WM8962_ADCL_RETUNE_C3_0: + case WM8962_ADCL_RETUNE_C4_1: + case WM8962_ADCL_RETUNE_C4_0: + case WM8962_ADCL_RETUNE_C5_1: + case WM8962_ADCL_RETUNE_C5_0: + case WM8962_ADCL_RETUNE_C6_1: + case WM8962_ADCL_RETUNE_C6_0: + case WM8962_ADCL_RETUNE_C7_1: + case WM8962_ADCL_RETUNE_C7_0: + case WM8962_ADCL_RETUNE_C8_1: + case WM8962_ADCL_RETUNE_C8_0: + case WM8962_ADCL_RETUNE_C9_1: + case WM8962_ADCL_RETUNE_C9_0: + case WM8962_ADCL_RETUNE_C10_1: + case WM8962_ADCL_RETUNE_C10_0: + case WM8962_ADCL_RETUNE_C11_1: + case WM8962_ADCL_RETUNE_C11_0: + case WM8962_ADCL_RETUNE_C12_1: + case WM8962_ADCL_RETUNE_C12_0: + case WM8962_ADCL_RETUNE_C13_1: + case WM8962_ADCL_RETUNE_C13_0: + case WM8962_ADCL_RETUNE_C14_1: + case WM8962_ADCL_RETUNE_C14_0: + case WM8962_ADCL_RETUNE_C15_1: + case WM8962_ADCL_RETUNE_C15_0: + case WM8962_ADCL_RETUNE_C16_1: + case WM8962_ADCL_RETUNE_C16_0: + case WM8962_ADCL_RETUNE_C17_1: + case WM8962_ADCL_RETUNE_C17_0: + case WM8962_ADCL_RETUNE_C18_1: + case WM8962_ADCL_RETUNE_C18_0: + case WM8962_ADCL_RETUNE_C19_1: + case WM8962_ADCL_RETUNE_C19_0: + case WM8962_ADCL_RETUNE_C20_1: + case WM8962_ADCL_RETUNE_C20_0: + case WM8962_ADCL_RETUNE_C21_1: + case WM8962_ADCL_RETUNE_C21_0: + case WM8962_ADCL_RETUNE_C22_1: + case WM8962_ADCL_RETUNE_C22_0: + case WM8962_ADCL_RETUNE_C23_1: + case WM8962_ADCL_RETUNE_C23_0: + case WM8962_ADCL_RETUNE_C24_1: + case WM8962_ADCL_RETUNE_C24_0: + case WM8962_ADCL_RETUNE_C25_1: + case WM8962_ADCL_RETUNE_C25_0: + case WM8962_ADCL_RETUNE_C26_1: + case WM8962_ADCL_RETUNE_C26_0: + case WM8962_ADCL_RETUNE_C27_1: + case WM8962_ADCL_RETUNE_C27_0: + case WM8962_ADCL_RETUNE_C28_1: + case WM8962_ADCL_RETUNE_C28_0: + case WM8962_ADCL_RETUNE_C29_1: + case WM8962_ADCL_RETUNE_C29_0: + case WM8962_ADCL_RETUNE_C30_1: + case WM8962_ADCL_RETUNE_C30_0: + case WM8962_ADCL_RETUNE_C31_1: + case WM8962_ADCL_RETUNE_C31_0: + case WM8962_ADCL_RETUNE_C32_1: + case WM8962_ADCL_RETUNE_C32_0: + case WM8962_RETUNEADC_PG2_1: + case WM8962_RETUNEADC_PG2_0: + case WM8962_RETUNEADC_PG_1: + case WM8962_RETUNEADC_PG_0: + case WM8962_ADCR_RETUNE_C1_1: + case WM8962_ADCR_RETUNE_C1_0: + case WM8962_ADCR_RETUNE_C2_1: + case WM8962_ADCR_RETUNE_C2_0: + case WM8962_ADCR_RETUNE_C3_1: + case WM8962_ADCR_RETUNE_C3_0: + case WM8962_ADCR_RETUNE_C4_1: + case WM8962_ADCR_RETUNE_C4_0: + case WM8962_ADCR_RETUNE_C5_1: + case WM8962_ADCR_RETUNE_C5_0: + case WM8962_ADCR_RETUNE_C6_1: + case WM8962_ADCR_RETUNE_C6_0: + case WM8962_ADCR_RETUNE_C7_1: + case WM8962_ADCR_RETUNE_C7_0: + case WM8962_ADCR_RETUNE_C8_1: + case WM8962_ADCR_RETUNE_C8_0: + case WM8962_ADCR_RETUNE_C9_1: + case WM8962_ADCR_RETUNE_C9_0: + case WM8962_ADCR_RETUNE_C10_1: + case WM8962_ADCR_RETUNE_C10_0: + case WM8962_ADCR_RETUNE_C11_1: + case WM8962_ADCR_RETUNE_C11_0: + case WM8962_ADCR_RETUNE_C12_1: + case WM8962_ADCR_RETUNE_C12_0: + case WM8962_ADCR_RETUNE_C13_1: + case WM8962_ADCR_RETUNE_C13_0: + case WM8962_ADCR_RETUNE_C14_1: + case WM8962_ADCR_RETUNE_C14_0: + case WM8962_ADCR_RETUNE_C15_1: + case WM8962_ADCR_RETUNE_C15_0: + case WM8962_ADCR_RETUNE_C16_1: + case WM8962_ADCR_RETUNE_C16_0: + case WM8962_ADCR_RETUNE_C17_1: + case WM8962_ADCR_RETUNE_C17_0: + case WM8962_ADCR_RETUNE_C18_1: + case WM8962_ADCR_RETUNE_C18_0: + case WM8962_ADCR_RETUNE_C19_1: + case WM8962_ADCR_RETUNE_C19_0: + case WM8962_ADCR_RETUNE_C20_1: + case WM8962_ADCR_RETUNE_C20_0: + case WM8962_ADCR_RETUNE_C21_1: + case WM8962_ADCR_RETUNE_C21_0: + case WM8962_ADCR_RETUNE_C22_1: + case WM8962_ADCR_RETUNE_C22_0: + case WM8962_ADCR_RETUNE_C23_1: + case WM8962_ADCR_RETUNE_C23_0: + case WM8962_ADCR_RETUNE_C24_1: + case WM8962_ADCR_RETUNE_C24_0: + case WM8962_ADCR_RETUNE_C25_1: + case WM8962_ADCR_RETUNE_C25_0: + case WM8962_ADCR_RETUNE_C26_1: + case WM8962_ADCR_RETUNE_C26_0: + case WM8962_ADCR_RETUNE_C27_1: + case WM8962_ADCR_RETUNE_C27_0: + case WM8962_ADCR_RETUNE_C28_1: + case WM8962_ADCR_RETUNE_C28_0: + case WM8962_ADCR_RETUNE_C29_1: + case WM8962_ADCR_RETUNE_C29_0: + case WM8962_ADCR_RETUNE_C30_1: + case WM8962_ADCR_RETUNE_C30_0: + case WM8962_ADCR_RETUNE_C31_1: + case WM8962_ADCR_RETUNE_C31_0: + case WM8962_ADCR_RETUNE_C32_1: + case WM8962_ADCR_RETUNE_C32_0: + case WM8962_DACL_RETUNE_C1_1: + case WM8962_DACL_RETUNE_C1_0: + case WM8962_DACL_RETUNE_C2_1: + case WM8962_DACL_RETUNE_C2_0: + case WM8962_DACL_RETUNE_C3_1: + case WM8962_DACL_RETUNE_C3_0: + case WM8962_DACL_RETUNE_C4_1: + case WM8962_DACL_RETUNE_C4_0: + case WM8962_DACL_RETUNE_C5_1: + case WM8962_DACL_RETUNE_C5_0: + case WM8962_DACL_RETUNE_C6_1: + case WM8962_DACL_RETUNE_C6_0: + case WM8962_DACL_RETUNE_C7_1: + case WM8962_DACL_RETUNE_C7_0: + case WM8962_DACL_RETUNE_C8_1: + case WM8962_DACL_RETUNE_C8_0: + case WM8962_DACL_RETUNE_C9_1: + case WM8962_DACL_RETUNE_C9_0: + case WM8962_DACL_RETUNE_C10_1: + case WM8962_DACL_RETUNE_C10_0: + case WM8962_DACL_RETUNE_C11_1: + case WM8962_DACL_RETUNE_C11_0: + case WM8962_DACL_RETUNE_C12_1: + case WM8962_DACL_RETUNE_C12_0: + case WM8962_DACL_RETUNE_C13_1: + case WM8962_DACL_RETUNE_C13_0: + case WM8962_DACL_RETUNE_C14_1: + case WM8962_DACL_RETUNE_C14_0: + case WM8962_DACL_RETUNE_C15_1: + case WM8962_DACL_RETUNE_C15_0: + case WM8962_DACL_RETUNE_C16_1: + case WM8962_DACL_RETUNE_C16_0: + case WM8962_DACL_RETUNE_C17_1: + case WM8962_DACL_RETUNE_C17_0: + case WM8962_DACL_RETUNE_C18_1: + case WM8962_DACL_RETUNE_C18_0: + case WM8962_DACL_RETUNE_C19_1: + case WM8962_DACL_RETUNE_C19_0: + case WM8962_DACL_RETUNE_C20_1: + case WM8962_DACL_RETUNE_C20_0: + case WM8962_DACL_RETUNE_C21_1: + case WM8962_DACL_RETUNE_C21_0: + case WM8962_DACL_RETUNE_C22_1: + case WM8962_DACL_RETUNE_C22_0: + case WM8962_DACL_RETUNE_C23_1: + case WM8962_DACL_RETUNE_C23_0: + case WM8962_DACL_RETUNE_C24_1: + case WM8962_DACL_RETUNE_C24_0: + case WM8962_DACL_RETUNE_C25_1: + case WM8962_DACL_RETUNE_C25_0: + case WM8962_DACL_RETUNE_C26_1: + case WM8962_DACL_RETUNE_C26_0: + case WM8962_DACL_RETUNE_C27_1: + case WM8962_DACL_RETUNE_C27_0: + case WM8962_DACL_RETUNE_C28_1: + case WM8962_DACL_RETUNE_C28_0: + case WM8962_DACL_RETUNE_C29_1: + case WM8962_DACL_RETUNE_C29_0: + case WM8962_DACL_RETUNE_C30_1: + case WM8962_DACL_RETUNE_C30_0: + case WM8962_DACL_RETUNE_C31_1: + case WM8962_DACL_RETUNE_C31_0: + case WM8962_DACL_RETUNE_C32_1: + case WM8962_DACL_RETUNE_C32_0: + case WM8962_RETUNEDAC_PG2_1: + case WM8962_RETUNEDAC_PG2_0: + case WM8962_RETUNEDAC_PG_1: + case WM8962_RETUNEDAC_PG_0: + case WM8962_DACR_RETUNE_C1_1: + case WM8962_DACR_RETUNE_C1_0: + case WM8962_DACR_RETUNE_C2_1: + case WM8962_DACR_RETUNE_C2_0: + case WM8962_DACR_RETUNE_C3_1: + case WM8962_DACR_RETUNE_C3_0: + case WM8962_DACR_RETUNE_C4_1: + case WM8962_DACR_RETUNE_C4_0: + case WM8962_DACR_RETUNE_C5_1: + case WM8962_DACR_RETUNE_C5_0: + case WM8962_DACR_RETUNE_C6_1: + case WM8962_DACR_RETUNE_C6_0: + case WM8962_DACR_RETUNE_C7_1: + case WM8962_DACR_RETUNE_C7_0: + case WM8962_DACR_RETUNE_C8_1: + case WM8962_DACR_RETUNE_C8_0: + case WM8962_DACR_RETUNE_C9_1: + case WM8962_DACR_RETUNE_C9_0: + case WM8962_DACR_RETUNE_C10_1: + case WM8962_DACR_RETUNE_C10_0: + case WM8962_DACR_RETUNE_C11_1: + case WM8962_DACR_RETUNE_C11_0: + case WM8962_DACR_RETUNE_C12_1: + case WM8962_DACR_RETUNE_C12_0: + case WM8962_DACR_RETUNE_C13_1: + case WM8962_DACR_RETUNE_C13_0: + case WM8962_DACR_RETUNE_C14_1: + case WM8962_DACR_RETUNE_C14_0: + case WM8962_DACR_RETUNE_C15_1: + case WM8962_DACR_RETUNE_C15_0: + case WM8962_DACR_RETUNE_C16_1: + case WM8962_DACR_RETUNE_C16_0: + case WM8962_DACR_RETUNE_C17_1: + case WM8962_DACR_RETUNE_C17_0: + case WM8962_DACR_RETUNE_C18_1: + case WM8962_DACR_RETUNE_C18_0: + case WM8962_DACR_RETUNE_C19_1: + case WM8962_DACR_RETUNE_C19_0: + case WM8962_DACR_RETUNE_C20_1: + case WM8962_DACR_RETUNE_C20_0: + case WM8962_DACR_RETUNE_C21_1: + case WM8962_DACR_RETUNE_C21_0: + case WM8962_DACR_RETUNE_C22_1: + case WM8962_DACR_RETUNE_C22_0: + case WM8962_DACR_RETUNE_C23_1: + case WM8962_DACR_RETUNE_C23_0: + case WM8962_DACR_RETUNE_C24_1: + case WM8962_DACR_RETUNE_C24_0: + case WM8962_DACR_RETUNE_C25_1: + case WM8962_DACR_RETUNE_C25_0: + case WM8962_DACR_RETUNE_C26_1: + case WM8962_DACR_RETUNE_C26_0: + case WM8962_DACR_RETUNE_C27_1: + case WM8962_DACR_RETUNE_C27_0: + case WM8962_DACR_RETUNE_C28_1: + case WM8962_DACR_RETUNE_C28_0: + case WM8962_DACR_RETUNE_C29_1: + case WM8962_DACR_RETUNE_C29_0: + case WM8962_DACR_RETUNE_C30_1: + case WM8962_DACR_RETUNE_C30_0: + case WM8962_DACR_RETUNE_C31_1: + case WM8962_DACR_RETUNE_C31_0: + case WM8962_DACR_RETUNE_C32_1: + case WM8962_DACR_RETUNE_C32_0: + case WM8962_VSS_XHD2_1: + case WM8962_VSS_XHD2_0: + case WM8962_VSS_XHD3_1: + case WM8962_VSS_XHD3_0: + case WM8962_VSS_XHN1_1: + case WM8962_VSS_XHN1_0: + case WM8962_VSS_XHN2_1: + case WM8962_VSS_XHN2_0: + case WM8962_VSS_XHN3_1: + case WM8962_VSS_XHN3_0: + case WM8962_VSS_XLA_1: + case WM8962_VSS_XLA_0: + case WM8962_VSS_XLB_1: + case WM8962_VSS_XLB_0: + case WM8962_VSS_XLG_1: + case WM8962_VSS_XLG_0: + case WM8962_VSS_PG2_1: + case WM8962_VSS_PG2_0: + case WM8962_VSS_PG_1: + case WM8962_VSS_PG_0: + case WM8962_VSS_XTD1_1: + case WM8962_VSS_XTD1_0: + case WM8962_VSS_XTD2_1: + case WM8962_VSS_XTD2_0: + case WM8962_VSS_XTD3_1: + case WM8962_VSS_XTD3_0: + case WM8962_VSS_XTD4_1: + case WM8962_VSS_XTD4_0: + case WM8962_VSS_XTD5_1: + case WM8962_VSS_XTD5_0: + case WM8962_VSS_XTD6_1: + case WM8962_VSS_XTD6_0: + case WM8962_VSS_XTD7_1: + case WM8962_VSS_XTD7_0: + case WM8962_VSS_XTD8_1: + case WM8962_VSS_XTD8_0: + case WM8962_VSS_XTD9_1: + case WM8962_VSS_XTD9_0: + case WM8962_VSS_XTD10_1: + case WM8962_VSS_XTD10_0: + case WM8962_VSS_XTD11_1: + case WM8962_VSS_XTD11_0: + case WM8962_VSS_XTD12_1: + case WM8962_VSS_XTD12_0: + case WM8962_VSS_XTD13_1: + case WM8962_VSS_XTD13_0: + case WM8962_VSS_XTD14_1: + case WM8962_VSS_XTD14_0: + case WM8962_VSS_XTD15_1: + case WM8962_VSS_XTD15_0: + case WM8962_VSS_XTD16_1: + case WM8962_VSS_XTD16_0: + case WM8962_VSS_XTD17_1: + case WM8962_VSS_XTD17_0: + case WM8962_VSS_XTD18_1: + case WM8962_VSS_XTD18_0: + case WM8962_VSS_XTD19_1: + case WM8962_VSS_XTD19_0: + case WM8962_VSS_XTD20_1: + case WM8962_VSS_XTD20_0: + case WM8962_VSS_XTD21_1: + case WM8962_VSS_XTD21_0: + case WM8962_VSS_XTD22_1: + case WM8962_VSS_XTD22_0: + case WM8962_VSS_XTD23_1: + case WM8962_VSS_XTD23_0: + case WM8962_VSS_XTD24_1: + case WM8962_VSS_XTD24_0: + case WM8962_VSS_XTD25_1: + case WM8962_VSS_XTD25_0: + case WM8962_VSS_XTD26_1: + case WM8962_VSS_XTD26_0: + case WM8962_VSS_XTD27_1: + case WM8962_VSS_XTD27_0: + case WM8962_VSS_XTD28_1: + case WM8962_VSS_XTD28_0: + case WM8962_VSS_XTD29_1: + case WM8962_VSS_XTD29_0: + case WM8962_VSS_XTD30_1: + case WM8962_VSS_XTD30_0: + case WM8962_VSS_XTD31_1: + case WM8962_VSS_XTD31_0: + case WM8962_VSS_XTD32_1: + case WM8962_VSS_XTD32_0: + case WM8962_VSS_XTS1_1: + case WM8962_VSS_XTS1_0: + case WM8962_VSS_XTS2_1: + case WM8962_VSS_XTS2_0: + case WM8962_VSS_XTS3_1: + case WM8962_VSS_XTS3_0: + case WM8962_VSS_XTS4_1: + case WM8962_VSS_XTS4_0: + case WM8962_VSS_XTS5_1: + case WM8962_VSS_XTS5_0: + case WM8962_VSS_XTS6_1: + case WM8962_VSS_XTS6_0: + case WM8962_VSS_XTS7_1: + case WM8962_VSS_XTS7_0: + case WM8962_VSS_XTS8_1: + case WM8962_VSS_XTS8_0: + case WM8962_VSS_XTS9_1: + case WM8962_VSS_XTS9_0: + case WM8962_VSS_XTS10_1: + case WM8962_VSS_XTS10_0: + case WM8962_VSS_XTS11_1: + case WM8962_VSS_XTS11_0: + case WM8962_VSS_XTS12_1: + case WM8962_VSS_XTS12_0: + case WM8962_VSS_XTS13_1: + case WM8962_VSS_XTS13_0: + case WM8962_VSS_XTS14_1: + case WM8962_VSS_XTS14_0: + case WM8962_VSS_XTS15_1: + case WM8962_VSS_XTS15_0: + case WM8962_VSS_XTS16_1: + case WM8962_VSS_XTS16_0: + case WM8962_VSS_XTS17_1: + case WM8962_VSS_XTS17_0: + case WM8962_VSS_XTS18_1: + case WM8962_VSS_XTS18_0: + case WM8962_VSS_XTS19_1: + case WM8962_VSS_XTS19_0: + case WM8962_VSS_XTS20_1: + case WM8962_VSS_XTS20_0: + case WM8962_VSS_XTS21_1: + case WM8962_VSS_XTS21_0: + case WM8962_VSS_XTS22_1: + case WM8962_VSS_XTS22_0: + case WM8962_VSS_XTS23_1: + case WM8962_VSS_XTS23_0: + case WM8962_VSS_XTS24_1: + case WM8962_VSS_XTS24_0: + case WM8962_VSS_XTS25_1: + case WM8962_VSS_XTS25_0: + case WM8962_VSS_XTS26_1: + case WM8962_VSS_XTS26_0: + case WM8962_VSS_XTS27_1: + case WM8962_VSS_XTS27_0: + case WM8962_VSS_XTS28_1: + case WM8962_VSS_XTS28_0: + case WM8962_VSS_XTS29_1: + case WM8962_VSS_XTS29_0: + case WM8962_VSS_XTS30_1: + case WM8962_VSS_XTS30_0: + case WM8962_VSS_XTS31_1: + case WM8962_VSS_XTS31_0: + case WM8962_VSS_XTS32_1: + case WM8962_VSS_XTS32_0: + return true; + default: + return false; + } } static int wm8962_reset(struct wm8962_priv *wm8962) -- cgit v1.2.3-18-g5258 From b33005f3ef6a85be3202ee1b8a2513ed1ef4019d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 6 Jan 2012 11:30:10 +0800 Subject: ASoC: jz4740: Convert qi_lb60 to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/jz4740/qi_lb60.c | 56 +++++++++++++++++++++------------------------- 1 file changed, 26 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 0097c3b13a1..e8aaff18d7c 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -91,56 +91,52 @@ static struct snd_soc_card qi_lb60 = { .num_dapm_routes = ARRAY_SIZE(qi_lb60_routes), }; -static struct platform_device *qi_lb60_snd_device; - static const struct gpio qi_lb60_gpios[] = { { QI_LB60_SND_GPIO, GPIOF_OUT_INIT_LOW, "SND" }, { QI_LB60_AMP_GPIO, GPIOF_OUT_INIT_LOW, "AMP" }, }; -static int __init qi_lb60_init(void) +static int __devinit qi_lb60_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &qi_lb60; int ret; - qi_lb60_snd_device = platform_device_alloc("soc-audio", -1); - - if (!qi_lb60_snd_device) - return -ENOMEM; - ret = gpio_request_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); - if (ret) { - pr_err("qi_lb60 snd: Failed to request gpios: %d\n", ret); - goto err_device_put; - } + if (ret) + return ret; - platform_set_drvdata(qi_lb60_snd_device, &qi_lb60); + card->dev = &pdev->dev; - ret = platform_device_add(qi_lb60_snd_device); + ret = snd_soc_register_card(card); if (ret) { - pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret); - goto err_unset_pdata; + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); } - - return 0; - -err_unset_pdata: - platform_set_drvdata(qi_lb60_snd_device, NULL); -/*err_gpio_free_array:*/ - gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); -err_device_put: - platform_device_put(qi_lb60_snd_device); - return ret; } -module_init(qi_lb60_init); -static void __exit qi_lb60_exit(void) +static int __devexit qi_lb60_remove(struct platform_device *pdev) { - platform_device_unregister(qi_lb60_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); + return 0; } -module_exit(qi_lb60_exit); + +static struct platform_driver qi_lb60_driver = { + .driver = { + .name = "qi-lb60-audio", + .owner = THIS_MODULE, + }, + .probe = qi_lb60_probe, + .remove = __devexit_p(qi_lb60_remove), +}; + +module_platform_driver(qi_lb60_driver); MODULE_AUTHOR("Lars-Peter Clausen "); MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:qi-lb60-audio"); -- cgit v1.2.3-18-g5258 From d19fd5db3e6b72e84c2012a28f6d7cf6f737193a Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 16 Jan 2012 10:44:13 +0000 Subject: ASoC: wm8983: Remove useless snd_kcontrol This must be a leftover from a previous driver. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index cebde568d19..367388fdc48 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -249,9 +249,6 @@ static const char *eq5_cutoff_text[] = { static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5, eq5_cutoff_text); -static const char *speaker_mode_text[] = { "Class A/B", "Class D" }; -static const SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text); - static const char *depth_3d_text[] = { "Off", "6.67%", @@ -369,8 +366,6 @@ static const struct snd_kcontrol_new wm8983_snd_controls[] = { SOC_SINGLE_TLV("EQ5 Volume", WM8983_EQ5_HIGH_SHELF, 0, 24, 1, eq_tlv), SOC_ENUM("3D Depth", depth_3d), - - SOC_ENUM("Speaker Mode", speaker_mode) }; static const struct snd_kcontrol_new left_out_mixer[] = { -- cgit v1.2.3-18-g5258 From 5f52ee48751e63ed555b56a82db446745f60bc82 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jan 2012 16:31:00 -0800 Subject: ASoC: Add WM8962 DAC and ADC L/R swap support Simple switches since there's no per-channel control. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a20e2b7ab26..cc4049e9174 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1714,6 +1714,8 @@ SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8962_LEFT_DAC_VOLUME, WM8962_RIGHT_DAC_VOLUME, 1, 127, 0, digital_tlv), SOC_SINGLE("DAC High Performance Switch", WM8962_ADC_DAC_CONTROL_2, 0, 1, 0), +SOC_SINGLE("DAC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 5, 1, 0), +SOC_SINGLE("ADC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 8, 1, 0), SOC_SINGLE("ADC High Performance Switch", WM8962_ADDITIONAL_CONTROL_1, 5, 1, 0), -- cgit v1.2.3-18-g5258 From 58ba9b25454fe9b6ded804f69cb7ed4500b685fc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Jan 2012 18:38:51 +0000 Subject: ASoC: Allow drivers to specify how many bits are significant on a DAI Most devices accept data in formats that don't correspond directly to their internal format. ALSA allows us to set a msbits constraint which tells userspace about this in case it finds it useful (for example, in order to avoid wasting effort dithering bits that will be ignored when raising the sample size of data) so provide a mechanism for drivers to specify the number of bits that are actually significant on a DAI and add the appropriate constraints along with all the others. This is done slightly awkwardly as the constraint is specified per sample size - we loop over every possible sample size, including ones that the device doesn't support and including ones that have fewer bits than are actually used, but this is harmless as the upper layers do the right thing in these cases. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-pcm.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cdc860a5ff3..8bb17937d59 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -62,6 +62,39 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, return 0; } +/* + * List of sample sizes that might go over the bus for parameter + * application. There ought to be a wildcard sample size for things + * like the DAC/ADC resolution to use but there isn't right now. + */ +static int sample_sizes[] = { + 8, 16, 24, 32, +}; + +static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret, i, bits; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + bits = dai->driver->playback.sig_bits; + else + bits = dai->driver->capture.sig_bits; + + if (!bits) + return; + + for (i = 0; i < ARRAY_SIZE(sample_sizes); i++) { + ret = snd_pcm_hw_constraint_msbits(substream->runtime, + 0, sample_sizes[i], + bits); + if (ret != 0) + dev_warn(dai->dev, + "Failed to set MSB %d/%d: %d\n", + bits, sample_sizes[i], ret); + } +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -187,6 +220,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } + soc_pcm_apply_msb(substream, codec_dai); + soc_pcm_apply_msb(substream, cpu_dai); + /* Symmetry only applies if we've already got an active stream. */ if (cpu_dai->active) { ret = soc_pcm_apply_symmetry(substream, cpu_dai); -- cgit v1.2.3-18-g5258 From a4b5233792443a2caed0db91003e5a75c50bc6c9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Jan 2012 18:39:21 +0000 Subject: ASoC: 24 bits are significant on the WM8996 audio interfaces Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 86f5b6bd7af..8e8f8d1fef9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3086,6 +3086,7 @@ static struct snd_soc_dai_driver wm8996_dai[] = { .channels_max = 6, .rates = WM8996_RATES, .formats = WM8996_FORMATS, + .sig_bits = 24, }, .capture = { .stream_name = "AIF1 Capture", @@ -3093,6 +3094,7 @@ static struct snd_soc_dai_driver wm8996_dai[] = { .channels_max = 6, .rates = WM8996_RATES, .formats = WM8996_FORMATS, + .sig_bits = 24, }, .ops = &wm8996_dai_ops, }, @@ -3104,6 +3106,7 @@ static struct snd_soc_dai_driver wm8996_dai[] = { .channels_max = 2, .rates = WM8996_RATES, .formats = WM8996_FORMATS, + .sig_bits = 24, }, .capture = { .stream_name = "AIF2 Capture", @@ -3111,6 +3114,7 @@ static struct snd_soc_dai_driver wm8996_dai[] = { .channels_max = 2, .rates = WM8996_RATES, .formats = WM8996_FORMATS, + .sig_bits = 24, }, .ops = &wm8996_dai_ops, }, -- cgit v1.2.3-18-g5258 From 164548d3b3733b10990274e1e92848656e9d6d1e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 17 Jan 2012 16:42:05 +0000 Subject: ASoC: Implement basic WM8993 interrupt support If an interrupt is supplied then use it for thermal warning and FLL lock notifications. When using the interrupt raise the timeout for the FLL lock substantially to reduce the chances of spurious warnings. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 94 ++++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 85 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index e7ae9fda3f5..eca93521124 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -204,9 +204,11 @@ static struct { struct wm8993_priv { struct wm_hubs_data hubs_data; + struct device *dev; struct regmap *regmap; struct regulator_bulk_data supplies[WM8993_NUM_SUPPLIES]; struct wm8993_platform_data pdata; + struct completion fll_lock; int master; int sysclk_source; int tdm_slots; @@ -225,6 +227,7 @@ static bool wm8993_volatile(struct device *dev, unsigned int reg) { switch (reg) { case WM8993_SOFTWARE_RESET: + case WM8993_GPIO_CTRL_1: case WM8993_DC_SERVO_0: case WM8993_DC_SERVO_READBACK_0: case WM8993_DC_SERVO_READBACK_1: @@ -467,8 +470,10 @@ static int _wm8993_set_fll(struct snd_soc_codec *codec, int fll_id, int source, unsigned int Fref, unsigned int Fout) { struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); + struct i2c_client *i2c = to_i2c_client(codec->dev); u16 reg1, reg4, reg5; struct _fll_div fll_div; + unsigned int timeout; int ret; /* Any change? */ @@ -539,14 +544,22 @@ static int _wm8993_set_fll(struct snd_soc_codec *codec, int fll_id, int source, reg5 |= fll_div.fll_clk_ref_div << WM8993_FLL_CLK_REF_DIV_SHIFT; snd_soc_write(codec, WM8993_FLL_CONTROL_5, reg5); + /* If we've got an interrupt wired up make sure we get it */ + if (i2c->irq) + timeout = msecs_to_jiffies(20); + else if (Fref < 1000000) + timeout = msecs_to_jiffies(3); + else + timeout = msecs_to_jiffies(1); + + try_wait_for_completion(&wm8993->fll_lock); + /* Enable the FLL */ snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); - /* Both overestimates */ - if (Fref < 1000000) - msleep(3); - else - msleep(1); + timeout = wait_for_completion_timeout(&wm8993->fll_lock, timeout); + if (i2c->irq && !timeout) + dev_warn(codec->dev, "Timed out waiting for FLL\n"); dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); @@ -1471,6 +1484,45 @@ out: return 0; } +static irqreturn_t wm8993_irq(int irq, void *data) +{ + struct wm8993_priv *wm8993 = data; + int mask, val, ret; + + ret = regmap_read(wm8993->regmap, WM8993_GPIO_CTRL_1, &val); + if (ret != 0) { + dev_err(wm8993->dev, "Failed to read interrupt status: %d\n", + ret); + return IRQ_NONE; + } + + ret = regmap_read(wm8993->regmap, WM8993_GPIOCTRL_2, &mask); + if (ret != 0) { + dev_err(wm8993->dev, "Failed to read interrupt mask: %d\n", + ret); + return IRQ_NONE; + } + + /* The IRQ pin status is visible in the register too */ + val &= ~(mask | WM8993_IRQ); + if (!val) + return IRQ_NONE; + + if (val & WM8993_TEMPOK_EINT) + dev_crit(wm8993->dev, "Thermal warning\n"); + + if (val & WM8993_FLL_LOCK_EINT) { + dev_dbg(wm8993->dev, "FLL locked\n"); + complete(&wm8993->fll_lock); + } + + ret = regmap_write(wm8993->regmap, WM8993_GPIO_CTRL_1, val); + if (ret != 0) + dev_err(wm8993->dev, "Failed to ack interrupt: %d\n", ret); + + return IRQ_HANDLED; +} + static const struct snd_soc_dai_ops wm8993_ops = { .set_sysclk = wm8993_set_sysclk, .set_fmt = wm8993_set_dai_fmt, @@ -1671,6 +1723,9 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, if (wm8993 == NULL) return -ENOMEM; + wm8993->dev = &i2c->dev; + init_completion(&wm8993->fll_lock); + wm8993->regmap = regmap_init_i2c(i2c, &wm8993_regmap); if (IS_ERR(wm8993->regmap)) { ret = PTR_ERR(wm8993->regmap); @@ -1713,6 +1768,22 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_enable; + if (i2c->irq) { + /* Put GPIO1 into interrupt mode (only GPIO1 can output IRQ) */ + ret = regmap_update_bits(wm8993->regmap, WM8993_GPIO1, + WM8993_GPIO1_PD | + WM8993_GPIO1_SEL_MASK, 7); + if (ret != 0) + goto err_enable; + + ret = request_threaded_irq(i2c->irq, NULL, wm8993_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, + "wm8993", wm8993); + if (ret != 0) + goto err_enable; + + } + regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); regcache_cache_only(wm8993->regmap, true); @@ -1721,11 +1792,14 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, &soc_codec_dev_wm8993, &wm8993_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - goto err_enable; + goto err_irq; } return 0; +err_irq: + if (i2c->irq) + free_irq(i2c->irq, wm8993); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); err_get: @@ -1735,11 +1809,13 @@ err: return ret; } -static __devexit int wm8993_i2c_remove(struct i2c_client *client) +static __devexit int wm8993_i2c_remove(struct i2c_client *i2c) { - struct wm8993_priv *wm8993 = i2c_get_clientdata(client); + struct wm8993_priv *wm8993 = i2c_get_clientdata(i2c); - snd_soc_unregister_codec(&client->dev); + snd_soc_unregister_codec(&i2c->dev); + if (i2c->irq) + free_irq(i2c->irq, wm8993); regmap_exit(wm8993->regmap); regulator_bulk_disable(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); -- cgit v1.2.3-18-g5258 From 85f883933cd7353ab2227d5d2041312dce323e6b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 17 Jan 2012 15:08:35 +0000 Subject: ASoC: Make WM8993 I2C usage unconditional The WM8993 only supports I2C so don't ifdef the I2C support in the driver. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 6 ------ 1 file changed, 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index eca93521124..5502543b8a2 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1710,7 +1710,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8993 = { .set_bias_level = wm8993_set_bias_level, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1838,27 +1837,22 @@ static struct i2c_driver wm8993_i2c_driver = { .remove = __devexit_p(wm8993_i2c_remove), .id_table = wm8993_i2c_id, }; -#endif static int __init wm8993_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8993_i2c_driver); if (ret != 0) { pr_err("WM8993: Unable to register I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8993_modinit); static void __exit wm8993_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8993_i2c_driver); -#endif } module_exit(wm8993_exit); -- cgit v1.2.3-18-g5258 From 99b0292d94975429eacc0c1ce4c985f7207394ba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 17 Jan 2012 11:50:26 +0000 Subject: ASoC: 24 bits are significant on wm_hubs DAIs Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 2 ++ sound/soc/codecs/wm8994.c | 8 +++++++- 2 files changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 5502543b8a2..dd687c3a84f 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1547,6 +1547,7 @@ static struct snd_soc_dai_driver wm8993_dai = { .channels_max = 2, .rates = WM8993_RATES, .formats = WM8993_FORMATS, + .sig_bits = 24, }, .capture = { .stream_name = "Capture", @@ -1554,6 +1555,7 @@ static struct snd_soc_dai_driver wm8993_dai = { .channels_max = 2, .rates = WM8993_RATES, .formats = WM8993_FORMATS, + .sig_bits = 24, }, .ops = &wm8993_ops, .symmetric_rates = 1, diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 93d27b66025..b047bfada70 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2645,6 +2645,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, + .sig_bits = 24, }, .capture = { .stream_name = "AIF1 Capture", @@ -2652,6 +2653,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, + .sig_bits = 24, }, .ops = &wm8994_aif1_dai_ops, }, @@ -2664,6 +2666,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, + .sig_bits = 24, }, .capture = { .stream_name = "AIF2 Capture", @@ -2671,6 +2674,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, + .sig_bits = 24, }, .probe = wm8994_aif2_probe, .ops = &wm8994_aif2_dai_ops, @@ -2684,6 +2688,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, + .sig_bits = 24, }, .capture = { .stream_name = "AIF3 Capture", @@ -2691,7 +2696,8 @@ static struct snd_soc_dai_driver wm8994_dai[] = { .channels_max = 2, .rates = WM8994_RATES, .formats = WM8994_FORMATS, - }, + .sig_bits = 24, + }, .ops = &wm8994_aif3_dai_ops, } }; -- cgit v1.2.3-18-g5258 From 2688738ebac66b4e276321248eb3e12d59cbcd7f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 17 Jan 2012 19:18:27 +0000 Subject: ASoC: When releasing WM5100 put /RESET into reset Reset is active low, make sure we leave it asserted when release the device. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 66f0611e68b..c112f5eaa11 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2764,7 +2764,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, err_reset: wm5100_free_gpio(i2c); if (wm5100->pdata.reset) { - gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_set_value_cansleep(wm5100->pdata.reset, 0); gpio_free(wm5100->pdata.reset); } err_ldo: @@ -2797,7 +2797,7 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); wm5100_free_gpio(client); if (wm5100->pdata.reset) { - gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_set_value_cansleep(wm5100->pdata.reset, 0); gpio_free(wm5100->pdata.reset); } if (wm5100->pdata.ldo_ena) { -- cgit v1.2.3-18-g5258 From 0132615da5bd97c74e4a015721039ef17a4841de Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 17 Jan 2012 19:27:04 +0000 Subject: ASoC: Say we can't read WM5100 ID register Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index c112f5eaa11..de8604229bc 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2702,7 +2702,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, ret = regmap_read(wm5100->regmap, WM5100_SOFTWARE_RESET, ®); if (ret < 0) { - dev_err(&i2c->dev, "Failed to read ID register\n"); + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); goto err_reset; } switch (reg) { -- cgit v1.2.3-18-g5258 From f2c6e757f60d3d3e90dc5d1f1ff1a241dc0ea916 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 16 Jan 2012 17:32:21 -0200 Subject: ASoC: sgtl5000: Print revision number in hex Throughout the sgtl5000 driver source code and also in the sgtl5000 datasheet the revision code is shown in hexadecimal. Print it hex format, for consistency. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7f4ba819a9f..04ea4850cd4 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1248,7 +1248,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - dev_info(codec->dev, "sgtl5000 revision %d\n", rev); + dev_info(codec->dev, "sgtl5000 revision 0x%x\n", rev); /* * workaround for revision 0x11 and later, -- cgit v1.2.3-18-g5258 From 8d725b2bcb82ff46236bf745c9ab7cc4dde74699 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 18 Jan 2012 12:18:25 +0100 Subject: ASoC: tlv320dac33: Use core to set the msbits constraint Core can set the msbits constraint in behalf of the dai. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f0aad26cdb3..21ccf0a616a 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -806,8 +806,6 @@ static int dac33_startup(struct snd_pcm_substream *substream, /* Stream started, save the substream pointer */ dac33->substream = substream; - snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); - return 0; } @@ -1516,6 +1514,7 @@ static struct snd_soc_dai_driver dac33_dai = { .channels_max = 2, .rates = DAC33_RATES, .formats = DAC33_FORMATS,}, + .sig_bits = 24, .ops = &dac33_dai_ops, }; -- cgit v1.2.3-18-g5258 From 8819f65ceca118561c9d59a04cfd91625c74a262 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 18 Jan 2012 12:18:26 +0100 Subject: ASoC: twl4030: Use core to set the msbits constraint Core can set the msbits constraint in behalf of the dai. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 18e71014cc2..a193f5fa4b3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1689,7 +1689,6 @@ static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); if (twl4030->master_substream) { twl4030->slave_substream = substream; /* The DAI has one configuration for playback and capture, so @@ -2175,13 +2174,15 @@ static struct snd_soc_dai_driver twl4030_dai[] = { .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES | SNDRV_PCM_RATE_96000, - .formats = TWL4030_FORMATS,}, + .formats = TWL4030_FORMATS, + .sig_bits = 24,}, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 4, .rates = TWL4030_RATES, - .formats = TWL4030_FORMATS,}, + .formats = TWL4030_FORMATS, + .sig_bits = 24,}, .ops = &twl4030_dai_hifi_ops, }, { -- cgit v1.2.3-18-g5258 From 7df6f2551f2255f55adab7338a3a1e6594a12f63 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 18 Jan 2012 12:18:23 +0100 Subject: ASoC: omap-dmic: Use core to set the msbits constraint Core can set the msbits constraint in behalf of the dai. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 0855c1cfa7f..4dcb5a7e40e 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -113,12 +113,10 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream, mutex_lock(&dmic->mutex); - if (!dai->active) { - snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); + if (!dai->active) dmic->active = 1; - } else { + else ret = -EBUSY; - } mutex_unlock(&dmic->mutex); @@ -445,6 +443,7 @@ static struct snd_soc_dai_driver omap_dmic_dai = { .channels_max = 6, .rates = SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, }, .ops = &omap_dmic_dai_ops, }; -- cgit v1.2.3-18-g5258 From b4badd4960c9937fcbdcab2a57f0dd7e4ad45c8e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 18 Jan 2012 12:18:24 +0100 Subject: ASoC: omap-mcpdm: Set 24msbits constraint McPDM internal FIFO is 24 bit wide. From the 32 bit sample 8 bit is discarded. Let application know about this via msbits constraint. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 0e25df4fa9e..39705561131 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -419,12 +419,14 @@ static struct snd_soc_dai_driver omap_mcpdm_dai = { .channels_max = 5, .rates = OMAP_MCPDM_RATES, .formats = OMAP_MCPDM_FORMATS, + .sig_bits = 24, }, .capture = { .channels_min = 1, .channels_max = 3, .rates = OMAP_MCPDM_RATES, .formats = OMAP_MCPDM_FORMATS, + .sig_bits = 24, }, .ops = &omap_mcpdm_dai_ops, }; -- cgit v1.2.3-18-g5258 From 218240e27f89b477564a638ff77d45147e42a8fd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Jan 2012 13:47:02 +0000 Subject: ASoC: Remove redundant set_bias_level() from WM5100 remove() The framework should bring the device down before it calls the driver. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index de8604229bc..8ea2089f7aa 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2562,7 +2562,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - wm5100_set_bias_level(codec, SND_SOC_BIAS_OFF); if (wm5100->pdata.hp_pol) { gpio_free(wm5100->pdata.hp_pol); } -- cgit v1.2.3-18-g5258 From 46c1a877c6fc29519760a3aaedf807332cd8a781 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Jan 2012 14:53:08 +0000 Subject: ASoC: Make WM5100 interrupt path use regmap directly This will allow us to move the interrupt allocation out of the ASoC part of the driver and simplifies the locking by removing any reliance in the bulk of the interrupt path on the big CODEC lock. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 188 ++++++++++++++++++++++++++-------------------- 1 file changed, 108 insertions(+), 80 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8ea2089f7aa..4b2c724ed9b 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -50,6 +50,7 @@ struct wm5100_fll { /* codec private data */ struct wm5100_priv { + struct device *dev; struct regmap *regmap; struct snd_soc_codec *codec; @@ -855,48 +856,48 @@ static int wm5100_dbvdd_ev(struct snd_soc_dapm_widget *w, } } -static void wm5100_log_status3(struct snd_soc_codec *codec, int val) +static void wm5100_log_status3(struct wm5100_priv *wm5100, int val) { if (val & WM5100_SPK_SHUTDOWN_WARN_EINT) - dev_crit(codec->dev, "Speaker shutdown warning\n"); + dev_crit(wm5100->dev, "Speaker shutdown warning\n"); if (val & WM5100_SPK_SHUTDOWN_EINT) - dev_crit(codec->dev, "Speaker shutdown\n"); + dev_crit(wm5100->dev, "Speaker shutdown\n"); if (val & WM5100_CLKGEN_ERR_EINT) - dev_crit(codec->dev, "SYSCLK underclocked\n"); + dev_crit(wm5100->dev, "SYSCLK underclocked\n"); if (val & WM5100_CLKGEN_ERR_ASYNC_EINT) - dev_crit(codec->dev, "ASYNCCLK underclocked\n"); + dev_crit(wm5100->dev, "ASYNCCLK underclocked\n"); } -static void wm5100_log_status4(struct snd_soc_codec *codec, int val) +static void wm5100_log_status4(struct wm5100_priv *wm5100, int val) { if (val & WM5100_AIF3_ERR_EINT) - dev_err(codec->dev, "AIF3 configuration error\n"); + dev_err(wm5100->dev, "AIF3 configuration error\n"); if (val & WM5100_AIF2_ERR_EINT) - dev_err(codec->dev, "AIF2 configuration error\n"); + dev_err(wm5100->dev, "AIF2 configuration error\n"); if (val & WM5100_AIF1_ERR_EINT) - dev_err(codec->dev, "AIF1 configuration error\n"); + dev_err(wm5100->dev, "AIF1 configuration error\n"); if (val & WM5100_CTRLIF_ERR_EINT) - dev_err(codec->dev, "Control interface error\n"); + dev_err(wm5100->dev, "Control interface error\n"); if (val & WM5100_ISRC2_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ISRC2 underclocked\n"); + dev_err(wm5100->dev, "ISRC2 underclocked\n"); if (val & WM5100_ISRC1_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ISRC1 underclocked\n"); + dev_err(wm5100->dev, "ISRC1 underclocked\n"); if (val & WM5100_FX_UNDERCLOCKED_EINT) - dev_err(codec->dev, "FX underclocked\n"); + dev_err(wm5100->dev, "FX underclocked\n"); if (val & WM5100_AIF3_UNDERCLOCKED_EINT) - dev_err(codec->dev, "AIF3 underclocked\n"); + dev_err(wm5100->dev, "AIF3 underclocked\n"); if (val & WM5100_AIF2_UNDERCLOCKED_EINT) - dev_err(codec->dev, "AIF2 underclocked\n"); + dev_err(wm5100->dev, "AIF2 underclocked\n"); if (val & WM5100_AIF1_UNDERCLOCKED_EINT) - dev_err(codec->dev, "AIF1 underclocked\n"); + dev_err(wm5100->dev, "AIF1 underclocked\n"); if (val & WM5100_ASRC_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ASRC underclocked\n"); + dev_err(wm5100->dev, "ASRC underclocked\n"); if (val & WM5100_DAC_UNDERCLOCKED_EINT) - dev_err(codec->dev, "DAC underclocked\n"); + dev_err(wm5100->dev, "DAC underclocked\n"); if (val & WM5100_ADC_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ADC underclocked\n"); + dev_err(wm5100->dev, "ADC underclocked\n"); if (val & WM5100_MIXER_UNDERCLOCKED_EINT) - dev_err(codec->dev, "Mixer underclocked\n"); + dev_err(wm5100->dev, "Mixer underclocked\n"); } static int wm5100_post_ev(struct snd_soc_dapm_widget *w, @@ -904,16 +905,17 @@ static int wm5100_post_ev(struct snd_soc_dapm_widget *w, int event) { struct snd_soc_codec *codec = w->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); int ret; ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_3); ret &= WM5100_SPK_SHUTDOWN_WARN_STS | WM5100_SPK_SHUTDOWN_STS | WM5100_CLKGEN_ERR_STS | WM5100_CLKGEN_ERR_ASYNC_STS; - wm5100_log_status3(codec, ret); + wm5100_log_status3(wm5100, ret); ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_4); - wm5100_log_status4(codec, ret); + wm5100_log_status4(wm5100, ret); return 0; } @@ -2123,55 +2125,59 @@ static int wm5100_dig_vu[] = { WM5100_DAC_DIGITAL_VOLUME_6R, }; -static void wm5100_set_detect_mode(struct snd_soc_codec *codec, int the_mode) +static void wm5100_set_detect_mode(struct wm5100_priv *wm5100, int the_mode) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); struct wm5100_jack_mode *mode = &wm5100->pdata.jack_modes[the_mode]; BUG_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes)); gpio_set_value_cansleep(wm5100->pdata.hp_pol, mode->hp_pol); - snd_soc_update_bits(codec, WM5100_ACCESSORY_DETECT_MODE_1, - WM5100_ACCDET_BIAS_SRC_MASK | - WM5100_ACCDET_SRC, - (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | - mode->micd_src << WM5100_ACCDET_SRC_SHIFT); - snd_soc_update_bits(codec, WM5100_MISC_CONTROL, - WM5100_HPCOM_SRC, - mode->micd_src << WM5100_HPCOM_SRC_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_ACCESSORY_DETECT_MODE_1, + WM5100_ACCDET_BIAS_SRC_MASK | + WM5100_ACCDET_SRC, + (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | + mode->micd_src << WM5100_ACCDET_SRC_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_MISC_CONTROL, + WM5100_HPCOM_SRC, + mode->micd_src << WM5100_HPCOM_SRC_SHIFT); wm5100->jack_mode = the_mode; - dev_dbg(codec->dev, "Set microphone polarity to %d\n", + dev_dbg(wm5100->dev, "Set microphone polarity to %d\n", wm5100->jack_mode); } -static void wm5100_micd_irq(struct snd_soc_codec *codec) +static void wm5100_micd_irq(struct wm5100_priv *wm5100) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - int val; + unsigned int val; + int ret; - val = snd_soc_read(codec, WM5100_MIC_DETECT_3); + ret = regmap_read(wm5100->regmap, WM5100_MIC_DETECT_3, &val); + if (ret != 0) { + dev_err(wm5100->dev, "Failed to read micropone status: %d\n", + ret); + return; + } - dev_dbg(codec->dev, "Microphone event: %x\n", val); + dev_dbg(wm5100->dev, "Microphone event: %x\n", val); if (!(val & WM5100_ACCDET_VALID)) { - dev_warn(codec->dev, "Microphone detection state invalid\n"); + dev_warn(wm5100->dev, "Microphone detection state invalid\n"); return; } /* No accessory, reset everything and report removal */ if (!(val & WM5100_ACCDET_STS)) { - dev_dbg(codec->dev, "Jack removal detected\n"); + dev_dbg(wm5100->dev, "Jack removal detected\n"); wm5100->jack_mic = false; wm5100->jack_detecting = true; snd_soc_jack_report(wm5100->jack, 0, SND_JACK_LINEOUT | SND_JACK_HEADSET | SND_JACK_BTN_0); - snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - WM5100_ACCDET_RATE_MASK); + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + WM5100_ACCDET_RATE_MASK); return; } @@ -2181,7 +2187,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) */ if (val & 0x400) { if (wm5100->jack_detecting) { - dev_dbg(codec->dev, "Microphone detected\n"); + dev_dbg(wm5100->dev, "Microphone detected\n"); wm5100->jack_mic = true; snd_soc_jack_report(wm5100->jack, SND_JACK_HEADSET, @@ -2189,11 +2195,11 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) /* Increase poll rate to give better responsiveness * for buttons */ - snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - 5 << WM5100_ACCDET_RATE_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 5 << WM5100_ACCDET_RATE_SHIFT); } else { - dev_dbg(codec->dev, "Mic button up\n"); + dev_dbg(wm5100->dev, "Mic button up\n"); snd_soc_jack_report(wm5100->jack, 0, SND_JACK_BTN_0); } @@ -2206,7 +2212,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) * plain headphones. */ if (wm5100->jack_detecting && (val & 0x3f8)) { - wm5100_set_detect_mode(codec, !wm5100->jack_mode); + wm5100_set_detect_mode(wm5100, !wm5100->jack_mode); return; } @@ -2216,20 +2222,20 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) */ if (val & 0x3fc) { if (wm5100->jack_mic) { - dev_dbg(codec->dev, "Mic button detected\n"); + dev_dbg(wm5100->dev, "Mic button detected\n"); snd_soc_jack_report(wm5100->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); } else if (wm5100->jack_detecting) { - dev_dbg(codec->dev, "Headphone detected\n"); + dev_dbg(wm5100->dev, "Headphone detected\n"); snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); /* Increase the detection rate a bit for * responsiveness. */ - snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - 7 << WM5100_ACCDET_RATE_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 7 << WM5100_ACCDET_RATE_SHIFT); } } } @@ -2242,7 +2248,7 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) wm5100->jack = jack; wm5100->jack_detecting = true; - wm5100_set_detect_mode(codec, 0); + wm5100_set_detect_mode(wm5100, 0); /* Slowest detection rate, gives debounce for initial * detection */ @@ -2281,52 +2287,70 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) static irqreturn_t wm5100_irq(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_priv *wm5100 = data; irqreturn_t status = IRQ_NONE; - int irq_val; + unsigned int irq_val, mask_val; + int ret; - irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3); - if (irq_val < 0) { - dev_err(codec->dev, "Failed to read IRQ status 3: %d\n", - irq_val); + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_3, &irq_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ status 3: %d\n", + ret); irq_val = 0; } - irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3_MASK); - snd_soc_write(codec, WM5100_INTERRUPT_STATUS_3, irq_val); + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_3_MASK, + &mask_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ mask 3: %d\n", + ret); + mask_val = 0xffff; + } + + irq_val &= ~mask_val; + + regmap_write(wm5100->regmap, WM5100_INTERRUPT_STATUS_3, irq_val); if (irq_val) status = IRQ_HANDLED; - wm5100_log_status3(codec, irq_val); + wm5100_log_status3(wm5100, irq_val); if (irq_val & WM5100_FLL1_LOCK_EINT) { - dev_dbg(codec->dev, "FLL1 locked\n"); + dev_dbg(wm5100->dev, "FLL1 locked\n"); complete(&wm5100->fll[0].lock); } if (irq_val & WM5100_FLL2_LOCK_EINT) { - dev_dbg(codec->dev, "FLL2 locked\n"); + dev_dbg(wm5100->dev, "FLL2 locked\n"); complete(&wm5100->fll[1].lock); } if (irq_val & WM5100_ACCDET_EINT) - wm5100_micd_irq(codec); + wm5100_micd_irq(wm5100); - irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4); - if (irq_val < 0) { - dev_err(codec->dev, "Failed to read IRQ status 4: %d\n", - irq_val); + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_4, &irq_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ status 4: %d\n", + ret); irq_val = 0; } - irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4_MASK); + + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_4_MASK, + &mask_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ mask 4: %d\n", + ret); + mask_val = 0xffff; + } + + irq_val &= ~mask_val; if (irq_val) status = IRQ_HANDLED; - snd_soc_write(codec, WM5100_INTERRUPT_STATUS_4, irq_val); + regmap_write(wm5100->regmap, WM5100_INTERRUPT_STATUS_4, irq_val); - wm5100_log_status4(codec, irq_val); + wm5100_log_status4(wm5100, irq_val); return status; } @@ -2485,11 +2509,12 @@ static int wm5100_probe(struct snd_soc_codec *codec) if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) ret = request_threaded_irq(i2c->irq, NULL, - wm5100_edge_irq, - irq_flags, "wm5100", codec); + wm5100_edge_irq, irq_flags, + "wm5100", wm5100); else ret = request_threaded_irq(i2c->irq, NULL, wm5100_irq, - irq_flags, "wm5100", codec); + irq_flags, "wm5100", + wm5100); if (ret != 0) { dev_err(codec->dev, "Failed to request IRQ %d: %d\n", @@ -2552,7 +2577,7 @@ static int wm5100_probe(struct snd_soc_codec *codec) err_gpio: if (i2c->irq) - free_irq(i2c->irq, codec); + free_irq(i2c->irq, wm5100); return ret; } @@ -2566,7 +2591,8 @@ static int wm5100_remove(struct snd_soc_codec *codec) gpio_free(wm5100->pdata.hp_pol); } if (i2c->irq) - free_irq(i2c->irq, codec); + free_irq(i2c->irq, wm5100); + return 0; } @@ -2622,6 +2648,8 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (wm5100 == NULL) return -ENOMEM; + wm5100->dev = &i2c->dev; + wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap); if (IS_ERR(wm5100->regmap)) { ret = PTR_ERR(wm5100->regmap); -- cgit v1.2.3-18-g5258 From 09452f23eb01241fa19c2e99585af5e340a0961b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Jan 2012 15:05:46 +0000 Subject: ASoC: Push WM5100 interrupt request into I2C probe This is more what the device model wants us to do and will allow use by non-audio functions before the audio part of the device has come up. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 121 ++++++++++++++++++++++++---------------------- 1 file changed, 63 insertions(+), 58 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4b2c724ed9b..c291f8ea32e 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2475,7 +2475,7 @@ static int wm5100_probe(struct snd_soc_codec *codec) { struct i2c_client *i2c = to_i2c_client(codec->dev); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - int ret, i, irq_flags; + int ret, i; wm5100->codec = codec; codec->control_data = wm5100->regmap; @@ -2499,61 +2499,10 @@ static int wm5100_probe(struct snd_soc_codec *codec) /* TODO: check if we're symmetric */ - if (i2c->irq) { - if (wm5100->pdata.irq_flags) - irq_flags = wm5100->pdata.irq_flags; - else - irq_flags = IRQF_TRIGGER_LOW; - - irq_flags |= IRQF_ONESHOT; - - if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) - ret = request_threaded_irq(i2c->irq, NULL, - wm5100_edge_irq, irq_flags, - "wm5100", wm5100); - else - ret = request_threaded_irq(i2c->irq, NULL, wm5100_irq, - irq_flags, "wm5100", - wm5100); - - if (ret != 0) { - dev_err(codec->dev, "Failed to request IRQ %d: %d\n", - i2c->irq, ret); - } else { - /* Enable default interrupts */ - snd_soc_update_bits(codec, - WM5100_INTERRUPT_STATUS_3_MASK, - WM5100_IM_SPK_SHUTDOWN_WARN_EINT | - WM5100_IM_SPK_SHUTDOWN_EINT | - WM5100_IM_ASRC2_LOCK_EINT | - WM5100_IM_ASRC1_LOCK_EINT | - WM5100_IM_FLL2_LOCK_EINT | - WM5100_IM_FLL1_LOCK_EINT | - WM5100_CLKGEN_ERR_EINT | - WM5100_CLKGEN_ERR_ASYNC_EINT, 0); - - snd_soc_update_bits(codec, - WM5100_INTERRUPT_STATUS_4_MASK, - WM5100_AIF3_ERR_EINT | - WM5100_AIF2_ERR_EINT | - WM5100_AIF1_ERR_EINT | - WM5100_CTRLIF_ERR_EINT | - WM5100_ISRC2_UNDERCLOCKED_EINT | - WM5100_ISRC1_UNDERCLOCKED_EINT | - WM5100_FX_UNDERCLOCKED_EINT | - WM5100_AIF3_UNDERCLOCKED_EINT | - WM5100_AIF2_UNDERCLOCKED_EINT | - WM5100_AIF1_UNDERCLOCKED_EINT | - WM5100_ASRC_UNDERCLOCKED_EINT | - WM5100_DAC_UNDERCLOCKED_EINT | - WM5100_ADC_UNDERCLOCKED_EINT | - WM5100_MIXER_UNDERCLOCKED_EINT, 0); - } - } else { + if (i2c->irq) snd_soc_dapm_new_controls(&codec->dapm, wm5100_dapm_widgets_noirq, ARRAY_SIZE(wm5100_dapm_widgets_noirq)); - } if (wm5100->pdata.hp_pol) { ret = gpio_request_one(wm5100->pdata.hp_pol, @@ -2641,7 +2590,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, struct wm5100_pdata *pdata = dev_get_platdata(&i2c->dev); struct wm5100_priv *wm5100; unsigned int reg; - int ret, i; + int ret, i, irq_flags; wm5100 = devm_kzalloc(&i2c->dev, sizeof(struct wm5100_priv), GFP_KERNEL); @@ -2778,6 +2727,58 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, WM5100_IN1_DMIC_SUP_SHIFT)); } + if (i2c->irq) { + if (wm5100->pdata.irq_flags) + irq_flags = wm5100->pdata.irq_flags; + else + irq_flags = IRQF_TRIGGER_LOW; + + irq_flags |= IRQF_ONESHOT; + + if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) + ret = request_threaded_irq(i2c->irq, NULL, + wm5100_edge_irq, irq_flags, + "wm5100", wm5100); + else + ret = request_threaded_irq(i2c->irq, NULL, wm5100_irq, + irq_flags, "wm5100", + wm5100); + + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request IRQ %d: %d\n", + i2c->irq, ret); + } else { + /* Enable default interrupts */ + regmap_update_bits(wm5100->regmap, + WM5100_INTERRUPT_STATUS_3_MASK, + WM5100_IM_SPK_SHUTDOWN_WARN_EINT | + WM5100_IM_SPK_SHUTDOWN_EINT | + WM5100_IM_ASRC2_LOCK_EINT | + WM5100_IM_ASRC1_LOCK_EINT | + WM5100_IM_FLL2_LOCK_EINT | + WM5100_IM_FLL1_LOCK_EINT | + WM5100_CLKGEN_ERR_EINT | + WM5100_CLKGEN_ERR_ASYNC_EINT, 0); + + regmap_update_bits(wm5100->regmap, + WM5100_INTERRUPT_STATUS_4_MASK, + WM5100_AIF3_ERR_EINT | + WM5100_AIF2_ERR_EINT | + WM5100_AIF1_ERR_EINT | + WM5100_CTRLIF_ERR_EINT | + WM5100_ISRC2_UNDERCLOCKED_EINT | + WM5100_ISRC1_UNDERCLOCKED_EINT | + WM5100_FX_UNDERCLOCKED_EINT | + WM5100_AIF3_UNDERCLOCKED_EINT | + WM5100_AIF2_UNDERCLOCKED_EINT | + WM5100_AIF1_UNDERCLOCKED_EINT | + WM5100_ASRC_UNDERCLOCKED_EINT | + WM5100_DAC_UNDERCLOCKED_EINT | + WM5100_ADC_UNDERCLOCKED_EINT | + WM5100_MIXER_UNDERCLOCKED_EINT, 0); + } + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm5100, wm5100_dai, ARRAY_SIZE(wm5100_dai)); @@ -2789,6 +2790,8 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, return ret; err_reset: + if (i2c->irq) + free_irq(i2c->irq, wm5100); wm5100_free_gpio(i2c); if (wm5100->pdata.reset) { gpio_set_value_cansleep(wm5100->pdata.reset, 0); @@ -2817,12 +2820,14 @@ err: return ret; } -static __devexit int wm5100_i2c_remove(struct i2c_client *client) +static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) { - struct wm5100_priv *wm5100 = i2c_get_clientdata(client); + struct wm5100_priv *wm5100 = i2c_get_clientdata(i2c); - snd_soc_unregister_codec(&client->dev); - wm5100_free_gpio(client); + snd_soc_unregister_codec(&i2c->dev); + if (i2c->irq) + free_irq(i2c->irq, wm5100); + wm5100_free_gpio(i2c); if (wm5100->pdata.reset) { gpio_set_value_cansleep(wm5100->pdata.reset, 0); gpio_free(wm5100->pdata.reset); -- cgit v1.2.3-18-g5258 From 278047fd654dde7ed95c8604fcefeeacc5c0bb2b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 19 Jan 2012 18:04:18 +0000 Subject: ASoC: Don't tell applications about msbits unless we're ignoring input On the off chance that an application both pays attention and gets confused. Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8bb17937d59..326890148a2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -85,9 +85,11 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, return; for (i = 0; i < ARRAY_SIZE(sample_sizes); i++) { - ret = snd_pcm_hw_constraint_msbits(substream->runtime, - 0, sample_sizes[i], - bits); + if (bits >= sample_sizes[i]) + continue; + + ret = snd_pcm_hw_constraint_msbits(substream->runtime, 0, + sample_sizes[i], bits); if (ret != 0) dev_warn(dai->dev, "Failed to set MSB %d/%d: %d\n", -- cgit v1.2.3-18-g5258 From 8a713da8d1ce9ceaf738b32e2b24f22d4432f886 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Dec 2011 12:33:55 +0000 Subject: ASoC: Use regmap update bits operation for drivers using regmap If a driver is using regmap directly ensure that we're coherent with non-ASoC register updates by using the regmap API directly to do our read/modify/write cycles. This will bypass the ASoC cache but drivers using regmap directly should not be using the ASoC cache. Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 25 +++++++++++++++---------- sound/soc/soc-dapm.c | 27 +++++++++++++++++---------- sound/soc/soc-io.c | 1 + 3 files changed, 33 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 41c8e45a23e..35a1e639d7f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1869,23 +1869,28 @@ EXPORT_SYMBOL_GPL(snd_soc_bulk_write_raw); int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, unsigned int mask, unsigned int value) { - int change; + bool change; unsigned int old, new; int ret; - ret = snd_soc_read(codec, reg); - if (ret < 0) - return ret; - - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = snd_soc_write(codec, reg, new); + if (codec->using_regmap) { + ret = regmap_update_bits_check(codec->control_data, reg, + mask, value, &change); + } else { + ret = snd_soc_read(codec, reg); if (ret < 0) return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) + ret = snd_soc_write(codec, reg, new); } + if (ret < 0) + return ret; + return change; } EXPORT_SYMBOL_GPL(snd_soc_update_bits); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1f55ded4047..31a06b2b444 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -197,21 +197,28 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { - int change; + bool change; unsigned int old, new; int ret; - ret = soc_widget_read(w, reg); - if (ret < 0) - return ret; - - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = soc_widget_write(w, reg, new); + if (w->codec && w->codec->using_regmap) { + ret = regmap_update_bits_check(w->codec->control_data, + reg, mask, value, &change); + if (ret != 0) + return ret; + } else { + ret = soc_widget_read(w, reg); if (ret < 0) return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) { + ret = soc_widget_write(w, reg, new); + if (ret < 0) + return ret; + } } return change; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index c8610cbf34a..39ba5070ff9 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -140,6 +140,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ + codec->using_regmap = true; break; default: -- cgit v1.2.3-18-g5258 From 3a4cbf88963963aacbeef63a1a795f8ea05d1d30 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 20 Jan 2012 17:52:39 +0000 Subject: ASoC: Fix build of tlv320dac33 The problem was introduced due to the obscure formatting some of the older drivers use. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 21ccf0a616a..c06c3e4b912 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1513,8 +1513,9 @@ static struct snd_soc_dai_driver dac33_dai = { .channels_min = 2, .channels_max = 2, .rates = DAC33_RATES, - .formats = DAC33_FORMATS,}, + .formats = DAC33_FORMATS, .sig_bits = 24, + }, .ops = &dac33_dai_ops, }; -- cgit v1.2.3-18-g5258 From 78adaeb2ae7d5e9e1a6e93e06db26d07fdd829fb Mon Sep 17 00:00:00 2001 From: Javier Martin Date: Fri, 20 Jan 2012 10:16:57 +0100 Subject: ASoC: Add external amplifier controls for Visstrim_M10. Visstrim_M10 has an external class D amplifier. This patch provides support for controlling the 4 possible gain levels and per channel muting. Signed-off-by: Javier Martin Signed-off-by: Mark Brown --- sound/soc/imx/mx27vis-aic32x4.c | 80 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 80 insertions(+) (limited to 'sound') diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index d37e23cfc94..155899c08c0 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -25,16 +25,37 @@ #include #include #include +#include #include #include #include #include +#include #include #include +#include #include "../codecs/tlv320aic32x4.h" #include "imx-ssi.h" +#define MX27VIS_AMP_GAIN 0 +#define MX27VIS_AMP_MUTE 1 + +#define MX27VIS_PIN_G0 (GPIO_PORTF + 9) +#define MX27VIS_PIN_G1 (GPIO_PORTF + 8) +#define MX27VIS_PIN_SDL (GPIO_PORTE + 5) +#define MX27VIS_PIN_SDR (GPIO_PORTF + 7) + +static int mx27vis_amp_gain; +static int mx27vis_amp_mute; + +static const int mx27vis_amp_pins[] = { + MX27VIS_PIN_G0 | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_G1 | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_SDL | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_SDR | GPIO_GPIO | GPIO_OUT, +}; + static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -74,8 +95,60 @@ static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { .hw_params = mx27vis_aic32x4_hw_params, }; +static int mx27vis_amp_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int value = ucontrol->value.integer.value[0]; + unsigned int reg = mc->reg; + int max = mc->max; + + if (value > max) + return -EINVAL; + + switch (reg) { + case MX27VIS_AMP_GAIN: + gpio_set_value(MX27VIS_PIN_G0, value & 1); + gpio_set_value(MX27VIS_PIN_G1, value >> 1); + mx27vis_amp_gain = value; + break; + case MX27VIS_AMP_MUTE: + gpio_set_value(MX27VIS_PIN_SDL, value & 1); + gpio_set_value(MX27VIS_PIN_SDR, value >> 1); + mx27vis_amp_mute = value; + break; + } + return 0; +} + +static int mx27vis_amp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + switch (reg) { + case MX27VIS_AMP_GAIN: + ucontrol->value.integer.value[0] = mx27vis_amp_gain; + break; + case MX27VIS_AMP_MUTE: + ucontrol->value.integer.value[0] = mx27vis_amp_mute; + break; + } + return 0; +} + +/* From 6dB to 24dB in steps of 6dB */ +static const DECLARE_TLV_DB_SCALE(mx27vis_amp_tlv, 600, 600, 0); + static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = { SOC_DAPM_PIN_SWITCH("External Mic"), + SOC_SINGLE_EXT_TLV("LO Ext Boost", MX27VIS_AMP_GAIN, 0, 3, 0, + mx27vis_amp_get, mx27vis_amp_set, mx27vis_amp_tlv), + SOC_DOUBLE_EXT("LO Ext Mute Switch", MX27VIS_AMP_MUTE, 0, 1, 1, 0, + mx27vis_amp_get, mx27vis_amp_set), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { @@ -146,6 +219,13 @@ static int __init mx27vis_aic32x4_init(void) MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) ); + ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, + ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); + if (ret) { + printk(KERN_ERR "ASoC: unable to setup gpios\n"); + platform_device_put(mx27vis_aic32x4_snd_device); + } + return ret; } -- cgit v1.2.3-18-g5258 From a1fea9404f6b400dcbda952599649e6d37aad1c0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 15:23:26 +0000 Subject: ASoC: wm8985: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index c0c86b3c6ad..e62a4c55a9c 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1079,7 +1079,7 @@ static int __devinit wm8985_spi_probe(struct spi_device *spi) struct wm8985_priv *wm8985; int ret; - wm8985 = kzalloc(sizeof *wm8985, GFP_KERNEL); + wm8985 = devm_kzalloc(&spi->dev, sizeof *wm8985, GFP_KERNEL); if (!wm8985) return -ENOMEM; @@ -1088,15 +1088,12 @@ static int __devinit wm8985_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8985, &wm8985_dai, 1); - if (ret < 0) - kfree(wm8985); return ret; } static int __devexit wm8985_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -1117,7 +1114,7 @@ static __devinit int wm8985_i2c_probe(struct i2c_client *i2c, struct wm8985_priv *wm8985; int ret; - wm8985 = kzalloc(sizeof *wm8985, GFP_KERNEL); + wm8985 = devm_kzalloc(&i2c->dev, sizeof *wm8985, GFP_KERNEL); if (!wm8985) return -ENOMEM; @@ -1126,15 +1123,12 @@ static __devinit int wm8985_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8985, &wm8985_dai, 1); - if (ret < 0) - kfree(wm8985); return ret; } static __devexit int wm8985_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3-18-g5258 From eb8f7693df0426b3c7aa6e6e401486962a033d5e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 15:36:08 +0000 Subject: ASoC: wm8985: Convert to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index e62a4c55a9c..ee3aba3098c 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -428,7 +428,7 @@ static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPKR") }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8985_dapm_routes[] = { { "Right Output Mixer", "PCM Switch", "Right DAC" }, { "Right Output Mixer", "Aux Switch", "AUXR" }, { "Right Output Mixer", "Line Switch", "Right Boost Mixer" }, @@ -531,17 +531,6 @@ static int eqmode_put(struct snd_kcontrol *kcontrol, return 0; } -static int wm8985_add_widgets(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, wm8985_dapm_widgets, - ARRAY_SIZE(wm8985_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, - ARRAY_SIZE(audio_map)); - return 0; -} - static int wm8985_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, WM8985_SOFTWARE_RESET, 0x0); @@ -1017,10 +1006,6 @@ static int wm8985_probe(struct snd_soc_codec *codec) cache[WM8985_BIAS_CTRL] |= WM8985_BIASCUT; codec->cache_sync = 1; - snd_soc_add_controls(codec, wm8985_snd_controls, - ARRAY_SIZE(wm8985_snd_controls)); - wm8985_add_widgets(codec); - wm8985_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -1068,9 +1053,16 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8985 = { .suspend = wm8985_suspend, .resume = wm8985_resume, .set_bias_level = wm8985_set_bias_level, + + .controls = wm8985_snd_controls, + .num_controls = ARRAY_SIZE(wm8985_snd_controls), + .dapm_widgets = wm8985_dapm_widgets, .reg_cache_size = ARRAY_SIZE(wm8985_reg_defs), .reg_word_size = sizeof(u16), .reg_cache_default = wm8985_reg_defs + .num_dapm_widgets = ARRAY_SIZE(wm8985_dapm_widgets), + .dapm_routes = wm8985_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8985_dapm_routes), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-18-g5258 From 8b71d441f75d180d3174b2e1b649db385552c266 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 15:36:30 +0000 Subject: ASoC: wm8985: Use standard cache sync implementation Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 21 +-------------------- 1 file changed, 1 insertion(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index ee3aba3098c..297119ffec6 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -834,25 +834,6 @@ static int wm8985_set_sysclk(struct snd_soc_dai *dai, return 0; } -static void wm8985_sync_cache(struct snd_soc_codec *codec) -{ - short i; - u16 *cache; - - if (!codec->cache_sync) - return; - codec->cache_only = 0; - /* restore cache */ - cache = codec->reg_cache; - for (i = 0; i < codec->driver->reg_cache_size; i++) { - if (i == WM8985_SOFTWARE_RESET - || cache[i] == wm8985_reg_defs[i]) - continue; - snd_soc_write(codec, i, cache[i]); - } - codec->cache_sync = 0; -} - static int wm8985_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -879,7 +860,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec, return ret; } - wm8985_sync_cache(codec); + snd_soc_cache_sync(codec); /* enable anti-pop features */ snd_soc_update_bits(codec, WM8985_OUT4_TO_ADC, -- cgit v1.2.3-18-g5258 From 9f8cbae4163ab132cd7a56385341efdd41fcd429 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 15:39:34 +0000 Subject: ASoC: wm8985 Don't directly reference the cache data structure In preparation for conversion to regmap. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 297119ffec6..bbe19b2ae51 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -946,7 +946,6 @@ static int wm8985_probe(struct snd_soc_codec *codec) size_t i; struct wm8985_priv *wm8985; int ret; - u16 *cache; wm8985 = snd_soc_codec_get_drvdata(codec); @@ -979,13 +978,13 @@ static int wm8985_probe(struct snd_soc_codec *codec) goto err_reg_enable; } - cache = codec->reg_cache; /* latch volume update bits */ for (i = 0; i < ARRAY_SIZE(volume_update_regs); ++i) - cache[volume_update_regs[i]] |= 0x100; + snd_soc_update_bits(codec, volume_update_regs[i], + 0x100, 0x100); /* enable BIASCUT */ - cache[WM8985_BIAS_CTRL] |= WM8985_BIASCUT; - codec->cache_sync = 1; + snd_soc_update_bits(codec, WM8985_BIAS_CTRL, WM8985_BIASCUT, + WM8985_BIASCUT); wm8985_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- cgit v1.2.3-18-g5258 From 411a3450c9539043c794a5f4a6bdb03bb040670a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 15:41:48 +0000 Subject: ASoC: wm8985: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 253 ++++++++++++++++++++++++++++++++-------------- 1 file changed, 177 insertions(+), 76 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index bbe19b2ae51..14f666398d0 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -39,73 +40,127 @@ static const char *wm8985_supply_names[WM8985_NUM_SUPPLIES] = { "AVDD2" }; -static const u16 wm8985_reg_defs[] = { - 0x0000, /* R0 - Software Reset */ - 0x0000, /* R1 - Power management 1 */ - 0x0000, /* R2 - Power management 2 */ - 0x0000, /* R3 - Power management 3 */ - 0x0050, /* R4 - Audio Interface */ - 0x0000, /* R5 - Companding control */ - 0x0140, /* R6 - Clock Gen control */ - 0x0000, /* R7 - Additional control */ - 0x0000, /* R8 - GPIO Control */ - 0x0000, /* R9 - Jack Detect Control 1 */ - 0x0000, /* R10 - DAC Control */ - 0x00FF, /* R11 - Left DAC digital Vol */ - 0x00FF, /* R12 - Right DAC digital vol */ - 0x0000, /* R13 - Jack Detect Control 2 */ - 0x0100, /* R14 - ADC Control */ - 0x00FF, /* R15 - Left ADC Digital Vol */ - 0x00FF, /* R16 - Right ADC Digital Vol */ - 0x0000, /* R17 */ - 0x012C, /* R18 - EQ1 - low shelf */ - 0x002C, /* R19 - EQ2 - peak 1 */ - 0x002C, /* R20 - EQ3 - peak 2 */ - 0x002C, /* R21 - EQ4 - peak 3 */ - 0x002C, /* R22 - EQ5 - high shelf */ - 0x0000, /* R23 */ - 0x0032, /* R24 - DAC Limiter 1 */ - 0x0000, /* R25 - DAC Limiter 2 */ - 0x0000, /* R26 */ - 0x0000, /* R27 - Notch Filter 1 */ - 0x0000, /* R28 - Notch Filter 2 */ - 0x0000, /* R29 - Notch Filter 3 */ - 0x0000, /* R30 - Notch Filter 4 */ - 0x0000, /* R31 */ - 0x0038, /* R32 - ALC control 1 */ - 0x000B, /* R33 - ALC control 2 */ - 0x0032, /* R34 - ALC control 3 */ - 0x0000, /* R35 - Noise Gate */ - 0x0008, /* R36 - PLL N */ - 0x000C, /* R37 - PLL K 1 */ - 0x0093, /* R38 - PLL K 2 */ - 0x00E9, /* R39 - PLL K 3 */ - 0x0000, /* R40 */ - 0x0000, /* R41 - 3D control */ - 0x0000, /* R42 - OUT4 to ADC */ - 0x0000, /* R43 - Beep control */ - 0x0033, /* R44 - Input ctrl */ - 0x0010, /* R45 - Left INP PGA gain ctrl */ - 0x0010, /* R46 - Right INP PGA gain ctrl */ - 0x0100, /* R47 - Left ADC BOOST ctrl */ - 0x0100, /* R48 - Right ADC BOOST ctrl */ - 0x0002, /* R49 - Output ctrl */ - 0x0001, /* R50 - Left mixer ctrl */ - 0x0001, /* R51 - Right mixer ctrl */ - 0x0039, /* R52 - LOUT1 (HP) volume ctrl */ - 0x0039, /* R53 - ROUT1 (HP) volume ctrl */ - 0x0039, /* R54 - LOUT2 (SPK) volume ctrl */ - 0x0039, /* R55 - ROUT2 (SPK) volume ctrl */ - 0x0001, /* R56 - OUT3 mixer ctrl */ - 0x0001, /* R57 - OUT4 (MONO) mix ctrl */ - 0x0001, /* R58 */ - 0x0000, /* R59 */ - 0x0004, /* R60 - OUTPUT ctrl */ - 0x0000, /* R61 - BIAS CTRL */ - 0x0180, /* R62 */ - 0x0000 /* R63 */ +static const struct reg_default wm8985_reg_defaults[] = { + { 1, 0x0000 }, /* R1 - Power management 1 */ + { 2, 0x0000 }, /* R2 - Power management 2 */ + { 3, 0x0000 }, /* R3 - Power management 3 */ + { 4, 0x0050 }, /* R4 - Audio Interface */ + { 5, 0x0000 }, /* R5 - Companding control */ + { 6, 0x0140 }, /* R6 - Clock Gen control */ + { 7, 0x0000 }, /* R7 - Additional control */ + { 8, 0x0000 }, /* R8 - GPIO Control */ + { 9, 0x0000 }, /* R9 - Jack Detect Control 1 */ + { 10, 0x0000 }, /* R10 - DAC Control */ + { 11, 0x00FF }, /* R11 - Left DAC digital Vol */ + { 12, 0x00FF }, /* R12 - Right DAC digital vol */ + { 13, 0x0000 }, /* R13 - Jack Detect Control 2 */ + { 14, 0x0100 }, /* R14 - ADC Control */ + { 15, 0x00FF }, /* R15 - Left ADC Digital Vol */ + { 16, 0x00FF }, /* R16 - Right ADC Digital Vol */ + { 18, 0x012C }, /* R18 - EQ1 - low shelf */ + { 19, 0x002C }, /* R19 - EQ2 - peak 1 */ + { 20, 0x002C }, /* R20 - EQ3 - peak 2 */ + { 21, 0x002C }, /* R21 - EQ4 - peak 3 */ + { 22, 0x002C }, /* R22 - EQ5 - high shelf */ + { 24, 0x0032 }, /* R24 - DAC Limiter 1 */ + { 25, 0x0000 }, /* R25 - DAC Limiter 2 */ + { 27, 0x0000 }, /* R27 - Notch Filter 1 */ + { 28, 0x0000 }, /* R28 - Notch Filter 2 */ + { 29, 0x0000 }, /* R29 - Notch Filter 3 */ + { 30, 0x0000 }, /* R30 - Notch Filter 4 */ + { 32, 0x0038 }, /* R32 - ALC control 1 */ + { 33, 0x000B }, /* R33 - ALC control 2 */ + { 34, 0x0032 }, /* R34 - ALC control 3 */ + { 35, 0x0000 }, /* R35 - Noise Gate */ + { 36, 0x0008 }, /* R36 - PLL N */ + { 37, 0x000C }, /* R37 - PLL K 1 */ + { 38, 0x0093 }, /* R38 - PLL K 2 */ + { 39, 0x00E9 }, /* R39 - PLL K 3 */ + { 41, 0x0000 }, /* R41 - 3D control */ + { 42, 0x0000 }, /* R42 - OUT4 to ADC */ + { 43, 0x0000 }, /* R43 - Beep control */ + { 44, 0x0033 }, /* R44 - Input ctrl */ + { 45, 0x0010 }, /* R45 - Left INP PGA gain ctrl */ + { 46, 0x0010 }, /* R46 - Right INP PGA gain ctrl */ + { 47, 0x0100 }, /* R47 - Left ADC BOOST ctrl */ + { 48, 0x0100 }, /* R48 - Right ADC BOOST ctrl */ + { 49, 0x0002 }, /* R49 - Output ctrl */ + { 50, 0x0001 }, /* R50 - Left mixer ctrl */ + { 51, 0x0001 }, /* R51 - Right mixer ctrl */ + { 52, 0x0039 }, /* R52 - LOUT1 (HP) volume ctrl */ + { 53, 0x0039 }, /* R53 - ROUT1 (HP) volume ctrl */ + { 54, 0x0039 }, /* R54 - LOUT2 (SPK) volume ctrl */ + { 55, 0x0039 }, /* R55 - ROUT2 (SPK) volume ctrl */ + { 56, 0x0001 }, /* R56 - OUT3 mixer ctrl */ + { 57, 0x0001 }, /* R57 - OUT4 (MONO) mix ctrl */ + { 60, 0x0004 }, /* R60 - OUTPUT ctrl */ + { 61, 0x0000 }, /* R61 - BIAS CTRL */ }; +static bool wm8985_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8985_SOFTWARE_RESET: + case WM8985_POWER_MANAGEMENT_1: + case WM8985_POWER_MANAGEMENT_2: + case WM8985_POWER_MANAGEMENT_3: + case WM8985_AUDIO_INTERFACE: + case WM8985_COMPANDING_CONTROL: + case WM8985_CLOCK_GEN_CONTROL: + case WM8985_ADDITIONAL_CONTROL: + case WM8985_GPIO_CONTROL: + case WM8985_JACK_DETECT_CONTROL_1: + case WM8985_DAC_CONTROL: + case WM8985_LEFT_DAC_DIGITAL_VOL: + case WM8985_RIGHT_DAC_DIGITAL_VOL: + case WM8985_JACK_DETECT_CONTROL_2: + case WM8985_ADC_CONTROL: + case WM8985_LEFT_ADC_DIGITAL_VOL: + case WM8985_RIGHT_ADC_DIGITAL_VOL: + case WM8985_EQ1_LOW_SHELF: + case WM8985_EQ2_PEAK_1: + case WM8985_EQ3_PEAK_2: + case WM8985_EQ4_PEAK_3: + case WM8985_EQ5_HIGH_SHELF: + case WM8985_DAC_LIMITER_1: + case WM8985_DAC_LIMITER_2: + case WM8985_NOTCH_FILTER_1: + case WM8985_NOTCH_FILTER_2: + case WM8985_NOTCH_FILTER_3: + case WM8985_NOTCH_FILTER_4: + case WM8985_ALC_CONTROL_1: + case WM8985_ALC_CONTROL_2: + case WM8985_ALC_CONTROL_3: + case WM8985_NOISE_GATE: + case WM8985_PLL_N: + case WM8985_PLL_K_1: + case WM8985_PLL_K_2: + case WM8985_PLL_K_3: + case WM8985_3D_CONTROL: + case WM8985_OUT4_TO_ADC: + case WM8985_BEEP_CONTROL: + case WM8985_INPUT_CTRL: + case WM8985_LEFT_INP_PGA_GAIN_CTRL: + case WM8985_RIGHT_INP_PGA_GAIN_CTRL: + case WM8985_LEFT_ADC_BOOST_CTRL: + case WM8985_RIGHT_ADC_BOOST_CTRL: + case WM8985_OUTPUT_CTRL0: + case WM8985_LEFT_MIXER_CTRL: + case WM8985_RIGHT_MIXER_CTRL: + case WM8985_LOUT1_HP_VOLUME_CTRL: + case WM8985_ROUT1_HP_VOLUME_CTRL: + case WM8985_LOUT2_SPK_VOLUME_CTRL: + case WM8985_ROUT2_SPK_VOLUME_CTRL: + case WM8985_OUT3_MIXER_CTRL: + case WM8985_OUT4_MONO_MIX_CTRL: + case WM8985_OUTPUT_CTRL1: + case WM8985_BIAS_CTRL: + return true; + default: + return false; + } +} + /* * latch bit 8 of these registers to ensure instant * volume updates @@ -124,7 +179,7 @@ static const int volume_update_regs[] = { }; struct wm8985_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8985_NUM_SUPPLIES]; unsigned int sysclk; unsigned int bclk; @@ -860,7 +915,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec, return ret; } - snd_soc_cache_sync(codec); + regcache_sync(wm8985->regmap); /* enable anti-pop features */ snd_soc_update_bits(codec, WM8985_OUT4_TO_ADC, @@ -903,7 +958,7 @@ static int wm8985_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8985_POWER_MANAGEMENT_2, 0); snd_soc_write(codec, WM8985_POWER_MANAGEMENT_3, 0); - codec->cache_sync = 1; + regcache_mark_dirty(wm8985->regmap); regulator_bulk_disable(ARRAY_SIZE(wm8985->supplies), wm8985->supplies); @@ -948,8 +1003,9 @@ static int wm8985_probe(struct snd_soc_codec *codec) int ret; wm8985 = snd_soc_codec_get_drvdata(codec); + codec->control_data = wm8985->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8985->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); return ret; @@ -1037,14 +1093,23 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8985 = { .controls = wm8985_snd_controls, .num_controls = ARRAY_SIZE(wm8985_snd_controls), .dapm_widgets = wm8985_dapm_widgets, - .reg_cache_size = ARRAY_SIZE(wm8985_reg_defs), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8985_reg_defs .num_dapm_widgets = ARRAY_SIZE(wm8985_dapm_widgets), .dapm_routes = wm8985_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8985_dapm_routes), }; +static const struct regmap_config wm8985_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8985_MAX_REGISTER, + .writeable_reg = wm8985_writeable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8985_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8985_reg_defaults), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8985_spi_probe(struct spi_device *spi) { @@ -1055,17 +1120,35 @@ static int __devinit wm8985_spi_probe(struct spi_device *spi) if (!wm8985) return -ENOMEM; - wm8985->control_type = SND_SOC_SPI; spi_set_drvdata(spi, wm8985); + wm8985->regmap = regmap_init_spi(spi, &wm8985_regmap); + if (IS_ERR(wm8985->regmap)) { + ret = PTR_ERR(wm8985->regmap); + dev_err(&spi->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8985, &wm8985_dai, 1); + if (ret != 0) + goto err; + + return 0; + +err: + regmap_exit(wm8985->regmap); return ret; } static int __devexit wm8985_spi_remove(struct spi_device *spi) { + struct wm8985_priv *wm8985 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); + regmap_exit(wm8985->regmap); + return 0; } @@ -1090,17 +1173,35 @@ static __devinit int wm8985_i2c_probe(struct i2c_client *i2c, if (!wm8985) return -ENOMEM; - wm8985->control_type = SND_SOC_I2C; i2c_set_clientdata(i2c, wm8985); + wm8985->regmap = regmap_init_i2c(i2c, &wm8985_regmap); + if (IS_ERR(wm8985->regmap)) { + ret = PTR_ERR(wm8985->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8985, &wm8985_dai, 1); + if (ret != 0) + goto err; + + return 0; + +err: + regmap_exit(wm8985->regmap); return ret; } -static __devexit int wm8985_i2c_remove(struct i2c_client *client) +static __devexit int wm8985_i2c_remove(struct i2c_client *i2c) { - snd_soc_unregister_codec(&client->dev); + struct wm8985_priv *wm8985 = i2c_get_clientdata(i2c); + + snd_soc_unregister_codec(&i2c->dev); + regmap_exit(wm8985->regmap); + return 0; } -- cgit v1.2.3-18-g5258 From dd21353f35082fa77d1c8672fffebf324954eb09 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 23:24:25 +0000 Subject: ASoC: wm8988: Convert to table based DAPM and control init Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index ab52963dd04..40aebafb35e 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -317,7 +317,7 @@ static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RINPUT2"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8988_dapm_routes[] = { { "Left Line Mux", "Line 1", "LINPUT1" }, { "Left Line Mux", "Line 2", "LINPUT2" }, @@ -743,7 +743,6 @@ static int wm8988_resume(struct snd_soc_codec *codec) static int wm8988_probe(struct snd_soc_codec *codec) { struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8988->control_type); @@ -767,12 +766,6 @@ static int wm8988_probe(struct snd_soc_codec *codec) wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8988_snd_controls, - ARRAY_SIZE(wm8988_snd_controls)); - snd_soc_dapm_new_controls(dapm, wm8988_dapm_widgets, - ARRAY_SIZE(wm8988_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -791,6 +784,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .reg_cache_size = ARRAY_SIZE(wm8988_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8988_reg, + + .controls = wm8988_snd_controls, + .num_controls = ARRAY_SIZE(wm8988_snd_controls), + .dapm_widgets = wm8988_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8988_dapm_widgets), + .dapm_routes = wm8988_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8988_dapm_routes), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3-18-g5258 From 82fa3670575143031517531936b1cc308d4981fa Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 23:25:16 +0000 Subject: ASoC: wm8988: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 40aebafb35e..4ef9d4cb7d7 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -799,7 +799,8 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi) struct wm8988_priv *wm8988; int ret; - wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL); + wm8988 = devm_kzalloc(&spi->dev, sizeof(struct wm8988_priv), + GFP_KERNEL); if (wm8988 == NULL) return -ENOMEM; @@ -808,15 +809,13 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8988, &wm8988_dai, 1); - if (ret < 0) - kfree(wm8988); + return ret; } static int __devexit wm8988_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -837,7 +836,8 @@ static __devinit int wm8988_i2c_probe(struct i2c_client *i2c, struct wm8988_priv *wm8988; int ret; - wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL); + wm8988 = devm_kzalloc(&i2c->dev, sizeof(struct wm8988_priv), + GFP_KERNEL); if (wm8988 == NULL) return -ENOMEM; @@ -846,15 +846,12 @@ static __devinit int wm8988_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8988, &wm8988_dai, 1); - if (ret < 0) - kfree(wm8988); return ret; } static __devexit int wm8988_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- cgit v1.2.3-18-g5258 From 899896379670f885072564669e395feb649105a6 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Sun, 22 Jan 2012 14:49:42 -0200 Subject: ASoC: sgtl5000: Convert to table based DAPM and control init Convert to table based DAPM and control init. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 04ea4850cd4..18e61a0be26 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -227,7 +227,7 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { }; /* routes for sgtl5000 */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ @@ -1353,15 +1353,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) if (ret) goto err; - snd_soc_add_controls(codec, sgtl5000_snd_controls, - ARRAY_SIZE(sgtl5000_snd_controls)); - - snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets, - ARRAY_SIZE(sgtl5000_dapm_widgets)); - - snd_soc_dapm_add_routes(&codec->dapm, audio_map, - ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(&codec->dapm); return 0; @@ -1402,6 +1393,12 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .reg_cache_step = 2, .reg_cache_default = sgtl5000_regs, .volatile_register = sgtl5000_volatile_register, + .controls = sgtl5000_snd_controls, + .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), + .dapm_widgets = sgtl500_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sgtl500_dapm_widgets), + .dapm_routes = sgtl500_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sgtl500_dapm_routes), }; static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, -- cgit v1.2.3-18-g5258 From 5e0ac527fd8bd81be1aaaaa484832846193f9a17 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Jan 2012 10:16:31 +0000 Subject: ASoC: sgtl5000: It's sgtl5000 not sgtl500 Reported-by: Stephen Rothwell Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 18e61a0be26..d1926266fe0 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1395,10 +1395,10 @@ static struct snd_soc_codec_driver sgtl5000_driver = { .volatile_register = sgtl5000_volatile_register, .controls = sgtl5000_snd_controls, .num_controls = ARRAY_SIZE(sgtl5000_snd_controls), - .dapm_widgets = sgtl500_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sgtl500_dapm_widgets), - .dapm_routes = sgtl500_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(sgtl500_dapm_routes), + .dapm_widgets = sgtl5000_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sgtl5000_dapm_widgets), + .dapm_routes = sgtl5000_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sgtl5000_dapm_routes), }; static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, -- cgit v1.2.3-18-g5258 From 5091f5b797564930371c218dbc57cc4d99732c1e Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 23 Jan 2012 11:18:17 +0800 Subject: ASoC: Add __devinit/__devexit annotations at necessary places MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix below build warning when CONFIG_HOTPLUG is not set. CC sound/soc/codecs/alc5623.o sound/soc/codecs/alc5623.c:1062: warning: ‘alc5623_i2c_remove’ defined but not used CC sound/soc/codecs/alc5632.o sound/soc/codecs/alc5632.c:1112: warning: ‘alc5632_i2c_remove’ defined but not used Signed-off-by: Axel Lin Acked-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 4 ++-- sound/soc/codecs/alc5632.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 3feee569cee..08f24198c8d 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -992,7 +992,7 @@ static struct snd_soc_codec_driver soc_codec_device_alc5623 = { * low = 0x1a * high = 0x1b */ -static int alc5623_i2c_probe(struct i2c_client *client, +static __devinit int alc5623_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct alc5623_platform_data *pdata; @@ -1059,7 +1059,7 @@ static int alc5623_i2c_probe(struct i2c_client *client, return ret; } -static int alc5623_i2c_remove(struct i2c_client *client) +static __devexit int alc5623_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); return 0; diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 390e437d7c5..af9c27ae02f 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1109,7 +1109,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client, return ret; } -static int alc5632_i2c_remove(struct i2c_client *client) +static __devexit int alc5632_i2c_remove(struct i2c_client *client) { struct alc5632_priv *alc5632 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); -- cgit v1.2.3-18-g5258 From 7bfe059e38b06a0d813d92b9b3e500455f6a2c99 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Jan 2012 17:53:39 +0100 Subject: ALSA: hda - explicitly set buffer-align flag for Nvidia controllers It turned out that Nvidial (HDMI) controllers require the buffer alignment. Thus it's better to mark it requiring the alignment, so that we can switch to non-aligned behavior as default in future. Also, change the module paramter to be bint, in order to let user overriding the default value. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa4442e8e1a..d3bd3e74806 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -121,8 +121,8 @@ module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #endif -static bool align_buffer_size = 1; -module_param(align_buffer_size, bool, 0644); +static int align_buffer_size = -1; +module_param(align_buffer_size, bint, 0644); MODULE_PARM_DESC(align_buffer_size, "Force buffer and period sizes to be multiple of 128 bytes."); @@ -515,6 +515,7 @@ enum { #define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */ #define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */ #define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */ +#define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */ /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -527,7 +528,8 @@ enum { /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ - (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI) + (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\ + AZX_DCAPS_ALIGN_BUFSIZE) static char *driver_short_names[] __devinitdata = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -2774,9 +2776,16 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } /* disable buffer size rounding to 128-byte multiples if supported */ - chip->align_buffer_size = align_buffer_size; - if (chip->driver_caps & AZX_DCAPS_BUFSIZE) - chip->align_buffer_size = 0; + if (align_buffer_size >= 0) + chip->align_buffer_size = !!align_buffer_size; + else { + if (chip->driver_caps & AZX_DCAPS_BUFSIZE) + chip->align_buffer_size = 0; + else if (chip->driver_caps & AZX_DCAPS_ALIGN_BUFSIZE) + chip->align_buffer_size = 1; + else + chip->align_buffer_size = 1; + } /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) -- cgit v1.2.3-18-g5258 From 05d448e2c9bc587216549d690332c72a74271abd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 21 Nov 2011 12:10:03 +0000 Subject: ASoC: Convert WM8731 to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 109 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 86 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 8821af70e66..a32caa72bd7 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -41,7 +42,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; unsigned int sysclk; int sysclk_type; @@ -52,16 +53,30 @@ struct wm8731_priv { /* * wm8731 register cache - * We can't read the WM8731 register space when we are - * using 2 wire for device control, so we cache them instead. - * There is no point in caching the reset register */ -static const u16 wm8731_reg[WM8731_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0079, 0x0079, - 0x000a, 0x0008, 0x009f, 0x000a, - 0x0000, 0x0000 +static const struct reg_default wm8731_reg_defaults[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x0079 }, + { 3, 0x0079 }, + { 4, 0x000a }, + { 5, 0x0008 }, + { 6, 0x009f }, + { 7, 0x000a }, + { 8, 0x0000 }, + { 9, 0x0000 }, }; +static bool wm8731_volatile(struct device *dev, unsigned int reg) +{ + return reg == WM8731_RESET; +} + +static bool wm8731_writeable(struct device *dev, unsigned int reg) +{ + return reg <= WM8731_RESET; +} + #define wm8731_reset(c) snd_soc_write(c, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; @@ -441,7 +456,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, if (ret != 0) return ret; - snd_soc_cache_sync(codec); + regcache_sync(wm8731->regmap); } /* Clear PWROFF, gate CLKOUT, everything else as-is */ @@ -452,7 +467,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - codec->cache_sync = 1; + regcache_mark_dirty(wm8731->regmap); break; } codec->dapm.bias_level = level; @@ -513,7 +528,8 @@ static int wm8731_probe(struct snd_soc_codec *codec) struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8731->control_type); + codec->control_data = wm8731->regmap; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -585,9 +601,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { .suspend = wm8731_suspend, .resume = wm8731_resume, .set_bias_level = wm8731_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8731_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8731_reg, .dapm_widgets = wm8731_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets), .dapm_routes = wm8731_intercon, @@ -603,6 +616,19 @@ static const struct of_device_id wm8731_of_match[] = { MODULE_DEVICE_TABLE(of, wm8731_of_match); +static const struct regmap_config wm8731_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8731_RESET, + .volatile_reg = wm8731_volatile, + .writeable_reg = wm8731_writeable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8731_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8731_reg_defaults), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8731_spi_probe(struct spi_device *spi) { @@ -613,20 +639,39 @@ static int __devinit wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; - wm8731->control_type = SND_SOC_SPI; + wm8731->regmap = regmap_init_spi(spi, &wm8731_regmap); + if (IS_ERR(wm8731->regmap)) { + ret = PTR_ERR(wm8731->regmap); + dev_err(&spi->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + spi_set_drvdata(spi, wm8731); ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); - if (ret < 0) - kfree(wm8731); + if (ret != 0) { + dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret); + goto err_regmap; + } + + return 0; + +err_regmap: + regmap_exit(wm8731->regmap); +err: + kfree(wm8731); return ret; } static int __devexit wm8731_spi_remove(struct spi_device *spi) { + struct wm8731_priv *wm8731 = spi_get_drvdata(spi); + snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); + regmap_exit(wm8731->regmap); + kfree(wm8731); return 0; } @@ -652,20 +697,38 @@ static __devinit int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + wm8731->regmap = regmap_init_i2c(i2c, &wm8731_regmap); + if (IS_ERR(wm8731->regmap)) { + ret = PTR_ERR(wm8731->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + i2c_set_clientdata(i2c, wm8731); - wm8731->control_type = SND_SOC_I2C; - ret = snd_soc_register_codec(&i2c->dev, + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8731, &wm8731_dai, 1); - if (ret < 0) - kfree(wm8731); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + goto err_regmap; + } + + return 0; + +err_regmap: + regmap_exit(wm8731->regmap); +err: + kfree(wm8731); return ret; } static __devexit int wm8731_i2c_remove(struct i2c_client *client) { + struct wm8731_priv *wm8731 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); + regmap_exit(wm8731->regmap); + kfree(wm8731); return 0; } -- cgit v1.2.3-18-g5258 From e8770dd878970140b7ef486ec0fe86d43eb50265 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Jan 2012 12:11:20 +0000 Subject: ASoC: wm5100: Fix mismerge of IRQ frees We only want them at the device level, not at the CODEC level. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index c291f8ea32e..81056d8dc89 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2525,8 +2525,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) return 0; err_gpio: - if (i2c->irq) - free_irq(i2c->irq, wm5100); return ret; } @@ -2539,8 +2537,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) if (wm5100->pdata.hp_pol) { gpio_free(wm5100->pdata.hp_pol); } - if (i2c->irq) - free_irq(i2c->irq, wm5100); return 0; } -- cgit v1.2.3-18-g5258 From bb92b7c4ed4f7d5102bb1623cc8a1a9960ddfc08 Mon Sep 17 00:00:00 2001 From: Raymond Yau Date: Tue, 17 Jan 2012 11:32:17 +0800 Subject: ALSA: Au88x0 - Implement subdevice volume controls - add "PCM Playback Volume" controls for 16 playback subdevices This allow application to change the volume of each subdevice by using hardware mixer of au88x0 and default is zero gain/attenunation. Signed-off-by: Raymond Yau Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0.h | 13 ++++- sound/pci/au88x0/au88x0_core.c | 18 +++--- sound/pci/au88x0/au88x0_pcm.c | 127 +++++++++++++++++++++++++++++++++++++++-- 3 files changed, 145 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index bb938153a96..466a5c8e835 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -26,7 +26,7 @@ #include #include #include - +#include #endif #ifndef CHIP_AU8820 @@ -107,6 +107,14 @@ #define NR_WTPB 0x20 /* WT channels per each bank. */ #define NR_PCM 0x10 +struct pcm_vol { + struct snd_kcontrol *kctl; + int active; + int dma; + int mixin[4]; + int vol[4]; +}; + /* Structs */ typedef struct { //int this_08; /* Still unknown */ @@ -168,6 +176,7 @@ struct snd_vortex { /* Xtalk canceler */ int xt_mode; /* 1: speakers, 0:headphones. */ #endif + struct pcm_vol pcm_vol[NR_PCM]; int isquad; /* cache of extended ID codec flag. */ @@ -239,7 +248,7 @@ static int vortex_alsafmt_aspfmt(int alsafmt); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); static int vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, - int dir, int type); + int dir, int type, int subdev); static char vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype); #ifndef CHIP_AU8810 diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 6933a27a5d7..1181c5ec2d4 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2050,8 +2050,6 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) } /* Default Connections */ -static int -vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type); static void vortex_connect_default(vortex_t * vortex, int en) { @@ -2111,15 +2109,13 @@ static void vortex_connect_default(vortex_t * vortex, int en) Return: Return allocated DMA or same DMA passed as "dma" when dma >= 0. */ static int -vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type) +vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, + int type, int subdev) { stream_t *stream; int i, en; + struct pcm_vol *p; - if ((nr_ch == 3) - || ((dir == SNDRV_PCM_STREAM_CAPTURE) && (nr_ch > 2))) - return -EBUSY; - if (dma >= 0) { en = 0; vortex_adb_checkinout(vortex, @@ -2250,6 +2246,14 @@ vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type) MIX_DEFIGAIN); #endif } + if (stream->type == VORTEX_PCM_ADB && en) { + p = &vortex->pcm_vol[subdev]; + p->dma = dma; + for (i = 0; i < nr_ch; i++) + p->mixin[i] = mix[i]; + for (i = 0; i < ch_top; i++) + p->vol[i] = 0; + } } #ifndef CHIP_AU8820 else { diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 0ef2f971220..e59f120742a 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -122,6 +122,18 @@ static struct snd_pcm_hw_constraint_list hw_constraints_au8830_channels = { .mask = 0, }; #endif + +static void vortex_notify_pcm_vol_change(struct snd_card *card, + struct snd_kcontrol *kctl, int activate) +{ + if (activate) + kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + else + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &(kctl->id)); +} + /* open callback */ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) { @@ -230,12 +242,14 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream, if (stream != NULL) vortex_adb_allocroute(chip, stream->dma, stream->nr_ch, stream->dir, - stream->type); + stream->type, + substream->number); /* Alloc routes. */ dma = vortex_adb_allocroute(chip, -1, params_channels(hw_params), - substream->stream, type); + substream->stream, type, + substream->number); if (dma < 0) { spin_unlock_irq(&chip->lock); return dma; @@ -246,6 +260,11 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream, vortex_adbdma_setbuffers(chip, dma, params_period_bytes(hw_params), params_periods(hw_params)); + if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { + chip->pcm_vol[substream->number].active = 1; + vortex_notify_pcm_vol_change(chip->card, + chip->pcm_vol[substream->number].kctl, 1); + } } #ifndef CHIP_AU8810 else { @@ -275,10 +294,18 @@ static int snd_vortex_pcm_hw_free(struct snd_pcm_substream *substream) spin_lock_irq(&chip->lock); // Delete audio routes. if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { - if (stream != NULL) + if (stream != NULL) { + if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { + chip->pcm_vol[substream->number].active = 0; + vortex_notify_pcm_vol_change(chip->card, + chip->pcm_vol[substream->number].kctl, + 0); + } vortex_adb_allocroute(chip, stream->dma, stream->nr_ch, stream->dir, - stream->type); + stream->type, + substream->number); + } } #ifndef CHIP_AU8810 else { @@ -506,6 +533,83 @@ static struct snd_kcontrol_new snd_vortex_mixer_spdif[] __devinitdata = { }, }; +/* subdevice PCM Volume control */ + +static int snd_vortex_pcm_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + vortex_t *vortex = snd_kcontrol_chip(kcontrol); + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = (VORTEX_IS_QUAD(vortex) ? 4 : 2); + uinfo->value.integer.min = -128; + uinfo->value.integer.max = 32; + return 0; +} + +static int snd_vortex_pcm_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int i; + vortex_t *vortex = snd_kcontrol_chip(kcontrol); + int subdev = kcontrol->id.subdevice; + struct pcm_vol *p = &vortex->pcm_vol[subdev]; + int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2); + for (i = 0; i < max_chn; i++) + ucontrol->value.integer.value[i] = p->vol[i]; + return 0; +} + +static int snd_vortex_pcm_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int i; + int changed = 0; + int mixin; + unsigned char vol; + vortex_t *vortex = snd_kcontrol_chip(kcontrol); + int subdev = kcontrol->id.subdevice; + struct pcm_vol *p = &vortex->pcm_vol[subdev]; + int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2); + for (i = 0; i < max_chn; i++) { + if (p->vol[i] != ucontrol->value.integer.value[i]) { + p->vol[i] = ucontrol->value.integer.value[i]; + if (p->active) { + switch (vortex->dma_adb[p->dma].nr_ch) { + case 1: + mixin = p->mixin[0]; + break; + case 2: + default: + mixin = p->mixin[(i < 2) ? i : (i - 2)]; + break; + case 4: + mixin = p->mixin[i]; + break; + }; + vol = p->vol[i]; + vortex_mix_setinputvolumebyte(vortex, + vortex->mixplayb[i], mixin, vol); + } + changed = 1; + } + } + return changed; +} + +static const DECLARE_TLV_DB_MINMAX(vortex_pcm_vol_db_scale, -9600, 2400); + +static struct snd_kcontrol_new snd_vortex_pcm_vol __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_INACTIVE, + .info = snd_vortex_pcm_vol_info, + .get = snd_vortex_pcm_vol_get, + .put = snd_vortex_pcm_vol_put, + .tlv = { .p = vortex_pcm_vol_db_scale }, +}; + /* create a pcm device */ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) { @@ -555,5 +659,20 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) return err; } } + if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_ADB) { + for (i = 0; i < NR_PCM; i++) { + chip->pcm_vol[i].active = 0; + chip->pcm_vol[i].dma = -1; + kctl = snd_ctl_new1(&snd_vortex_pcm_vol, chip); + if (!kctl) + return -ENOMEM; + chip->pcm_vol[i].kctl = kctl; + kctl->id.device = 0; + kctl->id.subdevice = i; + err = snd_ctl_add(chip->card, kctl); + if (err < 0) + return err; + } + } return 0; } -- cgit v1.2.3-18-g5258 From 15b52f10ec6131e1aff49e7823a67732cdc066a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 12:14:53 +0000 Subject: ASoC: Convert the WM5100 revision A updates to a regmap patch Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 46 +++++++++++++++++----------------------------- 1 file changed, 17 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 81056d8dc89..e40c81eaec3 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1310,10 +1310,7 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "PWM2", NULL, "PWM2 Driver" }, }; -static struct { - int reg; - int val; -} wm5100_reva_patches[] = { +static const __devinitdata struct reg_default wm5100_reva_patches[] = { { WM5100_AUDIO_IF_1_10, 0 }, { WM5100_AUDIO_IF_1_11, 1 }, { WM5100_AUDIO_IF_1_12, 2 }, @@ -1376,31 +1373,6 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, } regcache_cache_only(wm5100->regmap, false); - - switch (wm5100->rev) { - case 0: - regcache_cache_bypass(wm5100->regmap, true); - snd_soc_write(codec, 0x11, 0x3); - snd_soc_write(codec, 0x203, 0xc); - snd_soc_write(codec, 0x206, 0); - snd_soc_write(codec, 0x207, 0xf0); - snd_soc_write(codec, 0x208, 0x3c); - snd_soc_write(codec, 0x209, 0); - snd_soc_write(codec, 0x211, 0x20d8); - snd_soc_write(codec, 0x11, 0); - - for (i = 0; - i < ARRAY_SIZE(wm5100_reva_patches); - i++) - snd_soc_write(codec, - wm5100_reva_patches[i].reg, - wm5100_reva_patches[i].val); - regcache_cache_bypass(wm5100->regmap, false); - break; - default: - break; - } - regcache_sync(wm5100->regmap); } break; @@ -2703,6 +2675,22 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, goto err_reset; } + switch (wm5100->rev) { + case 0: + ret = regmap_register_patch(wm5100->regmap, + wm5100_reva_patches, + ARRAY_SIZE(wm5100_reva_patches)); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register patches: %d\n", + ret); + goto err_reset; + } + break; + default: + break; + } + + wm5100_init_gpio(i2c); for (i = 0; i < ARRAY_SIZE(wm5100->pdata.gpio_defaults); i++) { -- cgit v1.2.3-18-g5258 From 5509f2f80c711add6bbcec9af7f4bbba2e2cc22b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Jan 2012 19:51:34 +0000 Subject: ASoC: wm5100: Fix warnings from recent patches Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index e40c81eaec3..714256e609c 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1346,7 +1346,7 @@ static int wm5100_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - int ret, i; + int ret; switch (level) { case SND_SOC_BIAS_ON: @@ -2504,7 +2504,6 @@ err_gpio: static int wm5100_remove(struct snd_soc_codec *codec) { struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = to_i2c_client(codec->dev); if (wm5100->pdata.hp_pol) { gpio_free(wm5100->pdata.hp_pol); -- cgit v1.2.3-18-g5258 From 182c51ce7944a214dd77a0b5c0462241e49dd418 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 24 Jan 2012 21:07:55 +0000 Subject: ASoC: wm8962: Optimise power consumption for IN4 DC measurement usage When the hardware is configured with one or both of the IN4 inputs used for DC measurement (with no DC blocking capacitor connected) then we can improve power consumption slightly in idle modes by applying a register write sequence. Provide platform data to enable this, implemented using a regmap patch. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index cc4049e9174..2a654fd42d1 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3638,6 +3638,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .volatile_register = wm8962_soc_volatile, }; +/* Improve power consumption for IN4 DC measurement mode */ +static const struct reg_default wm8962_dc_measure[] = { + { 0xfd, 0x1 }, + { 0xcc, 0x40 }, + { 0xfd, 0 }, +}; + static const struct regmap_config wm8962_regmap = { .reg_bits = 16, .val_bits = 16, @@ -3653,6 +3660,7 @@ static const struct regmap_config wm8962_regmap = { static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct wm8962_pdata *pdata = dev_get_platdata(&i2c->dev); struct wm8962_priv *wm8962; unsigned int reg; int ret, i; @@ -3731,6 +3739,16 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, goto err_regmap; } + if (pdata && pdata->in4_dc_measure) { + ret = regmap_register_patch(wm8962->regmap, + wm8962_dc_measure, + ARRAY_SIZE(wm8962_dc_measure)); + if (ret != 0) + dev_err(&i2c->dev, + "Failed to configure for DC mesurement: %d\n", + ret); + } + regcache_cache_only(wm8962->regmap, true); ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.2.3-18-g5258 From 88e339541d28153b6d2bfad9b25b3462fcd2bcaa Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 25 Jan 2012 10:09:41 +0200 Subject: ASoC: soc-pcm: msbits constraint: Drop 8 and 16 bit sample sizes As per discussion we can safely ignore the 8 and 16 bit sample sizes when applying the msbits constraint. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 326890148a2..93be95b7864 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -68,7 +68,7 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, * like the DAC/ADC resolution to use but there isn't right now. */ static int sample_sizes[] = { - 8, 16, 24, 32, + 24, 32, }; static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, -- cgit v1.2.3-18-g5258 From 8747a6b7d5685ebb64f1ec4d58d9b1969df3e34d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 7 Dec 2011 15:31:26 +0200 Subject: ASoC: sdp4430: Correct author e-mail address Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/sdp4430.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 175ba9a04ed..ceadb137bf4 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -1,7 +1,7 @@ /* * sdp4430.c -- SoC audio for TI OMAP4430 SDP * - * Author: Misael Lopez Cruz + * Author: Misael Lopez Cruz * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License @@ -273,7 +273,7 @@ static void __exit sdp4430_soc_exit(void) } module_exit(sdp4430_soc_exit); -MODULE_AUTHOR("Misael Lopez Cruz "); +MODULE_AUTHOR("Misael Lopez Cruz "); MODULE_DESCRIPTION("ALSA SoC SDP4430"); MODULE_LICENSE("GPL"); -- cgit v1.2.3-18-g5258 From e15422edd463fbba7f38ffd8a2e2ca8564da1160 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 7 Dec 2011 15:38:38 +0200 Subject: ASoC: OMAP4: Rename the sdp4430 machine driver The same machine driver will support other boards with similar audio configuration (OMAP4, ABE, twl6040). Rename the driver to have more generic name. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/Kconfig | 2 +- sound/soc/omap/Makefile | 4 +- sound/soc/omap/omap-abe-twl6040.c | 279 ++++++++++++++++++++++++++++++++++++++ sound/soc/omap/sdp4430.c | 279 -------------------------------------- 4 files changed, 282 insertions(+), 282 deletions(-) create mode 100644 sound/soc/omap/omap-abe-twl6040.c delete mode 100644 sound/soc/omap/sdp4430.c (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index fb1bf2581ef..4eae92987fe 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -97,7 +97,7 @@ config SND_OMAP_SOC_SDP3430 Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. -config SND_OMAP_SOC_SDP4430 +config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for Texas Instruments SDP4430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP select SND_OMAP_SOC_DMIC diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 1fd723fb559..123ac18303e 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -20,7 +20,7 @@ snd-soc-overo-objs := overo.o snd-soc-omap3evm-objs := omap3evm.o snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o -snd-soc-sdp4430-objs := sdp4430.o +snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o @@ -36,7 +36,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o -obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c new file mode 100644 index 00000000000..ceadb137bf4 --- /dev/null +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -0,0 +1,279 @@ +/* + * sdp4430.c -- SoC audio for TI OMAP4430 SDP + * + * Author: Misael Lopez Cruz + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include + +#include +#include +#include +#include + +#include +#include +#include + +#include "omap-dmic.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" +#include "../codecs/twl6040.h" + +static int sdp4430_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int clk_id, freq; + int ret; + + clk_id = twl6040_get_clk_id(rtd->codec); + if (clk_id == TWL6040_SYSCLK_SEL_HPPLL) + freq = 38400000; + else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL) + freq = 32768; + else + return -EINVAL; + + /* set the codec mclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq, + SND_SOC_CLOCK_IN); + if (ret) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + return ret; +} + +static struct snd_soc_ops sdp4430_ops = { + .hw_params = sdp4430_hw_params, +}; + +static int sdp4430_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret = 0; + + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS, + 19200000, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set DMIC cpu system clock\n"); + return ret; + } + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + printk(KERN_ERR "can't set DMIC output clock\n"); + return ret; + } + return 0; +} + +static struct snd_soc_ops sdp4430_dmic_ops = { + .hw_params = sdp4430_dmic_hw_params, +}; + +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + +/* SDP4430 machine DAPM */ +static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_SPK("Earphone Spk", NULL), + SND_SOC_DAPM_INPUT("FM Stereo In"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Main Mic Bias"}, + {"SUBMIC", NULL, "Main Mic Bias"}, + {"Main Mic Bias", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Headset Stereophone (Headphone): HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, + + /* Earphone speaker */ + {"Earphone Spk", NULL, "EP"}, + + /* Aux/FM Stereo In: AFML, AFMR */ + {"AFML", NULL, "FM Stereo In"}, + {"AFMR", NULL, "FM Stereo In"}, +}; + +static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret, hs_trim; + + /* + * Configure McPDM offset cancellation based on the HSOTRIM value from + * twl6040. + */ + hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM); + omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim), + TWL6040_HSF_TRIM_RIGHT(hs_trim)); + + /* Headset jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + + if (machine_is_omap_4430sdp()) + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); + else + snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + + return ret; +} + +static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Digital Mic", NULL), +}; + +static const struct snd_soc_dapm_route dmic_audio_map[] = { + {"DMic", NULL, "Digital Mic1 Bias"}, + {"Digital Mic1 Bias", NULL, "Digital Mic"}, +}; + +static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets, + ARRAY_SIZE(sdp4430_dmic_dapm_widgets)); + if (ret) + return ret; + + return snd_soc_dapm_add_routes(dapm, dmic_audio_map, + ARRAY_SIZE(dmic_audio_map)); +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp4430_dai[] = { + { + .name = "TWL6040", + .stream_name = "TWL6040", + .cpu_dai_name = "omap-mcpdm", + .codec_dai_name = "twl6040-legacy", + .platform_name = "omap-pcm-audio", + .codec_name = "twl6040-codec", + .init = sdp4430_twl6040_init, + .ops = &sdp4430_ops, + }, + { + .name = "DMIC", + .stream_name = "DMIC Capture", + .cpu_dai_name = "omap-dmic", + .codec_dai_name = "dmic-hifi", + .platform_name = "omap-pcm-audio", + .codec_name = "dmic-codec", + .init = sdp4430_dmic_init, + .ops = &sdp4430_dmic_ops, + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_sdp4430 = { + .name = "SDP4430", + .owner = THIS_MODULE, + .dai_link = sdp4430_dai, + .num_links = ARRAY_SIZE(sdp4430_dai), + + .dapm_widgets = sdp4430_twl6040_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), +}; + +static struct platform_device *sdp4430_snd_device; + +static int __init sdp4430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_4430sdp()) + return -ENODEV; + printk(KERN_INFO "SDP4430 SoC init\n"); + + sdp4430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp4430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430); + + ret = platform_device_add(sdp4430_snd_device); + if (ret) + goto err; + + return 0; + +err: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp4430_snd_device); + return ret; +} +module_init(sdp4430_soc_init); + +static void __exit sdp4430_soc_exit(void) +{ + platform_device_unregister(sdp4430_snd_device); +} +module_exit(sdp4430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz "); +MODULE_DESCRIPTION("ALSA SoC SDP4430"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c deleted file mode 100644 index ceadb137bf4..00000000000 --- a/sound/soc/omap/sdp4430.c +++ /dev/null @@ -1,279 +0,0 @@ -/* - * sdp4430.c -- SoC audio for TI OMAP4430 SDP - * - * Author: Misael Lopez Cruz - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License - * version 2 as published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include -#include -#include -#include - -#include -#include -#include -#include - -#include -#include -#include - -#include "omap-dmic.h" -#include "omap-mcpdm.h" -#include "omap-pcm.h" -#include "../codecs/twl6040.h" - -static int sdp4430_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int clk_id, freq; - int ret; - - clk_id = twl6040_get_clk_id(rtd->codec); - if (clk_id == TWL6040_SYSCLK_SEL_HPPLL) - freq = 38400000; - else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL) - freq = 32768; - else - return -EINVAL; - - /* set the codec mclk */ - ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq, - SND_SOC_CLOCK_IN); - if (ret) { - printk(KERN_ERR "can't set codec system clock\n"); - return ret; - } - return ret; -} - -static struct snd_soc_ops sdp4430_ops = { - .hw_params = sdp4430_hw_params, -}; - -static int sdp4430_dmic_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0; - - ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS, - 19200000, SND_SOC_CLOCK_IN); - if (ret < 0) { - printk(KERN_ERR "can't set DMIC cpu system clock\n"); - return ret; - } - ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000, - SND_SOC_CLOCK_OUT); - if (ret < 0) { - printk(KERN_ERR "can't set DMIC output clock\n"); - return ret; - } - return 0; -} - -static struct snd_soc_ops sdp4430_dmic_ops = { - .hw_params = sdp4430_dmic_hw_params, -}; - -/* Headset jack */ -static struct snd_soc_jack hs_jack; - -/*Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin hs_jack_pins[] = { - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headset Stereophone", - .mask = SND_JACK_HEADPHONE, - }, -}; - -/* SDP4430 machine DAPM */ -static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Ext Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_HP("Headset Stereophone", NULL), - SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("FM Stereo In"), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - /* External Mics: MAINMIC, SUBMIC with bias*/ - {"MAINMIC", NULL, "Main Mic Bias"}, - {"SUBMIC", NULL, "Main Mic Bias"}, - {"Main Mic Bias", NULL, "Ext Mic"}, - - /* External Speakers: HFL, HFR */ - {"Ext Spk", NULL, "HFL"}, - {"Ext Spk", NULL, "HFR"}, - - /* Headset Mic: HSMIC with bias */ - {"HSMIC", NULL, "Headset Mic Bias"}, - {"Headset Mic Bias", NULL, "Headset Mic"}, - - /* Headset Stereophone (Headphone): HSOL, HSOR */ - {"Headset Stereophone", NULL, "HSOL"}, - {"Headset Stereophone", NULL, "HSOR"}, - - /* Earphone speaker */ - {"Earphone Spk", NULL, "EP"}, - - /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "FM Stereo In"}, - {"AFMR", NULL, "FM Stereo In"}, -}; - -static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - int ret, hs_trim; - - /* - * Configure McPDM offset cancellation based on the HSOTRIM value from - * twl6040. - */ - hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM); - omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim), - TWL6040_HSF_TRIM_RIGHT(hs_trim)); - - /* Headset jack detection */ - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET, &hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); - - if (machine_is_omap_4430sdp()) - twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); - else - snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET); - - return ret; -} - -static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Digital Mic", NULL), -}; - -static const struct snd_soc_dapm_route dmic_audio_map[] = { - {"DMic", NULL, "Digital Mic1 Bias"}, - {"Digital Mic1 Bias", NULL, "Digital Mic"}, -}; - -static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets, - ARRAY_SIZE(sdp4430_dmic_dapm_widgets)); - if (ret) - return ret; - - return snd_soc_dapm_add_routes(dapm, dmic_audio_map, - ARRAY_SIZE(dmic_audio_map)); -} - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link sdp4430_dai[] = { - { - .name = "TWL6040", - .stream_name = "TWL6040", - .cpu_dai_name = "omap-mcpdm", - .codec_dai_name = "twl6040-legacy", - .platform_name = "omap-pcm-audio", - .codec_name = "twl6040-codec", - .init = sdp4430_twl6040_init, - .ops = &sdp4430_ops, - }, - { - .name = "DMIC", - .stream_name = "DMIC Capture", - .cpu_dai_name = "omap-dmic", - .codec_dai_name = "dmic-hifi", - .platform_name = "omap-pcm-audio", - .codec_name = "dmic-codec", - .init = sdp4430_dmic_init, - .ops = &sdp4430_dmic_ops, - }, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_sdp4430 = { - .name = "SDP4430", - .owner = THIS_MODULE, - .dai_link = sdp4430_dai, - .num_links = ARRAY_SIZE(sdp4430_dai), - - .dapm_widgets = sdp4430_twl6040_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), -}; - -static struct platform_device *sdp4430_snd_device; - -static int __init sdp4430_soc_init(void) -{ - int ret; - - if (!machine_is_omap_4430sdp()) - return -ENODEV; - printk(KERN_INFO "SDP4430 SoC init\n"); - - sdp4430_snd_device = platform_device_alloc("soc-audio", -1); - if (!sdp4430_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430); - - ret = platform_device_add(sdp4430_snd_device); - if (ret) - goto err; - - return 0; - -err: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(sdp4430_snd_device); - return ret; -} -module_init(sdp4430_soc_init); - -static void __exit sdp4430_soc_exit(void) -{ - platform_device_unregister(sdp4430_snd_device); -} -module_exit(sdp4430_soc_exit); - -MODULE_AUTHOR("Misael Lopez Cruz "); -MODULE_DESCRIPTION("ALSA SoC SDP4430"); -MODULE_LICENSE("GPL"); - -- cgit v1.2.3-18-g5258 From 13f81d65959eb3512b39e8c338e81e018d1c515b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 7 Dec 2011 15:54:50 +0200 Subject: ASoC: omap-abe-twl6040: Correct internal prefix, Kconfig entry Change the internal prefixes within the driver from sdp4430. At he same time correct the Kconfig text as well. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/Kconfig | 7 +++-- sound/soc/omap/omap-abe-twl6040.c | 65 ++++++++++++++++++++------------------- 2 files changed, 37 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4eae92987fe..98410b833f1 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -98,15 +98,16 @@ config SND_OMAP_SOC_SDP3430 SDP3430. config SND_OMAP_SOC_OMAP_ABE_TWL6040 - tristate "SoC Audio support for Texas Instruments SDP4430" + tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 select SND_SOC_DMIC help - Say Y if you want to add support for SoC audio on Texas Instruments - SDP4430. + Say Y if you want to add support for SoC audio on OMAP boards using + ABE and twl6040 codec. This driver currently supports: + - SDP4430/Blaze boards config SND_OMAP_SOC_OMAP4_HDMI tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index ceadb137bf4..376ca351e09 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -1,5 +1,6 @@ /* - * sdp4430.c -- SoC audio for TI OMAP4430 SDP + * omap-abe-twl6040.c -- SoC audio for TI OMAP based boards with ABE and + * twl6040 codec * * Author: Misael Lopez Cruz * @@ -38,7 +39,7 @@ #include "omap-pcm.h" #include "../codecs/twl6040.h" -static int sdp4430_hw_params(struct snd_pcm_substream *substream, +static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -64,11 +65,11 @@ static int sdp4430_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops sdp4430_ops = { - .hw_params = sdp4430_hw_params, +static struct snd_soc_ops omap_abe_ops = { + .hw_params = omap_abe_hw_params, }; -static int sdp4430_dmic_hw_params(struct snd_pcm_substream *substream, +static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -90,8 +91,8 @@ static int sdp4430_dmic_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops sdp4430_dmic_ops = { - .hw_params = sdp4430_dmic_hw_params, +static struct snd_soc_ops omap_abe_dmic_ops = { + .hw_params = omap_abe_dmic_hw_params, }; /* Headset jack */ @@ -110,7 +111,7 @@ static struct snd_soc_jack_pin hs_jack_pins[] = { }; /* SDP4430 machine DAPM */ -static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { +static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Ext Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -145,7 +146,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFMR", NULL, "FM Stereo In"}, }; -static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) +static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; int ret, hs_trim; @@ -175,7 +176,7 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static const struct snd_soc_dapm_widget sdp4430_dmic_dapm_widgets[] = { +static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { SND_SOC_DAPM_MIC("Digital Mic", NULL), }; @@ -184,14 +185,14 @@ static const struct snd_soc_dapm_route dmic_audio_map[] = { {"Digital Mic1 Bias", NULL, "Digital Mic"}, }; -static int sdp4430_dmic_init(struct snd_soc_pcm_runtime *rtd) +static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - ret = snd_soc_dapm_new_controls(dapm, sdp4430_dmic_dapm_widgets, - ARRAY_SIZE(sdp4430_dmic_dapm_widgets)); + ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets, + ARRAY_SIZE(dmic_dapm_widgets)); if (ret) return ret; @@ -208,8 +209,8 @@ static struct snd_soc_dai_link sdp4430_dai[] = { .codec_dai_name = "twl6040-legacy", .platform_name = "omap-pcm-audio", .codec_name = "twl6040-codec", - .init = sdp4430_twl6040_init, - .ops = &sdp4430_ops, + .init = omap_abe_twl6040_init, + .ops = &omap_abe_ops, }, { .name = "DMIC", @@ -218,27 +219,27 @@ static struct snd_soc_dai_link sdp4430_dai[] = { .codec_dai_name = "dmic-hifi", .platform_name = "omap-pcm-audio", .codec_name = "dmic-codec", - .init = sdp4430_dmic_init, - .ops = &sdp4430_dmic_ops, + .init = omap_abe_dmic_init, + .ops = &omap_abe_dmic_ops, }, }; /* Audio machine driver */ -static struct snd_soc_card snd_soc_sdp4430 = { +static struct snd_soc_card omap_abe_card = { .name = "SDP4430", .owner = THIS_MODULE, .dai_link = sdp4430_dai, .num_links = ARRAY_SIZE(sdp4430_dai), - .dapm_widgets = sdp4430_twl6040_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(sdp4430_twl6040_dapm_widgets), + .dapm_widgets = twl6040_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static struct platform_device *sdp4430_snd_device; +static struct platform_device *omap_abe_snd_device; -static int __init sdp4430_soc_init(void) +static int __init omap_abe_soc_init(void) { int ret; @@ -246,15 +247,15 @@ static int __init sdp4430_soc_init(void) return -ENODEV; printk(KERN_INFO "SDP4430 SoC init\n"); - sdp4430_snd_device = platform_device_alloc("soc-audio", -1); - if (!sdp4430_snd_device) { + omap_abe_snd_device = platform_device_alloc("soc-audio", -1); + if (!omap_abe_snd_device) { printk(KERN_ERR "Platform device allocation failed\n"); return -ENOMEM; } - platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430); + platform_set_drvdata(omap_abe_snd_device, &omap_abe_card); - ret = platform_device_add(sdp4430_snd_device); + ret = platform_device_add(omap_abe_snd_device); if (ret) goto err; @@ -262,18 +263,18 @@ static int __init sdp4430_soc_init(void) err: printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(sdp4430_snd_device); + platform_device_put(omap_abe_snd_device); return ret; } -module_init(sdp4430_soc_init); +module_init(omap_abe_soc_init); -static void __exit sdp4430_soc_exit(void) +static void __exit omap_abe_soc_exit(void) { - platform_device_unregister(sdp4430_snd_device); + platform_device_unregister(omap_abe_snd_device); } -module_exit(sdp4430_soc_exit); +module_exit(omap_abe_soc_exit); MODULE_AUTHOR("Misael Lopez Cruz "); -MODULE_DESCRIPTION("ALSA SoC SDP4430"); +MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec"); MODULE_LICENSE("GPL"); -- cgit v1.2.3-18-g5258 From d1be1d2fd1158b45dca99e3924621e1068c87b82 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 8 Dec 2011 11:07:42 +0200 Subject: ASoC: omap-abe-twl6040: Convert to platform deriver Convert the OMAP4 ABE/TWL6040 machine driver to platform driver. For the card name use the string provided via platform data. The card's name for OMAP4 SDP4430 has been changed: SDP4430 -> OMAP4-SDP4430 Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 60 +++++++++++++++++++++++---------------- 1 file changed, 36 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 376ca351e09..6b7047066f4 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include @@ -226,7 +227,6 @@ static struct snd_soc_dai_link sdp4430_dai[] = { /* Audio machine driver */ static struct snd_soc_card omap_abe_card = { - .name = "SDP4430", .owner = THIS_MODULE, .dai_link = sdp4430_dai, .num_links = ARRAY_SIZE(sdp4430_dai), @@ -237,44 +237,56 @@ static struct snd_soc_card omap_abe_card = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static struct platform_device *omap_abe_snd_device; - -static int __init omap_abe_soc_init(void) +static __devinit int omap_abe_probe(struct platform_device *pdev) { + struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); + struct snd_soc_card *card = &omap_abe_card; int ret; - if (!machine_is_omap_4430sdp()) - return -ENODEV; - printk(KERN_INFO "SDP4430 SoC init\n"); + card->dev = &pdev->dev; - omap_abe_snd_device = platform_device_alloc("soc-audio", -1); - if (!omap_abe_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; + if (!pdata) { + dev_err(&pdev->dev, "Missing pdata\n"); + return -ENODEV; } - platform_set_drvdata(omap_abe_snd_device, &omap_abe_card); + if (pdata->card_name) { + card->name = pdata->card_name; + } else { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } - ret = platform_device_add(omap_abe_snd_device); + ret = snd_soc_register_card(card); if (ret) - goto err; - - return 0; + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); -err: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(omap_abe_snd_device); return ret; } -module_init(omap_abe_soc_init); -static void __exit omap_abe_soc_exit(void) +static int __devexit omap_abe_remove(struct platform_device *pdev) { - platform_device_unregister(omap_abe_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; } -module_exit(omap_abe_soc_exit); + +static struct platform_driver omap_abe_driver = { + .driver = { + .name = "omap-abe-twl6040", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = omap_abe_probe, + .remove = __devexit_p(omap_abe_remove), +}; + +module_platform_driver(omap_abe_driver); MODULE_AUTHOR("Misael Lopez Cruz "); MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:omap-abe-twl6040"); -- cgit v1.2.3-18-g5258 From 778cee7afd063b1321dd9a2d2ecd8822d76bb33a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Dec 2011 12:47:21 +0200 Subject: ASoC: twl6040: Convert MICBIAS to SUPPLY widget In order to avoid breakage change the omap-abe-twl6040 machine driver's routing. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/codecs/twl6040.c | 16 ++++++++-------- sound/soc/omap/omap-abe-twl6040.c | 14 +++++++------- 2 files changed, 15 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 284dd2e9997..1a64edf671a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1138,14 +1138,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { TWL6040_REG_MICRCTL, 2, 0), /* Microphone bias */ - SND_SOC_DAPM_MICBIAS("Headset Mic Bias", - TWL6040_REG_AMICBCTL, 0, 0), - SND_SOC_DAPM_MICBIAS("Main Mic Bias", - TWL6040_REG_AMICBCTL, 4, 0), - SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias", - TWL6040_REG_DMICBCTL, 0, 0), - SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias", - TWL6040_REG_DMICBCTL, 4, 0), + SND_SOC_DAPM_SUPPLY("Headset Mic Bias", + TWL6040_REG_AMICBCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Main Mic Bias", + TWL6040_REG_AMICBCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias", + TWL6040_REG_DMICBCTL, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias", + TWL6040_REG_DMICBCTL, 4, 0, NULL, 0), /* DACs */ SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0), diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 6b7047066f4..ca7152cb6a2 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -123,17 +123,17 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { static const struct snd_soc_dapm_route audio_map[] = { /* External Mics: MAINMIC, SUBMIC with bias*/ - {"MAINMIC", NULL, "Main Mic Bias"}, - {"SUBMIC", NULL, "Main Mic Bias"}, - {"Main Mic Bias", NULL, "Ext Mic"}, + {"MAINMIC", NULL, "Ext Mic"}, + {"SUBMIC", NULL, "Ext Mic"}, + {"Ext Mic", NULL, "Main Mic Bias"}, /* External Speakers: HFL, HFR */ {"Ext Spk", NULL, "HFL"}, {"Ext Spk", NULL, "HFR"}, /* Headset Mic: HSMIC with bias */ - {"HSMIC", NULL, "Headset Mic Bias"}, - {"Headset Mic Bias", NULL, "Headset Mic"}, + {"HSMIC", NULL, "Headset Mic"}, + {"Headset Mic", NULL, "Headset Mic Bias"}, /* Headset Stereophone (Headphone): HSOL, HSOR */ {"Headset Stereophone", NULL, "HSOL"}, @@ -182,8 +182,8 @@ static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = { }; static const struct snd_soc_dapm_route dmic_audio_map[] = { - {"DMic", NULL, "Digital Mic1 Bias"}, - {"Digital Mic1 Bias", NULL, "Digital Mic"}, + {"DMic", NULL, "Digital Mic"}, + {"Digital Mic", NULL, "Digital Mic1 Bias"}, }; static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) -- cgit v1.2.3-18-g5258 From b3208839106476fe31b4bf3e6a2470cae509b2a5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Dec 2011 12:55:49 +0200 Subject: ASoC: omap-abe-twl6040: Add complete DAPM routing SDP4430 is a reference platform, and as such it has all possible audio routing implemented. Correct the DAPM routing to be complete. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 46 ++++++++++++++++++++++++--------------- 1 file changed, 28 insertions(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index ca7152cb6a2..f2b5fec8f68 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -113,38 +113,48 @@ static struct snd_soc_jack_pin hs_jack_pins[] = { /* SDP4430 machine DAPM */ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { - SND_SOC_DAPM_MIC("Ext Mic", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), + /* Outputs */ SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("FM Stereo In"), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_SPK("Vibrator", NULL), + + /* Inputs */ + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Handset Mic", NULL), + SND_SOC_DAPM_MIC("Sub Handset Mic", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { - /* External Mics: MAINMIC, SUBMIC with bias*/ - {"MAINMIC", NULL, "Ext Mic"}, - {"SUBMIC", NULL, "Ext Mic"}, - {"Ext Mic", NULL, "Main Mic Bias"}, + /* Routings for outputs */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, + + {"Earphone Spk", NULL, "EP"}, - /* External Speakers: HFL, HFR */ {"Ext Spk", NULL, "HFL"}, {"Ext Spk", NULL, "HFR"}, - /* Headset Mic: HSMIC with bias */ + {"Line Out", NULL, "AUXL"}, + {"Line Out", NULL, "AUXR"}, + + {"Vibrator", NULL, "VIBRAL"}, + {"Vibrator", NULL, "VIBRAR"}, + + /* Routings for inputs */ {"HSMIC", NULL, "Headset Mic"}, {"Headset Mic", NULL, "Headset Mic Bias"}, - /* Headset Stereophone (Headphone): HSOL, HSOR */ - {"Headset Stereophone", NULL, "HSOL"}, - {"Headset Stereophone", NULL, "HSOR"}, + {"MAINMIC", NULL, "Main Handset Mic"}, + {"Main Handset Mic", NULL, "Main Mic Bias"}, - /* Earphone speaker */ - {"Earphone Spk", NULL, "EP"}, + {"SUBMIC", NULL, "Sub Handset Mic"}, + {"Sub Handset Mic", NULL, "Main Mic Bias"}, - /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "FM Stereo In"}, - {"AFMR", NULL, "FM Stereo In"}, + {"AFML", NULL, "Line In"}, + {"AFMR", NULL, "Line In"}, }; static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) -- cgit v1.2.3-18-g5258 From 8d946dd7dc5cc3f0eae3d6df2c9e927497e38850 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Dec 2011 13:10:02 +0200 Subject: ASoC: omap-abe-twl6040: DAI link selection based on platform data We can have machines without DMIC connected. In this case there is no need to create amother (unusable) capture PCM on the card. The existence of the DMIC connection can be checked via pdata->has_dmic. Select the correct dai_link structure for the card based on pdata->has_dmic. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index f2b5fec8f68..5b781f904cd 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -212,7 +212,7 @@ static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) } /* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link sdp4430_dai[] = { +static struct snd_soc_dai_link twl6040_dmic_dai[] = { { .name = "TWL6040", .stream_name = "TWL6040", @@ -235,11 +235,22 @@ static struct snd_soc_dai_link sdp4430_dai[] = { }, }; +static struct snd_soc_dai_link twl6040_only_dai[] = { + { + .name = "TWL6040", + .stream_name = "TWL6040", + .cpu_dai_name = "omap-mcpdm", + .codec_dai_name = "twl6040-legacy", + .platform_name = "omap-pcm-audio", + .codec_name = "twl6040-codec", + .init = omap_abe_twl6040_init, + .ops = &omap_abe_ops, + }, +}; + /* Audio machine driver */ static struct snd_soc_card omap_abe_card = { .owner = THIS_MODULE, - .dai_link = sdp4430_dai, - .num_links = ARRAY_SIZE(sdp4430_dai), .dapm_widgets = twl6040_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), @@ -267,6 +278,14 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } + if (pdata->has_dmic) { + card->dai_link = twl6040_dmic_dai; + card->num_links = ARRAY_SIZE(twl6040_dmic_dai); + } else { + card->dai_link = twl6040_only_dai; + card->num_links = ARRAY_SIZE(twl6040_only_dai); + } + ret = snd_soc_register_card(card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", -- cgit v1.2.3-18-g5258 From 4ca07cb0cf698151a21e9a9e68a67ca9656d3632 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Dec 2011 13:58:01 +0200 Subject: ASoC: omap-abe-twl6040: Configure card according to platform data Disable the not connected pins on the board based on the received platform data. DO not register the jack function on boards, which does not have means to detect it (jack is always connected). Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 44 ++++++++++++++++++++++++++++----------- 1 file changed, 32 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 5b781f904cd..1da26031e26 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -157,10 +157,32 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFMR", NULL, "Line In"}, }; +static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm, + int connected, char *pin) +{ + if (!connected) + snd_soc_dapm_disable_pin(dapm, pin); +} + static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - int ret, hs_trim; + struct snd_soc_card *card = codec->card; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); + int hs_trim; + int ret = 0; + + /* Disable not connected paths if not used */ + twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); + twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); + twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); + twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); + twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator"); + twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); + twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); /* * Configure McPDM offset cancellation based on the HSOTRIM value from @@ -170,19 +192,17 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim), TWL6040_HSF_TRIM_RIGHT(hs_trim)); - /* Headset jack detection */ - ret = snd_soc_jack_new(codec, "Headset Jack", - SND_JACK_HEADSET, &hs_jack); - if (ret) - return ret; - - ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), - hs_jack_pins); + /* Headset jack detection only if it is supported */ + if (pdata->jack_detection) { + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; - if (machine_is_omap_4430sdp()) + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); - else - snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + } return ret; } -- cgit v1.2.3-18-g5258 From 62ba98ce4a74ef606e6a7f6c8541fc5e3127f944 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 24 Jan 2012 12:36:10 +0200 Subject: ASoC: omap-abe-twl6040: Use provided MCLK frequency from pdata Avoid using hardwired configuration for MCLK frequency. Different board design might use other MCLK frequency. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 1da26031e26..93bb8eee22b 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -45,12 +45,15 @@ static int omap_abe_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = codec->card; + struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); int clk_id, freq; int ret; clk_id = twl6040_get_clk_id(rtd->codec); if (clk_id == TWL6040_SYSCLK_SEL_HPPLL) - freq = 38400000; + freq = pdata->mclk_freq; else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL) freq = 32768; else @@ -298,6 +301,11 @@ static __devinit int omap_abe_probe(struct platform_device *pdev) return -ENODEV; } + if (!pdata->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency missing\n"); + return -ENODEV; + } + if (pdata->has_dmic) { card->dai_link = twl6040_dmic_dai; card->num_links = ARRAY_SIZE(twl6040_dmic_dai); -- cgit v1.2.3-18-g5258 From 6b21ed851624a03f11ea9ed3f229f56419e03686 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 22 Dec 2011 14:19:05 +0200 Subject: ASoC: Kconfig: OMAP4: Enable support for PandaBoards Enable ASoC audio support for OMAP4 based machines with twl6040 codec via the omap-abe-twl6040 machine driver. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown --- sound/soc/omap/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 98410b833f1..47b23fea20c 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -99,7 +99,7 @@ config SND_OMAP_SOC_SDP3430 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP + depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4 select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 @@ -108,6 +108,8 @@ config SND_OMAP_SOC_OMAP_ABE_TWL6040 Say Y if you want to add support for SoC audio on OMAP boards using ABE and twl6040 codec. This driver currently supports: - SDP4430/Blaze boards + - PandaBoard (4430) + - PandaBoardES (4460) config SND_OMAP_SOC_OMAP4_HDMI tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" -- cgit v1.2.3-18-g5258 From 62ea874abc11f02dbeb05314eb82f7d38e82e894 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 21:14:48 +0000 Subject: ASoC: Provide REGULATOR_SUPPLY widget type Modern devices allow systems to enable and disable individual supplies on the device, allowing additional power saving by switching off regulators which power portions of the device which are not currently in use. Add a new SND_SOC_DAPM_REGULATOR_SUPPLY widget type factoring out the code for managing such widgets from individual drivers. The widget name will be used as the supply name when requesting the regulator from the regulator API. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 49 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 31a06b2b444..30f9b5c71ee 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -40,6 +40,7 @@ #include #include #include +#include #include #include #include @@ -55,6 +56,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, + [snd_soc_dapm_regulator_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, @@ -90,6 +92,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, + [snd_soc_dapm_regulator_supply] = 11, [snd_soc_dapm_supply] = 11, [snd_soc_dapm_post] = 12, }; @@ -352,6 +355,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_hp: @@ -680,8 +684,13 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) DAPM_UPDATE_STAT(widget, path_checks); - if (widget->id == snd_soc_dapm_supply) + switch (widget->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: return 0; + default: + break; + } switch (widget->id) { case snd_soc_dapm_adc: @@ -745,8 +754,13 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) DAPM_UPDATE_STAT(widget, path_checks); - if (widget->id == snd_soc_dapm_supply) + switch (widget->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: return 0; + default: + break; + } /* active stream ? */ switch (widget->id) { @@ -828,6 +842,19 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_reg_event); +/* + * Handler for regulator supply widget. + */ +int dapm_regulator_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + return regulator_enable(w->priv); + else + return regulator_disable_deferred(w->priv, w->shift); +} +EXPORT_SYMBOL_GPL(dapm_regulator_event); + static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) { if (w->power_checked) @@ -1308,6 +1335,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, } switch (w->id) { case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -1411,6 +1439,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) */ switch (w->id) { case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1769,6 +1798,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -2007,6 +2037,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_pre: case snd_soc_dapm_post: case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: list_add(&path->list, &dapm->card->paths); @@ -2673,10 +2704,25 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w; size_t name_len; + int ret; if ((w = dapm_cnew_widget(widget)) == NULL) return -ENOMEM; + switch (w->id) { + case snd_soc_dapm_regulator_supply: + w->priv = devm_regulator_get(dapm->dev, w->name); + if (IS_ERR(w->priv)) { + ret = PTR_ERR(w->priv); + dev_err(dapm->dev, "Failed to request %s: %d\n", + w->name, ret); + return ret; + } + break; + default: + break; + } + name_len = strlen(widget->name) + 1; if (dapm->codec && dapm->codec->name_prefix) name_len += 1 + strlen(dapm->codec->name_prefix); @@ -2722,6 +2768,7 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: w->power_check = dapm_supply_check_power; break; default: -- cgit v1.2.3-18-g5258 From 1cf733569e6f484a09cb7e4b8602a48c32864594 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 22:10:24 +0000 Subject: ASoC: wm5100: Move regulator supplies over to DAPM infrastructure Saves a nice block of code. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 131 ++++------------------------------------------ 1 file changed, 10 insertions(+), 121 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 714256e609c..c1c8bdb7bb0 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -55,9 +55,6 @@ struct wm5100_priv { struct snd_soc_codec *codec; struct regulator_bulk_data core_supplies[WM5100_NUM_CORE_SUPPLIES]; - struct regulator *cpvdd; - struct regulator *dbvdd2; - struct regulator *dbvdd3; int rev; @@ -777,85 +774,6 @@ static int wm5100_out_ev(struct snd_soc_dapm_widget *w, return 0; } -static int wm5100_cp_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, - int event) -{ - struct snd_soc_codec *codec = w->codec; - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - int ret; - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - ret = regulator_enable(wm5100->cpvdd); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable CPVDD: %d\n", - ret); - return ret; - } - return ret; - - case SND_SOC_DAPM_POST_PMD: - ret = regulator_disable_deferred(wm5100->cpvdd, 20); - if (ret != 0) { - dev_err(codec->dev, "Failed to disable CPVDD: %d\n", - ret); - return ret; - } - return ret; - - default: - BUG(); - return 0; - } -} - -static int wm5100_dbvdd_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, - int event) -{ - struct snd_soc_codec *codec = w->codec; - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - struct regulator *regulator; - int ret; - - switch (w->shift) { - case 2: - regulator = wm5100->dbvdd2; - break; - case 3: - regulator = wm5100->dbvdd3; - break; - default: - BUG(); - return 0; - } - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - ret = regulator_enable(regulator); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", - w->shift, ret); - return ret; - } - return ret; - - case SND_SOC_DAPM_POST_PMD: - ret = regulator_disable(regulator); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", - w->shift, ret); - return ret; - } - return ret; - - default: - BUG(); - return 0; - } -} - static void wm5100_log_status3(struct wm5100_priv *wm5100, int val) { if (val & WM5100_SPK_SHUTDOWN_WARN_EINT) @@ -926,18 +844,16 @@ SND_SOC_DAPM_SUPPLY("SYSCLK", WM5100_CLOCKING_3, WM5100_SYSCLK_ENA_SHIFT, 0, SND_SOC_DAPM_SUPPLY("ASYNCCLK", WM5100_CLOCKING_6, WM5100_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), + SND_SOC_DAPM_SUPPLY("CP1", WM5100_HP_CHARGE_PUMP_1, WM5100_CP1_ENA_SHIFT, 0, - wm5100_cp_ev, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + NULL, 0), SND_SOC_DAPM_SUPPLY("CP2", WM5100_MIC_CHARGE_PUMP_1, WM5100_CP2_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("CP2 Active", WM5100_MIC_CHARGE_PUMP_1, - WM5100_CP2_BYPASS_SHIFT, 1, wm5100_cp_ev, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_SUPPLY("DBVDD2", SND_SOC_NOPM, 2, 0, wm5100_dbvdd_ev, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_SUPPLY("DBVDD3", SND_SOC_NOPM, 3, 0, wm5100_dbvdd_ev, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + WM5100_CP2_BYPASS_SHIFT, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM5100_MIC_BIAS_CTRL_1, WM5100_MICB1_ENA_SHIFT, 0, NULL, 0), @@ -1148,6 +1064,9 @@ SND_SOC_DAPM_POST("Post", wm5100_post_ev), }; static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { + { "CP1", NULL, "CPVDD" }, + { "CP2 Active", NULL, "CPVDD" }, + { "IN1L", NULL, "SYSCLK" }, { "IN1R", NULL, "SYSCLK" }, { "IN2L", NULL, "SYSCLK" }, @@ -2593,33 +2512,12 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, goto err_regmap; } - wm5100->cpvdd = regulator_get(&i2c->dev, "CPVDD"); - if (IS_ERR(wm5100->cpvdd)) { - ret = PTR_ERR(wm5100->cpvdd); - dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); - goto err_core; - } - - wm5100->dbvdd2 = regulator_get(&i2c->dev, "DBVDD2"); - if (IS_ERR(wm5100->dbvdd2)) { - ret = PTR_ERR(wm5100->dbvdd2); - dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); - goto err_cpvdd; - } - - wm5100->dbvdd3 = regulator_get(&i2c->dev, "DBVDD3"); - if (IS_ERR(wm5100->dbvdd3)) { - ret = PTR_ERR(wm5100->dbvdd3); - dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); - goto err_dbvdd2; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", ret); - goto err_dbvdd3; + goto err_core; } if (wm5100->pdata.ldo_ena) { @@ -2788,12 +2686,6 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -err_dbvdd3: - regulator_put(wm5100->dbvdd3); -err_dbvdd2: - regulator_put(wm5100->dbvdd2); -err_cpvdd: - regulator_put(wm5100->cpvdd); err_core: regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); @@ -2819,9 +2711,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } - regulator_put(wm5100->dbvdd3); - regulator_put(wm5100->dbvdd2); - regulator_put(wm5100->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); regmap_exit(wm5100->regmap); -- cgit v1.2.3-18-g5258 From d5315a23ccdf921c838d26df6360b439c8d7ac83 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 19:29:41 +0000 Subject: ASoC: wm2200: Add WM2200 CODEC driver The WM2200 is a low power mobile CODEC with enhanced Wolfson myZone Ambient Noise Cancellation (ANC) intended for mobile telephony applications. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm2200.c | 2285 ++++++++++++++++++++++++++++ sound/soc/codecs/wm2200.h | 3674 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 5965 insertions(+) create mode 100644 sound/soc/codecs/wm2200.c create mode 100644 sound/soc/codecs/wm2200.h (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7c205e77d83..39aec3c12cf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WL1273 if MFD_WL1273_CORE select SND_SOC_WM1250_EV1 if I2C select SND_SOC_WM2000 if I2C + select SND_SOC_WM2200 if I2C select SND_SOC_WM5100 if I2C select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 @@ -292,6 +293,9 @@ config SND_SOC_WM1250_EV1 config SND_SOC_WM2000 tristate +config SND_SOC_WM2200 + tristate + config SND_SOC_WM5100 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index de8078178f8..716c5842bfd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -51,6 +51,7 @@ snd-soc-uda1380-objs := uda1380.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm1250-ev1-objs := wm1250-ev1.o snd-soc-wm2000-objs := wm2000.o +snd-soc-wm2200-objs := wm2200.o snd-soc-wm5100-objs := wm5100.o wm5100-tables.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o @@ -153,6 +154,7 @@ obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o +obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c new file mode 100644 index 00000000000..956490d320a --- /dev/null +++ b/sound/soc/codecs/wm2200.c @@ -0,0 +1,2285 @@ +/* + * wm2200.c -- WM2200 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm2200.h" + +/* The code assumes DCVDD is generated internally */ +#define WM2200_NUM_CORE_SUPPLIES 2 +static const char *wm2200_core_supply_names[WM2200_NUM_CORE_SUPPLIES] = { + "DBVDD", + "LDOVDD", +}; + +struct wm2200_fll { + int fref; + int fout; + int src; + struct completion lock; +}; + +/* codec private data */ +struct wm2200_priv { + struct regmap *regmap; + struct device *dev; + struct snd_soc_codec *codec; + struct wm2200_pdata pdata; + struct regulator_bulk_data core_supplies[WM2200_NUM_CORE_SUPPLIES]; + + struct completion fll_lock; + int fll_fout; + int fll_fref; + int fll_src; + + int rev; + int sysclk; +}; + +static struct reg_default wm2200_reg_defaults[] = { + { 0x000B, 0x0000 }, /* R11 - Tone Generator 1 */ + { 0x0102, 0x0000 }, /* R258 - Clocking 3 */ + { 0x0103, 0x0011 }, /* R259 - Clocking 4 */ + { 0x0111, 0x0000 }, /* R273 - FLL Control 1 */ + { 0x0112, 0x0000 }, /* R274 - FLL Control 2 */ + { 0x0113, 0x0000 }, /* R275 - FLL Control 3 */ + { 0x0114, 0x0000 }, /* R276 - FLL Control 4 */ + { 0x0116, 0x0177 }, /* R278 - FLL Control 6 */ + { 0x0117, 0x0004 }, /* R279 - FLL Control 7 */ + { 0x0119, 0x0000 }, /* R281 - FLL EFS 1 */ + { 0x011A, 0x0002 }, /* R282 - FLL EFS 2 */ + { 0x0200, 0x0000 }, /* R512 - Mic Charge Pump 1 */ + { 0x0201, 0x03FF }, /* R513 - Mic Charge Pump 2 */ + { 0x0202, 0x9BDE }, /* R514 - DM Charge Pump 1 */ + { 0x020C, 0x0000 }, /* R524 - Mic Bias Ctrl 1 */ + { 0x020D, 0x0000 }, /* R525 - Mic Bias Ctrl 2 */ + { 0x020F, 0x0000 }, /* R527 - Ear Piece Ctrl 1 */ + { 0x0210, 0x0000 }, /* R528 - Ear Piece Ctrl 2 */ + { 0x0301, 0x0000 }, /* R769 - Input Enables */ + { 0x0302, 0x2240 }, /* R770 - IN1L Control */ + { 0x0303, 0x0040 }, /* R771 - IN1R Control */ + { 0x0304, 0x2240 }, /* R772 - IN2L Control */ + { 0x0305, 0x0040 }, /* R773 - IN2R Control */ + { 0x0306, 0x2240 }, /* R774 - IN3L Control */ + { 0x0307, 0x0040 }, /* R775 - IN3R Control */ + { 0x030A, 0x0000 }, /* R778 - RXANC_SRC */ + { 0x030B, 0x0022 }, /* R779 - Input Volume Ramp */ + { 0x030C, 0x0180 }, /* R780 - ADC Digital Volume 1L */ + { 0x030D, 0x0180 }, /* R781 - ADC Digital Volume 1R */ + { 0x030E, 0x0180 }, /* R782 - ADC Digital Volume 2L */ + { 0x030F, 0x0180 }, /* R783 - ADC Digital Volume 2R */ + { 0x0310, 0x0180 }, /* R784 - ADC Digital Volume 3L */ + { 0x0311, 0x0180 }, /* R785 - ADC Digital Volume 3R */ + { 0x0400, 0x0000 }, /* R1024 - Output Enables */ + { 0x0401, 0x0000 }, /* R1025 - DAC Volume Limit 1L */ + { 0x0402, 0x0000 }, /* R1026 - DAC Volume Limit 1R */ + { 0x0403, 0x0000 }, /* R1027 - DAC Volume Limit 2L */ + { 0x0404, 0x0000 }, /* R1028 - DAC Volume Limit 2R */ + { 0x0409, 0x0000 }, /* R1033 - DAC AEC Control 1 */ + { 0x040A, 0x0022 }, /* R1034 - Output Volume Ramp */ + { 0x040B, 0x0180 }, /* R1035 - DAC Digital Volume 1L */ + { 0x040C, 0x0180 }, /* R1036 - DAC Digital Volume 1R */ + { 0x040D, 0x0180 }, /* R1037 - DAC Digital Volume 2L */ + { 0x040E, 0x0180 }, /* R1038 - DAC Digital Volume 2R */ + { 0x0417, 0x0069 }, /* R1047 - PDM 1 */ + { 0x0418, 0x0000 }, /* R1048 - PDM 2 */ + { 0x0500, 0x0000 }, /* R1280 - Audio IF 1_1 */ + { 0x0501, 0x0008 }, /* R1281 - Audio IF 1_2 */ + { 0x0502, 0x0000 }, /* R1282 - Audio IF 1_3 */ + { 0x0503, 0x0000 }, /* R1283 - Audio IF 1_4 */ + { 0x0504, 0x0000 }, /* R1284 - Audio IF 1_5 */ + { 0x0505, 0x0001 }, /* R1285 - Audio IF 1_6 */ + { 0x0506, 0x0001 }, /* R1286 - Audio IF 1_7 */ + { 0x0507, 0x0000 }, /* R1287 - Audio IF 1_8 */ + { 0x0508, 0x0000 }, /* R1288 - Audio IF 1_9 */ + { 0x0509, 0x0000 }, /* R1289 - Audio IF 1_10 */ + { 0x050A, 0x0000 }, /* R1290 - Audio IF 1_11 */ + { 0x050B, 0x0000 }, /* R1291 - Audio IF 1_12 */ + { 0x050C, 0x0000 }, /* R1292 - Audio IF 1_13 */ + { 0x050D, 0x0000 }, /* R1293 - Audio IF 1_14 */ + { 0x050E, 0x0000 }, /* R1294 - Audio IF 1_15 */ + { 0x050F, 0x0000 }, /* R1295 - Audio IF 1_16 */ + { 0x0510, 0x0000 }, /* R1296 - Audio IF 1_17 */ + { 0x0511, 0x0000 }, /* R1297 - Audio IF 1_18 */ + { 0x0512, 0x0000 }, /* R1298 - Audio IF 1_19 */ + { 0x0513, 0x0000 }, /* R1299 - Audio IF 1_20 */ + { 0x0514, 0x0000 }, /* R1300 - Audio IF 1_21 */ + { 0x0515, 0x0001 }, /* R1301 - Audio IF 1_22 */ + { 0x0600, 0x0000 }, /* R1536 - OUT1LMIX Input 1 Source */ + { 0x0601, 0x0080 }, /* R1537 - OUT1LMIX Input 1 Volume */ + { 0x0602, 0x0000 }, /* R1538 - OUT1LMIX Input 2 Source */ + { 0x0603, 0x0080 }, /* R1539 - OUT1LMIX Input 2 Volume */ + { 0x0604, 0x0000 }, /* R1540 - OUT1LMIX Input 3 Source */ + { 0x0605, 0x0080 }, /* R1541 - OUT1LMIX Input 3 Volume */ + { 0x0606, 0x0000 }, /* R1542 - OUT1LMIX Input 4 Source */ + { 0x0607, 0x0080 }, /* R1543 - OUT1LMIX Input 4 Volume */ + { 0x0608, 0x0000 }, /* R1544 - OUT1RMIX Input 1 Source */ + { 0x0609, 0x0080 }, /* R1545 - OUT1RMIX Input 1 Volume */ + { 0x060A, 0x0000 }, /* R1546 - OUT1RMIX Input 2 Source */ + { 0x060B, 0x0080 }, /* R1547 - OUT1RMIX Input 2 Volume */ + { 0x060C, 0x0000 }, /* R1548 - OUT1RMIX Input 3 Source */ + { 0x060D, 0x0080 }, /* R1549 - OUT1RMIX Input 3 Volume */ + { 0x060E, 0x0000 }, /* R1550 - OUT1RMIX Input 4 Source */ + { 0x060F, 0x0080 }, /* R1551 - OUT1RMIX Input 4 Volume */ + { 0x0610, 0x0000 }, /* R1552 - OUT2LMIX Input 1 Source */ + { 0x0611, 0x0080 }, /* R1553 - OUT2LMIX Input 1 Volume */ + { 0x0612, 0x0000 }, /* R1554 - OUT2LMIX Input 2 Source */ + { 0x0613, 0x0080 }, /* R1555 - OUT2LMIX Input 2 Volume */ + { 0x0614, 0x0000 }, /* R1556 - OUT2LMIX Input 3 Source */ + { 0x0615, 0x0080 }, /* R1557 - OUT2LMIX Input 3 Volume */ + { 0x0616, 0x0000 }, /* R1558 - OUT2LMIX Input 4 Source */ + { 0x0617, 0x0080 }, /* R1559 - OUT2LMIX Input 4 Volume */ + { 0x0618, 0x0000 }, /* R1560 - OUT2RMIX Input 1 Source */ + { 0x0619, 0x0080 }, /* R1561 - OUT2RMIX Input 1 Volume */ + { 0x061A, 0x0000 }, /* R1562 - OUT2RMIX Input 2 Source */ + { 0x061B, 0x0080 }, /* R1563 - OUT2RMIX Input 2 Volume */ + { 0x061C, 0x0000 }, /* R1564 - OUT2RMIX Input 3 Source */ + { 0x061D, 0x0080 }, /* R1565 - OUT2RMIX Input 3 Volume */ + { 0x061E, 0x0000 }, /* R1566 - OUT2RMIX Input 4 Source */ + { 0x061F, 0x0080 }, /* R1567 - OUT2RMIX Input 4 Volume */ + { 0x0620, 0x0000 }, /* R1568 - AIF1TX1MIX Input 1 Source */ + { 0x0621, 0x0080 }, /* R1569 - AIF1TX1MIX Input 1 Volume */ + { 0x0622, 0x0000 }, /* R1570 - AIF1TX1MIX Input 2 Source */ + { 0x0623, 0x0080 }, /* R1571 - AIF1TX1MIX Input 2 Volume */ + { 0x0624, 0x0000 }, /* R1572 - AIF1TX1MIX Input 3 Source */ + { 0x0625, 0x0080 }, /* R1573 - AIF1TX1MIX Input 3 Volume */ + { 0x0626, 0x0000 }, /* R1574 - AIF1TX1MIX Input 4 Source */ + { 0x0627, 0x0080 }, /* R1575 - AIF1TX1MIX Input 4 Volume */ + { 0x0628, 0x0000 }, /* R1576 - AIF1TX2MIX Input 1 Source */ + { 0x0629, 0x0080 }, /* R1577 - AIF1TX2MIX Input 1 Volume */ + { 0x062A, 0x0000 }, /* R1578 - AIF1TX2MIX Input 2 Source */ + { 0x062B, 0x0080 }, /* R1579 - AIF1TX2MIX Input 2 Volume */ + { 0x062C, 0x0000 }, /* R1580 - AIF1TX2MIX Input 3 Source */ + { 0x062D, 0x0080 }, /* R1581 - AIF1TX2MIX Input 3 Volume */ + { 0x062E, 0x0000 }, /* R1582 - AIF1TX2MIX Input 4 Source */ + { 0x062F, 0x0080 }, /* R1583 - AIF1TX2MIX Input 4 Volume */ + { 0x0630, 0x0000 }, /* R1584 - AIF1TX3MIX Input 1 Source */ + { 0x0631, 0x0080 }, /* R1585 - AIF1TX3MIX Input 1 Volume */ + { 0x0632, 0x0000 }, /* R1586 - AIF1TX3MIX Input 2 Source */ + { 0x0633, 0x0080 }, /* R1587 - AIF1TX3MIX Input 2 Volume */ + { 0x0634, 0x0000 }, /* R1588 - AIF1TX3MIX Input 3 Source */ + { 0x0635, 0x0080 }, /* R1589 - AIF1TX3MIX Input 3 Volume */ + { 0x0636, 0x0000 }, /* R1590 - AIF1TX3MIX Input 4 Source */ + { 0x0637, 0x0080 }, /* R1591 - AIF1TX3MIX Input 4 Volume */ + { 0x0638, 0x0000 }, /* R1592 - AIF1TX4MIX Input 1 Source */ + { 0x0639, 0x0080 }, /* R1593 - AIF1TX4MIX Input 1 Volume */ + { 0x063A, 0x0000 }, /* R1594 - AIF1TX4MIX Input 2 Source */ + { 0x063B, 0x0080 }, /* R1595 - AIF1TX4MIX Input 2 Volume */ + { 0x063C, 0x0000 }, /* R1596 - AIF1TX4MIX Input 3 Source */ + { 0x063D, 0x0080 }, /* R1597 - AIF1TX4MIX Input 3 Volume */ + { 0x063E, 0x0000 }, /* R1598 - AIF1TX4MIX Input 4 Source */ + { 0x063F, 0x0080 }, /* R1599 - AIF1TX4MIX Input 4 Volume */ + { 0x0640, 0x0000 }, /* R1600 - AIF1TX5MIX Input 1 Source */ + { 0x0641, 0x0080 }, /* R1601 - AIF1TX5MIX Input 1 Volume */ + { 0x0642, 0x0000 }, /* R1602 - AIF1TX5MIX Input 2 Source */ + { 0x0643, 0x0080 }, /* R1603 - AIF1TX5MIX Input 2 Volume */ + { 0x0644, 0x0000 }, /* R1604 - AIF1TX5MIX Input 3 Source */ + { 0x0645, 0x0080 }, /* R1605 - AIF1TX5MIX Input 3 Volume */ + { 0x0646, 0x0000 }, /* R1606 - AIF1TX5MIX Input 4 Source */ + { 0x0647, 0x0080 }, /* R1607 - AIF1TX5MIX Input 4 Volume */ + { 0x0648, 0x0000 }, /* R1608 - AIF1TX6MIX Input 1 Source */ + { 0x0649, 0x0080 }, /* R1609 - AIF1TX6MIX Input 1 Volume */ + { 0x064A, 0x0000 }, /* R1610 - AIF1TX6MIX Input 2 Source */ + { 0x064B, 0x0080 }, /* R1611 - AIF1TX6MIX Input 2 Volume */ + { 0x064C, 0x0000 }, /* R1612 - AIF1TX6MIX Input 3 Source */ + { 0x064D, 0x0080 }, /* R1613 - AIF1TX6MIX Input 3 Volume */ + { 0x064E, 0x0000 }, /* R1614 - AIF1TX6MIX Input 4 Source */ + { 0x064F, 0x0080 }, /* R1615 - AIF1TX6MIX Input 4 Volume */ + { 0x0650, 0x0000 }, /* R1616 - EQLMIX Input 1 Source */ + { 0x0651, 0x0080 }, /* R1617 - EQLMIX Input 1 Volume */ + { 0x0652, 0x0000 }, /* R1618 - EQLMIX Input 2 Source */ + { 0x0653, 0x0080 }, /* R1619 - EQLMIX Input 2 Volume */ + { 0x0654, 0x0000 }, /* R1620 - EQLMIX Input 3 Source */ + { 0x0655, 0x0080 }, /* R1621 - EQLMIX Input 3 Volume */ + { 0x0656, 0x0000 }, /* R1622 - EQLMIX Input 4 Source */ + { 0x0657, 0x0080 }, /* R1623 - EQLMIX Input 4 Volume */ + { 0x0658, 0x0000 }, /* R1624 - EQRMIX Input 1 Source */ + { 0x0659, 0x0080 }, /* R1625 - EQRMIX Input 1 Volume */ + { 0x065A, 0x0000 }, /* R1626 - EQRMIX Input 2 Source */ + { 0x065B, 0x0080 }, /* R1627 - EQRMIX Input 2 Volume */ + { 0x065C, 0x0000 }, /* R1628 - EQRMIX Input 3 Source */ + { 0x065D, 0x0080 }, /* R1629 - EQRMIX Input 3 Volume */ + { 0x065E, 0x0000 }, /* R1630 - EQRMIX Input 4 Source */ + { 0x065F, 0x0080 }, /* R1631 - EQRMIX Input 4 Volume */ + { 0x0660, 0x0000 }, /* R1632 - LHPF1MIX Input 1 Source */ + { 0x0661, 0x0080 }, /* R1633 - LHPF1MIX Input 1 Volume */ + { 0x0662, 0x0000 }, /* R1634 - LHPF1MIX Input 2 Source */ + { 0x0663, 0x0080 }, /* R1635 - LHPF1MIX Input 2 Volume */ + { 0x0664, 0x0000 }, /* R1636 - LHPF1MIX Input 3 Source */ + { 0x0665, 0x0080 }, /* R1637 - LHPF1MIX Input 3 Volume */ + { 0x0666, 0x0000 }, /* R1638 - LHPF1MIX Input 4 Source */ + { 0x0667, 0x0080 }, /* R1639 - LHPF1MIX Input 4 Volume */ + { 0x0668, 0x0000 }, /* R1640 - LHPF2MIX Input 1 Source */ + { 0x0669, 0x0080 }, /* R1641 - LHPF2MIX Input 1 Volume */ + { 0x066A, 0x0000 }, /* R1642 - LHPF2MIX Input 2 Source */ + { 0x066B, 0x0080 }, /* R1643 - LHPF2MIX Input 2 Volume */ + { 0x066C, 0x0000 }, /* R1644 - LHPF2MIX Input 3 Source */ + { 0x066D, 0x0080 }, /* R1645 - LHPF2MIX Input 3 Volume */ + { 0x066E, 0x0000 }, /* R1646 - LHPF2MIX Input 4 Source */ + { 0x066F, 0x0080 }, /* R1647 - LHPF2MIX Input 4 Volume */ + { 0x0670, 0x0000 }, /* R1648 - DSP1LMIX Input 1 Source */ + { 0x0671, 0x0080 }, /* R1649 - DSP1LMIX Input 1 Volume */ + { 0x0672, 0x0000 }, /* R1650 - DSP1LMIX Input 2 Source */ + { 0x0673, 0x0080 }, /* R1651 - DSP1LMIX Input 2 Volume */ + { 0x0674, 0x0000 }, /* R1652 - DSP1LMIX Input 3 Source */ + { 0x0675, 0x0080 }, /* R1653 - DSP1LMIX Input 3 Volume */ + { 0x0676, 0x0000 }, /* R1654 - DSP1LMIX Input 4 Source */ + { 0x0677, 0x0080 }, /* R1655 - DSP1LMIX Input 4 Volume */ + { 0x0678, 0x0000 }, /* R1656 - DSP1RMIX Input 1 Source */ + { 0x0679, 0x0080 }, /* R1657 - DSP1RMIX Input 1 Volume */ + { 0x067A, 0x0000 }, /* R1658 - DSP1RMIX Input 2 Source */ + { 0x067B, 0x0080 }, /* R1659 - DSP1RMIX Input 2 Volume */ + { 0x067C, 0x0000 }, /* R1660 - DSP1RMIX Input 3 Source */ + { 0x067D, 0x0080 }, /* R1661 - DSP1RMIX Input 3 Volume */ + { 0x067E, 0x0000 }, /* R1662 - DSP1RMIX Input 4 Source */ + { 0x067F, 0x0080 }, /* R1663 - DSP1RMIX Input 4 Volume */ + { 0x0680, 0x0000 }, /* R1664 - DSP1AUX1MIX Input 1 Source */ + { 0x0681, 0x0000 }, /* R1665 - DSP1AUX2MIX Input 1 Source */ + { 0x0682, 0x0000 }, /* R1666 - DSP1AUX3MIX Input 1 Source */ + { 0x0683, 0x0000 }, /* R1667 - DSP1AUX4MIX Input 1 Source */ + { 0x0684, 0x0000 }, /* R1668 - DSP1AUX5MIX Input 1 Source */ + { 0x0685, 0x0000 }, /* R1669 - DSP1AUX6MIX Input 1 Source */ + { 0x0686, 0x0000 }, /* R1670 - DSP2LMIX Input 1 Source */ + { 0x0687, 0x0080 }, /* R1671 - DSP2LMIX Input 1 Volume */ + { 0x0688, 0x0000 }, /* R1672 - DSP2LMIX Input 2 Source */ + { 0x0689, 0x0080 }, /* R1673 - DSP2LMIX Input 2 Volume */ + { 0x068A, 0x0000 }, /* R1674 - DSP2LMIX Input 3 Source */ + { 0x068B, 0x0080 }, /* R1675 - DSP2LMIX Input 3 Volume */ + { 0x068C, 0x0000 }, /* R1676 - DSP2LMIX Input 4 Source */ + { 0x068D, 0x0080 }, /* R1677 - DSP2LMIX Input 4 Volume */ + { 0x068E, 0x0000 }, /* R1678 - DSP2RMIX Input 1 Source */ + { 0x068F, 0x0080 }, /* R1679 - DSP2RMIX Input 1 Volume */ + { 0x0690, 0x0000 }, /* R1680 - DSP2RMIX Input 2 Source */ + { 0x0691, 0x0080 }, /* R1681 - DSP2RMIX Input 2 Volume */ + { 0x0692, 0x0000 }, /* R1682 - DSP2RMIX Input 3 Source */ + { 0x0693, 0x0080 }, /* R1683 - DSP2RMIX Input 3 Volume */ + { 0x0694, 0x0000 }, /* R1684 - DSP2RMIX Input 4 Source */ + { 0x0695, 0x0080 }, /* R1685 - DSP2RMIX Input 4 Volume */ + { 0x0696, 0x0000 }, /* R1686 - DSP2AUX1MIX Input 1 Source */ + { 0x0697, 0x0000 }, /* R1687 - DSP2AUX2MIX Input 1 Source */ + { 0x0698, 0x0000 }, /* R1688 - DSP2AUX3MIX Input 1 Source */ + { 0x0699, 0x0000 }, /* R1689 - DSP2AUX4MIX Input 1 Source */ + { 0x069A, 0x0000 }, /* R1690 - DSP2AUX5MIX Input 1 Source */ + { 0x069B, 0x0000 }, /* R1691 - DSP2AUX6MIX Input 1 Source */ + { 0x0700, 0xA101 }, /* R1792 - GPIO CTRL 1 */ + { 0x0701, 0xA101 }, /* R1793 - GPIO CTRL 2 */ + { 0x0702, 0xA101 }, /* R1794 - GPIO CTRL 3 */ + { 0x0703, 0xA101 }, /* R1795 - GPIO CTRL 4 */ + { 0x0709, 0x0000 }, /* R1801 - Misc Pad Ctrl 1 */ + { 0x0801, 0x00FF }, /* R2049 - Interrupt Status 1 Mask */ + { 0x0804, 0xFFFF }, /* R2052 - Interrupt Status 2 Mask */ + { 0x0808, 0x0000 }, /* R2056 - Interrupt Control */ + { 0x0900, 0x0000 }, /* R2304 - EQL_1 */ + { 0x0901, 0x0000 }, /* R2305 - EQL_2 */ + { 0x0902, 0x0000 }, /* R2306 - EQL_3 */ + { 0x0903, 0x0000 }, /* R2307 - EQL_4 */ + { 0x0904, 0x0000 }, /* R2308 - EQL_5 */ + { 0x0905, 0x0000 }, /* R2309 - EQL_6 */ + { 0x0906, 0x0000 }, /* R2310 - EQL_7 */ + { 0x0907, 0x0000 }, /* R2311 - EQL_8 */ + { 0x0908, 0x0000 }, /* R2312 - EQL_9 */ + { 0x0909, 0x0000 }, /* R2313 - EQL_10 */ + { 0x090A, 0x0000 }, /* R2314 - EQL_11 */ + { 0x090B, 0x0000 }, /* R2315 - EQL_12 */ + { 0x090C, 0x0000 }, /* R2316 - EQL_13 */ + { 0x090D, 0x0000 }, /* R2317 - EQL_14 */ + { 0x090E, 0x0000 }, /* R2318 - EQL_15 */ + { 0x090F, 0x0000 }, /* R2319 - EQL_16 */ + { 0x0910, 0x0000 }, /* R2320 - EQL_17 */ + { 0x0911, 0x0000 }, /* R2321 - EQL_18 */ + { 0x0912, 0x0000 }, /* R2322 - EQL_19 */ + { 0x0913, 0x0000 }, /* R2323 - EQL_20 */ + { 0x0916, 0x0000 }, /* R2326 - EQR_1 */ + { 0x0917, 0x0000 }, /* R2327 - EQR_2 */ + { 0x0918, 0x0000 }, /* R2328 - EQR_3 */ + { 0x0919, 0x0000 }, /* R2329 - EQR_4 */ + { 0x091A, 0x0000 }, /* R2330 - EQR_5 */ + { 0x091B, 0x0000 }, /* R2331 - EQR_6 */ + { 0x091C, 0x0000 }, /* R2332 - EQR_7 */ + { 0x091D, 0x0000 }, /* R2333 - EQR_8 */ + { 0x091E, 0x0000 }, /* R2334 - EQR_9 */ + { 0x091F, 0x0000 }, /* R2335 - EQR_10 */ + { 0x0920, 0x0000 }, /* R2336 - EQR_11 */ + { 0x0921, 0x0000 }, /* R2337 - EQR_12 */ + { 0x0922, 0x0000 }, /* R2338 - EQR_13 */ + { 0x0923, 0x0000 }, /* R2339 - EQR_14 */ + { 0x0924, 0x0000 }, /* R2340 - EQR_15 */ + { 0x0925, 0x0000 }, /* R2341 - EQR_16 */ + { 0x0926, 0x0000 }, /* R2342 - EQR_17 */ + { 0x0927, 0x0000 }, /* R2343 - EQR_18 */ + { 0x0928, 0x0000 }, /* R2344 - EQR_19 */ + { 0x0929, 0x0000 }, /* R2345 - EQR_20 */ + { 0x093E, 0x0000 }, /* R2366 - HPLPF1_1 */ + { 0x093F, 0x0000 }, /* R2367 - HPLPF1_2 */ + { 0x0942, 0x0000 }, /* R2370 - HPLPF2_1 */ + { 0x0943, 0x0000 }, /* R2371 - HPLPF2_2 */ + { 0x0A00, 0x0000 }, /* R2560 - DSP1 Control 1 */ + { 0x0A02, 0x0000 }, /* R2562 - DSP1 Control 2 */ + { 0x0A03, 0x0000 }, /* R2563 - DSP1 Control 3 */ + { 0x0A04, 0x0000 }, /* R2564 - DSP1 Control 4 */ + { 0x0A06, 0x0000 }, /* R2566 - DSP1 Control 5 */ + { 0x0A07, 0x0000 }, /* R2567 - DSP1 Control 6 */ + { 0x0A08, 0x0000 }, /* R2568 - DSP1 Control 7 */ + { 0x0A09, 0x0000 }, /* R2569 - DSP1 Control 8 */ + { 0x0A0A, 0x0000 }, /* R2570 - DSP1 Control 9 */ + { 0x0A0B, 0x0000 }, /* R2571 - DSP1 Control 10 */ + { 0x0A0C, 0x0000 }, /* R2572 - DSP1 Control 11 */ + { 0x0A0D, 0x0000 }, /* R2573 - DSP1 Control 12 */ + { 0x0A0F, 0x0000 }, /* R2575 - DSP1 Control 13 */ + { 0x0A10, 0x0000 }, /* R2576 - DSP1 Control 14 */ + { 0x0A11, 0x0000 }, /* R2577 - DSP1 Control 15 */ + { 0x0A12, 0x0000 }, /* R2578 - DSP1 Control 16 */ + { 0x0A13, 0x0000 }, /* R2579 - DSP1 Control 17 */ + { 0x0A14, 0x0000 }, /* R2580 - DSP1 Control 18 */ + { 0x0A16, 0x0000 }, /* R2582 - DSP1 Control 19 */ + { 0x0A17, 0x0000 }, /* R2583 - DSP1 Control 20 */ + { 0x0A18, 0x0000 }, /* R2584 - DSP1 Control 21 */ + { 0x0A1A, 0x1800 }, /* R2586 - DSP1 Control 22 */ + { 0x0A1B, 0x1000 }, /* R2587 - DSP1 Control 23 */ + { 0x0A1C, 0x0400 }, /* R2588 - DSP1 Control 24 */ + { 0x0A1E, 0x0000 }, /* R2590 - DSP1 Control 25 */ + { 0x0A20, 0x0000 }, /* R2592 - DSP1 Control 26 */ + { 0x0A21, 0x0000 }, /* R2593 - DSP1 Control 27 */ + { 0x0A22, 0x0000 }, /* R2594 - DSP1 Control 28 */ + { 0x0A23, 0x0000 }, /* R2595 - DSP1 Control 29 */ + { 0x0A24, 0x0000 }, /* R2596 - DSP1 Control 30 */ + { 0x0A26, 0x0000 }, /* R2598 - DSP1 Control 31 */ + { 0x0B00, 0x0000 }, /* R2816 - DSP2 Control 1 */ + { 0x0B02, 0x0000 }, /* R2818 - DSP2 Control 2 */ + { 0x0B03, 0x0000 }, /* R2819 - DSP2 Control 3 */ + { 0x0B04, 0x0000 }, /* R2820 - DSP2 Control 4 */ + { 0x0B06, 0x0000 }, /* R2822 - DSP2 Control 5 */ + { 0x0B07, 0x0000 }, /* R2823 - DSP2 Control 6 */ + { 0x0B08, 0x0000 }, /* R2824 - DSP2 Control 7 */ + { 0x0B09, 0x0000 }, /* R2825 - DSP2 Control 8 */ + { 0x0B0A, 0x0000 }, /* R2826 - DSP2 Control 9 */ + { 0x0B0B, 0x0000 }, /* R2827 - DSP2 Control 10 */ + { 0x0B0C, 0x0000 }, /* R2828 - DSP2 Control 11 */ + { 0x0B0D, 0x0000 }, /* R2829 - DSP2 Control 12 */ + { 0x0B0F, 0x0000 }, /* R2831 - DSP2 Control 13 */ + { 0x0B10, 0x0000 }, /* R2832 - DSP2 Control 14 */ + { 0x0B11, 0x0000 }, /* R2833 - DSP2 Control 15 */ + { 0x0B12, 0x0000 }, /* R2834 - DSP2 Control 16 */ + { 0x0B13, 0x0000 }, /* R2835 - DSP2 Control 17 */ + { 0x0B14, 0x0000 }, /* R2836 - DSP2 Control 18 */ + { 0x0B16, 0x0000 }, /* R2838 - DSP2 Control 19 */ + { 0x0B17, 0x0000 }, /* R2839 - DSP2 Control 20 */ + { 0x0B18, 0x0000 }, /* R2840 - DSP2 Control 21 */ + { 0x0B1A, 0x0800 }, /* R2842 - DSP2 Control 22 */ + { 0x0B1B, 0x1000 }, /* R2843 - DSP2 Control 23 */ + { 0x0B1C, 0x0400 }, /* R2844 - DSP2 Control 24 */ + { 0x0B1E, 0x0000 }, /* R2846 - DSP2 Control 25 */ + { 0x0B20, 0x0000 }, /* R2848 - DSP2 Control 26 */ + { 0x0B21, 0x0000 }, /* R2849 - DSP2 Control 27 */ + { 0x0B22, 0x0000 }, /* R2850 - DSP2 Control 28 */ + { 0x0B23, 0x0000 }, /* R2851 - DSP2 Control 29 */ + { 0x0B24, 0x0000 }, /* R2852 - DSP2 Control 30 */ + { 0x0B26, 0x0000 }, /* R2854 - DSP2 Control 31 */ +}; + +static bool wm2200_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM2200_SOFTWARE_RESET: + case WM2200_DEVICE_REVISION: + case WM2200_ADPS1_IRQ0: + case WM2200_ADPS1_IRQ1: + case WM2200_INTERRUPT_STATUS_1: + case WM2200_INTERRUPT_STATUS_2: + case WM2200_INTERRUPT_RAW_STATUS_2: + return true; + default: + return false; + } +} + +static bool wm2200_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM2200_SOFTWARE_RESET: + case WM2200_DEVICE_REVISION: + case WM2200_TONE_GENERATOR_1: + case WM2200_CLOCKING_3: + case WM2200_CLOCKING_4: + case WM2200_FLL_CONTROL_1: + case WM2200_FLL_CONTROL_2: + case WM2200_FLL_CONTROL_3: + case WM2200_FLL_CONTROL_4: + case WM2200_FLL_CONTROL_6: + case WM2200_FLL_CONTROL_7: + case WM2200_FLL_EFS_1: + case WM2200_FLL_EFS_2: + case WM2200_MIC_CHARGE_PUMP_1: + case WM2200_MIC_CHARGE_PUMP_2: + case WM2200_DM_CHARGE_PUMP_1: + case WM2200_MIC_BIAS_CTRL_1: + case WM2200_MIC_BIAS_CTRL_2: + case WM2200_EAR_PIECE_CTRL_1: + case WM2200_EAR_PIECE_CTRL_2: + case WM2200_INPUT_ENABLES: + case WM2200_IN1L_CONTROL: + case WM2200_IN1R_CONTROL: + case WM2200_IN2L_CONTROL: + case WM2200_IN2R_CONTROL: + case WM2200_IN3L_CONTROL: + case WM2200_IN3R_CONTROL: + case WM2200_RXANC_SRC: + case WM2200_INPUT_VOLUME_RAMP: + case WM2200_ADC_DIGITAL_VOLUME_1L: + case WM2200_ADC_DIGITAL_VOLUME_1R: + case WM2200_ADC_DIGITAL_VOLUME_2L: + case WM2200_ADC_DIGITAL_VOLUME_2R: + case WM2200_ADC_DIGITAL_VOLUME_3L: + case WM2200_ADC_DIGITAL_VOLUME_3R: + case WM2200_OUTPUT_ENABLES: + case WM2200_DAC_VOLUME_LIMIT_1L: + case WM2200_DAC_VOLUME_LIMIT_1R: + case WM2200_DAC_VOLUME_LIMIT_2L: + case WM2200_DAC_VOLUME_LIMIT_2R: + case WM2200_DAC_AEC_CONTROL_1: + case WM2200_OUTPUT_VOLUME_RAMP: + case WM2200_DAC_DIGITAL_VOLUME_1L: + case WM2200_DAC_DIGITAL_VOLUME_1R: + case WM2200_DAC_DIGITAL_VOLUME_2L: + case WM2200_DAC_DIGITAL_VOLUME_2R: + case WM2200_PDM_1: + case WM2200_PDM_2: + case WM2200_AUDIO_IF_1_1: + case WM2200_AUDIO_IF_1_2: + case WM2200_AUDIO_IF_1_3: + case WM2200_AUDIO_IF_1_4: + case WM2200_AUDIO_IF_1_5: + case WM2200_AUDIO_IF_1_6: + case WM2200_AUDIO_IF_1_7: + case WM2200_AUDIO_IF_1_8: + case WM2200_AUDIO_IF_1_9: + case WM2200_AUDIO_IF_1_10: + case WM2200_AUDIO_IF_1_11: + case WM2200_AUDIO_IF_1_12: + case WM2200_AUDIO_IF_1_13: + case WM2200_AUDIO_IF_1_14: + case WM2200_AUDIO_IF_1_15: + case WM2200_AUDIO_IF_1_16: + case WM2200_AUDIO_IF_1_17: + case WM2200_AUDIO_IF_1_18: + case WM2200_AUDIO_IF_1_19: + case WM2200_AUDIO_IF_1_20: + case WM2200_AUDIO_IF_1_21: + case WM2200_AUDIO_IF_1_22: + case WM2200_OUT1LMIX_INPUT_1_SOURCE: + case WM2200_OUT1LMIX_INPUT_1_VOLUME: + case WM2200_OUT1LMIX_INPUT_2_SOURCE: + case WM2200_OUT1LMIX_INPUT_2_VOLUME: + case WM2200_OUT1LMIX_INPUT_3_SOURCE: + case WM2200_OUT1LMIX_INPUT_3_VOLUME: + case WM2200_OUT1LMIX_INPUT_4_SOURCE: + case WM2200_OUT1LMIX_INPUT_4_VOLUME: + case WM2200_OUT1RMIX_INPUT_1_SOURCE: + case WM2200_OUT1RMIX_INPUT_1_VOLUME: + case WM2200_OUT1RMIX_INPUT_2_SOURCE: + case WM2200_OUT1RMIX_INPUT_2_VOLUME: + case WM2200_OUT1RMIX_INPUT_3_SOURCE: + case WM2200_OUT1RMIX_INPUT_3_VOLUME: + case WM2200_OUT1RMIX_INPUT_4_SOURCE: + case WM2200_OUT1RMIX_INPUT_4_VOLUME: + case WM2200_OUT2LMIX_INPUT_1_SOURCE: + case WM2200_OUT2LMIX_INPUT_1_VOLUME: + case WM2200_OUT2LMIX_INPUT_2_SOURCE: + case WM2200_OUT2LMIX_INPUT_2_VOLUME: + case WM2200_OUT2LMIX_INPUT_3_SOURCE: + case WM2200_OUT2LMIX_INPUT_3_VOLUME: + case WM2200_OUT2LMIX_INPUT_4_SOURCE: + case WM2200_OUT2LMIX_INPUT_4_VOLUME: + case WM2200_OUT2RMIX_INPUT_1_SOURCE: + case WM2200_OUT2RMIX_INPUT_1_VOLUME: + case WM2200_OUT2RMIX_INPUT_2_SOURCE: + case WM2200_OUT2RMIX_INPUT_2_VOLUME: + case WM2200_OUT2RMIX_INPUT_3_SOURCE: + case WM2200_OUT2RMIX_INPUT_3_VOLUME: + case WM2200_OUT2RMIX_INPUT_4_SOURCE: + case WM2200_OUT2RMIX_INPUT_4_VOLUME: + case WM2200_AIF1TX1MIX_INPUT_1_SOURCE: + case WM2200_AIF1TX1MIX_INPUT_1_VOLUME: + case WM2200_AIF1TX1MIX_INPUT_2_SOURCE: + case WM2200_AIF1TX1MIX_INPUT_2_VOLUME: + case WM2200_AIF1TX1MIX_INPUT_3_SOURCE: + case WM2200_AIF1TX1MIX_INPUT_3_VOLUME: + case WM2200_AIF1TX1MIX_INPUT_4_SOURCE: + case WM2200_AIF1TX1MIX_INPUT_4_VOLUME: + case WM2200_AIF1TX2MIX_INPUT_1_SOURCE: + case WM2200_AIF1TX2MIX_INPUT_1_VOLUME: + case WM2200_AIF1TX2MIX_INPUT_2_SOURCE: + case WM2200_AIF1TX2MIX_INPUT_2_VOLUME: + case WM2200_AIF1TX2MIX_INPUT_3_SOURCE: + case WM2200_AIF1TX2MIX_INPUT_3_VOLUME: + case WM2200_AIF1TX2MIX_INPUT_4_SOURCE: + case WM2200_AIF1TX2MIX_INPUT_4_VOLUME: + case WM2200_AIF1TX3MIX_INPUT_1_SOURCE: + case WM2200_AIF1TX3MIX_INPUT_1_VOLUME: + case WM2200_AIF1TX3MIX_INPUT_2_SOURCE: + case WM2200_AIF1TX3MIX_INPUT_2_VOLUME: + case WM2200_AIF1TX3MIX_INPUT_3_SOURCE: + case WM2200_AIF1TX3MIX_INPUT_3_VOLUME: + case WM2200_AIF1TX3MIX_INPUT_4_SOURCE: + case WM2200_AIF1TX3MIX_INPUT_4_VOLUME: + case WM2200_AIF1TX4MIX_INPUT_1_SOURCE: + case WM2200_AIF1TX4MIX_INPUT_1_VOLUME: + case WM2200_AIF1TX4MIX_INPUT_2_SOURCE: + case WM2200_AIF1TX4MIX_INPUT_2_VOLUME: + case WM2200_AIF1TX4MIX_INPUT_3_SOURCE: + case WM2200_AIF1TX4MIX_INPUT_3_VOLUME: + case WM2200_AIF1TX4MIX_INPUT_4_SOURCE: + case WM2200_AIF1TX4MIX_INPUT_4_VOLUME: + case WM2200_AIF1TX5MIX_INPUT_1_SOURCE: + case WM2200_AIF1TX5MIX_INPUT_1_VOLUME: + case WM2200_AIF1TX5MIX_INPUT_2_SOURCE: + case WM2200_AIF1TX5MIX_INPUT_2_VOLUME: + case WM2200_AIF1TX5MIX_INPUT_3_SOURCE: + case WM2200_AIF1TX5MIX_INPUT_3_VOLUME: + case WM2200_AIF1TX5MIX_INPUT_4_SOURCE: + case WM2200_AIF1TX5MIX_INPUT_4_VOLUME: + case WM2200_AIF1TX6MIX_INPUT_1_SOURCE: + case WM2200_AIF1TX6MIX_INPUT_1_VOLUME: + case WM2200_AIF1TX6MIX_INPUT_2_SOURCE: + case WM2200_AIF1TX6MIX_INPUT_2_VOLUME: + case WM2200_AIF1TX6MIX_INPUT_3_SOURCE: + case WM2200_AIF1TX6MIX_INPUT_3_VOLUME: + case WM2200_AIF1TX6MIX_INPUT_4_SOURCE: + case WM2200_AIF1TX6MIX_INPUT_4_VOLUME: + case WM2200_EQLMIX_INPUT_1_SOURCE: + case WM2200_EQLMIX_INPUT_1_VOLUME: + case WM2200_EQLMIX_INPUT_2_SOURCE: + case WM2200_EQLMIX_INPUT_2_VOLUME: + case WM2200_EQLMIX_INPUT_3_SOURCE: + case WM2200_EQLMIX_INPUT_3_VOLUME: + case WM2200_EQLMIX_INPUT_4_SOURCE: + case WM2200_EQLMIX_INPUT_4_VOLUME: + case WM2200_EQRMIX_INPUT_1_SOURCE: + case WM2200_EQRMIX_INPUT_1_VOLUME: + case WM2200_EQRMIX_INPUT_2_SOURCE: + case WM2200_EQRMIX_INPUT_2_VOLUME: + case WM2200_EQRMIX_INPUT_3_SOURCE: + case WM2200_EQRMIX_INPUT_3_VOLUME: + case WM2200_EQRMIX_INPUT_4_SOURCE: + case WM2200_EQRMIX_INPUT_4_VOLUME: + case WM2200_LHPF1MIX_INPUT_1_SOURCE: + case WM2200_LHPF1MIX_INPUT_1_VOLUME: + case WM2200_LHPF1MIX_INPUT_2_SOURCE: + case WM2200_LHPF1MIX_INPUT_2_VOLUME: + case WM2200_LHPF1MIX_INPUT_3_SOURCE: + case WM2200_LHPF1MIX_INPUT_3_VOLUME: + case WM2200_LHPF1MIX_INPUT_4_SOURCE: + case WM2200_LHPF1MIX_INPUT_4_VOLUME: + case WM2200_LHPF2MIX_INPUT_1_SOURCE: + case WM2200_LHPF2MIX_INPUT_1_VOLUME: + case WM2200_LHPF2MIX_INPUT_2_SOURCE: + case WM2200_LHPF2MIX_INPUT_2_VOLUME: + case WM2200_LHPF2MIX_INPUT_3_SOURCE: + case WM2200_LHPF2MIX_INPUT_3_VOLUME: + case WM2200_LHPF2MIX_INPUT_4_SOURCE: + case WM2200_LHPF2MIX_INPUT_4_VOLUME: + case WM2200_DSP1LMIX_INPUT_1_SOURCE: + case WM2200_DSP1LMIX_INPUT_1_VOLUME: + case WM2200_DSP1LMIX_INPUT_2_SOURCE: + case WM2200_DSP1LMIX_INPUT_2_VOLUME: + case WM2200_DSP1LMIX_INPUT_3_SOURCE: + case WM2200_DSP1LMIX_INPUT_3_VOLUME: + case WM2200_DSP1LMIX_INPUT_4_SOURCE: + case WM2200_DSP1LMIX_INPUT_4_VOLUME: + case WM2200_DSP1RMIX_INPUT_1_SOURCE: + case WM2200_DSP1RMIX_INPUT_1_VOLUME: + case WM2200_DSP1RMIX_INPUT_2_SOURCE: + case WM2200_DSP1RMIX_INPUT_2_VOLUME: + case WM2200_DSP1RMIX_INPUT_3_SOURCE: + case WM2200_DSP1RMIX_INPUT_3_VOLUME: + case WM2200_DSP1RMIX_INPUT_4_SOURCE: + case WM2200_DSP1RMIX_INPUT_4_VOLUME: + case WM2200_DSP1AUX1MIX_INPUT_1_SOURCE: + case WM2200_DSP1AUX2MIX_INPUT_1_SOURCE: + case WM2200_DSP1AUX3MIX_INPUT_1_SOURCE: + case WM2200_DSP1AUX4MIX_INPUT_1_SOURCE: + case WM2200_DSP1AUX5MIX_INPUT_1_SOURCE: + case WM2200_DSP1AUX6MIX_INPUT_1_SOURCE: + case WM2200_DSP2LMIX_INPUT_1_SOURCE: + case WM2200_DSP2LMIX_INPUT_1_VOLUME: + case WM2200_DSP2LMIX_INPUT_2_SOURCE: + case WM2200_DSP2LMIX_INPUT_2_VOLUME: + case WM2200_DSP2LMIX_INPUT_3_SOURCE: + case WM2200_DSP2LMIX_INPUT_3_VOLUME: + case WM2200_DSP2LMIX_INPUT_4_SOURCE: + case WM2200_DSP2LMIX_INPUT_4_VOLUME: + case WM2200_DSP2RMIX_INPUT_1_SOURCE: + case WM2200_DSP2RMIX_INPUT_1_VOLUME: + case WM2200_DSP2RMIX_INPUT_2_SOURCE: + case WM2200_DSP2RMIX_INPUT_2_VOLUME: + case WM2200_DSP2RMIX_INPUT_3_SOURCE: + case WM2200_DSP2RMIX_INPUT_3_VOLUME: + case WM2200_DSP2RMIX_INPUT_4_SOURCE: + case WM2200_DSP2RMIX_INPUT_4_VOLUME: + case WM2200_DSP2AUX1MIX_INPUT_1_SOURCE: + case WM2200_DSP2AUX2MIX_INPUT_1_SOURCE: + case WM2200_DSP2AUX3MIX_INPUT_1_SOURCE: + case WM2200_DSP2AUX4MIX_INPUT_1_SOURCE: + case WM2200_DSP2AUX5MIX_INPUT_1_SOURCE: + case WM2200_DSP2AUX6MIX_INPUT_1_SOURCE: + case WM2200_GPIO_CTRL_1: + case WM2200_GPIO_CTRL_2: + case WM2200_GPIO_CTRL_3: + case WM2200_GPIO_CTRL_4: + case WM2200_ADPS1_IRQ0: + case WM2200_ADPS1_IRQ1: + case WM2200_MISC_PAD_CTRL_1: + case WM2200_INTERRUPT_STATUS_1: + case WM2200_INTERRUPT_STATUS_1_MASK: + case WM2200_INTERRUPT_STATUS_2: + case WM2200_INTERRUPT_RAW_STATUS_2: + case WM2200_INTERRUPT_STATUS_2_MASK: + case WM2200_INTERRUPT_CONTROL: + case WM2200_EQL_1: + case WM2200_EQL_2: + case WM2200_EQL_3: + case WM2200_EQL_4: + case WM2200_EQL_5: + case WM2200_EQL_6: + case WM2200_EQL_7: + case WM2200_EQL_8: + case WM2200_EQL_9: + case WM2200_EQL_10: + case WM2200_EQL_11: + case WM2200_EQL_12: + case WM2200_EQL_13: + case WM2200_EQL_14: + case WM2200_EQL_15: + case WM2200_EQL_16: + case WM2200_EQL_17: + case WM2200_EQL_18: + case WM2200_EQL_19: + case WM2200_EQL_20: + case WM2200_EQR_1: + case WM2200_EQR_2: + case WM2200_EQR_3: + case WM2200_EQR_4: + case WM2200_EQR_5: + case WM2200_EQR_6: + case WM2200_EQR_7: + case WM2200_EQR_8: + case WM2200_EQR_9: + case WM2200_EQR_10: + case WM2200_EQR_11: + case WM2200_EQR_12: + case WM2200_EQR_13: + case WM2200_EQR_14: + case WM2200_EQR_15: + case WM2200_EQR_16: + case WM2200_EQR_17: + case WM2200_EQR_18: + case WM2200_EQR_19: + case WM2200_EQR_20: + case WM2200_HPLPF1_1: + case WM2200_HPLPF1_2: + case WM2200_HPLPF2_1: + case WM2200_HPLPF2_2: + case WM2200_DSP1_CONTROL_1: + case WM2200_DSP1_CONTROL_2: + case WM2200_DSP1_CONTROL_3: + case WM2200_DSP1_CONTROL_4: + case WM2200_DSP1_CONTROL_5: + case WM2200_DSP1_CONTROL_6: + case WM2200_DSP1_CONTROL_7: + case WM2200_DSP1_CONTROL_8: + case WM2200_DSP1_CONTROL_9: + case WM2200_DSP1_CONTROL_10: + case WM2200_DSP1_CONTROL_11: + case WM2200_DSP1_CONTROL_12: + case WM2200_DSP1_CONTROL_13: + case WM2200_DSP1_CONTROL_14: + case WM2200_DSP1_CONTROL_15: + case WM2200_DSP1_CONTROL_16: + case WM2200_DSP1_CONTROL_17: + case WM2200_DSP1_CONTROL_18: + case WM2200_DSP1_CONTROL_19: + case WM2200_DSP1_CONTROL_20: + case WM2200_DSP1_CONTROL_21: + case WM2200_DSP1_CONTROL_22: + case WM2200_DSP1_CONTROL_23: + case WM2200_DSP1_CONTROL_24: + case WM2200_DSP1_CONTROL_25: + case WM2200_DSP1_CONTROL_26: + case WM2200_DSP1_CONTROL_27: + case WM2200_DSP1_CONTROL_28: + case WM2200_DSP1_CONTROL_29: + case WM2200_DSP1_CONTROL_30: + case WM2200_DSP1_CONTROL_31: + case WM2200_DSP2_CONTROL_1: + case WM2200_DSP2_CONTROL_2: + case WM2200_DSP2_CONTROL_3: + case WM2200_DSP2_CONTROL_4: + case WM2200_DSP2_CONTROL_5: + case WM2200_DSP2_CONTROL_6: + case WM2200_DSP2_CONTROL_7: + case WM2200_DSP2_CONTROL_8: + case WM2200_DSP2_CONTROL_9: + case WM2200_DSP2_CONTROL_10: + case WM2200_DSP2_CONTROL_11: + case WM2200_DSP2_CONTROL_12: + case WM2200_DSP2_CONTROL_13: + case WM2200_DSP2_CONTROL_14: + case WM2200_DSP2_CONTROL_15: + case WM2200_DSP2_CONTROL_16: + case WM2200_DSP2_CONTROL_17: + case WM2200_DSP2_CONTROL_18: + case WM2200_DSP2_CONTROL_19: + case WM2200_DSP2_CONTROL_20: + case WM2200_DSP2_CONTROL_21: + case WM2200_DSP2_CONTROL_22: + case WM2200_DSP2_CONTROL_23: + case WM2200_DSP2_CONTROL_24: + case WM2200_DSP2_CONTROL_25: + case WM2200_DSP2_CONTROL_26: + case WM2200_DSP2_CONTROL_27: + case WM2200_DSP2_CONTROL_28: + case WM2200_DSP2_CONTROL_29: + case WM2200_DSP2_CONTROL_30: + case WM2200_DSP2_CONTROL_31: + return true; + default: + return false; + } +} + +static const struct reg_default wm2200_reva_patch[] = { + { 0x07, 0x0003 }, + { 0x102, 0x0200 }, + { 0x203, 0x0084 }, + { 0x201, 0x83FF }, + { 0x20C, 0x0062 }, + { 0x20D, 0x0062 }, + { 0x207, 0x2002 }, + { 0x208, 0x20C0 }, + { 0x21D, 0x01C0 }, + { 0x50A, 0x0001 }, + { 0x50B, 0x0002 }, + { 0x50C, 0x0003 }, + { 0x50D, 0x0004 }, + { 0x50E, 0x0005 }, + { 0x510, 0x0001 }, + { 0x511, 0x0002 }, + { 0x512, 0x0003 }, + { 0x513, 0x0004 }, + { 0x514, 0x0005 }, + { 0x515, 0x0000 }, + { 0x201, 0x8084 }, + { 0x202, 0xBBDE }, + { 0x203, 0x00EC }, + { 0x500, 0x8000 }, + { 0x507, 0x1820 }, + { 0x508, 0x1820 }, + { 0x505, 0x0300 }, + { 0x506, 0x0300 }, + { 0x302, 0x2280 }, + { 0x303, 0x0080 }, + { 0x304, 0x2280 }, + { 0x305, 0x0080 }, + { 0x306, 0x2280 }, + { 0x307, 0x0080 }, + { 0x401, 0x0080 }, + { 0x402, 0x0080 }, + { 0x417, 0x3069 }, + { 0x900, 0x6318 }, + { 0x901, 0x6300 }, + { 0x902, 0x0FC8 }, + { 0x903, 0x03FE }, + { 0x904, 0x00E0 }, + { 0x905, 0x1EC4 }, + { 0x906, 0xF136 }, + { 0x907, 0x0409 }, + { 0x908, 0x04CC }, + { 0x909, 0x1C9B }, + { 0x90A, 0xF337 }, + { 0x90B, 0x040B }, + { 0x90C, 0x0CBB }, + { 0x90D, 0x16F8 }, + { 0x90E, 0xF7D9 }, + { 0x90F, 0x040A }, + { 0x910, 0x1F14 }, + { 0x911, 0x058C }, + { 0x912, 0x0563 }, + { 0x913, 0x4000 }, + { 0x916, 0x6318 }, + { 0x917, 0x6300 }, + { 0x918, 0x0FC8 }, + { 0x919, 0x03FE }, + { 0x91A, 0x00E0 }, + { 0x91B, 0x1EC4 }, + { 0x91C, 0xF136 }, + { 0x91D, 0x0409 }, + { 0x91E, 0x04CC }, + { 0x91F, 0x1C9B }, + { 0x920, 0xF337 }, + { 0x921, 0x040B }, + { 0x922, 0x0CBB }, + { 0x923, 0x16F8 }, + { 0x924, 0xF7D9 }, + { 0x925, 0x040A }, + { 0x926, 0x1F14 }, + { 0x927, 0x058C }, + { 0x928, 0x0563 }, + { 0x929, 0x4000 }, + { 0x709, 0x2000 }, + { 0x207, 0x200E }, + { 0x208, 0x20D4 }, + { 0x20A, 0x0080 }, + { 0x07, 0x0000 }, +}; + +static int wm2200_reset(struct wm2200_priv *wm2200) +{ + if (wm2200->pdata.reset) { + gpio_set_value_cansleep(wm2200->pdata.reset, 0); + gpio_set_value_cansleep(wm2200->pdata.reset, 1); + + return 0; + } else { + return regmap_write(wm2200->regmap, WM2200_SOFTWARE_RESET, + 0x2200); + } +} + +static DECLARE_TLV_DB_SCALE(in_tlv, -6300, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(out_tlv, -6400, 100, 0); + +static const char *wm2200_mixer_texts[] = { + "None", + "Tone Generator", + "AEC loopback", + "IN1L", + "IN1R", + "IN2L", + "IN2R", + "IN3L", + "IN3R", + "AIF1RX1", + "AIF1RX2", + "AIF1RX3", + "AIF1RX4", + "AIF1RX5", + "AIF1RX6", + "EQL", + "EQR", + "LHPF1", + "LHPF2", + "LHPF3", + "LHPF4", + "DSP1.1", + "DSP1.2", + "DSP1.3", + "DSP1.4", + "DSP1.5", + "DSP1.6", + "DSP2.1", + "DSP2.2", + "DSP2.3", + "DSP2.4", + "DSP2.5", + "DSP2.6", +}; + +static int wm2200_mixer_values[] = { + 0x00, + 0x04, /* Tone */ + 0x08, /* AEC */ + 0x10, /* Input */ + 0x11, + 0x12, + 0x13, + 0x14, + 0x15, + 0x20, /* AIF */ + 0x21, + 0x22, + 0x23, + 0x24, + 0x25, + 0x50, /* EQ */ + 0x51, + 0x52, + 0x60, /* LHPF1 */ + 0x61, /* LHPF2 */ + 0x68, /* DSP1 */ + 0x69, + 0x6a, + 0x6b, + 0x6c, + 0x6d, + 0x70, /* DSP2 */ + 0x71, + 0x72, + 0x73, + 0x74, + 0x75, +}; + +#define WM2200_MIXER_CONTROLS(name, base) \ + SOC_SINGLE_TLV(name " Input 1 Volume", base + 1 , \ + WM2200_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \ + SOC_SINGLE_TLV(name " Input 2 Volume", base + 3 , \ + WM2200_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \ + SOC_SINGLE_TLV(name " Input 3 Volume", base + 5 , \ + WM2200_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \ + SOC_SINGLE_TLV(name " Input 4 Volume", base + 7 , \ + WM2200_MIXER_VOL_SHIFT, 80, 0, mixer_tlv) + +#define WM2200_MUX_ENUM_DECL(name, reg) \ + SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \ + wm2200_mixer_texts, wm2200_mixer_values) + +#define WM2200_MUX_CTL_DECL(name) \ + const struct snd_kcontrol_new name##_mux = \ + SOC_DAPM_VALUE_ENUM("Route", name##_enum) + +#define WM2200_MIXER_ENUMS(name, base_reg) \ + static WM2200_MUX_ENUM_DECL(name##_in1_enum, base_reg); \ + static WM2200_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \ + static WM2200_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \ + static WM2200_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \ + static WM2200_MUX_CTL_DECL(name##_in1); \ + static WM2200_MUX_CTL_DECL(name##_in2); \ + static WM2200_MUX_CTL_DECL(name##_in3); \ + static WM2200_MUX_CTL_DECL(name##_in4) + +static const struct snd_kcontrol_new wm2200_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL, + WM2200_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", WM2200_IN2L_CONTROL, + WM2200_IN2_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN3 High Performance Switch", WM2200_IN3L_CONTROL, + WM2200_IN3_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R_TLV("IN1 Volume", WM2200_IN1L_CONTROL, WM2200_IN1R_CONTROL, + WM2200_IN1L_PGA_VOL_SHIFT, 0x5f, 0, in_tlv), +SOC_DOUBLE_R_TLV("IN2 Volume", WM2200_IN2L_CONTROL, WM2200_IN2R_CONTROL, + WM2200_IN2L_PGA_VOL_SHIFT, 0x5f, 0, in_tlv), +SOC_DOUBLE_R_TLV("IN3 Volume", WM2200_IN3L_CONTROL, WM2200_IN3R_CONTROL, + WM2200_IN3L_PGA_VOL_SHIFT, 0x5f, 0, in_tlv), + +SOC_DOUBLE_R("IN1 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, + WM2200_ADC_DIGITAL_VOLUME_1R, WM2200_IN1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, + WM2200_ADC_DIGITAL_VOLUME_2R, WM2200_IN2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L, + WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("IN1 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_1L, + WM2200_ADC_DIGITAL_VOLUME_1R, WM2200_IN1L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN2 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_2L, + WM2200_ADC_DIGITAL_VOLUME_2R, WM2200_IN2L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN3 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_3L, + WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_SINGLE("OUT1 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_1L, + WM2200_OUT1_OSR_SHIFT, 1, 0), +SOC_SINGLE("OUT2 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_2L, + WM2200_OUT2_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("OUT1 Digital Switch", WM2200_DAC_DIGITAL_VOLUME_1L, + WM2200_DAC_DIGITAL_VOLUME_1R, WM2200_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R_TLV("OUT1 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_1L, + WM2200_DAC_DIGITAL_VOLUME_1R, WM2200_OUT1L_VOL_SHIFT, 0x9f, 0, + digital_tlv), +SOC_DOUBLE_R_TLV("OUT1 Volume", WM2200_DAC_VOLUME_LIMIT_1L, + WM2200_DAC_VOLUME_LIMIT_1R, WM2200_OUT1L_PGA_VOL_SHIFT, + 0x46, 0, out_tlv), + +SOC_DOUBLE_R("OUT2 Digital Switch", WM2200_DAC_DIGITAL_VOLUME_2L, + WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, + WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0, + digital_tlv), +SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, + WM2200_SPK1R_MUTE_SHIFT, 1, 0), +}; + +WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(OUT1R, WM2200_OUT1RMIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(OUT2L, WM2200_OUT2LMIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(OUT2R, WM2200_OUT2RMIX_INPUT_1_SOURCE); + +WM2200_MIXER_ENUMS(AIF1TX1, WM2200_AIF1TX1MIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(AIF1TX2, WM2200_AIF1TX2MIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(AIF1TX3, WM2200_AIF1TX3MIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(AIF1TX4, WM2200_AIF1TX4MIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(AIF1TX5, WM2200_AIF1TX5MIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(AIF1TX6, WM2200_AIF1TX6MIX_INPUT_1_SOURCE); + +WM2200_MIXER_ENUMS(EQL, WM2200_EQLMIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(EQR, WM2200_EQRMIX_INPUT_1_SOURCE); + +WM2200_MIXER_ENUMS(DSP1L, WM2200_DSP1LMIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(DSP1R, WM2200_DSP1RMIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(DSP2L, WM2200_DSP2LMIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(DSP2R, WM2200_DSP2RMIX_INPUT_1_SOURCE); + +WM2200_MIXER_ENUMS(LHPF1, WM2200_LHPF1MIX_INPUT_1_SOURCE); +WM2200_MIXER_ENUMS(LHPF2, WM2200_LHPF2MIX_INPUT_1_SOURCE); + +#define WM2200_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +#define WM2200_MIXER_WIDGETS(name, name_str) \ + WM2200_MUX(name_str " Input 1", &name##_in1_mux), \ + WM2200_MUX(name_str " Input 2", &name##_in2_mux), \ + WM2200_MUX(name_str " Input 3", &name##_in3_mux), \ + WM2200_MUX(name_str " Input 4", &name##_in4_mux), \ + SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0) + +#define WM2200_MIXER_INPUT_ROUTES(name) \ + { name, "Tone Generator", "Tone Generator" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "IN3L", "IN3L PGA" }, \ + { name, "IN3R", "IN3R PGA" }, \ + { name, "DSP1.1", "DSP1" }, \ + { name, "DSP1.2", "DSP1" }, \ + { name, "DSP1.3", "DSP1" }, \ + { name, "DSP1.4", "DSP1" }, \ + { name, "DSP1.5", "DSP1" }, \ + { name, "DSP1.6", "DSP1" }, \ + { name, "DSP2.1", "DSP2" }, \ + { name, "DSP2.2", "DSP2" }, \ + { name, "DSP2.3", "DSP2" }, \ + { name, "DSP2.4", "DSP2" }, \ + { name, "DSP2.5", "DSP2" }, \ + { name, "DSP2.6", "DSP2" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "EQL", "EQL" }, \ + { name, "EQR", "EQR" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" } + +#define WM2200_MIXER_ROUTES(widget, name) \ + { widget, NULL, name " Mixer" }, \ + { name " Mixer", NULL, name " Input 1" }, \ + { name " Mixer", NULL, name " Input 2" }, \ + { name " Mixer", NULL, name " Input 3" }, \ + { name " Mixer", NULL, name " Input 4" }, \ + WM2200_MIXER_INPUT_ROUTES(name " Input 1"), \ + WM2200_MIXER_INPUT_ROUTES(name " Input 2"), \ + WM2200_MIXER_INPUT_ROUTES(name " Input 3"), \ + WM2200_MIXER_INPUT_ROUTES(name " Input 4") + +static const struct snd_soc_dapm_widget wm2200_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", WM2200_CLOCKING_3, WM2200_SYSCLK_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_SUPPLY("CP1", WM2200_DM_CHARGE_PUMP_1, WM2200_CPDM_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_SUPPLY("CP2", WM2200_MIC_CHARGE_PUMP_1, WM2200_CPMIC_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM2200_MIC_BIAS_CTRL_1, WM2200_MICB1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM2200_MIC_BIAS_CTRL_2, WM2200_MICB2_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 20), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_PGA("Tone Generator", WM2200_TONE_GENERATOR_1, + WM2200_TONE_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("IN1L PGA", WM2200_INPUT_ENABLES, WM2200_IN1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("IN1R PGA", WM2200_INPUT_ENABLES, WM2200_IN1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("IN2L PGA", WM2200_INPUT_ENABLES, WM2200_IN2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("IN2R PGA", WM2200_INPUT_ENABLES, WM2200_IN2R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("IN3L PGA", WM2200_INPUT_ENABLES, WM2200_IN3L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("IN3R PGA", WM2200_INPUT_ENABLES, WM2200_IN3R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", "Playback", 0, + WM2200_AUDIO_IF_1_22, WM2200_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", "Playback", 1, + WM2200_AUDIO_IF_1_22, WM2200_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", "Playback", 2, + WM2200_AUDIO_IF_1_22, WM2200_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", "Playback", 3, + WM2200_AUDIO_IF_1_22, WM2200_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", "Playback", 4, + WM2200_AUDIO_IF_1_22, WM2200_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", "Playback", 5, + WM2200_AUDIO_IF_1_22, WM2200_AIF1RX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA("EQL", WM2200_EQL_1, WM2200_EQL_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQR", WM2200_EQR_1, WM2200_EQR_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", WM2200_HPLPF1_1, WM2200_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", WM2200_HPLPF2_1, WM2200_LHPF2_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA_E("DSP1", SND_SOC_NOPM, 0, 0, NULL, 0, NULL, 0), +SND_SOC_DAPM_PGA_E("DSP2", SND_SOC_NOPM, 1, 0, NULL, 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, + WM2200_AUDIO_IF_1_22, WM2200_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 1, + WM2200_AUDIO_IF_1_22, WM2200_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 2, + WM2200_AUDIO_IF_1_22, WM2200_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 3, + WM2200_AUDIO_IF_1_22, WM2200_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 4, + WM2200_AUDIO_IF_1_22, WM2200_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 5, + WM2200_AUDIO_IF_1_22, WM2200_AIF1TX6_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_S("OUT1L", 0, WM2200_OUTPUT_ENABLES, + WM2200_OUT1L_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("OUT1R", 0, WM2200_OUTPUT_ENABLES, + WM2200_OUT1R_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("EPD_LP", 1, WM2200_EAR_PIECE_CTRL_1, + WM2200_EPD_LP_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("EPD_OUTP_LP", 1, WM2200_EAR_PIECE_CTRL_1, + WM2200_EPD_OUTP_LP_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("EPD_RMV_SHRT_LP", 1, WM2200_EAR_PIECE_CTRL_1, + WM2200_EPD_RMV_SHRT_LP_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("EPD_LN", 1, WM2200_EAR_PIECE_CTRL_1, + WM2200_EPD_LN_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("EPD_OUTP_LN", 1, WM2200_EAR_PIECE_CTRL_1, + WM2200_EPD_OUTP_LN_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("EPD_RMV_SHRT_LN", 1, WM2200_EAR_PIECE_CTRL_1, + WM2200_EPD_RMV_SHRT_LN_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("EPD_RP", 1, WM2200_EAR_PIECE_CTRL_2, + WM2200_EPD_RP_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("EPD_OUTP_RP", 1, WM2200_EAR_PIECE_CTRL_2, + WM2200_EPD_OUTP_RP_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("EPD_RMV_SHRT_RP", 1, WM2200_EAR_PIECE_CTRL_2, + WM2200_EPD_RMV_SHRT_RP_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("EPD_RN", 1, WM2200_EAR_PIECE_CTRL_2, + WM2200_EPD_RN_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("EPD_OUTP_RN", 1, WM2200_EAR_PIECE_CTRL_2, + WM2200_EPD_OUTP_RN_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("EPD_RMV_SHRT_RN", 1, WM2200_EAR_PIECE_CTRL_2, + WM2200_EPD_RMV_SHRT_RN_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("OUT2L", WM2200_OUTPUT_ENABLES, WM2200_OUT2L_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("OUT2R", WM2200_OUTPUT_ENABLES, WM2200_OUT2R_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_OUTPUT("EPOUTLN"), +SND_SOC_DAPM_OUTPUT("EPOUTLP"), +SND_SOC_DAPM_OUTPUT("EPOUTRN"), +SND_SOC_DAPM_OUTPUT("EPOUTRP"), +SND_SOC_DAPM_OUTPUT("SPK"), + +WM2200_MIXER_WIDGETS(EQL, "EQL"), +WM2200_MIXER_WIDGETS(EQR, "EQR"), + +WM2200_MIXER_WIDGETS(LHPF1, "LHPF1"), +WM2200_MIXER_WIDGETS(LHPF2, "LHPF2"), + +WM2200_MIXER_WIDGETS(DSP1L, "DSP1L"), +WM2200_MIXER_WIDGETS(DSP1R, "DSP1R"), +WM2200_MIXER_WIDGETS(DSP2L, "DSP2L"), +WM2200_MIXER_WIDGETS(DSP2R, "DSP2R"), + +WM2200_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +WM2200_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +WM2200_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +WM2200_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +WM2200_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +WM2200_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), + +WM2200_MIXER_WIDGETS(OUT1L, "OUT1L"), +WM2200_MIXER_WIDGETS(OUT1R, "OUT1R"), +WM2200_MIXER_WIDGETS(OUT2L, "OUT2L"), +WM2200_MIXER_WIDGETS(OUT2R, "OUT2R"), +}; + +static const struct snd_soc_dapm_route wm2200_dapm_routes[] = { + /* Everything needs SYSCLK but only hook up things on the edge + * of the chip */ + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + { "IN3L", NULL, "SYSCLK" }, + { "IN3R", NULL, "SYSCLK" }, + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "AIF1RX1", NULL, "SYSCLK" }, + { "AIF1RX2", NULL, "SYSCLK" }, + { "AIF1RX3", NULL, "SYSCLK" }, + { "AIF1RX4", NULL, "SYSCLK" }, + { "AIF1RX5", NULL, "SYSCLK" }, + { "AIF1RX6", NULL, "SYSCLK" }, + { "AIF1TX1", NULL, "SYSCLK" }, + { "AIF1TX2", NULL, "SYSCLK" }, + { "AIF1TX3", NULL, "SYSCLK" }, + { "AIF1TX4", NULL, "SYSCLK" }, + { "AIF1TX5", NULL, "SYSCLK" }, + { "AIF1TX6", NULL, "SYSCLK" }, + + { "IN1L", NULL, "AVDD" }, + { "IN1R", NULL, "AVDD" }, + { "IN2L", NULL, "AVDD" }, + { "IN2R", NULL, "AVDD" }, + { "IN3L", NULL, "AVDD" }, + { "IN3R", NULL, "AVDD" }, + { "OUT1L", NULL, "AVDD" }, + { "OUT1R", NULL, "AVDD" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + + { "Tone Generator", NULL, "TONE" }, + + { "CP2", NULL, "CPVDD" }, + { "MICBIAS1", NULL, "CP2" }, + { "MICBIAS2", NULL, "CP2" }, + + { "CP1", NULL, "CPVDD" }, + { "EPD_LN", NULL, "CP1" }, + { "EPD_LP", NULL, "CP1" }, + { "EPD_RN", NULL, "CP1" }, + { "EPD_RP", NULL, "CP1" }, + + { "EPD_LP", NULL, "OUT1L" }, + { "EPD_OUTP_LP", NULL, "EPD_LP" }, + { "EPD_RMV_SHRT_LP", NULL, "EPD_OUTP_LP" }, + { "EPOUTLP", NULL, "EPD_RMV_SHRT_LP" }, + + { "EPD_LN", NULL, "OUT1L" }, + { "EPD_OUTP_LN", NULL, "EPD_LN" }, + { "EPD_RMV_SHRT_LN", NULL, "EPD_OUTP_LN" }, + { "EPOUTLN", NULL, "EPD_RMV_SHRT_LN" }, + + { "EPD_RP", NULL, "OUT1R" }, + { "EPD_OUTP_RP", NULL, "EPD_RP" }, + { "EPD_RMV_SHRT_RP", NULL, "EPD_OUTP_RP" }, + { "EPOUTRP", NULL, "EPD_RMV_SHRT_RP" }, + + { "EPD_RN", NULL, "OUT1R" }, + { "EPD_OUTP_RN", NULL, "EPD_RN" }, + { "EPD_RMV_SHRT_RN", NULL, "EPD_OUTP_RN" }, + { "EPOUTRN", NULL, "EPD_RMV_SHRT_RN" }, + + { "SPK", NULL, "OUT2L" }, + { "SPK", NULL, "OUT2R" }, + + WM2200_MIXER_ROUTES("DSP1", "DSP1L"), + WM2200_MIXER_ROUTES("DSP1", "DSP1R"), + WM2200_MIXER_ROUTES("DSP2", "DSP2L"), + WM2200_MIXER_ROUTES("DSP2", "DSP2R"), + + WM2200_MIXER_ROUTES("OUT1L", "OUT1L"), + WM2200_MIXER_ROUTES("OUT1R", "OUT1R"), + WM2200_MIXER_ROUTES("OUT2L", "OUT2L"), + WM2200_MIXER_ROUTES("OUT2R", "OUT2R"), + + WM2200_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + WM2200_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + WM2200_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + WM2200_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + WM2200_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + WM2200_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + + WM2200_MIXER_ROUTES("EQL", "EQL"), + WM2200_MIXER_ROUTES("EQR", "EQR"), + + WM2200_MIXER_ROUTES("LHPF1", "LHPF1"), + WM2200_MIXER_ROUTES("LHPF2", "LHPF2"), +}; + +static int wm2200_probe(struct snd_soc_codec *codec) +{ + struct wm2200_priv *wm2200 = dev_get_drvdata(codec->dev); + int ret; + + wm2200->codec = codec; + codec->control_data = wm2200->regmap; + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return ret; +} + +static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int lrclk, bclk, fmt_val; + + lrclk = 0; + bclk = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + fmt_val = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + fmt_val = 1; + break; + case SND_SOC_DAIFMT_I2S: + fmt_val = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + fmt_val = 3; + break; + default: + dev_err(codec->dev, "Unsupported DAI format %d\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk |= WM2200_AIF1TX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= WM2200_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + lrclk |= WM2200_AIF1TX_LRCLK_MSTR; + bclk |= WM2200_AIF1_BCLK_MSTR; + break; + default: + dev_err(codec->dev, "Unsupported master mode %d\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= WM2200_AIF1_BCLK_INV; + lrclk |= WM2200_AIF1TX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= WM2200_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk |= WM2200_AIF1TX_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_1, WM2200_AIF1_BCLK_MSTR | + WM2200_AIF1_BCLK_INV, bclk); + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_2, + WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV, + lrclk); + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_3, + WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV, + lrclk); + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5, + WM2200_AIF1_FMT_MASK << 1, fmt_val << 1); + + return 0; +} + +static int wm2200_sr_code[] = { + 0, + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 0, + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +#define WM2200_NUM_BCLK_RATES 12 + +static int wm2200_bclk_rates_dat[WM2200_NUM_BCLK_RATES] = { + 6144000, + 3072000, + 2048000, + 1536000, + 768000, + 512000, + 384000, + 256000, + 192000, + 128000, + 96000, + 64000, +}; + +static int wm2200_bclk_rates_cd[WM2200_NUM_BCLK_RATES] = { + 5644800, + 2882400, + 1881600, + 1411200, + 705600, + 470400, + 352800, + 176400, + 117600, + 88200, + 58800, +}; + +static int wm2200_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm2200_priv *wm2200 = snd_soc_codec_get_drvdata(codec); + int i, bclk, lrclk, wl, fl, sr_code; + int *bclk_rates; + + /* Data sizes if not using TDM */ + wl = snd_pcm_format_width(params_format(params)); + if (wl < 0) + return wl; + fl = snd_soc_params_to_frame_size(params); + if (fl < 0) + return fl; + + dev_dbg(codec->dev, "Word length %d bits, frame length %d bits\n", + wl, fl); + + /* Target BCLK rate */ + bclk = snd_soc_params_to_bclk(params); + if (bclk < 0) + return bclk; + + if (!wm2200->sysclk) { + dev_err(codec->dev, "SYSCLK has no rate set\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(wm2200_sr_code); i++) + if (wm2200_sr_code[i] == params_rate(params)) + break; + if (i == ARRAY_SIZE(wm2200_sr_code)) { + dev_err(codec->dev, "Unsupported sample rate: %dHz\n", + params_rate(params)); + return -EINVAL; + } + sr_code = i; + + dev_dbg(codec->dev, "Target BCLK is %dHz, using %dHz SYSCLK\n", + bclk, wm2200->sysclk); + + if (wm2200->sysclk % 4000) + bclk_rates = wm2200_bclk_rates_cd; + else + bclk_rates = wm2200_bclk_rates_dat; + + for (i = 0; i < WM2200_NUM_BCLK_RATES; i++) + if (bclk_rates[i] >= bclk && (bclk_rates[i] % bclk == 0)) + break; + if (i == WM2200_NUM_BCLK_RATES) { + dev_err(codec->dev, + "No valid BCLK for %dHz found from %dHz SYSCLK\n", + bclk, wm2200->sysclk); + return -EINVAL; + } + + bclk = i; + dev_dbg(codec->dev, "Setting %dHz BCLK\n", bclk_rates[bclk]); + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_1, + WM2200_AIF1_BCLK_DIV_MASK, bclk); + + lrclk = bclk_rates[bclk] / params_rate(params); + dev_dbg(codec->dev, "Setting %dHz LRCLK\n", bclk_rates[bclk] / lrclk); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK || + dai->symmetric_rates) + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_7, + WM2200_AIF1RX_BCPF_MASK, lrclk); + else + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_6, + WM2200_AIF1TX_BCPF_MASK, lrclk); + + i = (wl << WM2200_AIF1TX_WL_SHIFT) | wl; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_9, + WM2200_AIF1RX_WL_MASK | + WM2200_AIF1RX_SLOT_LEN_MASK, i); + else + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_8, + WM2200_AIF1TX_WL_MASK | + WM2200_AIF1TX_SLOT_LEN_MASK, i); + + snd_soc_update_bits(codec, WM2200_CLOCKING_4, + WM2200_SAMPLE_RATE_1_MASK, sr_code); + + return 0; +} + +static const struct snd_soc_dai_ops wm2200_dai_ops = { + .set_fmt = wm2200_set_fmt, + .hw_params = wm2200_hw_params, +}; + +static int wm2200_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct wm2200_priv *wm2200 = snd_soc_codec_get_drvdata(codec); + int fval; + + switch (clk_id) { + case WM2200_CLK_SYSCLK: + break; + + default: + dev_err(codec->dev, "Unknown clock %d\n", clk_id); + return -EINVAL; + } + + switch (source) { + case WM2200_CLKSRC_MCLK1: + case WM2200_CLKSRC_MCLK2: + case WM2200_CLKSRC_FLL: + case WM2200_CLKSRC_BCLK1: + break; + default: + dev_err(codec->dev, "Invalid source %d\n", source); + return -EINVAL; + } + + switch (freq) { + case 22579200: + case 24576000: + fval = 2; + break; + default: + dev_err(codec->dev, "Invalid clock rate: %d\n", freq); + return -EINVAL; + } + + /* TODO: Check if MCLKs are in use and enable/disable pulls to + * match. + */ + + snd_soc_update_bits(codec, WM2200_CLOCKING_3, WM2200_SYSCLK_FREQ_MASK | + WM2200_SYSCLK_SRC_MASK, + fval << WM2200_SYSCLK_FREQ_SHIFT | source); + + wm2200->sysclk = freq; + + return 0; +} + +struct _fll_div { + u16 fll_fratio; + u16 fll_outdiv; + u16 fll_refclk_div; + u16 n; + u16 theta; + u16 lambda; +}; + +static struct { + unsigned int min; + unsigned int max; + u16 fll_fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, + unsigned int Fout) +{ + unsigned int target; + unsigned int div; + unsigned int fratio, gcd_fll; + int i; + + /* Fref must be <=13.5MHz */ + div = 1; + fll_div->fll_refclk_div = 0; + while ((Fref / div) > 13500000) { + div *= 2; + fll_div->fll_refclk_div++; + + if (div > 8) { + pr_err("Can't scale %dMHz input down to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + pr_debug("FLL Fref=%u Fout=%u\n", Fref, Fout); + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 2; + while (Fout * div < 90000000) { + div++; + if (div > 64) { + pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + fll_div->fll_outdiv = div - 1; + + pr_debug("FLL Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + fll_div->fll_fratio = fll_fratios[i].fll_fratio; + fratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref); + return -EINVAL; + } + + fll_div->n = target / (fratio * Fref); + + if (target % Fref == 0) { + fll_div->theta = 0; + fll_div->lambda = 0; + } else { + gcd_fll = gcd(target, fratio * Fref); + + fll_div->theta = (target - (fll_div->n * fratio * Fref)) + / gcd_fll; + fll_div->lambda = (fratio * Fref) / gcd_fll; + } + + pr_debug("FLL N=%x THETA=%x LAMBDA=%x\n", + fll_div->n, fll_div->theta, fll_div->lambda); + pr_debug("FLL_FRATIO=%x(%d) FLL_OUTDIV=%x FLL_REFCLK_DIV=%x\n", + fll_div->fll_fratio, fratio, fll_div->fll_outdiv, + fll_div->fll_refclk_div); + + return 0; +} + +static int wm2200_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct i2c_client *i2c = to_i2c_client(codec->dev); + struct wm2200_priv *wm2200 = snd_soc_codec_get_drvdata(codec); + struct _fll_div factors; + int ret, i, timeout; + + if (!Fout) { + dev_dbg(codec->dev, "FLL disabled"); + + if (wm2200->fll_fout) + pm_runtime_put(codec->dev); + + wm2200->fll_fout = 0; + snd_soc_update_bits(codec, WM2200_FLL_CONTROL_1, + WM2200_FLL_ENA, 0); + return 0; + } + + switch (source) { + case WM2200_FLL_SRC_MCLK1: + case WM2200_FLL_SRC_MCLK2: + case WM2200_FLL_SRC_BCLK: + break; + default: + dev_err(codec->dev, "Invalid FLL source %d\n", source); + return -EINVAL; + } + + ret = fll_factors(&factors, Fref, Fout); + if (ret < 0) + return ret; + + /* Disable the FLL while we reconfigure */ + snd_soc_update_bits(codec, WM2200_FLL_CONTROL_1, WM2200_FLL_ENA, 0); + + snd_soc_update_bits(codec, WM2200_FLL_CONTROL_2, + WM2200_FLL_OUTDIV_MASK | WM2200_FLL_FRATIO_MASK, + (factors.fll_outdiv << WM2200_FLL_OUTDIV_SHIFT) | + factors.fll_fratio); + if (factors.theta) { + snd_soc_update_bits(codec, WM2200_FLL_CONTROL_3, + WM2200_FLL_FRACN_ENA, + WM2200_FLL_FRACN_ENA); + snd_soc_update_bits(codec, WM2200_FLL_EFS_2, + WM2200_FLL_EFS_ENA, + WM2200_FLL_EFS_ENA); + } else { + snd_soc_update_bits(codec, WM2200_FLL_CONTROL_3, + WM2200_FLL_FRACN_ENA, 0); + snd_soc_update_bits(codec, WM2200_FLL_EFS_2, + WM2200_FLL_EFS_ENA, 0); + } + + snd_soc_update_bits(codec, WM2200_FLL_CONTROL_4, WM2200_FLL_THETA_MASK, + factors.theta); + snd_soc_update_bits(codec, WM2200_FLL_CONTROL_6, WM2200_FLL_N_MASK, + factors.n); + snd_soc_update_bits(codec, WM2200_FLL_CONTROL_7, + WM2200_FLL_CLK_REF_DIV_MASK | + WM2200_FLL_CLK_REF_SRC_MASK, + (factors.fll_refclk_div + << WM2200_FLL_CLK_REF_DIV_SHIFT) | source); + snd_soc_update_bits(codec, WM2200_FLL_EFS_1, + WM2200_FLL_LAMBDA_MASK, factors.lambda); + + /* Clear any pending completions */ + try_wait_for_completion(&wm2200->fll_lock); + + pm_runtime_get_sync(codec->dev); + + snd_soc_update_bits(codec, WM2200_FLL_CONTROL_1, + WM2200_FLL_ENA, WM2200_FLL_ENA); + + if (i2c->irq) + timeout = 2; + else + timeout = 50; + + snd_soc_update_bits(codec, WM2200_CLOCKING_3, WM2200_SYSCLK_ENA, + WM2200_SYSCLK_ENA); + + /* Poll for the lock; will use the interrupt to exit quickly */ + for (i = 0; i < timeout; i++) { + if (i2c->irq) { + ret = wait_for_completion_timeout(&wm2200->fll_lock, + msecs_to_jiffies(25)); + if (ret > 0) + break; + } else { + msleep(1); + } + + ret = snd_soc_read(codec, + WM2200_INTERRUPT_RAW_STATUS_2); + if (ret < 0) { + dev_err(codec->dev, + "Failed to read FLL status: %d\n", + ret); + continue; + } + if (ret & WM2200_FLL_LOCK_STS) + break; + } + if (i == timeout) { + dev_err(codec->dev, "FLL lock timed out\n"); + pm_runtime_put(codec->dev); + return -ETIMEDOUT; + } + + wm2200->fll_src = source; + wm2200->fll_fref = Fref; + wm2200->fll_fout = Fout; + + dev_dbg(codec->dev, "FLL running %dHz->%dHz\n", Fref, Fout); + + return 0; +} + +static int wm2200_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val = 0; + int ret; + + ret = snd_soc_read(codec, WM2200_GPIO_CTRL_1); + if (ret >= 0) { + if ((ret & WM2200_GP1_FN_MASK) != 0) { + dai->symmetric_rates = true; + val = WM2200_AIF1TX_LRCLK_SRC; + } + } else { + dev_err(codec->dev, "Failed to read GPIO 1 config: %d\n", ret); + } + + snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_2, + WM2200_AIF1TX_LRCLK_SRC, val); + + return 0; +} + +#define WM2200_RATES SNDRV_PCM_RATE_8000_48000 + +#define WM2200_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm2200_dai = { + .name = "wm2200", + .probe = wm2200_dai_probe, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM2200_RATES, + .formats = WM2200_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM2200_RATES, + .formats = WM2200_FORMATS, + }, + .ops = &wm2200_dai_ops, +}; + +static struct snd_soc_codec_driver soc_codec_wm2200 = { + .probe = wm2200_probe, + + .idle_bias_off = true, + .set_sysclk = wm2200_set_sysclk, + .set_pll = wm2200_set_fll, + + .controls = wm2200_snd_controls, + .num_controls = ARRAY_SIZE(wm2200_snd_controls), + .dapm_widgets = wm2200_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm2200_dapm_widgets), + .dapm_routes = wm2200_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm2200_dapm_routes), +}; + +static irqreturn_t wm2200_irq(int irq, void *data) +{ + struct wm2200_priv *wm2200 = data; + unsigned int val, mask; + int ret; + + ret = regmap_read(wm2200->regmap, WM2200_INTERRUPT_STATUS_2, &val); + if (ret != 0) { + dev_err(wm2200->dev, "Failed to read IRQ status: %d\n", ret); + return IRQ_NONE; + } + + ret = regmap_read(wm2200->regmap, WM2200_INTERRUPT_STATUS_2_MASK, + &mask); + if (ret != 0) { + dev_warn(wm2200->dev, "Failed to read IRQ mask: %d\n", ret); + mask = 0; + } + + val &= ~mask; + + if (val & WM2200_FLL_LOCK_EINT) { + dev_dbg(wm2200->dev, "FLL locked\n"); + complete(&wm2200->fll_lock); + } + + if (val) { + regmap_write(wm2200->regmap, WM2200_INTERRUPT_STATUS_2, val); + + return IRQ_HANDLED; + } else { + return IRQ_NONE; + } +} + +static const struct regmap_config wm2200_regmap = { + .reg_bits = 16, + .val_bits = 16, + + .max_register = WM2200_MAX_REGISTER, + .reg_defaults = wm2200_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm2200_reg_defaults), + .volatile_reg = wm2200_volatile_register, + .readable_reg = wm2200_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static const unsigned int wm2200_dig_vu[] = { + WM2200_DAC_DIGITAL_VOLUME_1L, + WM2200_DAC_DIGITAL_VOLUME_1R, + WM2200_DAC_DIGITAL_VOLUME_2L, + WM2200_DAC_DIGITAL_VOLUME_2R, + WM2200_ADC_DIGITAL_VOLUME_1L, + WM2200_ADC_DIGITAL_VOLUME_1R, + WM2200_ADC_DIGITAL_VOLUME_2L, + WM2200_ADC_DIGITAL_VOLUME_2R, + WM2200_ADC_DIGITAL_VOLUME_3L, + WM2200_ADC_DIGITAL_VOLUME_3R, +}; + +static const unsigned int wm2200_mic_ctrl_reg[] = { + WM2200_IN1L_CONTROL, + WM2200_IN2L_CONTROL, + WM2200_IN3L_CONTROL, +}; + +static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm2200_pdata *pdata = dev_get_platdata(&i2c->dev); + struct wm2200_priv *wm2200; + unsigned int reg; + int ret, i; + + wm2200 = devm_kzalloc(&i2c->dev, sizeof(struct wm2200_priv), + GFP_KERNEL); + if (wm2200 == NULL) + return -ENOMEM; + + wm2200->dev = &i2c->dev; + init_completion(&wm2200->fll_lock); + + wm2200->regmap = regmap_init_i2c(i2c, &wm2200_regmap); + if (IS_ERR(wm2200->regmap)) { + ret = PTR_ERR(wm2200->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + goto err; + } + + if (pdata) + wm2200->pdata = *pdata; + + i2c_set_clientdata(i2c, wm2200); + + for (i = 0; i < ARRAY_SIZE(wm2200->core_supplies); i++) + wm2200->core_supplies[i].supply = wm2200_core_supply_names[i]; + + ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request core supplies: %d\n", + ret); + goto err_regmap; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", + ret); + goto err_core; + } + + if (wm2200->pdata.ldo_ena) { + ret = gpio_request_one(wm2200->pdata.ldo_ena, + GPIOF_OUT_INIT_HIGH, "WM2200 LDOENA"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", + wm2200->pdata.ldo_ena, ret); + goto err_enable; + } + msleep(2); + } + + if (wm2200->pdata.reset) { + ret = gpio_request_one(wm2200->pdata.reset, + GPIOF_OUT_INIT_HIGH, "WM2200 /RESET"); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", + wm2200->pdata.reset, ret); + goto err_ldo; + } + } + + ret = regmap_read(wm2200->regmap, WM2200_SOFTWARE_RESET, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); + goto err_reset; + } + switch (reg) { + case 0x2200: + break; + + default: + dev_err(&i2c->dev, "Device is not a WM2200, ID is %x\n", reg); + ret = -EINVAL; + goto err_reset; + } + + ret = regmap_read(wm2200->regmap, WM2200_DEVICE_REVISION, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read revision register\n"); + goto err_reset; + } + + wm2200->rev = ret & WM2200_DEVICE_REVISION_MASK; + + dev_info(&i2c->dev, "revision %c\n", wm2200->rev + 'A'); + + switch (wm2200->rev) { + case 0: + ret = regmap_register_patch(wm2200->regmap, wm2200_reva_patch, + ARRAY_SIZE(wm2200_reva_patch)); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register patch: %d\n", + ret); + } + break; + default: + break; + } + + ret = wm2200_reset(wm2200); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + goto err_reset; + } + + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.gpio_defaults); i++) { + if (!wm2200->pdata.gpio_defaults[i]) + continue; + + regmap_write(wm2200->regmap, WM2200_GPIO_CTRL_1 + i, + wm2200->pdata.gpio_defaults[i]); + } + + for (i = 0; i < ARRAY_SIZE(wm2200_dig_vu); i++) + regmap_update_bits(wm2200->regmap, wm2200_dig_vu[i], + WM2200_OUT_VU, WM2200_OUT_VU); + + /* Assign slots 1-6 to channels 1-6 for both TX and RX */ + for (i = 0; i < 6; i++) { + regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_10 + i, i); + regmap_write(wm2200->regmap, WM2200_AUDIO_IF_1_16 + i, i); + } + + for (i = 0; i < ARRAY_SIZE(wm2200->pdata.in_mode); i++) { + regmap_update_bits(wm2200->regmap, wm2200_mic_ctrl_reg[i], + WM2200_IN1_MODE_MASK | + WM2200_IN1_DMIC_SUP_MASK, + (wm2200->pdata.in_mode[i] << + WM2200_IN1_MODE_SHIFT) | + (wm2200->pdata.dmic_sup[i] << + WM2200_IN1_DMIC_SUP_SHIFT)); + } + + if (i2c->irq) { + ret = request_threaded_irq(i2c->irq, NULL, wm2200_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, + "wm2200", wm2200); + if (ret == 0) + regmap_update_bits(wm2200->regmap, + WM2200_INTERRUPT_STATUS_2_MASK, + WM2200_FLL_LOCK_EINT, 0); + else + dev_err(&i2c->dev, "Failed to request IRQ %d: %d\n", + i2c->irq, ret); + } + + pm_runtime_set_active(&i2c->dev); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_wm2200, + &wm2200_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + goto err_pm_runtime; + } + + return 0; + +err_pm_runtime: + pm_runtime_disable(&i2c->dev); +err_reset: + if (wm2200->pdata.reset) { + gpio_set_value_cansleep(wm2200->pdata.reset, 0); + gpio_free(wm2200->pdata.reset); + } +err_ldo: + if (wm2200->pdata.ldo_ena) { + gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); + gpio_free(wm2200->pdata.ldo_ena); + } +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); +err_core: + regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); +err_regmap: + regmap_exit(wm2200->regmap); +err: + return ret; +} + +static __devexit int wm2200_i2c_remove(struct i2c_client *i2c) +{ + struct wm2200_priv *wm2200 = i2c_get_clientdata(i2c); + + snd_soc_unregister_codec(&i2c->dev); + if (i2c->irq) + free_irq(i2c->irq, wm2200); + if (wm2200->pdata.reset) { + gpio_set_value_cansleep(wm2200->pdata.reset, 0); + gpio_free(wm2200->pdata.reset); + } + if (wm2200->pdata.ldo_ena) { + gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); + gpio_free(wm2200->pdata.ldo_ena); + } + regulator_bulk_free(ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); + regmap_exit(wm2200->regmap); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int wm2200_runtime_suspend(struct device *dev) +{ + struct wm2200_priv *wm2200 = dev_get_drvdata(dev); + + regcache_cache_only(wm2200->regmap, true); + regcache_mark_dirty(wm2200->regmap); + if (wm2200->pdata.ldo_ena) + gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); + + return 0; +} + +static int wm2200_runtime_resume(struct device *dev) +{ + struct wm2200_priv *wm2200 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(wm2200->core_supplies), + wm2200->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (wm2200->pdata.ldo_ena) { + gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 1); + msleep(2); + } + + regcache_cache_only(wm2200->regmap, false); + regcache_sync(wm2200->regmap); + + return 0; +} +#endif + +static struct dev_pm_ops wm2200_pm = { + SET_RUNTIME_PM_OPS(wm2200_runtime_suspend, wm2200_runtime_resume, + NULL) +}; + +static const struct i2c_device_id wm2200_i2c_id[] = { + { "wm2200", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm2200_i2c_id); + +static struct i2c_driver wm2200_i2c_driver = { + .driver = { + .name = "wm2200", + .owner = THIS_MODULE, + .pm = &wm2200_pm, + }, + .probe = wm2200_i2c_probe, + .remove = __devexit_p(wm2200_i2c_remove), + .id_table = wm2200_i2c_id, +}; + +static int __init wm2200_modinit(void) +{ + return i2c_add_driver(&wm2200_i2c_driver); +} +module_init(wm2200_modinit); + +static void __exit wm2200_exit(void) +{ + i2c_del_driver(&wm2200_i2c_driver); +} +module_exit(wm2200_exit); + +MODULE_DESCRIPTION("ASoC WM2200 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm2200.h b/sound/soc/codecs/wm2200.h new file mode 100644 index 00000000000..5d719d6b4a8 --- /dev/null +++ b/sound/soc/codecs/wm2200.h @@ -0,0 +1,3674 @@ +/* + * wm2200.h - WM2200 audio codec interface + * + * Copyright 2012 Wolfson Microelectronics PLC. + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _WM2200_H +#define _WM2200_H + +#define WM2200_CLK_SYSCLK 1 + +#define WM2200_CLKSRC_MCLK1 0 +#define WM2200_CLKSRC_MCLK2 1 +#define WM2200_CLKSRC_FLL 4 +#define WM2200_CLKSRC_BCLK1 8 + +#define WM2200_FLL_SRC_MCLK1 0 +#define WM2200_FLL_SRC_MCLK2 1 +#define WM2200_FLL_SRC_BCLK 2 + +/* + * Register values. + */ +#define WM2200_SOFTWARE_RESET 0x00 +#define WM2200_DEVICE_REVISION 0x01 +#define WM2200_TONE_GENERATOR_1 0x0B +#define WM2200_CLOCKING_3 0x102 +#define WM2200_CLOCKING_4 0x103 +#define WM2200_FLL_CONTROL_1 0x111 +#define WM2200_FLL_CONTROL_2 0x112 +#define WM2200_FLL_CONTROL_3 0x113 +#define WM2200_FLL_CONTROL_4 0x114 +#define WM2200_FLL_CONTROL_6 0x116 +#define WM2200_FLL_CONTROL_7 0x117 +#define WM2200_FLL_EFS_1 0x119 +#define WM2200_FLL_EFS_2 0x11A +#define WM2200_MIC_CHARGE_PUMP_1 0x200 +#define WM2200_MIC_CHARGE_PUMP_2 0x201 +#define WM2200_DM_CHARGE_PUMP_1 0x202 +#define WM2200_MIC_BIAS_CTRL_1 0x20C +#define WM2200_MIC_BIAS_CTRL_2 0x20D +#define WM2200_EAR_PIECE_CTRL_1 0x20F +#define WM2200_EAR_PIECE_CTRL_2 0x210 +#define WM2200_INPUT_ENABLES 0x301 +#define WM2200_IN1L_CONTROL 0x302 +#define WM2200_IN1R_CONTROL 0x303 +#define WM2200_IN2L_CONTROL 0x304 +#define WM2200_IN2R_CONTROL 0x305 +#define WM2200_IN3L_CONTROL 0x306 +#define WM2200_IN3R_CONTROL 0x307 +#define WM2200_RXANC_SRC 0x30A +#define WM2200_INPUT_VOLUME_RAMP 0x30B +#define WM2200_ADC_DIGITAL_VOLUME_1L 0x30C +#define WM2200_ADC_DIGITAL_VOLUME_1R 0x30D +#define WM2200_ADC_DIGITAL_VOLUME_2L 0x30E +#define WM2200_ADC_DIGITAL_VOLUME_2R 0x30F +#define WM2200_ADC_DIGITAL_VOLUME_3L 0x310 +#define WM2200_ADC_DIGITAL_VOLUME_3R 0x311 +#define WM2200_OUTPUT_ENABLES 0x400 +#define WM2200_DAC_VOLUME_LIMIT_1L 0x401 +#define WM2200_DAC_VOLUME_LIMIT_1R 0x402 +#define WM2200_DAC_VOLUME_LIMIT_2L 0x403 +#define WM2200_DAC_VOLUME_LIMIT_2R 0x404 +#define WM2200_DAC_AEC_CONTROL_1 0x409 +#define WM2200_OUTPUT_VOLUME_RAMP 0x40A +#define WM2200_DAC_DIGITAL_VOLUME_1L 0x40B +#define WM2200_DAC_DIGITAL_VOLUME_1R 0x40C +#define WM2200_DAC_DIGITAL_VOLUME_2L 0x40D +#define WM2200_DAC_DIGITAL_VOLUME_2R 0x40E +#define WM2200_PDM_1 0x417 +#define WM2200_PDM_2 0x418 +#define WM2200_AUDIO_IF_1_1 0x500 +#define WM2200_AUDIO_IF_1_2 0x501 +#define WM2200_AUDIO_IF_1_3 0x502 +#define WM2200_AUDIO_IF_1_4 0x503 +#define WM2200_AUDIO_IF_1_5 0x504 +#define WM2200_AUDIO_IF_1_6 0x505 +#define WM2200_AUDIO_IF_1_7 0x506 +#define WM2200_AUDIO_IF_1_8 0x507 +#define WM2200_AUDIO_IF_1_9 0x508 +#define WM2200_AUDIO_IF_1_10 0x509 +#define WM2200_AUDIO_IF_1_11 0x50A +#define WM2200_AUDIO_IF_1_12 0x50B +#define WM2200_AUDIO_IF_1_13 0x50C +#define WM2200_AUDIO_IF_1_14 0x50D +#define WM2200_AUDIO_IF_1_15 0x50E +#define WM2200_AUDIO_IF_1_16 0x50F +#define WM2200_AUDIO_IF_1_17 0x510 +#define WM2200_AUDIO_IF_1_18 0x511 +#define WM2200_AUDIO_IF_1_19 0x512 +#define WM2200_AUDIO_IF_1_20 0x513 +#define WM2200_AUDIO_IF_1_21 0x514 +#define WM2200_AUDIO_IF_1_22 0x515 +#define WM2200_OUT1LMIX_INPUT_1_SOURCE 0x600 +#define WM2200_OUT1LMIX_INPUT_1_VOLUME 0x601 +#define WM2200_OUT1LMIX_INPUT_2_SOURCE 0x602 +#define WM2200_OUT1LMIX_INPUT_2_VOLUME 0x603 +#define WM2200_OUT1LMIX_INPUT_3_SOURCE 0x604 +#define WM2200_OUT1LMIX_INPUT_3_VOLUME 0x605 +#define WM2200_OUT1LMIX_INPUT_4_SOURCE 0x606 +#define WM2200_OUT1LMIX_INPUT_4_VOLUME 0x607 +#define WM2200_OUT1RMIX_INPUT_1_SOURCE 0x608 +#define WM2200_OUT1RMIX_INPUT_1_VOLUME 0x609 +#define WM2200_OUT1RMIX_INPUT_2_SOURCE 0x60A +#define WM2200_OUT1RMIX_INPUT_2_VOLUME 0x60B +#define WM2200_OUT1RMIX_INPUT_3_SOURCE 0x60C +#define WM2200_OUT1RMIX_INPUT_3_VOLUME 0x60D +#define WM2200_OUT1RMIX_INPUT_4_SOURCE 0x60E +#define WM2200_OUT1RMIX_INPUT_4_VOLUME 0x60F +#define WM2200_OUT2LMIX_INPUT_1_SOURCE 0x610 +#define WM2200_OUT2LMIX_INPUT_1_VOLUME 0x611 +#define WM2200_OUT2LMIX_INPUT_2_SOURCE 0x612 +#define WM2200_OUT2LMIX_INPUT_2_VOLUME 0x613 +#define WM2200_OUT2LMIX_INPUT_3_SOURCE 0x614 +#define WM2200_OUT2LMIX_INPUT_3_VOLUME 0x615 +#define WM2200_OUT2LMIX_INPUT_4_SOURCE 0x616 +#define WM2200_OUT2LMIX_INPUT_4_VOLUME 0x617 +#define WM2200_OUT2RMIX_INPUT_1_SOURCE 0x618 +#define WM2200_OUT2RMIX_INPUT_1_VOLUME 0x619 +#define WM2200_OUT2RMIX_INPUT_2_SOURCE 0x61A +#define WM2200_OUT2RMIX_INPUT_2_VOLUME 0x61B +#define WM2200_OUT2RMIX_INPUT_3_SOURCE 0x61C +#define WM2200_OUT2RMIX_INPUT_3_VOLUME 0x61D +#define WM2200_OUT2RMIX_INPUT_4_SOURCE 0x61E +#define WM2200_OUT2RMIX_INPUT_4_VOLUME 0x61F +#define WM2200_AIF1TX1MIX_INPUT_1_SOURCE 0x620 +#define WM2200_AIF1TX1MIX_INPUT_1_VOLUME 0x621 +#define WM2200_AIF1TX1MIX_INPUT_2_SOURCE 0x622 +#define WM2200_AIF1TX1MIX_INPUT_2_VOLUME 0x623 +#define WM2200_AIF1TX1MIX_INPUT_3_SOURCE 0x624 +#define WM2200_AIF1TX1MIX_INPUT_3_VOLUME 0x625 +#define WM2200_AIF1TX1MIX_INPUT_4_SOURCE 0x626 +#define WM2200_AIF1TX1MIX_INPUT_4_VOLUME 0x627 +#define WM2200_AIF1TX2MIX_INPUT_1_SOURCE 0x628 +#define WM2200_AIF1TX2MIX_INPUT_1_VOLUME 0x629 +#define WM2200_AIF1TX2MIX_INPUT_2_SOURCE 0x62A +#define WM2200_AIF1TX2MIX_INPUT_2_VOLUME 0x62B +#define WM2200_AIF1TX2MIX_INPUT_3_SOURCE 0x62C +#define WM2200_AIF1TX2MIX_INPUT_3_VOLUME 0x62D +#define WM2200_AIF1TX2MIX_INPUT_4_SOURCE 0x62E +#define WM2200_AIF1TX2MIX_INPUT_4_VOLUME 0x62F +#define WM2200_AIF1TX3MIX_INPUT_1_SOURCE 0x630 +#define WM2200_AIF1TX3MIX_INPUT_1_VOLUME 0x631 +#define WM2200_AIF1TX3MIX_INPUT_2_SOURCE 0x632 +#define WM2200_AIF1TX3MIX_INPUT_2_VOLUME 0x633 +#define WM2200_AIF1TX3MIX_INPUT_3_SOURCE 0x634 +#define WM2200_AIF1TX3MIX_INPUT_3_VOLUME 0x635 +#define WM2200_AIF1TX3MIX_INPUT_4_SOURCE 0x636 +#define WM2200_AIF1TX3MIX_INPUT_4_VOLUME 0x637 +#define WM2200_AIF1TX4MIX_INPUT_1_SOURCE 0x638 +#define WM2200_AIF1TX4MIX_INPUT_1_VOLUME 0x639 +#define WM2200_AIF1TX4MIX_INPUT_2_SOURCE 0x63A +#define WM2200_AIF1TX4MIX_INPUT_2_VOLUME 0x63B +#define WM2200_AIF1TX4MIX_INPUT_3_SOURCE 0x63C +#define WM2200_AIF1TX4MIX_INPUT_3_VOLUME 0x63D +#define WM2200_AIF1TX4MIX_INPUT_4_SOURCE 0x63E +#define WM2200_AIF1TX4MIX_INPUT_4_VOLUME 0x63F +#define WM2200_AIF1TX5MIX_INPUT_1_SOURCE 0x640 +#define WM2200_AIF1TX5MIX_INPUT_1_VOLUME 0x641 +#define WM2200_AIF1TX5MIX_INPUT_2_SOURCE 0x642 +#define WM2200_AIF1TX5MIX_INPUT_2_VOLUME 0x643 +#define WM2200_AIF1TX5MIX_INPUT_3_SOURCE 0x644 +#define WM2200_AIF1TX5MIX_INPUT_3_VOLUME 0x645 +#define WM2200_AIF1TX5MIX_INPUT_4_SOURCE 0x646 +#define WM2200_AIF1TX5MIX_INPUT_4_VOLUME 0x647 +#define WM2200_AIF1TX6MIX_INPUT_1_SOURCE 0x648 +#define WM2200_AIF1TX6MIX_INPUT_1_VOLUME 0x649 +#define WM2200_AIF1TX6MIX_INPUT_2_SOURCE 0x64A +#define WM2200_AIF1TX6MIX_INPUT_2_VOLUME 0x64B +#define WM2200_AIF1TX6MIX_INPUT_3_SOURCE 0x64C +#define WM2200_AIF1TX6MIX_INPUT_3_VOLUME 0x64D +#define WM2200_AIF1TX6MIX_INPUT_4_SOURCE 0x64E +#define WM2200_AIF1TX6MIX_INPUT_4_VOLUME 0x64F +#define WM2200_EQLMIX_INPUT_1_SOURCE 0x650 +#define WM2200_EQLMIX_INPUT_1_VOLUME 0x651 +#define WM2200_EQLMIX_INPUT_2_SOURCE 0x652 +#define WM2200_EQLMIX_INPUT_2_VOLUME 0x653 +#define WM2200_EQLMIX_INPUT_3_SOURCE 0x654 +#define WM2200_EQLMIX_INPUT_3_VOLUME 0x655 +#define WM2200_EQLMIX_INPUT_4_SOURCE 0x656 +#define WM2200_EQLMIX_INPUT_4_VOLUME 0x657 +#define WM2200_EQRMIX_INPUT_1_SOURCE 0x658 +#define WM2200_EQRMIX_INPUT_1_VOLUME 0x659 +#define WM2200_EQRMIX_INPUT_2_SOURCE 0x65A +#define WM2200_EQRMIX_INPUT_2_VOLUME 0x65B +#define WM2200_EQRMIX_INPUT_3_SOURCE 0x65C +#define WM2200_EQRMIX_INPUT_3_VOLUME 0x65D +#define WM2200_EQRMIX_INPUT_4_SOURCE 0x65E +#define WM2200_EQRMIX_INPUT_4_VOLUME 0x65F +#define WM2200_LHPF1MIX_INPUT_1_SOURCE 0x660 +#define WM2200_LHPF1MIX_INPUT_1_VOLUME 0x661 +#define WM2200_LHPF1MIX_INPUT_2_SOURCE 0x662 +#define WM2200_LHPF1MIX_INPUT_2_VOLUME 0x663 +#define WM2200_LHPF1MIX_INPUT_3_SOURCE 0x664 +#define WM2200_LHPF1MIX_INPUT_3_VOLUME 0x665 +#define WM2200_LHPF1MIX_INPUT_4_SOURCE 0x666 +#define WM2200_LHPF1MIX_INPUT_4_VOLUME 0x667 +#define WM2200_LHPF2MIX_INPUT_1_SOURCE 0x668 +#define WM2200_LHPF2MIX_INPUT_1_VOLUME 0x669 +#define WM2200_LHPF2MIX_INPUT_2_SOURCE 0x66A +#define WM2200_LHPF2MIX_INPUT_2_VOLUME 0x66B +#define WM2200_LHPF2MIX_INPUT_3_SOURCE 0x66C +#define WM2200_LHPF2MIX_INPUT_3_VOLUME 0x66D +#define WM2200_LHPF2MIX_INPUT_4_SOURCE 0x66E +#define WM2200_LHPF2MIX_INPUT_4_VOLUME 0x66F +#define WM2200_DSP1LMIX_INPUT_1_SOURCE 0x670 +#define WM2200_DSP1LMIX_INPUT_1_VOLUME 0x671 +#define WM2200_DSP1LMIX_INPUT_2_SOURCE 0x672 +#define WM2200_DSP1LMIX_INPUT_2_VOLUME 0x673 +#define WM2200_DSP1LMIX_INPUT_3_SOURCE 0x674 +#define WM2200_DSP1LMIX_INPUT_3_VOLUME 0x675 +#define WM2200_DSP1LMIX_INPUT_4_SOURCE 0x676 +#define WM2200_DSP1LMIX_INPUT_4_VOLUME 0x677 +#define WM2200_DSP1RMIX_INPUT_1_SOURCE 0x678 +#define WM2200_DSP1RMIX_INPUT_1_VOLUME 0x679 +#define WM2200_DSP1RMIX_INPUT_2_SOURCE 0x67A +#define WM2200_DSP1RMIX_INPUT_2_VOLUME 0x67B +#define WM2200_DSP1RMIX_INPUT_3_SOURCE 0x67C +#define WM2200_DSP1RMIX_INPUT_3_VOLUME 0x67D +#define WM2200_DSP1RMIX_INPUT_4_SOURCE 0x67E +#define WM2200_DSP1RMIX_INPUT_4_VOLUME 0x67F +#define WM2200_DSP1AUX1MIX_INPUT_1_SOURCE 0x680 +#define WM2200_DSP1AUX2MIX_INPUT_1_SOURCE 0x681 +#define WM2200_DSP1AUX3MIX_INPUT_1_SOURCE 0x682 +#define WM2200_DSP1AUX4MIX_INPUT_1_SOURCE 0x683 +#define WM2200_DSP1AUX5MIX_INPUT_1_SOURCE 0x684 +#define WM2200_DSP1AUX6MIX_INPUT_1_SOURCE 0x685 +#define WM2200_DSP2LMIX_INPUT_1_SOURCE 0x686 +#define WM2200_DSP2LMIX_INPUT_1_VOLUME 0x687 +#define WM2200_DSP2LMIX_INPUT_2_SOURCE 0x688 +#define WM2200_DSP2LMIX_INPUT_2_VOLUME 0x689 +#define WM2200_DSP2LMIX_INPUT_3_SOURCE 0x68A +#define WM2200_DSP2LMIX_INPUT_3_VOLUME 0x68B +#define WM2200_DSP2LMIX_INPUT_4_SOURCE 0x68C +#define WM2200_DSP2LMIX_INPUT_4_VOLUME 0x68D +#define WM2200_DSP2RMIX_INPUT_1_SOURCE 0x68E +#define WM2200_DSP2RMIX_INPUT_1_VOLUME 0x68F +#define WM2200_DSP2RMIX_INPUT_2_SOURCE 0x690 +#define WM2200_DSP2RMIX_INPUT_2_VOLUME 0x691 +#define WM2200_DSP2RMIX_INPUT_3_SOURCE 0x692 +#define WM2200_DSP2RMIX_INPUT_3_VOLUME 0x693 +#define WM2200_DSP2RMIX_INPUT_4_SOURCE 0x694 +#define WM2200_DSP2RMIX_INPUT_4_VOLUME 0x695 +#define WM2200_DSP2AUX1MIX_INPUT_1_SOURCE 0x696 +#define WM2200_DSP2AUX2MIX_INPUT_1_SOURCE 0x697 +#define WM2200_DSP2AUX3MIX_INPUT_1_SOURCE 0x698 +#define WM2200_DSP2AUX4MIX_INPUT_1_SOURCE 0x699 +#define WM2200_DSP2AUX5MIX_INPUT_1_SOURCE 0x69A +#define WM2200_DSP2AUX6MIX_INPUT_1_SOURCE 0x69B +#define WM2200_GPIO_CTRL_1 0x700 +#define WM2200_GPIO_CTRL_2 0x701 +#define WM2200_GPIO_CTRL_3 0x702 +#define WM2200_GPIO_CTRL_4 0x703 +#define WM2200_ADPS1_IRQ0 0x707 +#define WM2200_ADPS1_IRQ1 0x708 +#define WM2200_MISC_PAD_CTRL_1 0x709 +#define WM2200_INTERRUPT_STATUS_1 0x800 +#define WM2200_INTERRUPT_STATUS_1_MASK 0x801 +#define WM2200_INTERRUPT_STATUS_2 0x802 +#define WM2200_INTERRUPT_RAW_STATUS_2 0x803 +#define WM2200_INTERRUPT_STATUS_2_MASK 0x804 +#define WM2200_INTERRUPT_CONTROL 0x808 +#define WM2200_EQL_1 0x900 +#define WM2200_EQL_2 0x901 +#define WM2200_EQL_3 0x902 +#define WM2200_EQL_4 0x903 +#define WM2200_EQL_5 0x904 +#define WM2200_EQL_6 0x905 +#define WM2200_EQL_7 0x906 +#define WM2200_EQL_8 0x907 +#define WM2200_EQL_9 0x908 +#define WM2200_EQL_10 0x909 +#define WM2200_EQL_11 0x90A +#define WM2200_EQL_12 0x90B +#define WM2200_EQL_13 0x90C +#define WM2200_EQL_14 0x90D +#define WM2200_EQL_15 0x90E +#define WM2200_EQL_16 0x90F +#define WM2200_EQL_17 0x910 +#define WM2200_EQL_18 0x911 +#define WM2200_EQL_19 0x912 +#define WM2200_EQL_20 0x913 +#define WM2200_EQR_1 0x916 +#define WM2200_EQR_2 0x917 +#define WM2200_EQR_3 0x918 +#define WM2200_EQR_4 0x919 +#define WM2200_EQR_5 0x91A +#define WM2200_EQR_6 0x91B +#define WM2200_EQR_7 0x91C +#define WM2200_EQR_8 0x91D +#define WM2200_EQR_9 0x91E +#define WM2200_EQR_10 0x91F +#define WM2200_EQR_11 0x920 +#define WM2200_EQR_12 0x921 +#define WM2200_EQR_13 0x922 +#define WM2200_EQR_14 0x923 +#define WM2200_EQR_15 0x924 +#define WM2200_EQR_16 0x925 +#define WM2200_EQR_17 0x926 +#define WM2200_EQR_18 0x927 +#define WM2200_EQR_19 0x928 +#define WM2200_EQR_20 0x929 +#define WM2200_HPLPF1_1 0x93E +#define WM2200_HPLPF1_2 0x93F +#define WM2200_HPLPF2_1 0x942 +#define WM2200_HPLPF2_2 0x943 +#define WM2200_DSP1_CONTROL_1 0xA00 +#define WM2200_DSP1_CONTROL_2 0xA02 +#define WM2200_DSP1_CONTROL_3 0xA03 +#define WM2200_DSP1_CONTROL_4 0xA04 +#define WM2200_DSP1_CONTROL_5 0xA06 +#define WM2200_DSP1_CONTROL_6 0xA07 +#define WM2200_DSP1_CONTROL_7 0xA08 +#define WM2200_DSP1_CONTROL_8 0xA09 +#define WM2200_DSP1_CONTROL_9 0xA0A +#define WM2200_DSP1_CONTROL_10 0xA0B +#define WM2200_DSP1_CONTROL_11 0xA0C +#define WM2200_DSP1_CONTROL_12 0xA0D +#define WM2200_DSP1_CONTROL_13 0xA0F +#define WM2200_DSP1_CONTROL_14 0xA10 +#define WM2200_DSP1_CONTROL_15 0xA11 +#define WM2200_DSP1_CONTROL_16 0xA12 +#define WM2200_DSP1_CONTROL_17 0xA13 +#define WM2200_DSP1_CONTROL_18 0xA14 +#define WM2200_DSP1_CONTROL_19 0xA16 +#define WM2200_DSP1_CONTROL_20 0xA17 +#define WM2200_DSP1_CONTROL_21 0xA18 +#define WM2200_DSP1_CONTROL_22 0xA1A +#define WM2200_DSP1_CONTROL_23 0xA1B +#define WM2200_DSP1_CONTROL_24 0xA1C +#define WM2200_DSP1_CONTROL_25 0xA1E +#define WM2200_DSP1_CONTROL_26 0xA20 +#define WM2200_DSP1_CONTROL_27 0xA21 +#define WM2200_DSP1_CONTROL_28 0xA22 +#define WM2200_DSP1_CONTROL_29 0xA23 +#define WM2200_DSP1_CONTROL_30 0xA24 +#define WM2200_DSP1_CONTROL_31 0xA26 +#define WM2200_DSP2_CONTROL_1 0xB00 +#define WM2200_DSP2_CONTROL_2 0xB02 +#define WM2200_DSP2_CONTROL_3 0xB03 +#define WM2200_DSP2_CONTROL_4 0xB04 +#define WM2200_DSP2_CONTROL_5 0xB06 +#define WM2200_DSP2_CONTROL_6 0xB07 +#define WM2200_DSP2_CONTROL_7 0xB08 +#define WM2200_DSP2_CONTROL_8 0xB09 +#define WM2200_DSP2_CONTROL_9 0xB0A +#define WM2200_DSP2_CONTROL_10 0xB0B +#define WM2200_DSP2_CONTROL_11 0xB0C +#define WM2200_DSP2_CONTROL_12 0xB0D +#define WM2200_DSP2_CONTROL_13 0xB0F +#define WM2200_DSP2_CONTROL_14 0xB10 +#define WM2200_DSP2_CONTROL_15 0xB11 +#define WM2200_DSP2_CONTROL_16 0xB12 +#define WM2200_DSP2_CONTROL_17 0xB13 +#define WM2200_DSP2_CONTROL_18 0xB14 +#define WM2200_DSP2_CONTROL_19 0xB16 +#define WM2200_DSP2_CONTROL_20 0xB17 +#define WM2200_DSP2_CONTROL_21 0xB18 +#define WM2200_DSP2_CONTROL_22 0xB1A +#define WM2200_DSP2_CONTROL_23 0xB1B +#define WM2200_DSP2_CONTROL_24 0xB1C +#define WM2200_DSP2_CONTROL_25 0xB1E +#define WM2200_DSP2_CONTROL_26 0xB20 +#define WM2200_DSP2_CONTROL_27 0xB21 +#define WM2200_DSP2_CONTROL_28 0xB22 +#define WM2200_DSP2_CONTROL_29 0xB23 +#define WM2200_DSP2_CONTROL_30 0xB24 +#define WM2200_DSP2_CONTROL_31 0xB26 +#define WM2200_ANC_CTRL1 0xD00 +#define WM2200_ANC_CTRL2 0xD01 +#define WM2200_ANC_CTRL3 0xD02 +#define WM2200_ANC_CTRL7 0xD08 +#define WM2200_ANC_CTRL8 0xD09 +#define WM2200_ANC_CTRL9 0xD0A +#define WM2200_ANC_CTRL10 0xD0B +#define WM2200_ANC_CTRL11 0xD0C +#define WM2200_ANC_CTRL12 0xD0D +#define WM2200_ANC_CTRL13 0xD0E +#define WM2200_ANC_CTRL14 0xD0F +#define WM2200_ANC_CTRL15 0xD10 +#define WM2200_ANC_CTRL16 0xD11 +#define WM2200_ANC_CTRL17 0xD12 +#define WM2200_ANC_CTRL18 0xD15 +#define WM2200_ANC_CTRL19 0xD16 +#define WM2200_ANC_CTRL20 0xD17 +#define WM2200_ANC_CTRL21 0xD18 +#define WM2200_ANC_CTRL22 0xD19 +#define WM2200_ANC_CTRL23 0xD1A +#define WM2200_ANC_CTRL24 0xD1B +#define WM2200_ANC_CTRL25 0xD1C +#define WM2200_ANC_CTRL26 0xD1D +#define WM2200_ANC_CTRL27 0xD1E +#define WM2200_ANC_CTRL28 0xD1F +#define WM2200_ANC_CTRL29 0xD20 +#define WM2200_ANC_CTRL30 0xD21 +#define WM2200_ANC_CTRL31 0xD23 +#define WM2200_ANC_CTRL32 0xD24 +#define WM2200_ANC_CTRL33 0xD25 +#define WM2200_ANC_CTRL34 0xD27 +#define WM2200_ANC_CTRL35 0xD28 +#define WM2200_ANC_CTRL36 0xD29 +#define WM2200_ANC_CTRL37 0xD2A +#define WM2200_ANC_CTRL38 0xD2B +#define WM2200_ANC_CTRL39 0xD2C +#define WM2200_ANC_CTRL40 0xD2D +#define WM2200_ANC_CTRL41 0xD2E +#define WM2200_ANC_CTRL42 0xD2F +#define WM2200_ANC_CTRL43 0xD30 +#define WM2200_ANC_CTRL44 0xD31 +#define WM2200_ANC_CTRL45 0xD32 +#define WM2200_ANC_CTRL46 0xD33 +#define WM2200_ANC_CTRL47 0xD34 +#define WM2200_ANC_CTRL48 0xD35 +#define WM2200_ANC_CTRL49 0xD36 +#define WM2200_ANC_CTRL50 0xD37 +#define WM2200_ANC_CTRL51 0xD38 +#define WM2200_ANC_CTRL52 0xD39 +#define WM2200_ANC_CTRL53 0xD3A +#define WM2200_ANC_CTRL54 0xD3B +#define WM2200_ANC_CTRL55 0xD3C +#define WM2200_ANC_CTRL56 0xD3D +#define WM2200_ANC_CTRL57 0xD3E +#define WM2200_ANC_CTRL58 0xD3F +#define WM2200_ANC_CTRL59 0xD40 +#define WM2200_ANC_CTRL60 0xD41 +#define WM2200_ANC_CTRL61 0xD42 +#define WM2200_ANC_CTRL62 0xD43 +#define WM2200_ANC_CTRL63 0xD44 +#define WM2200_ANC_CTRL64 0xD45 +#define WM2200_ANC_CTRL65 0xD46 +#define WM2200_ANC_CTRL66 0xD47 +#define WM2200_ANC_CTRL67 0xD48 +#define WM2200_ANC_CTRL68 0xD49 +#define WM2200_ANC_CTRL69 0xD4A +#define WM2200_ANC_CTRL70 0xD4B +#define WM2200_ANC_CTRL71 0xD4C +#define WM2200_ANC_CTRL72 0xD4D +#define WM2200_ANC_CTRL73 0xD4E +#define WM2200_ANC_CTRL74 0xD4F +#define WM2200_ANC_CTRL75 0xD50 +#define WM2200_ANC_CTRL76 0xD51 +#define WM2200_ANC_CTRL77 0xD52 +#define WM2200_ANC_CTRL78 0xD53 +#define WM2200_ANC_CTRL79 0xD54 +#define WM2200_ANC_CTRL80 0xD55 +#define WM2200_ANC_CTRL81 0xD56 +#define WM2200_ANC_CTRL82 0xD57 +#define WM2200_ANC_CTRL83 0xD58 +#define WM2200_ANC_CTRL84 0xD5B +#define WM2200_ANC_CTRL85 0xD5C +#define WM2200_ANC_CTRL86 0xD5F +#define WM2200_ANC_CTRL87 0xD60 +#define WM2200_ANC_CTRL88 0xD61 +#define WM2200_ANC_CTRL89 0xD62 +#define WM2200_ANC_CTRL90 0xD63 +#define WM2200_ANC_CTRL91 0xD64 +#define WM2200_ANC_CTRL92 0xD65 +#define WM2200_ANC_CTRL93 0xD66 +#define WM2200_ANC_CTRL94 0xD67 +#define WM2200_ANC_CTRL95 0xD68 +#define WM2200_ANC_CTRL96 0xD69 +#define WM2200_DSP1_DM_0 0x3000 +#define WM2200_DSP1_DM_1 0x3001 +#define WM2200_DSP1_DM_2 0x3002 +#define WM2200_DSP1_DM_3 0x3003 +#define WM2200_DSP1_DM_2044 0x37FC +#define WM2200_DSP1_DM_2045 0x37FD +#define WM2200_DSP1_DM_2046 0x37FE +#define WM2200_DSP1_DM_2047 0x37FF +#define WM2200_DSP1_PM_0 0x3800 +#define WM2200_DSP1_PM_1 0x3801 +#define WM2200_DSP1_PM_2 0x3802 +#define WM2200_DSP1_PM_3 0x3803 +#define WM2200_DSP1_PM_4 0x3804 +#define WM2200_DSP1_PM_5 0x3805 +#define WM2200_DSP1_PM_762 0x3AFA +#define WM2200_DSP1_PM_763 0x3AFB +#define WM2200_DSP1_PM_764 0x3AFC +#define WM2200_DSP1_PM_765 0x3AFD +#define WM2200_DSP1_PM_766 0x3AFE +#define WM2200_DSP1_PM_767 0x3AFF +#define WM2200_DSP1_ZM_0 0x3C00 +#define WM2200_DSP1_ZM_1 0x3C01 +#define WM2200_DSP1_ZM_2 0x3C02 +#define WM2200_DSP1_ZM_3 0x3C03 +#define WM2200_DSP1_ZM_1020 0x3FFC +#define WM2200_DSP1_ZM_1021 0x3FFD +#define WM2200_DSP1_ZM_1022 0x3FFE +#define WM2200_DSP1_ZM_1023 0x3FFF +#define WM2200_DSP2_DM_0 0x4000 +#define WM2200_DSP2_DM_1 0x4001 +#define WM2200_DSP2_DM_2 0x4002 +#define WM2200_DSP2_DM_3 0x4003 +#define WM2200_DSP2_DM_2044 0x47FC +#define WM2200_DSP2_DM_2045 0x47FD +#define WM2200_DSP2_DM_2046 0x47FE +#define WM2200_DSP2_DM_2047 0x47FF +#define WM2200_DSP2_PM_0 0x4800 +#define WM2200_DSP2_PM_1 0x4801 +#define WM2200_DSP2_PM_2 0x4802 +#define WM2200_DSP2_PM_3 0x4803 +#define WM2200_DSP2_PM_4 0x4804 +#define WM2200_DSP2_PM_5 0x4805 +#define WM2200_DSP2_PM_762 0x4AFA +#define WM2200_DSP2_PM_763 0x4AFB +#define WM2200_DSP2_PM_764 0x4AFC +#define WM2200_DSP2_PM_765 0x4AFD +#define WM2200_DSP2_PM_766 0x4AFE +#define WM2200_DSP2_PM_767 0x4AFF +#define WM2200_DSP2_ZM_0 0x4C00 +#define WM2200_DSP2_ZM_1 0x4C01 +#define WM2200_DSP2_ZM_2 0x4C02 +#define WM2200_DSP2_ZM_3 0x4C03 +#define WM2200_DSP2_ZM_1020 0x4FFC +#define WM2200_DSP2_ZM_1021 0x4FFD +#define WM2200_DSP2_ZM_1022 0x4FFE +#define WM2200_DSP2_ZM_1023 0x4FFF + +#define WM2200_REGISTER_COUNT 494 +#define WM2200_MAX_REGISTER 0x4FFF + +/* + * Field Definitions. + */ + +/* + * R0 (0x00) - software reset + */ +#define WM2200_SW_RESET_CHIP_ID1_MASK 0xFFFF /* SW_RESET_CHIP_ID1 - [15:0] */ +#define WM2200_SW_RESET_CHIP_ID1_SHIFT 0 /* SW_RESET_CHIP_ID1 - [15:0] */ +#define WM2200_SW_RESET_CHIP_ID1_WIDTH 16 /* SW_RESET_CHIP_ID1 - [15:0] */ + +/* + * R1 (0x01) - Device Revision + */ +#define WM2200_DEVICE_REVISION_MASK 0x000F /* DEVICE_REVISION - [3:0] */ +#define WM2200_DEVICE_REVISION_SHIFT 0 /* DEVICE_REVISION - [3:0] */ +#define WM2200_DEVICE_REVISION_WIDTH 4 /* DEVICE_REVISION - [3:0] */ + +/* + * R11 (0x0B) - Tone Generator 1 + */ +#define WM2200_TONE_ENA 0x0001 /* TONE_ENA */ +#define WM2200_TONE_ENA_MASK 0x0001 /* TONE_ENA */ +#define WM2200_TONE_ENA_SHIFT 0 /* TONE_ENA */ +#define WM2200_TONE_ENA_WIDTH 1 /* TONE_ENA */ + +/* + * R258 (0x102) - Clocking 3 + */ +#define WM2200_SYSCLK_FREQ_MASK 0x0700 /* SYSCLK_FREQ - [10:8] */ +#define WM2200_SYSCLK_FREQ_SHIFT 8 /* SYSCLK_FREQ - [10:8] */ +#define WM2200_SYSCLK_FREQ_WIDTH 3 /* SYSCLK_FREQ - [10:8] */ +#define WM2200_SYSCLK_ENA 0x0040 /* SYSCLK_ENA */ +#define WM2200_SYSCLK_ENA_MASK 0x0040 /* SYSCLK_ENA */ +#define WM2200_SYSCLK_ENA_SHIFT 6 /* SYSCLK_ENA */ +#define WM2200_SYSCLK_ENA_WIDTH 1 /* SYSCLK_ENA */ +#define WM2200_SYSCLK_SRC_MASK 0x000F /* SYSCLK_SRC - [3:0] */ +#define WM2200_SYSCLK_SRC_SHIFT 0 /* SYSCLK_SRC - [3:0] */ +#define WM2200_SYSCLK_SRC_WIDTH 4 /* SYSCLK_SRC - [3:0] */ + +/* + * R259 (0x103) - Clocking 4 + */ +#define WM2200_SAMPLE_RATE_1_MASK 0x001F /* SAMPLE_RATE_1 - [4:0] */ +#define WM2200_SAMPLE_RATE_1_SHIFT 0 /* SAMPLE_RATE_1 - [4:0] */ +#define WM2200_SAMPLE_RATE_1_WIDTH 5 /* SAMPLE_RATE_1 - [4:0] */ + +/* + * R273 (0x111) - FLL Control 1 + */ +#define WM2200_FLL_ENA 0x0001 /* FLL_ENA */ +#define WM2200_FLL_ENA_MASK 0x0001 /* FLL_ENA */ +#define WM2200_FLL_ENA_SHIFT 0 /* FLL_ENA */ +#define WM2200_FLL_ENA_WIDTH 1 /* FLL_ENA */ + +/* + * R274 (0x112) - FLL Control 2 + */ +#define WM2200_FLL_OUTDIV_MASK 0x3F00 /* FLL_OUTDIV - [13:8] */ +#define WM2200_FLL_OUTDIV_SHIFT 8 /* FLL_OUTDIV - [13:8] */ +#define WM2200_FLL_OUTDIV_WIDTH 6 /* FLL_OUTDIV - [13:8] */ +#define WM2200_FLL_FRATIO_MASK 0x0007 /* FLL_FRATIO - [2:0] */ +#define WM2200_FLL_FRATIO_SHIFT 0 /* FLL_FRATIO - [2:0] */ +#define WM2200_FLL_FRATIO_WIDTH 3 /* FLL_FRATIO - [2:0] */ + +/* + * R275 (0x113) - FLL Control 3 + */ +#define WM2200_FLL_FRACN_ENA 0x0001 /* FLL_FRACN_ENA */ +#define WM2200_FLL_FRACN_ENA_MASK 0x0001 /* FLL_FRACN_ENA */ +#define WM2200_FLL_FRACN_ENA_SHIFT 0 /* FLL_FRACN_ENA */ +#define WM2200_FLL_FRACN_ENA_WIDTH 1 /* FLL_FRACN_ENA */ + +/* + * R276 (0x114) - FLL Control 4 + */ +#define WM2200_FLL_THETA_MASK 0xFFFF /* FLL_THETA - [15:0] */ +#define WM2200_FLL_THETA_SHIFT 0 /* FLL_THETA - [15:0] */ +#define WM2200_FLL_THETA_WIDTH 16 /* FLL_THETA - [15:0] */ + +/* + * R278 (0x116) - FLL Control 6 + */ +#define WM2200_FLL_N_MASK 0x03FF /* FLL_N - [9:0] */ +#define WM2200_FLL_N_SHIFT 0 /* FLL_N - [9:0] */ +#define WM2200_FLL_N_WIDTH 10 /* FLL_N - [9:0] */ + +/* + * R279 (0x117) - FLL Control 7 + */ +#define WM2200_FLL_CLK_REF_DIV_MASK 0x0030 /* FLL_CLK_REF_DIV - [5:4] */ +#define WM2200_FLL_CLK_REF_DIV_SHIFT 4 /* FLL_CLK_REF_DIV - [5:4] */ +#define WM2200_FLL_CLK_REF_DIV_WIDTH 2 /* FLL_CLK_REF_DIV - [5:4] */ +#define WM2200_FLL_CLK_REF_SRC_MASK 0x0003 /* FLL_CLK_REF_SRC - [1:0] */ +#define WM2200_FLL_CLK_REF_SRC_SHIFT 0 /* FLL_CLK_REF_SRC - [1:0] */ +#define WM2200_FLL_CLK_REF_SRC_WIDTH 2 /* FLL_CLK_REF_SRC - [1:0] */ + +/* + * R281 (0x119) - FLL EFS 1 + */ +#define WM2200_FLL_LAMBDA_MASK 0xFFFF /* FLL_LAMBDA - [15:0] */ +#define WM2200_FLL_LAMBDA_SHIFT 0 /* FLL_LAMBDA - [15:0] */ +#define WM2200_FLL_LAMBDA_WIDTH 16 /* FLL_LAMBDA - [15:0] */ + +/* + * R282 (0x11A) - FLL EFS 2 + */ +#define WM2200_FLL_EFS_ENA 0x0001 /* FLL_EFS_ENA */ +#define WM2200_FLL_EFS_ENA_MASK 0x0001 /* FLL_EFS_ENA */ +#define WM2200_FLL_EFS_ENA_SHIFT 0 /* FLL_EFS_ENA */ +#define WM2200_FLL_EFS_ENA_WIDTH 1 /* FLL_EFS_ENA */ + +/* + * R512 (0x200) - Mic Charge Pump 1 + */ +#define WM2200_CPMIC_BYPASS_MODE 0x0020 /* CPMIC_BYPASS_MODE */ +#define WM2200_CPMIC_BYPASS_MODE_MASK 0x0020 /* CPMIC_BYPASS_MODE */ +#define WM2200_CPMIC_BYPASS_MODE_SHIFT 5 /* CPMIC_BYPASS_MODE */ +#define WM2200_CPMIC_BYPASS_MODE_WIDTH 1 /* CPMIC_BYPASS_MODE */ +#define WM2200_CPMIC_ENA 0x0001 /* CPMIC_ENA */ +#define WM2200_CPMIC_ENA_MASK 0x0001 /* CPMIC_ENA */ +#define WM2200_CPMIC_ENA_SHIFT 0 /* CPMIC_ENA */ +#define WM2200_CPMIC_ENA_WIDTH 1 /* CPMIC_ENA */ + +/* + * R513 (0x201) - Mic Charge Pump 2 + */ +#define WM2200_CPMIC_LDO_VSEL_OVERRIDE_MASK 0xF800 /* CPMIC_LDO_VSEL_OVERRIDE - [15:11] */ +#define WM2200_CPMIC_LDO_VSEL_OVERRIDE_SHIFT 11 /* CPMIC_LDO_VSEL_OVERRIDE - [15:11] */ +#define WM2200_CPMIC_LDO_VSEL_OVERRIDE_WIDTH 5 /* CPMIC_LDO_VSEL_OVERRIDE - [15:11] */ + +/* + * R514 (0x202) - DM Charge Pump 1 + */ +#define WM2200_CPDM_ENA 0x0001 /* CPDM_ENA */ +#define WM2200_CPDM_ENA_MASK 0x0001 /* CPDM_ENA */ +#define WM2200_CPDM_ENA_SHIFT 0 /* CPDM_ENA */ +#define WM2200_CPDM_ENA_WIDTH 1 /* CPDM_ENA */ + +/* + * R524 (0x20C) - Mic Bias Ctrl 1 + */ +#define WM2200_MICB1_DISCH 0x0040 /* MICB1_DISCH */ +#define WM2200_MICB1_DISCH_MASK 0x0040 /* MICB1_DISCH */ +#define WM2200_MICB1_DISCH_SHIFT 6 /* MICB1_DISCH */ +#define WM2200_MICB1_DISCH_WIDTH 1 /* MICB1_DISCH */ +#define WM2200_MICB1_RATE 0x0020 /* MICB1_RATE */ +#define WM2200_MICB1_RATE_MASK 0x0020 /* MICB1_RATE */ +#define WM2200_MICB1_RATE_SHIFT 5 /* MICB1_RATE */ +#define WM2200_MICB1_RATE_WIDTH 1 /* MICB1_RATE */ +#define WM2200_MICB1_LVL_MASK 0x001C /* MICB1_LVL - [4:2] */ +#define WM2200_MICB1_LVL_SHIFT 2 /* MICB1_LVL - [4:2] */ +#define WM2200_MICB1_LVL_WIDTH 3 /* MICB1_LVL - [4:2] */ +#define WM2200_MICB1_MODE 0x0002 /* MICB1_MODE */ +#define WM2200_MICB1_MODE_MASK 0x0002 /* MICB1_MODE */ +#define WM2200_MICB1_MODE_SHIFT 1 /* MICB1_MODE */ +#define WM2200_MICB1_MODE_WIDTH 1 /* MICB1_MODE */ +#define WM2200_MICB1_ENA 0x0001 /* MICB1_ENA */ +#define WM2200_MICB1_ENA_MASK 0x0001 /* MICB1_ENA */ +#define WM2200_MICB1_ENA_SHIFT 0 /* MICB1_ENA */ +#define WM2200_MICB1_ENA_WIDTH 1 /* MICB1_ENA */ + +/* + * R525 (0x20D) - Mic Bias Ctrl 2 + */ +#define WM2200_MICB2_DISCH 0x0040 /* MICB2_DISCH */ +#define WM2200_MICB2_DISCH_MASK 0x0040 /* MICB2_DISCH */ +#define WM2200_MICB2_DISCH_SHIFT 6 /* MICB2_DISCH */ +#define WM2200_MICB2_DISCH_WIDTH 1 /* MICB2_DISCH */ +#define WM2200_MICB2_RATE 0x0020 /* MICB2_RATE */ +#define WM2200_MICB2_RATE_MASK 0x0020 /* MICB2_RATE */ +#define WM2200_MICB2_RATE_SHIFT 5 /* MICB2_RATE */ +#define WM2200_MICB2_RATE_WIDTH 1 /* MICB2_RATE */ +#define WM2200_MICB2_LVL_MASK 0x001C /* MICB2_LVL - [4:2] */ +#define WM2200_MICB2_LVL_SHIFT 2 /* MICB2_LVL - [4:2] */ +#define WM2200_MICB2_LVL_WIDTH 3 /* MICB2_LVL - [4:2] */ +#define WM2200_MICB2_MODE 0x0002 /* MICB2_MODE */ +#define WM2200_MICB2_MODE_MASK 0x0002 /* MICB2_MODE */ +#define WM2200_MICB2_MODE_SHIFT 1 /* MICB2_MODE */ +#define WM2200_MICB2_MODE_WIDTH 1 /* MICB2_MODE */ +#define WM2200_MICB2_ENA 0x0001 /* MICB2_ENA */ +#define WM2200_MICB2_ENA_MASK 0x0001 /* MICB2_ENA */ +#define WM2200_MICB2_ENA_SHIFT 0 /* MICB2_ENA */ +#define WM2200_MICB2_ENA_WIDTH 1 /* MICB2_ENA */ + +/* + * R527 (0x20F) - Ear Piece Ctrl 1 + */ +#define WM2200_EPD_LP_ENA 0x4000 /* EPD_LP_ENA */ +#define WM2200_EPD_LP_ENA_MASK 0x4000 /* EPD_LP_ENA */ +#define WM2200_EPD_LP_ENA_SHIFT 14 /* EPD_LP_ENA */ +#define WM2200_EPD_LP_ENA_WIDTH 1 /* EPD_LP_ENA */ +#define WM2200_EPD_OUTP_LP_ENA 0x2000 /* EPD_OUTP_LP_ENA */ +#define WM2200_EPD_OUTP_LP_ENA_MASK 0x2000 /* EPD_OUTP_LP_ENA */ +#define WM2200_EPD_OUTP_LP_ENA_SHIFT 13 /* EPD_OUTP_LP_ENA */ +#define WM2200_EPD_OUTP_LP_ENA_WIDTH 1 /* EPD_OUTP_LP_ENA */ +#define WM2200_EPD_RMV_SHRT_LP 0x1000 /* EPD_RMV_SHRT_LP */ +#define WM2200_EPD_RMV_SHRT_LP_MASK 0x1000 /* EPD_RMV_SHRT_LP */ +#define WM2200_EPD_RMV_SHRT_LP_SHIFT 12 /* EPD_RMV_SHRT_LP */ +#define WM2200_EPD_RMV_SHRT_LP_WIDTH 1 /* EPD_RMV_SHRT_LP */ +#define WM2200_EPD_LN_ENA 0x0800 /* EPD_LN_ENA */ +#define WM2200_EPD_LN_ENA_MASK 0x0800 /* EPD_LN_ENA */ +#define WM2200_EPD_LN_ENA_SHIFT 11 /* EPD_LN_ENA */ +#define WM2200_EPD_LN_ENA_WIDTH 1 /* EPD_LN_ENA */ +#define WM2200_EPD_OUTP_LN_ENA 0x0400 /* EPD_OUTP_LN_ENA */ +#define WM2200_EPD_OUTP_LN_ENA_MASK 0x0400 /* EPD_OUTP_LN_ENA */ +#define WM2200_EPD_OUTP_LN_ENA_SHIFT 10 /* EPD_OUTP_LN_ENA */ +#define WM2200_EPD_OUTP_LN_ENA_WIDTH 1 /* EPD_OUTP_LN_ENA */ +#define WM2200_EPD_RMV_SHRT_LN 0x0200 /* EPD_RMV_SHRT_LN */ +#define WM2200_EPD_RMV_SHRT_LN_MASK 0x0200 /* EPD_RMV_SHRT_LN */ +#define WM2200_EPD_RMV_SHRT_LN_SHIFT 9 /* EPD_RMV_SHRT_LN */ +#define WM2200_EPD_RMV_SHRT_LN_WIDTH 1 /* EPD_RMV_SHRT_LN */ + +/* + * R528 (0x210) - Ear Piece Ctrl 2 + */ +#define WM2200_EPD_RP_ENA 0x4000 /* EPD_RP_ENA */ +#define WM2200_EPD_RP_ENA_MASK 0x4000 /* EPD_RP_ENA */ +#define WM2200_EPD_RP_ENA_SHIFT 14 /* EPD_RP_ENA */ +#define WM2200_EPD_RP_ENA_WIDTH 1 /* EPD_RP_ENA */ +#define WM2200_EPD_OUTP_RP_ENA 0x2000 /* EPD_OUTP_RP_ENA */ +#define WM2200_EPD_OUTP_RP_ENA_MASK 0x2000 /* EPD_OUTP_RP_ENA */ +#define WM2200_EPD_OUTP_RP_ENA_SHIFT 13 /* EPD_OUTP_RP_ENA */ +#define WM2200_EPD_OUTP_RP_ENA_WIDTH 1 /* EPD_OUTP_RP_ENA */ +#define WM2200_EPD_RMV_SHRT_RP 0x1000 /* EPD_RMV_SHRT_RP */ +#define WM2200_EPD_RMV_SHRT_RP_MASK 0x1000 /* EPD_RMV_SHRT_RP */ +#define WM2200_EPD_RMV_SHRT_RP_SHIFT 12 /* EPD_RMV_SHRT_RP */ +#define WM2200_EPD_RMV_SHRT_RP_WIDTH 1 /* EPD_RMV_SHRT_RP */ +#define WM2200_EPD_RN_ENA 0x0800 /* EPD_RN_ENA */ +#define WM2200_EPD_RN_ENA_MASK 0x0800 /* EPD_RN_ENA */ +#define WM2200_EPD_RN_ENA_SHIFT 11 /* EPD_RN_ENA */ +#define WM2200_EPD_RN_ENA_WIDTH 1 /* EPD_RN_ENA */ +#define WM2200_EPD_OUTP_RN_ENA 0x0400 /* EPD_OUTP_RN_ENA */ +#define WM2200_EPD_OUTP_RN_ENA_MASK 0x0400 /* EPD_OUTP_RN_ENA */ +#define WM2200_EPD_OUTP_RN_ENA_SHIFT 10 /* EPD_OUTP_RN_ENA */ +#define WM2200_EPD_OUTP_RN_ENA_WIDTH 1 /* EPD_OUTP_RN_ENA */ +#define WM2200_EPD_RMV_SHRT_RN 0x0200 /* EPD_RMV_SHRT_RN */ +#define WM2200_EPD_RMV_SHRT_RN_MASK 0x0200 /* EPD_RMV_SHRT_RN */ +#define WM2200_EPD_RMV_SHRT_RN_SHIFT 9 /* EPD_RMV_SHRT_RN */ +#define WM2200_EPD_RMV_SHRT_RN_WIDTH 1 /* EPD_RMV_SHRT_RN */ + +/* + * R769 (0x301) - Input Enables + */ +#define WM2200_IN3L_ENA 0x0020 /* IN3L_ENA */ +#define WM2200_IN3L_ENA_MASK 0x0020 /* IN3L_ENA */ +#define WM2200_IN3L_ENA_SHIFT 5 /* IN3L_ENA */ +#define WM2200_IN3L_ENA_WIDTH 1 /* IN3L_ENA */ +#define WM2200_IN3R_ENA 0x0010 /* IN3R_ENA */ +#define WM2200_IN3R_ENA_MASK 0x0010 /* IN3R_ENA */ +#define WM2200_IN3R_ENA_SHIFT 4 /* IN3R_ENA */ +#define WM2200_IN3R_ENA_WIDTH 1 /* IN3R_ENA */ +#define WM2200_IN2L_ENA 0x0008 /* IN2L_ENA */ +#define WM2200_IN2L_ENA_MASK 0x0008 /* IN2L_ENA */ +#define WM2200_IN2L_ENA_SHIFT 3 /* IN2L_ENA */ +#define WM2200_IN2L_ENA_WIDTH 1 /* IN2L_ENA */ +#define WM2200_IN2R_ENA 0x0004 /* IN2R_ENA */ +#define WM2200_IN2R_ENA_MASK 0x0004 /* IN2R_ENA */ +#define WM2200_IN2R_ENA_SHIFT 2 /* IN2R_ENA */ +#define WM2200_IN2R_ENA_WIDTH 1 /* IN2R_ENA */ +#define WM2200_IN1L_ENA 0x0002 /* IN1L_ENA */ +#define WM2200_IN1L_ENA_MASK 0x0002 /* IN1L_ENA */ +#define WM2200_IN1L_ENA_SHIFT 1 /* IN1L_ENA */ +#define WM2200_IN1L_ENA_WIDTH 1 /* IN1L_ENA */ +#define WM2200_IN1R_ENA 0x0001 /* IN1R_ENA */ +#define WM2200_IN1R_ENA_MASK 0x0001 /* IN1R_ENA */ +#define WM2200_IN1R_ENA_SHIFT 0 /* IN1R_ENA */ +#define WM2200_IN1R_ENA_WIDTH 1 /* IN1R_ENA */ + +/* + * R770 (0x302) - IN1L Control + */ +#define WM2200_IN1_OSR 0x2000 /* IN1_OSR */ +#define WM2200_IN1_OSR_MASK 0x2000 /* IN1_OSR */ +#define WM2200_IN1_OSR_SHIFT 13 /* IN1_OSR */ +#define WM2200_IN1_OSR_WIDTH 1 /* IN1_OSR */ +#define WM2200_IN1_DMIC_SUP_MASK 0x1800 /* IN1_DMIC_SUP - [12:11] */ +#define WM2200_IN1_DMIC_SUP_SHIFT 11 /* IN1_DMIC_SUP - [12:11] */ +#define WM2200_IN1_DMIC_SUP_WIDTH 2 /* IN1_DMIC_SUP - [12:11] */ +#define WM2200_IN1_MODE_MASK 0x0600 /* IN1_MODE - [10:9] */ +#define WM2200_IN1_MODE_SHIFT 9 /* IN1_MODE - [10:9] */ +#define WM2200_IN1_MODE_WIDTH 2 /* IN1_MODE - [10:9] */ +#define WM2200_IN1L_PGA_VOL_MASK 0x00FE /* IN1L_PGA_VOL - [7:1] */ +#define WM2200_IN1L_PGA_VOL_SHIFT 1 /* IN1L_PGA_VOL - [7:1] */ +#define WM2200_IN1L_PGA_VOL_WIDTH 7 /* IN1L_PGA_VOL - [7:1] */ + +/* + * R771 (0x303) - IN1R Control + */ +#define WM2200_IN1R_PGA_VOL_MASK 0x00FE /* IN1R_PGA_VOL - [7:1] */ +#define WM2200_IN1R_PGA_VOL_SHIFT 1 /* IN1R_PGA_VOL - [7:1] */ +#define WM2200_IN1R_PGA_VOL_WIDTH 7 /* IN1R_PGA_VOL - [7:1] */ + +/* + * R772 (0x304) - IN2L Control + */ +#define WM2200_IN2_OSR 0x2000 /* IN2_OSR */ +#define WM2200_IN2_OSR_MASK 0x2000 /* IN2_OSR */ +#define WM2200_IN2_OSR_SHIFT 13 /* IN2_OSR */ +#define WM2200_IN2_OSR_WIDTH 1 /* IN2_OSR */ +#define WM2200_IN2_DMIC_SUP_MASK 0x1800 /* IN2_DMIC_SUP - [12:11] */ +#define WM2200_IN2_DMIC_SUP_SHIFT 11 /* IN2_DMIC_SUP - [12:11] */ +#define WM2200_IN2_DMIC_SUP_WIDTH 2 /* IN2_DMIC_SUP - [12:11] */ +#define WM2200_IN2_MODE_MASK 0x0600 /* IN2_MODE - [10:9] */ +#define WM2200_IN2_MODE_SHIFT 9 /* IN2_MODE - [10:9] */ +#define WM2200_IN2_MODE_WIDTH 2 /* IN2_MODE - [10:9] */ +#define WM2200_IN2L_PGA_VOL_MASK 0x00FE /* IN2L_PGA_VOL - [7:1] */ +#define WM2200_IN2L_PGA_VOL_SHIFT 1 /* IN2L_PGA_VOL - [7:1] */ +#define WM2200_IN2L_PGA_VOL_WIDTH 7 /* IN2L_PGA_VOL - [7:1] */ + +/* + * R773 (0x305) - IN2R Control + */ +#define WM2200_IN2R_PGA_VOL_MASK 0x00FE /* IN2R_PGA_VOL - [7:1] */ +#define WM2200_IN2R_PGA_VOL_SHIFT 1 /* IN2R_PGA_VOL - [7:1] */ +#define WM2200_IN2R_PGA_VOL_WIDTH 7 /* IN2R_PGA_VOL - [7:1] */ + +/* + * R774 (0x306) - IN3L Control + */ +#define WM2200_IN3_OSR 0x2000 /* IN3_OSR */ +#define WM2200_IN3_OSR_MASK 0x2000 /* IN3_OSR */ +#define WM2200_IN3_OSR_SHIFT 13 /* IN3_OSR */ +#define WM2200_IN3_OSR_WIDTH 1 /* IN3_OSR */ +#define WM2200_IN3_DMIC_SUP_MASK 0x1800 /* IN3_DMIC_SUP - [12:11] */ +#define WM2200_IN3_DMIC_SUP_SHIFT 11 /* IN3_DMIC_SUP - [12:11] */ +#define WM2200_IN3_DMIC_SUP_WIDTH 2 /* IN3_DMIC_SUP - [12:11] */ +#define WM2200_IN3_MODE_MASK 0x0600 /* IN3_MODE - [10:9] */ +#define WM2200_IN3_MODE_SHIFT 9 /* IN3_MODE - [10:9] */ +#define WM2200_IN3_MODE_WIDTH 2 /* IN3_MODE - [10:9] */ +#define WM2200_IN3L_PGA_VOL_MASK 0x00FE /* IN3L_PGA_VOL - [7:1] */ +#define WM2200_IN3L_PGA_VOL_SHIFT 1 /* IN3L_PGA_VOL - [7:1] */ +#define WM2200_IN3L_PGA_VOL_WIDTH 7 /* IN3L_PGA_VOL - [7:1] */ + +/* + * R775 (0x307) - IN3R Control + */ +#define WM2200_IN3R_PGA_VOL_MASK 0x00FE /* IN3R_PGA_VOL - [7:1] */ +#define WM2200_IN3R_PGA_VOL_SHIFT 1 /* IN3R_PGA_VOL - [7:1] */ +#define WM2200_IN3R_PGA_VOL_WIDTH 7 /* IN3R_PGA_VOL - [7:1] */ + +/* + * R778 (0x30A) - RXANC_SRC + */ +#define WM2200_IN_RXANC_SEL_MASK 0x0007 /* IN_RXANC_SEL - [2:0] */ +#define WM2200_IN_RXANC_SEL_SHIFT 0 /* IN_RXANC_SEL - [2:0] */ +#define WM2200_IN_RXANC_SEL_WIDTH 3 /* IN_RXANC_SEL - [2:0] */ + +/* + * R779 (0x30B) - Input Volume Ramp + */ +#define WM2200_IN_VD_RAMP_MASK 0x0070 /* IN_VD_RAMP - [6:4] */ +#define WM2200_IN_VD_RAMP_SHIFT 4 /* IN_VD_RAMP - [6:4] */ +#define WM2200_IN_VD_RAMP_WIDTH 3 /* IN_VD_RAMP - [6:4] */ +#define WM2200_IN_VI_RAMP_MASK 0x0007 /* IN_VI_RAMP - [2:0] */ +#define WM2200_IN_VI_RAMP_SHIFT 0 /* IN_VI_RAMP - [2:0] */ +#define WM2200_IN_VI_RAMP_WIDTH 3 /* IN_VI_RAMP - [2:0] */ + +/* + * R780 (0x30C) - ADC Digital Volume 1L + */ +#define WM2200_IN_VU 0x0200 /* IN_VU */ +#define WM2200_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM2200_IN_VU_SHIFT 9 /* IN_VU */ +#define WM2200_IN_VU_WIDTH 1 /* IN_VU */ +#define WM2200_IN1L_MUTE 0x0100 /* IN1L_MUTE */ +#define WM2200_IN1L_MUTE_MASK 0x0100 /* IN1L_MUTE */ +#define WM2200_IN1L_MUTE_SHIFT 8 /* IN1L_MUTE */ +#define WM2200_IN1L_MUTE_WIDTH 1 /* IN1L_MUTE */ +#define WM2200_IN1L_DIG_VOL_MASK 0x00FF /* IN1L_DIG_VOL - [7:0] */ +#define WM2200_IN1L_DIG_VOL_SHIFT 0 /* IN1L_DIG_VOL - [7:0] */ +#define WM2200_IN1L_DIG_VOL_WIDTH 8 /* IN1L_DIG_VOL - [7:0] */ + +/* + * R781 (0x30D) - ADC Digital Volume 1R + */ +#define WM2200_IN_VU 0x0200 /* IN_VU */ +#define WM2200_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM2200_IN_VU_SHIFT 9 /* IN_VU */ +#define WM2200_IN_VU_WIDTH 1 /* IN_VU */ +#define WM2200_IN1R_MUTE 0x0100 /* IN1R_MUTE */ +#define WM2200_IN1R_MUTE_MASK 0x0100 /* IN1R_MUTE */ +#define WM2200_IN1R_MUTE_SHIFT 8 /* IN1R_MUTE */ +#define WM2200_IN1R_MUTE_WIDTH 1 /* IN1R_MUTE */ +#define WM2200_IN1R_DIG_VOL_MASK 0x00FF /* IN1R_DIG_VOL - [7:0] */ +#define WM2200_IN1R_DIG_VOL_SHIFT 0 /* IN1R_DIG_VOL - [7:0] */ +#define WM2200_IN1R_DIG_VOL_WIDTH 8 /* IN1R_DIG_VOL - [7:0] */ + +/* + * R782 (0x30E) - ADC Digital Volume 2L + */ +#define WM2200_IN_VU 0x0200 /* IN_VU */ +#define WM2200_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM2200_IN_VU_SHIFT 9 /* IN_VU */ +#define WM2200_IN_VU_WIDTH 1 /* IN_VU */ +#define WM2200_IN2L_MUTE 0x0100 /* IN2L_MUTE */ +#define WM2200_IN2L_MUTE_MASK 0x0100 /* IN2L_MUTE */ +#define WM2200_IN2L_MUTE_SHIFT 8 /* IN2L_MUTE */ +#define WM2200_IN2L_MUTE_WIDTH 1 /* IN2L_MUTE */ +#define WM2200_IN2L_DIG_VOL_MASK 0x00FF /* IN2L_DIG_VOL - [7:0] */ +#define WM2200_IN2L_DIG_VOL_SHIFT 0 /* IN2L_DIG_VOL - [7:0] */ +#define WM2200_IN2L_DIG_VOL_WIDTH 8 /* IN2L_DIG_VOL - [7:0] */ + +/* + * R783 (0x30F) - ADC Digital Volume 2R + */ +#define WM2200_IN_VU 0x0200 /* IN_VU */ +#define WM2200_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM2200_IN_VU_SHIFT 9 /* IN_VU */ +#define WM2200_IN_VU_WIDTH 1 /* IN_VU */ +#define WM2200_IN2R_MUTE 0x0100 /* IN2R_MUTE */ +#define WM2200_IN2R_MUTE_MASK 0x0100 /* IN2R_MUTE */ +#define WM2200_IN2R_MUTE_SHIFT 8 /* IN2R_MUTE */ +#define WM2200_IN2R_MUTE_WIDTH 1 /* IN2R_MUTE */ +#define WM2200_IN2R_DIG_VOL_MASK 0x00FF /* IN2R_DIG_VOL - [7:0] */ +#define WM2200_IN2R_DIG_VOL_SHIFT 0 /* IN2R_DIG_VOL - [7:0] */ +#define WM2200_IN2R_DIG_VOL_WIDTH 8 /* IN2R_DIG_VOL - [7:0] */ + +/* + * R784 (0x310) - ADC Digital Volume 3L + */ +#define WM2200_IN_VU 0x0200 /* IN_VU */ +#define WM2200_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM2200_IN_VU_SHIFT 9 /* IN_VU */ +#define WM2200_IN_VU_WIDTH 1 /* IN_VU */ +#define WM2200_IN3L_MUTE 0x0100 /* IN3L_MUTE */ +#define WM2200_IN3L_MUTE_MASK 0x0100 /* IN3L_MUTE */ +#define WM2200_IN3L_MUTE_SHIFT 8 /* IN3L_MUTE */ +#define WM2200_IN3L_MUTE_WIDTH 1 /* IN3L_MUTE */ +#define WM2200_IN3L_DIG_VOL_MASK 0x00FF /* IN3L_DIG_VOL - [7:0] */ +#define WM2200_IN3L_DIG_VOL_SHIFT 0 /* IN3L_DIG_VOL - [7:0] */ +#define WM2200_IN3L_DIG_VOL_WIDTH 8 /* IN3L_DIG_VOL - [7:0] */ + +/* + * R785 (0x311) - ADC Digital Volume 3R + */ +#define WM2200_IN_VU 0x0200 /* IN_VU */ +#define WM2200_IN_VU_MASK 0x0200 /* IN_VU */ +#define WM2200_IN_VU_SHIFT 9 /* IN_VU */ +#define WM2200_IN_VU_WIDTH 1 /* IN_VU */ +#define WM2200_IN3R_MUTE 0x0100 /* IN3R_MUTE */ +#define WM2200_IN3R_MUTE_MASK 0x0100 /* IN3R_MUTE */ +#define WM2200_IN3R_MUTE_SHIFT 8 /* IN3R_MUTE */ +#define WM2200_IN3R_MUTE_WIDTH 1 /* IN3R_MUTE */ +#define WM2200_IN3R_DIG_VOL_MASK 0x00FF /* IN3R_DIG_VOL - [7:0] */ +#define WM2200_IN3R_DIG_VOL_SHIFT 0 /* IN3R_DIG_VOL - [7:0] */ +#define WM2200_IN3R_DIG_VOL_WIDTH 8 /* IN3R_DIG_VOL - [7:0] */ + +/* + * R1024 (0x400) - Output Enables + */ +#define WM2200_OUT2L_ENA 0x0008 /* OUT2L_ENA */ +#define WM2200_OUT2L_ENA_MASK 0x0008 /* OUT2L_ENA */ +#define WM2200_OUT2L_ENA_SHIFT 3 /* OUT2L_ENA */ +#define WM2200_OUT2L_ENA_WIDTH 1 /* OUT2L_ENA */ +#define WM2200_OUT2R_ENA 0x0004 /* OUT2R_ENA */ +#define WM2200_OUT2R_ENA_MASK 0x0004 /* OUT2R_ENA */ +#define WM2200_OUT2R_ENA_SHIFT 2 /* OUT2R_ENA */ +#define WM2200_OUT2R_ENA_WIDTH 1 /* OUT2R_ENA */ +#define WM2200_OUT1L_ENA 0x0002 /* OUT1L_ENA */ +#define WM2200_OUT1L_ENA_MASK 0x0002 /* OUT1L_ENA */ +#define WM2200_OUT1L_ENA_SHIFT 1 /* OUT1L_ENA */ +#define WM2200_OUT1L_ENA_WIDTH 1 /* OUT1L_ENA */ +#define WM2200_OUT1R_ENA 0x0001 /* OUT1R_ENA */ +#define WM2200_OUT1R_ENA_MASK 0x0001 /* OUT1R_ENA */ +#define WM2200_OUT1R_ENA_SHIFT 0 /* OUT1R_ENA */ +#define WM2200_OUT1R_ENA_WIDTH 1 /* OUT1R_ENA */ + +/* + * R1025 (0x401) - DAC Volume Limit 1L + */ +#define WM2200_OUT1_OSR 0x2000 /* OUT1_OSR */ +#define WM2200_OUT1_OSR_MASK 0x2000 /* OUT1_OSR */ +#define WM2200_OUT1_OSR_SHIFT 13 /* OUT1_OSR */ +#define WM2200_OUT1_OSR_WIDTH 1 /* OUT1_OSR */ +#define WM2200_OUT1L_ANC_SRC 0x0800 /* OUT1L_ANC_SRC */ +#define WM2200_OUT1L_ANC_SRC_MASK 0x0800 /* OUT1L_ANC_SRC */ +#define WM2200_OUT1L_ANC_SRC_SHIFT 11 /* OUT1L_ANC_SRC */ +#define WM2200_OUT1L_ANC_SRC_WIDTH 1 /* OUT1L_ANC_SRC */ +#define WM2200_OUT1L_PGA_VOL_MASK 0x00FE /* OUT1L_PGA_VOL - [7:1] */ +#define WM2200_OUT1L_PGA_VOL_SHIFT 1 /* OUT1L_PGA_VOL - [7:1] */ +#define WM2200_OUT1L_PGA_VOL_WIDTH 7 /* OUT1L_PGA_VOL - [7:1] */ + +/* + * R1026 (0x402) - DAC Volume Limit 1R + */ +#define WM2200_OUT1R_ANC_SRC 0x0800 /* OUT1R_ANC_SRC */ +#define WM2200_OUT1R_ANC_SRC_MASK 0x0800 /* OUT1R_ANC_SRC */ +#define WM2200_OUT1R_ANC_SRC_SHIFT 11 /* OUT1R_ANC_SRC */ +#define WM2200_OUT1R_ANC_SRC_WIDTH 1 /* OUT1R_ANC_SRC */ +#define WM2200_OUT1R_PGA_VOL_MASK 0x00FE /* OUT1R_PGA_VOL - [7:1] */ +#define WM2200_OUT1R_PGA_VOL_SHIFT 1 /* OUT1R_PGA_VOL - [7:1] */ +#define WM2200_OUT1R_PGA_VOL_WIDTH 7 /* OUT1R_PGA_VOL - [7:1] */ + +/* + * R1027 (0x403) - DAC Volume Limit 2L + */ +#define WM2200_OUT2_OSR 0x2000 /* OUT2_OSR */ +#define WM2200_OUT2_OSR_MASK 0x2000 /* OUT2_OSR */ +#define WM2200_OUT2_OSR_SHIFT 13 /* OUT2_OSR */ +#define WM2200_OUT2_OSR_WIDTH 1 /* OUT2_OSR */ +#define WM2200_OUT2L_ANC_SRC 0x0800 /* OUT2L_ANC_SRC */ +#define WM2200_OUT2L_ANC_SRC_MASK 0x0800 /* OUT2L_ANC_SRC */ +#define WM2200_OUT2L_ANC_SRC_SHIFT 11 /* OUT2L_ANC_SRC */ +#define WM2200_OUT2L_ANC_SRC_WIDTH 1 /* OUT2L_ANC_SRC */ + +/* + * R1028 (0x404) - DAC Volume Limit 2R + */ +#define WM2200_OUT2R_ANC_SRC 0x0800 /* OUT2R_ANC_SRC */ +#define WM2200_OUT2R_ANC_SRC_MASK 0x0800 /* OUT2R_ANC_SRC */ +#define WM2200_OUT2R_ANC_SRC_SHIFT 11 /* OUT2R_ANC_SRC */ +#define WM2200_OUT2R_ANC_SRC_WIDTH 1 /* OUT2R_ANC_SRC */ + +/* + * R1033 (0x409) - DAC AEC Control 1 + */ +#define WM2200_AEC_LOOPBACK_ENA 0x0004 /* AEC_LOOPBACK_ENA */ +#define WM2200_AEC_LOOPBACK_ENA_MASK 0x0004 /* AEC_LOOPBACK_ENA */ +#define WM2200_AEC_LOOPBACK_ENA_SHIFT 2 /* AEC_LOOPBACK_ENA */ +#define WM2200_AEC_LOOPBACK_ENA_WIDTH 1 /* AEC_LOOPBACK_ENA */ +#define WM2200_AEC_LOOPBACK_SRC_MASK 0x0003 /* AEC_LOOPBACK_SRC - [1:0] */ +#define WM2200_AEC_LOOPBACK_SRC_SHIFT 0 /* AEC_LOOPBACK_SRC - [1:0] */ +#define WM2200_AEC_LOOPBACK_SRC_WIDTH 2 /* AEC_LOOPBACK_SRC - [1:0] */ + +/* + * R1034 (0x40A) - Output Volume Ramp + */ +#define WM2200_OUT_VD_RAMP_MASK 0x0070 /* OUT_VD_RAMP - [6:4] */ +#define WM2200_OUT_VD_RAMP_SHIFT 4 /* OUT_VD_RAMP - [6:4] */ +#define WM2200_OUT_VD_RAMP_WIDTH 3 /* OUT_VD_RAMP - [6:4] */ +#define WM2200_OUT_VI_RAMP_MASK 0x0007 /* OUT_VI_RAMP - [2:0] */ +#define WM2200_OUT_VI_RAMP_SHIFT 0 /* OUT_VI_RAMP - [2:0] */ +#define WM2200_OUT_VI_RAMP_WIDTH 3 /* OUT_VI_RAMP - [2:0] */ + +/* + * R1035 (0x40B) - DAC Digital Volume 1L + */ +#define WM2200_OUT_VU 0x0200 /* OUT_VU */ +#define WM2200_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM2200_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM2200_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM2200_OUT1L_MUTE 0x0100 /* OUT1L_MUTE */ +#define WM2200_OUT1L_MUTE_MASK 0x0100 /* OUT1L_MUTE */ +#define WM2200_OUT1L_MUTE_SHIFT 8 /* OUT1L_MUTE */ +#define WM2200_OUT1L_MUTE_WIDTH 1 /* OUT1L_MUTE */ +#define WM2200_OUT1L_VOL_MASK 0x00FF /* OUT1L_VOL - [7:0] */ +#define WM2200_OUT1L_VOL_SHIFT 0 /* OUT1L_VOL - [7:0] */ +#define WM2200_OUT1L_VOL_WIDTH 8 /* OUT1L_VOL - [7:0] */ + +/* + * R1036 (0x40C) - DAC Digital Volume 1R + */ +#define WM2200_OUT_VU 0x0200 /* OUT_VU */ +#define WM2200_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM2200_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM2200_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM2200_OUT1R_MUTE 0x0100 /* OUT1R_MUTE */ +#define WM2200_OUT1R_MUTE_MASK 0x0100 /* OUT1R_MUTE */ +#define WM2200_OUT1R_MUTE_SHIFT 8 /* OUT1R_MUTE */ +#define WM2200_OUT1R_MUTE_WIDTH 1 /* OUT1R_MUTE */ +#define WM2200_OUT1R_VOL_MASK 0x00FF /* OUT1R_VOL - [7:0] */ +#define WM2200_OUT1R_VOL_SHIFT 0 /* OUT1R_VOL - [7:0] */ +#define WM2200_OUT1R_VOL_WIDTH 8 /* OUT1R_VOL - [7:0] */ + +/* + * R1037 (0x40D) - DAC Digital Volume 2L + */ +#define WM2200_OUT_VU 0x0200 /* OUT_VU */ +#define WM2200_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM2200_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM2200_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM2200_OUT2L_MUTE 0x0100 /* OUT2L_MUTE */ +#define WM2200_OUT2L_MUTE_MASK 0x0100 /* OUT2L_MUTE */ +#define WM2200_OUT2L_MUTE_SHIFT 8 /* OUT2L_MUTE */ +#define WM2200_OUT2L_MUTE_WIDTH 1 /* OUT2L_MUTE */ +#define WM2200_OUT2L_VOL_MASK 0x00FF /* OUT2L_VOL - [7:0] */ +#define WM2200_OUT2L_VOL_SHIFT 0 /* OUT2L_VOL - [7:0] */ +#define WM2200_OUT2L_VOL_WIDTH 8 /* OUT2L_VOL - [7:0] */ + +/* + * R1038 (0x40E) - DAC Digital Volume 2R + */ +#define WM2200_OUT_VU 0x0200 /* OUT_VU */ +#define WM2200_OUT_VU_MASK 0x0200 /* OUT_VU */ +#define WM2200_OUT_VU_SHIFT 9 /* OUT_VU */ +#define WM2200_OUT_VU_WIDTH 1 /* OUT_VU */ +#define WM2200_OUT2R_MUTE 0x0100 /* OUT2R_MUTE */ +#define WM2200_OUT2R_MUTE_MASK 0x0100 /* OUT2R_MUTE */ +#define WM2200_OUT2R_MUTE_SHIFT 8 /* OUT2R_MUTE */ +#define WM2200_OUT2R_MUTE_WIDTH 1 /* OUT2R_MUTE */ +#define WM2200_OUT2R_VOL_MASK 0x00FF /* OUT2R_VOL - [7:0] */ +#define WM2200_OUT2R_VOL_SHIFT 0 /* OUT2R_VOL - [7:0] */ +#define WM2200_OUT2R_VOL_WIDTH 8 /* OUT2R_VOL - [7:0] */ + +/* + * R1047 (0x417) - PDM 1 + */ +#define WM2200_SPK1R_MUTE 0x2000 /* SPK1R_MUTE */ +#define WM2200_SPK1R_MUTE_MASK 0x2000 /* SPK1R_MUTE */ +#define WM2200_SPK1R_MUTE_SHIFT 13 /* SPK1R_MUTE */ +#define WM2200_SPK1R_MUTE_WIDTH 1 /* SPK1R_MUTE */ +#define WM2200_SPK1L_MUTE 0x1000 /* SPK1L_MUTE */ +#define WM2200_SPK1L_MUTE_MASK 0x1000 /* SPK1L_MUTE */ +#define WM2200_SPK1L_MUTE_SHIFT 12 /* SPK1L_MUTE */ +#define WM2200_SPK1L_MUTE_WIDTH 1 /* SPK1L_MUTE */ +#define WM2200_SPK1_MUTE_ENDIAN 0x0100 /* SPK1_MUTE_ENDIAN */ +#define WM2200_SPK1_MUTE_ENDIAN_MASK 0x0100 /* SPK1_MUTE_ENDIAN */ +#define WM2200_SPK1_MUTE_ENDIAN_SHIFT 8 /* SPK1_MUTE_ENDIAN */ +#define WM2200_SPK1_MUTE_ENDIAN_WIDTH 1 /* SPK1_MUTE_ENDIAN */ +#define WM2200_SPK1_MUTE_SEQL_MASK 0x00FF /* SPK1_MUTE_SEQL - [7:0] */ +#define WM2200_SPK1_MUTE_SEQL_SHIFT 0 /* SPK1_MUTE_SEQL - [7:0] */ +#define WM2200_SPK1_MUTE_SEQL_WIDTH 8 /* SPK1_MUTE_SEQL - [7:0] */ + +/* + * R1048 (0x418) - PDM 2 + */ +#define WM2200_SPK1_FMT 0x0001 /* SPK1_FMT */ +#define WM2200_SPK1_FMT_MASK 0x0001 /* SPK1_FMT */ +#define WM2200_SPK1_FMT_SHIFT 0 /* SPK1_FMT */ +#define WM2200_SPK1_FMT_WIDTH 1 /* SPK1_FMT */ + +/* + * R1280 (0x500) - Audio IF 1_1 + */ +#define WM2200_AIF1_BCLK_INV 0x0040 /* AIF1_BCLK_INV */ +#define WM2200_AIF1_BCLK_INV_MASK 0x0040 /* AIF1_BCLK_INV */ +#define WM2200_AIF1_BCLK_INV_SHIFT 6 /* AIF1_BCLK_INV */ +#define WM2200_AIF1_BCLK_INV_WIDTH 1 /* AIF1_BCLK_INV */ +#define WM2200_AIF1_BCLK_FRC 0x0020 /* AIF1_BCLK_FRC */ +#define WM2200_AIF1_BCLK_FRC_MASK 0x0020 /* AIF1_BCLK_FRC */ +#define WM2200_AIF1_BCLK_FRC_SHIFT 5 /* AIF1_BCLK_FRC */ +#define WM2200_AIF1_BCLK_FRC_WIDTH 1 /* AIF1_BCLK_FRC */ +#define WM2200_AIF1_BCLK_MSTR 0x0010 /* AIF1_BCLK_MSTR */ +#define WM2200_AIF1_BCLK_MSTR_MASK 0x0010 /* AIF1_BCLK_MSTR */ +#define WM2200_AIF1_BCLK_MSTR_SHIFT 4 /* AIF1_BCLK_MSTR */ +#define WM2200_AIF1_BCLK_MSTR_WIDTH 1 /* AIF1_BCLK_MSTR */ +#define WM2200_AIF1_BCLK_DIV_MASK 0x000F /* AIF1_BCLK_DIV - [3:0] */ +#define WM2200_AIF1_BCLK_DIV_SHIFT 0 /* AIF1_BCLK_DIV - [3:0] */ +#define WM2200_AIF1_BCLK_DIV_WIDTH 4 /* AIF1_BCLK_DIV - [3:0] */ + +/* + * R1281 (0x501) - Audio IF 1_2 + */ +#define WM2200_AIF1TX_DAT_TRI 0x0020 /* AIF1TX_DAT_TRI */ +#define WM2200_AIF1TX_DAT_TRI_MASK 0x0020 /* AIF1TX_DAT_TRI */ +#define WM2200_AIF1TX_DAT_TRI_SHIFT 5 /* AIF1TX_DAT_TRI */ +#define WM2200_AIF1TX_DAT_TRI_WIDTH 1 /* AIF1TX_DAT_TRI */ +#define WM2200_AIF1TX_LRCLK_SRC 0x0008 /* AIF1TX_LRCLK_SRC */ +#define WM2200_AIF1TX_LRCLK_SRC_MASK 0x0008 /* AIF1TX_LRCLK_SRC */ +#define WM2200_AIF1TX_LRCLK_SRC_SHIFT 3 /* AIF1TX_LRCLK_SRC */ +#define WM2200_AIF1TX_LRCLK_SRC_WIDTH 1 /* AIF1TX_LRCLK_SRC */ +#define WM2200_AIF1TX_LRCLK_INV 0x0004 /* AIF1TX_LRCLK_INV */ +#define WM2200_AIF1TX_LRCLK_INV_MASK 0x0004 /* AIF1TX_LRCLK_INV */ +#define WM2200_AIF1TX_LRCLK_INV_SHIFT 2 /* AIF1TX_LRCLK_INV */ +#define WM2200_AIF1TX_LRCLK_INV_WIDTH 1 /* AIF1TX_LRCLK_INV */ +#define WM2200_AIF1TX_LRCLK_FRC 0x0002 /* AIF1TX_LRCLK_FRC */ +#define WM2200_AIF1TX_LRCLK_FRC_MASK 0x0002 /* AIF1TX_LRCLK_FRC */ +#define WM2200_AIF1TX_LRCLK_FRC_SHIFT 1 /* AIF1TX_LRCLK_FRC */ +#define WM2200_AIF1TX_LRCLK_FRC_WIDTH 1 /* AIF1TX_LRCLK_FRC */ +#define WM2200_AIF1TX_LRCLK_MSTR 0x0001 /* AIF1TX_LRCLK_MSTR */ +#define WM2200_AIF1TX_LRCLK_MSTR_MASK 0x0001 /* AIF1TX_LRCLK_MSTR */ +#define WM2200_AIF1TX_LRCLK_MSTR_SHIFT 0 /* AIF1TX_LRCLK_MSTR */ +#define WM2200_AIF1TX_LRCLK_MSTR_WIDTH 1 /* AIF1TX_LRCLK_MSTR */ + +/* + * R1282 (0x502) - Audio IF 1_3 + */ +#define WM2200_AIF1RX_LRCLK_INV 0x0004 /* AIF1RX_LRCLK_INV */ +#define WM2200_AIF1RX_LRCLK_INV_MASK 0x0004 /* AIF1RX_LRCLK_INV */ +#define WM2200_AIF1RX_LRCLK_INV_SHIFT 2 /* AIF1RX_LRCLK_INV */ +#define WM2200_AIF1RX_LRCLK_INV_WIDTH 1 /* AIF1RX_LRCLK_INV */ +#define WM2200_AIF1RX_LRCLK_FRC 0x0002 /* AIF1RX_LRCLK_FRC */ +#define WM2200_AIF1RX_LRCLK_FRC_MASK 0x0002 /* AIF1RX_LRCLK_FRC */ +#define WM2200_AIF1RX_LRCLK_FRC_SHIFT 1 /* AIF1RX_LRCLK_FRC */ +#define WM2200_AIF1RX_LRCLK_FRC_WIDTH 1 /* AIF1RX_LRCLK_FRC */ +#define WM2200_AIF1RX_LRCLK_MSTR 0x0001 /* AIF1RX_LRCLK_MSTR */ +#define WM2200_AIF1RX_LRCLK_MSTR_MASK 0x0001 /* AIF1RX_LRCLK_MSTR */ +#define WM2200_AIF1RX_LRCLK_MSTR_SHIFT 0 /* AIF1RX_LRCLK_MSTR */ +#define WM2200_AIF1RX_LRCLK_MSTR_WIDTH 1 /* AIF1RX_LRCLK_MSTR */ + +/* + * R1283 (0x503) - Audio IF 1_4 + */ +#define WM2200_AIF1_TRI 0x0040 /* AIF1_TRI */ +#define WM2200_AIF1_TRI_MASK 0x0040 /* AIF1_TRI */ +#define WM2200_AIF1_TRI_SHIFT 6 /* AIF1_TRI */ +#define WM2200_AIF1_TRI_WIDTH 1 /* AIF1_TRI */ + +/* + * R1284 (0x504) - Audio IF 1_5 + */ +#define WM2200_AIF1_FMT_MASK 0x0007 /* AIF1_FMT - [2:0] */ +#define WM2200_AIF1_FMT_SHIFT 0 /* AIF1_FMT - [2:0] */ +#define WM2200_AIF1_FMT_WIDTH 3 /* AIF1_FMT - [2:0] */ + +/* + * R1285 (0x505) - Audio IF 1_6 + */ +#define WM2200_AIF1TX_BCPF_MASK 0x07FF /* AIF1TX_BCPF - [10:0] */ +#define WM2200_AIF1TX_BCPF_SHIFT 0 /* AIF1TX_BCPF - [10:0] */ +#define WM2200_AIF1TX_BCPF_WIDTH 11 /* AIF1TX_BCPF - [10:0] */ + +/* + * R1286 (0x506) - Audio IF 1_7 + */ +#define WM2200_AIF1RX_BCPF_MASK 0x07FF /* AIF1RX_BCPF - [10:0] */ +#define WM2200_AIF1RX_BCPF_SHIFT 0 /* AIF1RX_BCPF - [10:0] */ +#define WM2200_AIF1RX_BCPF_WIDTH 11 /* AIF1RX_BCPF - [10:0] */ + +/* + * R1287 (0x507) - Audio IF 1_8 + */ +#define WM2200_AIF1TX_WL_MASK 0x3F00 /* AIF1TX_WL - [13:8] */ +#define WM2200_AIF1TX_WL_SHIFT 8 /* AIF1TX_WL - [13:8] */ +#define WM2200_AIF1TX_WL_WIDTH 6 /* AIF1TX_WL - [13:8] */ +#define WM2200_AIF1TX_SLOT_LEN_MASK 0x00FF /* AIF1TX_SLOT_LEN - [7:0] */ +#define WM2200_AIF1TX_SLOT_LEN_SHIFT 0 /* AIF1TX_SLOT_LEN - [7:0] */ +#define WM2200_AIF1TX_SLOT_LEN_WIDTH 8 /* AIF1TX_SLOT_LEN - [7:0] */ + +/* + * R1288 (0x508) - Audio IF 1_9 + */ +#define WM2200_AIF1RX_WL_MASK 0x3F00 /* AIF1RX_WL - [13:8] */ +#define WM2200_AIF1RX_WL_SHIFT 8 /* AIF1RX_WL - [13:8] */ +#define WM2200_AIF1RX_WL_WIDTH 6 /* AIF1RX_WL - [13:8] */ +#define WM2200_AIF1RX_SLOT_LEN_MASK 0x00FF /* AIF1RX_SLOT_LEN - [7:0] */ +#define WM2200_AIF1RX_SLOT_LEN_SHIFT 0 /* AIF1RX_SLOT_LEN - [7:0] */ +#define WM2200_AIF1RX_SLOT_LEN_WIDTH 8 /* AIF1RX_SLOT_LEN - [7:0] */ + +/* + * R1289 (0x509) - Audio IF 1_10 + */ +#define WM2200_AIF1TX1_SLOT_MASK 0x003F /* AIF1TX1_SLOT - [5:0] */ +#define WM2200_AIF1TX1_SLOT_SHIFT 0 /* AIF1TX1_SLOT - [5:0] */ +#define WM2200_AIF1TX1_SLOT_WIDTH 6 /* AIF1TX1_SLOT - [5:0] */ + +/* + * R1290 (0x50A) - Audio IF 1_11 + */ +#define WM2200_AIF1TX2_SLOT_MASK 0x003F /* AIF1TX2_SLOT - [5:0] */ +#define WM2200_AIF1TX2_SLOT_SHIFT 0 /* AIF1TX2_SLOT - [5:0] */ +#define WM2200_AIF1TX2_SLOT_WIDTH 6 /* AIF1TX2_SLOT - [5:0] */ + +/* + * R1291 (0x50B) - Audio IF 1_12 + */ +#define WM2200_AIF1TX3_SLOT_MASK 0x003F /* AIF1TX3_SLOT - [5:0] */ +#define WM2200_AIF1TX3_SLOT_SHIFT 0 /* AIF1TX3_SLOT - [5:0] */ +#define WM2200_AIF1TX3_SLOT_WIDTH 6 /* AIF1TX3_SLOT - [5:0] */ + +/* + * R1292 (0x50C) - Audio IF 1_13 + */ +#define WM2200_AIF1TX4_SLOT_MASK 0x003F /* AIF1TX4_SLOT - [5:0] */ +#define WM2200_AIF1TX4_SLOT_SHIFT 0 /* AIF1TX4_SLOT - [5:0] */ +#define WM2200_AIF1TX4_SLOT_WIDTH 6 /* AIF1TX4_SLOT - [5:0] */ + +/* + * R1293 (0x50D) - Audio IF 1_14 + */ +#define WM2200_AIF1TX5_SLOT_MASK 0x003F /* AIF1TX5_SLOT - [5:0] */ +#define WM2200_AIF1TX5_SLOT_SHIFT 0 /* AIF1TX5_SLOT - [5:0] */ +#define WM2200_AIF1TX5_SLOT_WIDTH 6 /* AIF1TX5_SLOT - [5:0] */ + +/* + * R1294 (0x50E) - Audio IF 1_15 + */ +#define WM2200_AIF1TX6_SLOT_MASK 0x003F /* AIF1TX6_SLOT - [5:0] */ +#define WM2200_AIF1TX6_SLOT_SHIFT 0 /* AIF1TX6_SLOT - [5:0] */ +#define WM2200_AIF1TX6_SLOT_WIDTH 6 /* AIF1TX6_SLOT - [5:0] */ + +/* + * R1295 (0x50F) - Audio IF 1_16 + */ +#define WM2200_AIF1RX1_SLOT_MASK 0x003F /* AIF1RX1_SLOT - [5:0] */ +#define WM2200_AIF1RX1_SLOT_SHIFT 0 /* AIF1RX1_SLOT - [5:0] */ +#define WM2200_AIF1RX1_SLOT_WIDTH 6 /* AIF1RX1_SLOT - [5:0] */ + +/* + * R1296 (0x510) - Audio IF 1_17 + */ +#define WM2200_AIF1RX2_SLOT_MASK 0x003F /* AIF1RX2_SLOT - [5:0] */ +#define WM2200_AIF1RX2_SLOT_SHIFT 0 /* AIF1RX2_SLOT - [5:0] */ +#define WM2200_AIF1RX2_SLOT_WIDTH 6 /* AIF1RX2_SLOT - [5:0] */ + +/* + * R1297 (0x511) - Audio IF 1_18 + */ +#define WM2200_AIF1RX3_SLOT_MASK 0x003F /* AIF1RX3_SLOT - [5:0] */ +#define WM2200_AIF1RX3_SLOT_SHIFT 0 /* AIF1RX3_SLOT - [5:0] */ +#define WM2200_AIF1RX3_SLOT_WIDTH 6 /* AIF1RX3_SLOT - [5:0] */ + +/* + * R1298 (0x512) - Audio IF 1_19 + */ +#define WM2200_AIF1RX4_SLOT_MASK 0x003F /* AIF1RX4_SLOT - [5:0] */ +#define WM2200_AIF1RX4_SLOT_SHIFT 0 /* AIF1RX4_SLOT - [5:0] */ +#define WM2200_AIF1RX4_SLOT_WIDTH 6 /* AIF1RX4_SLOT - [5:0] */ + +/* + * R1299 (0x513) - Audio IF 1_20 + */ +#define WM2200_AIF1RX5_SLOT_MASK 0x003F /* AIF1RX5_SLOT - [5:0] */ +#define WM2200_AIF1RX5_SLOT_SHIFT 0 /* AIF1RX5_SLOT - [5:0] */ +#define WM2200_AIF1RX5_SLOT_WIDTH 6 /* AIF1RX5_SLOT - [5:0] */ + +/* + * R1300 (0x514) - Audio IF 1_21 + */ +#define WM2200_AIF1RX6_SLOT_MASK 0x003F /* AIF1RX6_SLOT - [5:0] */ +#define WM2200_AIF1RX6_SLOT_SHIFT 0 /* AIF1RX6_SLOT - [5:0] */ +#define WM2200_AIF1RX6_SLOT_WIDTH 6 /* AIF1RX6_SLOT - [5:0] */ + +/* + * R1301 (0x515) - Audio IF 1_22 + */ +#define WM2200_AIF1RX6_ENA 0x0800 /* AIF1RX6_ENA */ +#define WM2200_AIF1RX6_ENA_MASK 0x0800 /* AIF1RX6_ENA */ +#define WM2200_AIF1RX6_ENA_SHIFT 11 /* AIF1RX6_ENA */ +#define WM2200_AIF1RX6_ENA_WIDTH 1 /* AIF1RX6_ENA */ +#define WM2200_AIF1RX5_ENA 0x0400 /* AIF1RX5_ENA */ +#define WM2200_AIF1RX5_ENA_MASK 0x0400 /* AIF1RX5_ENA */ +#define WM2200_AIF1RX5_ENA_SHIFT 10 /* AIF1RX5_ENA */ +#define WM2200_AIF1RX5_ENA_WIDTH 1 /* AIF1RX5_ENA */ +#define WM2200_AIF1RX4_ENA 0x0200 /* AIF1RX4_ENA */ +#define WM2200_AIF1RX4_ENA_MASK 0x0200 /* AIF1RX4_ENA */ +#define WM2200_AIF1RX4_ENA_SHIFT 9 /* AIF1RX4_ENA */ +#define WM2200_AIF1RX4_ENA_WIDTH 1 /* AIF1RX4_ENA */ +#define WM2200_AIF1RX3_ENA 0x0100 /* AIF1RX3_ENA */ +#define WM2200_AIF1RX3_ENA_MASK 0x0100 /* AIF1RX3_ENA */ +#define WM2200_AIF1RX3_ENA_SHIFT 8 /* AIF1RX3_ENA */ +#define WM2200_AIF1RX3_ENA_WIDTH 1 /* AIF1RX3_ENA */ +#define WM2200_AIF1RX2_ENA 0x0080 /* AIF1RX2_ENA */ +#define WM2200_AIF1RX2_ENA_MASK 0x0080 /* AIF1RX2_ENA */ +#define WM2200_AIF1RX2_ENA_SHIFT 7 /* AIF1RX2_ENA */ +#define WM2200_AIF1RX2_ENA_WIDTH 1 /* AIF1RX2_ENA */ +#define WM2200_AIF1RX1_ENA 0x0040 /* AIF1RX1_ENA */ +#define WM2200_AIF1RX1_ENA_MASK 0x0040 /* AIF1RX1_ENA */ +#define WM2200_AIF1RX1_ENA_SHIFT 6 /* AIF1RX1_ENA */ +#define WM2200_AIF1RX1_ENA_WIDTH 1 /* AIF1RX1_ENA */ +#define WM2200_AIF1TX6_ENA 0x0020 /* AIF1TX6_ENA */ +#define WM2200_AIF1TX6_ENA_MASK 0x0020 /* AIF1TX6_ENA */ +#define WM2200_AIF1TX6_ENA_SHIFT 5 /* AIF1TX6_ENA */ +#define WM2200_AIF1TX6_ENA_WIDTH 1 /* AIF1TX6_ENA */ +#define WM2200_AIF1TX5_ENA 0x0010 /* AIF1TX5_ENA */ +#define WM2200_AIF1TX5_ENA_MASK 0x0010 /* AIF1TX5_ENA */ +#define WM2200_AIF1TX5_ENA_SHIFT 4 /* AIF1TX5_ENA */ +#define WM2200_AIF1TX5_ENA_WIDTH 1 /* AIF1TX5_ENA */ +#define WM2200_AIF1TX4_ENA 0x0008 /* AIF1TX4_ENA */ +#define WM2200_AIF1TX4_ENA_MASK 0x0008 /* AIF1TX4_ENA */ +#define WM2200_AIF1TX4_ENA_SHIFT 3 /* AIF1TX4_ENA */ +#define WM2200_AIF1TX4_ENA_WIDTH 1 /* AIF1TX4_ENA */ +#define WM2200_AIF1TX3_ENA 0x0004 /* AIF1TX3_ENA */ +#define WM2200_AIF1TX3_ENA_MASK 0x0004 /* AIF1TX3_ENA */ +#define WM2200_AIF1TX3_ENA_SHIFT 2 /* AIF1TX3_ENA */ +#define WM2200_AIF1TX3_ENA_WIDTH 1 /* AIF1TX3_ENA */ +#define WM2200_AIF1TX2_ENA 0x0002 /* AIF1TX2_ENA */ +#define WM2200_AIF1TX2_ENA_MASK 0x0002 /* AIF1TX2_ENA */ +#define WM2200_AIF1TX2_ENA_SHIFT 1 /* AIF1TX2_ENA */ +#define WM2200_AIF1TX2_ENA_WIDTH 1 /* AIF1TX2_ENA */ +#define WM2200_AIF1TX1_ENA 0x0001 /* AIF1TX1_ENA */ +#define WM2200_AIF1TX1_ENA_MASK 0x0001 /* AIF1TX1_ENA */ +#define WM2200_AIF1TX1_ENA_SHIFT 0 /* AIF1TX1_ENA */ +#define WM2200_AIF1TX1_ENA_WIDTH 1 /* AIF1TX1_ENA */ + +/* + * R1536 (0x600) - OUT1LMIX Input 1 Source + */ +#define WM2200_OUT1LMIX_SRC1_MASK 0x007F /* OUT1LMIX_SRC1 - [6:0] */ +#define WM2200_OUT1LMIX_SRC1_SHIFT 0 /* OUT1LMIX_SRC1 - [6:0] */ +#define WM2200_OUT1LMIX_SRC1_WIDTH 7 /* OUT1LMIX_SRC1 - [6:0] */ + +/* + * R1537 (0x601) - OUT1LMIX Input 1 Volume + */ +#define WM2200_OUT1LMIX_VOL1_MASK 0x00FE /* OUT1LMIX_VOL1 - [7:1] */ +#define WM2200_OUT1LMIX_VOL1_SHIFT 1 /* OUT1LMIX_VOL1 - [7:1] */ +#define WM2200_OUT1LMIX_VOL1_WIDTH 7 /* OUT1LMIX_VOL1 - [7:1] */ + +/* + * R1538 (0x602) - OUT1LMIX Input 2 Source + */ +#define WM2200_OUT1LMIX_SRC2_MASK 0x007F /* OUT1LMIX_SRC2 - [6:0] */ +#define WM2200_OUT1LMIX_SRC2_SHIFT 0 /* OUT1LMIX_SRC2 - [6:0] */ +#define WM2200_OUT1LMIX_SRC2_WIDTH 7 /* OUT1LMIX_SRC2 - [6:0] */ + +/* + * R1539 (0x603) - OUT1LMIX Input 2 Volume + */ +#define WM2200_OUT1LMIX_VOL2_MASK 0x00FE /* OUT1LMIX_VOL2 - [7:1] */ +#define WM2200_OUT1LMIX_VOL2_SHIFT 1 /* OUT1LMIX_VOL2 - [7:1] */ +#define WM2200_OUT1LMIX_VOL2_WIDTH 7 /* OUT1LMIX_VOL2 - [7:1] */ + +/* + * R1540 (0x604) - OUT1LMIX Input 3 Source + */ +#define WM2200_OUT1LMIX_SRC3_MASK 0x007F /* OUT1LMIX_SRC3 - [6:0] */ +#define WM2200_OUT1LMIX_SRC3_SHIFT 0 /* OUT1LMIX_SRC3 - [6:0] */ +#define WM2200_OUT1LMIX_SRC3_WIDTH 7 /* OUT1LMIX_SRC3 - [6:0] */ + +/* + * R1541 (0x605) - OUT1LMIX Input 3 Volume + */ +#define WM2200_OUT1LMIX_VOL3_MASK 0x00FE /* OUT1LMIX_VOL3 - [7:1] */ +#define WM2200_OUT1LMIX_VOL3_SHIFT 1 /* OUT1LMIX_VOL3 - [7:1] */ +#define WM2200_OUT1LMIX_VOL3_WIDTH 7 /* OUT1LMIX_VOL3 - [7:1] */ + +/* + * R1542 (0x606) - OUT1LMIX Input 4 Source + */ +#define WM2200_OUT1LMIX_SRC4_MASK 0x007F /* OUT1LMIX_SRC4 - [6:0] */ +#define WM2200_OUT1LMIX_SRC4_SHIFT 0 /* OUT1LMIX_SRC4 - [6:0] */ +#define WM2200_OUT1LMIX_SRC4_WIDTH 7 /* OUT1LMIX_SRC4 - [6:0] */ + +/* + * R1543 (0x607) - OUT1LMIX Input 4 Volume + */ +#define WM2200_OUT1LMIX_VOL4_MASK 0x00FE /* OUT1LMIX_VOL4 - [7:1] */ +#define WM2200_OUT1LMIX_VOL4_SHIFT 1 /* OUT1LMIX_VOL4 - [7:1] */ +#define WM2200_OUT1LMIX_VOL4_WIDTH 7 /* OUT1LMIX_VOL4 - [7:1] */ + +/* + * R1544 (0x608) - OUT1RMIX Input 1 Source + */ +#define WM2200_OUT1RMIX_SRC1_MASK 0x007F /* OUT1RMIX_SRC1 - [6:0] */ +#define WM2200_OUT1RMIX_SRC1_SHIFT 0 /* OUT1RMIX_SRC1 - [6:0] */ +#define WM2200_OUT1RMIX_SRC1_WIDTH 7 /* OUT1RMIX_SRC1 - [6:0] */ + +/* + * R1545 (0x609) - OUT1RMIX Input 1 Volume + */ +#define WM2200_OUT1RMIX_VOL1_MASK 0x00FE /* OUT1RMIX_VOL1 - [7:1] */ +#define WM2200_OUT1RMIX_VOL1_SHIFT 1 /* OUT1RMIX_VOL1 - [7:1] */ +#define WM2200_OUT1RMIX_VOL1_WIDTH 7 /* OUT1RMIX_VOL1 - [7:1] */ + +/* + * R1546 (0x60A) - OUT1RMIX Input 2 Source + */ +#define WM2200_OUT1RMIX_SRC2_MASK 0x007F /* OUT1RMIX_SRC2 - [6:0] */ +#define WM2200_OUT1RMIX_SRC2_SHIFT 0 /* OUT1RMIX_SRC2 - [6:0] */ +#define WM2200_OUT1RMIX_SRC2_WIDTH 7 /* OUT1RMIX_SRC2 - [6:0] */ + +/* + * R1547 (0x60B) - OUT1RMIX Input 2 Volume + */ +#define WM2200_OUT1RMIX_VOL2_MASK 0x00FE /* OUT1RMIX_VOL2 - [7:1] */ +#define WM2200_OUT1RMIX_VOL2_SHIFT 1 /* OUT1RMIX_VOL2 - [7:1] */ +#define WM2200_OUT1RMIX_VOL2_WIDTH 7 /* OUT1RMIX_VOL2 - [7:1] */ + +/* + * R1548 (0x60C) - OUT1RMIX Input 3 Source + */ +#define WM2200_OUT1RMIX_SRC3_MASK 0x007F /* OUT1RMIX_SRC3 - [6:0] */ +#define WM2200_OUT1RMIX_SRC3_SHIFT 0 /* OUT1RMIX_SRC3 - [6:0] */ +#define WM2200_OUT1RMIX_SRC3_WIDTH 7 /* OUT1RMIX_SRC3 - [6:0] */ + +/* + * R1549 (0x60D) - OUT1RMIX Input 3 Volume + */ +#define WM2200_OUT1RMIX_VOL3_MASK 0x00FE /* OUT1RMIX_VOL3 - [7:1] */ +#define WM2200_OUT1RMIX_VOL3_SHIFT 1 /* OUT1RMIX_VOL3 - [7:1] */ +#define WM2200_OUT1RMIX_VOL3_WIDTH 7 /* OUT1RMIX_VOL3 - [7:1] */ + +/* + * R1550 (0x60E) - OUT1RMIX Input 4 Source + */ +#define WM2200_OUT1RMIX_SRC4_MASK 0x007F /* OUT1RMIX_SRC4 - [6:0] */ +#define WM2200_OUT1RMIX_SRC4_SHIFT 0 /* OUT1RMIX_SRC4 - [6:0] */ +#define WM2200_OUT1RMIX_SRC4_WIDTH 7 /* OUT1RMIX_SRC4 - [6:0] */ + +/* + * R1551 (0x60F) - OUT1RMIX Input 4 Volume + */ +#define WM2200_OUT1RMIX_VOL4_MASK 0x00FE /* OUT1RMIX_VOL4 - [7:1] */ +#define WM2200_OUT1RMIX_VOL4_SHIFT 1 /* OUT1RMIX_VOL4 - [7:1] */ +#define WM2200_OUT1RMIX_VOL4_WIDTH 7 /* OUT1RMIX_VOL4 - [7:1] */ + +/* + * R1552 (0x610) - OUT2LMIX Input 1 Source + */ +#define WM2200_OUT2LMIX_SRC1_MASK 0x007F /* OUT2LMIX_SRC1 - [6:0] */ +#define WM2200_OUT2LMIX_SRC1_SHIFT 0 /* OUT2LMIX_SRC1 - [6:0] */ +#define WM2200_OUT2LMIX_SRC1_WIDTH 7 /* OUT2LMIX_SRC1 - [6:0] */ + +/* + * R1553 (0x611) - OUT2LMIX Input 1 Volume + */ +#define WM2200_OUT2LMIX_VOL1_MASK 0x00FE /* OUT2LMIX_VOL1 - [7:1] */ +#define WM2200_OUT2LMIX_VOL1_SHIFT 1 /* OUT2LMIX_VOL1 - [7:1] */ +#define WM2200_OUT2LMIX_VOL1_WIDTH 7 /* OUT2LMIX_VOL1 - [7:1] */ + +/* + * R1554 (0x612) - OUT2LMIX Input 2 Source + */ +#define WM2200_OUT2LMIX_SRC2_MASK 0x007F /* OUT2LMIX_SRC2 - [6:0] */ +#define WM2200_OUT2LMIX_SRC2_SHIFT 0 /* OUT2LMIX_SRC2 - [6:0] */ +#define WM2200_OUT2LMIX_SRC2_WIDTH 7 /* OUT2LMIX_SRC2 - [6:0] */ + +/* + * R1555 (0x613) - OUT2LMIX Input 2 Volume + */ +#define WM2200_OUT2LMIX_VOL2_MASK 0x00FE /* OUT2LMIX_VOL2 - [7:1] */ +#define WM2200_OUT2LMIX_VOL2_SHIFT 1 /* OUT2LMIX_VOL2 - [7:1] */ +#define WM2200_OUT2LMIX_VOL2_WIDTH 7 /* OUT2LMIX_VOL2 - [7:1] */ + +/* + * R1556 (0x614) - OUT2LMIX Input 3 Source + */ +#define WM2200_OUT2LMIX_SRC3_MASK 0x007F /* OUT2LMIX_SRC3 - [6:0] */ +#define WM2200_OUT2LMIX_SRC3_SHIFT 0 /* OUT2LMIX_SRC3 - [6:0] */ +#define WM2200_OUT2LMIX_SRC3_WIDTH 7 /* OUT2LMIX_SRC3 - [6:0] */ + +/* + * R1557 (0x615) - OUT2LMIX Input 3 Volume + */ +#define WM2200_OUT2LMIX_VOL3_MASK 0x00FE /* OUT2LMIX_VOL3 - [7:1] */ +#define WM2200_OUT2LMIX_VOL3_SHIFT 1 /* OUT2LMIX_VOL3 - [7:1] */ +#define WM2200_OUT2LMIX_VOL3_WIDTH 7 /* OUT2LMIX_VOL3 - [7:1] */ + +/* + * R1558 (0x616) - OUT2LMIX Input 4 Source + */ +#define WM2200_OUT2LMIX_SRC4_MASK 0x007F /* OUT2LMIX_SRC4 - [6:0] */ +#define WM2200_OUT2LMIX_SRC4_SHIFT 0 /* OUT2LMIX_SRC4 - [6:0] */ +#define WM2200_OUT2LMIX_SRC4_WIDTH 7 /* OUT2LMIX_SRC4 - [6:0] */ + +/* + * R1559 (0x617) - OUT2LMIX Input 4 Volume + */ +#define WM2200_OUT2LMIX_VOL4_MASK 0x00FE /* OUT2LMIX_VOL4 - [7:1] */ +#define WM2200_OUT2LMIX_VOL4_SHIFT 1 /* OUT2LMIX_VOL4 - [7:1] */ +#define WM2200_OUT2LMIX_VOL4_WIDTH 7 /* OUT2LMIX_VOL4 - [7:1] */ + +/* + * R1560 (0x618) - OUT2RMIX Input 1 Source + */ +#define WM2200_OUT2RMIX_SRC1_MASK 0x007F /* OUT2RMIX_SRC1 - [6:0] */ +#define WM2200_OUT2RMIX_SRC1_SHIFT 0 /* OUT2RMIX_SRC1 - [6:0] */ +#define WM2200_OUT2RMIX_SRC1_WIDTH 7 /* OUT2RMIX_SRC1 - [6:0] */ + +/* + * R1561 (0x619) - OUT2RMIX Input 1 Volume + */ +#define WM2200_OUT2RMIX_VOL1_MASK 0x00FE /* OUT2RMIX_VOL1 - [7:1] */ +#define WM2200_OUT2RMIX_VOL1_SHIFT 1 /* OUT2RMIX_VOL1 - [7:1] */ +#define WM2200_OUT2RMIX_VOL1_WIDTH 7 /* OUT2RMIX_VOL1 - [7:1] */ + +/* + * R1562 (0x61A) - OUT2RMIX Input 2 Source + */ +#define WM2200_OUT2RMIX_SRC2_MASK 0x007F /* OUT2RMIX_SRC2 - [6:0] */ +#define WM2200_OUT2RMIX_SRC2_SHIFT 0 /* OUT2RMIX_SRC2 - [6:0] */ +#define WM2200_OUT2RMIX_SRC2_WIDTH 7 /* OUT2RMIX_SRC2 - [6:0] */ + +/* + * R1563 (0x61B) - OUT2RMIX Input 2 Volume + */ +#define WM2200_OUT2RMIX_VOL2_MASK 0x00FE /* OUT2RMIX_VOL2 - [7:1] */ +#define WM2200_OUT2RMIX_VOL2_SHIFT 1 /* OUT2RMIX_VOL2 - [7:1] */ +#define WM2200_OUT2RMIX_VOL2_WIDTH 7 /* OUT2RMIX_VOL2 - [7:1] */ + +/* + * R1564 (0x61C) - OUT2RMIX Input 3 Source + */ +#define WM2200_OUT2RMIX_SRC3_MASK 0x007F /* OUT2RMIX_SRC3 - [6:0] */ +#define WM2200_OUT2RMIX_SRC3_SHIFT 0 /* OUT2RMIX_SRC3 - [6:0] */ +#define WM2200_OUT2RMIX_SRC3_WIDTH 7 /* OUT2RMIX_SRC3 - [6:0] */ + +/* + * R1565 (0x61D) - OUT2RMIX Input 3 Volume + */ +#define WM2200_OUT2RMIX_VOL3_MASK 0x00FE /* OUT2RMIX_VOL3 - [7:1] */ +#define WM2200_OUT2RMIX_VOL3_SHIFT 1 /* OUT2RMIX_VOL3 - [7:1] */ +#define WM2200_OUT2RMIX_VOL3_WIDTH 7 /* OUT2RMIX_VOL3 - [7:1] */ + +/* + * R1566 (0x61E) - OUT2RMIX Input 4 Source + */ +#define WM2200_OUT2RMIX_SRC4_MASK 0x007F /* OUT2RMIX_SRC4 - [6:0] */ +#define WM2200_OUT2RMIX_SRC4_SHIFT 0 /* OUT2RMIX_SRC4 - [6:0] */ +#define WM2200_OUT2RMIX_SRC4_WIDTH 7 /* OUT2RMIX_SRC4 - [6:0] */ + +/* + * R1567 (0x61F) - OUT2RMIX Input 4 Volume + */ +#define WM2200_OUT2RMIX_VOL4_MASK 0x00FE /* OUT2RMIX_VOL4 - [7:1] */ +#define WM2200_OUT2RMIX_VOL4_SHIFT 1 /* OUT2RMIX_VOL4 - [7:1] */ +#define WM2200_OUT2RMIX_VOL4_WIDTH 7 /* OUT2RMIX_VOL4 - [7:1] */ + +/* + * R1568 (0x620) - AIF1TX1MIX Input 1 Source + */ +#define WM2200_AIF1TX1MIX_SRC1_MASK 0x007F /* AIF1TX1MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX1MIX_SRC1_SHIFT 0 /* AIF1TX1MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX1MIX_SRC1_WIDTH 7 /* AIF1TX1MIX_SRC1 - [6:0] */ + +/* + * R1569 (0x621) - AIF1TX1MIX Input 1 Volume + */ +#define WM2200_AIF1TX1MIX_VOL1_MASK 0x00FE /* AIF1TX1MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX1MIX_VOL1_SHIFT 1 /* AIF1TX1MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX1MIX_VOL1_WIDTH 7 /* AIF1TX1MIX_VOL1 - [7:1] */ + +/* + * R1570 (0x622) - AIF1TX1MIX Input 2 Source + */ +#define WM2200_AIF1TX1MIX_SRC2_MASK 0x007F /* AIF1TX1MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX1MIX_SRC2_SHIFT 0 /* AIF1TX1MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX1MIX_SRC2_WIDTH 7 /* AIF1TX1MIX_SRC2 - [6:0] */ + +/* + * R1571 (0x623) - AIF1TX1MIX Input 2 Volume + */ +#define WM2200_AIF1TX1MIX_VOL2_MASK 0x00FE /* AIF1TX1MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX1MIX_VOL2_SHIFT 1 /* AIF1TX1MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX1MIX_VOL2_WIDTH 7 /* AIF1TX1MIX_VOL2 - [7:1] */ + +/* + * R1572 (0x624) - AIF1TX1MIX Input 3 Source + */ +#define WM2200_AIF1TX1MIX_SRC3_MASK 0x007F /* AIF1TX1MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX1MIX_SRC3_SHIFT 0 /* AIF1TX1MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX1MIX_SRC3_WIDTH 7 /* AIF1TX1MIX_SRC3 - [6:0] */ + +/* + * R1573 (0x625) - AIF1TX1MIX Input 3 Volume + */ +#define WM2200_AIF1TX1MIX_VOL3_MASK 0x00FE /* AIF1TX1MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX1MIX_VOL3_SHIFT 1 /* AIF1TX1MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX1MIX_VOL3_WIDTH 7 /* AIF1TX1MIX_VOL3 - [7:1] */ + +/* + * R1574 (0x626) - AIF1TX1MIX Input 4 Source + */ +#define WM2200_AIF1TX1MIX_SRC4_MASK 0x007F /* AIF1TX1MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX1MIX_SRC4_SHIFT 0 /* AIF1TX1MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX1MIX_SRC4_WIDTH 7 /* AIF1TX1MIX_SRC4 - [6:0] */ + +/* + * R1575 (0x627) - AIF1TX1MIX Input 4 Volume + */ +#define WM2200_AIF1TX1MIX_VOL4_MASK 0x00FE /* AIF1TX1MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX1MIX_VOL4_SHIFT 1 /* AIF1TX1MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX1MIX_VOL4_WIDTH 7 /* AIF1TX1MIX_VOL4 - [7:1] */ + +/* + * R1576 (0x628) - AIF1TX2MIX Input 1 Source + */ +#define WM2200_AIF1TX2MIX_SRC1_MASK 0x007F /* AIF1TX2MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX2MIX_SRC1_SHIFT 0 /* AIF1TX2MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX2MIX_SRC1_WIDTH 7 /* AIF1TX2MIX_SRC1 - [6:0] */ + +/* + * R1577 (0x629) - AIF1TX2MIX Input 1 Volume + */ +#define WM2200_AIF1TX2MIX_VOL1_MASK 0x00FE /* AIF1TX2MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX2MIX_VOL1_SHIFT 1 /* AIF1TX2MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX2MIX_VOL1_WIDTH 7 /* AIF1TX2MIX_VOL1 - [7:1] */ + +/* + * R1578 (0x62A) - AIF1TX2MIX Input 2 Source + */ +#define WM2200_AIF1TX2MIX_SRC2_MASK 0x007F /* AIF1TX2MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX2MIX_SRC2_SHIFT 0 /* AIF1TX2MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX2MIX_SRC2_WIDTH 7 /* AIF1TX2MIX_SRC2 - [6:0] */ + +/* + * R1579 (0x62B) - AIF1TX2MIX Input 2 Volume + */ +#define WM2200_AIF1TX2MIX_VOL2_MASK 0x00FE /* AIF1TX2MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX2MIX_VOL2_SHIFT 1 /* AIF1TX2MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX2MIX_VOL2_WIDTH 7 /* AIF1TX2MIX_VOL2 - [7:1] */ + +/* + * R1580 (0x62C) - AIF1TX2MIX Input 3 Source + */ +#define WM2200_AIF1TX2MIX_SRC3_MASK 0x007F /* AIF1TX2MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX2MIX_SRC3_SHIFT 0 /* AIF1TX2MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX2MIX_SRC3_WIDTH 7 /* AIF1TX2MIX_SRC3 - [6:0] */ + +/* + * R1581 (0x62D) - AIF1TX2MIX Input 3 Volume + */ +#define WM2200_AIF1TX2MIX_VOL3_MASK 0x00FE /* AIF1TX2MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX2MIX_VOL3_SHIFT 1 /* AIF1TX2MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX2MIX_VOL3_WIDTH 7 /* AIF1TX2MIX_VOL3 - [7:1] */ + +/* + * R1582 (0x62E) - AIF1TX2MIX Input 4 Source + */ +#define WM2200_AIF1TX2MIX_SRC4_MASK 0x007F /* AIF1TX2MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX2MIX_SRC4_SHIFT 0 /* AIF1TX2MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX2MIX_SRC4_WIDTH 7 /* AIF1TX2MIX_SRC4 - [6:0] */ + +/* + * R1583 (0x62F) - AIF1TX2MIX Input 4 Volume + */ +#define WM2200_AIF1TX2MIX_VOL4_MASK 0x00FE /* AIF1TX2MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX2MIX_VOL4_SHIFT 1 /* AIF1TX2MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX2MIX_VOL4_WIDTH 7 /* AIF1TX2MIX_VOL4 - [7:1] */ + +/* + * R1584 (0x630) - AIF1TX3MIX Input 1 Source + */ +#define WM2200_AIF1TX3MIX_SRC1_MASK 0x007F /* AIF1TX3MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX3MIX_SRC1_SHIFT 0 /* AIF1TX3MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX3MIX_SRC1_WIDTH 7 /* AIF1TX3MIX_SRC1 - [6:0] */ + +/* + * R1585 (0x631) - AIF1TX3MIX Input 1 Volume + */ +#define WM2200_AIF1TX3MIX_VOL1_MASK 0x00FE /* AIF1TX3MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX3MIX_VOL1_SHIFT 1 /* AIF1TX3MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX3MIX_VOL1_WIDTH 7 /* AIF1TX3MIX_VOL1 - [7:1] */ + +/* + * R1586 (0x632) - AIF1TX3MIX Input 2 Source + */ +#define WM2200_AIF1TX3MIX_SRC2_MASK 0x007F /* AIF1TX3MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX3MIX_SRC2_SHIFT 0 /* AIF1TX3MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX3MIX_SRC2_WIDTH 7 /* AIF1TX3MIX_SRC2 - [6:0] */ + +/* + * R1587 (0x633) - AIF1TX3MIX Input 2 Volume + */ +#define WM2200_AIF1TX3MIX_VOL2_MASK 0x00FE /* AIF1TX3MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX3MIX_VOL2_SHIFT 1 /* AIF1TX3MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX3MIX_VOL2_WIDTH 7 /* AIF1TX3MIX_VOL2 - [7:1] */ + +/* + * R1588 (0x634) - AIF1TX3MIX Input 3 Source + */ +#define WM2200_AIF1TX3MIX_SRC3_MASK 0x007F /* AIF1TX3MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX3MIX_SRC3_SHIFT 0 /* AIF1TX3MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX3MIX_SRC3_WIDTH 7 /* AIF1TX3MIX_SRC3 - [6:0] */ + +/* + * R1589 (0x635) - AIF1TX3MIX Input 3 Volume + */ +#define WM2200_AIF1TX3MIX_VOL3_MASK 0x00FE /* AIF1TX3MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX3MIX_VOL3_SHIFT 1 /* AIF1TX3MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX3MIX_VOL3_WIDTH 7 /* AIF1TX3MIX_VOL3 - [7:1] */ + +/* + * R1590 (0x636) - AIF1TX3MIX Input 4 Source + */ +#define WM2200_AIF1TX3MIX_SRC4_MASK 0x007F /* AIF1TX3MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX3MIX_SRC4_SHIFT 0 /* AIF1TX3MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX3MIX_SRC4_WIDTH 7 /* AIF1TX3MIX_SRC4 - [6:0] */ + +/* + * R1591 (0x637) - AIF1TX3MIX Input 4 Volume + */ +#define WM2200_AIF1TX3MIX_VOL4_MASK 0x00FE /* AIF1TX3MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX3MIX_VOL4_SHIFT 1 /* AIF1TX3MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX3MIX_VOL4_WIDTH 7 /* AIF1TX3MIX_VOL4 - [7:1] */ + +/* + * R1592 (0x638) - AIF1TX4MIX Input 1 Source + */ +#define WM2200_AIF1TX4MIX_SRC1_MASK 0x007F /* AIF1TX4MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX4MIX_SRC1_SHIFT 0 /* AIF1TX4MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX4MIX_SRC1_WIDTH 7 /* AIF1TX4MIX_SRC1 - [6:0] */ + +/* + * R1593 (0x639) - AIF1TX4MIX Input 1 Volume + */ +#define WM2200_AIF1TX4MIX_VOL1_MASK 0x00FE /* AIF1TX4MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX4MIX_VOL1_SHIFT 1 /* AIF1TX4MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX4MIX_VOL1_WIDTH 7 /* AIF1TX4MIX_VOL1 - [7:1] */ + +/* + * R1594 (0x63A) - AIF1TX4MIX Input 2 Source + */ +#define WM2200_AIF1TX4MIX_SRC2_MASK 0x007F /* AIF1TX4MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX4MIX_SRC2_SHIFT 0 /* AIF1TX4MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX4MIX_SRC2_WIDTH 7 /* AIF1TX4MIX_SRC2 - [6:0] */ + +/* + * R1595 (0x63B) - AIF1TX4MIX Input 2 Volume + */ +#define WM2200_AIF1TX4MIX_VOL2_MASK 0x00FE /* AIF1TX4MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX4MIX_VOL2_SHIFT 1 /* AIF1TX4MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX4MIX_VOL2_WIDTH 7 /* AIF1TX4MIX_VOL2 - [7:1] */ + +/* + * R1596 (0x63C) - AIF1TX4MIX Input 3 Source + */ +#define WM2200_AIF1TX4MIX_SRC3_MASK 0x007F /* AIF1TX4MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX4MIX_SRC3_SHIFT 0 /* AIF1TX4MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX4MIX_SRC3_WIDTH 7 /* AIF1TX4MIX_SRC3 - [6:0] */ + +/* + * R1597 (0x63D) - AIF1TX4MIX Input 3 Volume + */ +#define WM2200_AIF1TX4MIX_VOL3_MASK 0x00FE /* AIF1TX4MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX4MIX_VOL3_SHIFT 1 /* AIF1TX4MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX4MIX_VOL3_WIDTH 7 /* AIF1TX4MIX_VOL3 - [7:1] */ + +/* + * R1598 (0x63E) - AIF1TX4MIX Input 4 Source + */ +#define WM2200_AIF1TX4MIX_SRC4_MASK 0x007F /* AIF1TX4MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX4MIX_SRC4_SHIFT 0 /* AIF1TX4MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX4MIX_SRC4_WIDTH 7 /* AIF1TX4MIX_SRC4 - [6:0] */ + +/* + * R1599 (0x63F) - AIF1TX4MIX Input 4 Volume + */ +#define WM2200_AIF1TX4MIX_VOL4_MASK 0x00FE /* AIF1TX4MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX4MIX_VOL4_SHIFT 1 /* AIF1TX4MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX4MIX_VOL4_WIDTH 7 /* AIF1TX4MIX_VOL4 - [7:1] */ + +/* + * R1600 (0x640) - AIF1TX5MIX Input 1 Source + */ +#define WM2200_AIF1TX5MIX_SRC1_MASK 0x007F /* AIF1TX5MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX5MIX_SRC1_SHIFT 0 /* AIF1TX5MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX5MIX_SRC1_WIDTH 7 /* AIF1TX5MIX_SRC1 - [6:0] */ + +/* + * R1601 (0x641) - AIF1TX5MIX Input 1 Volume + */ +#define WM2200_AIF1TX5MIX_VOL1_MASK 0x00FE /* AIF1TX5MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX5MIX_VOL1_SHIFT 1 /* AIF1TX5MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX5MIX_VOL1_WIDTH 7 /* AIF1TX5MIX_VOL1 - [7:1] */ + +/* + * R1602 (0x642) - AIF1TX5MIX Input 2 Source + */ +#define WM2200_AIF1TX5MIX_SRC2_MASK 0x007F /* AIF1TX5MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX5MIX_SRC2_SHIFT 0 /* AIF1TX5MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX5MIX_SRC2_WIDTH 7 /* AIF1TX5MIX_SRC2 - [6:0] */ + +/* + * R1603 (0x643) - AIF1TX5MIX Input 2 Volume + */ +#define WM2200_AIF1TX5MIX_VOL2_MASK 0x00FE /* AIF1TX5MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX5MIX_VOL2_SHIFT 1 /* AIF1TX5MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX5MIX_VOL2_WIDTH 7 /* AIF1TX5MIX_VOL2 - [7:1] */ + +/* + * R1604 (0x644) - AIF1TX5MIX Input 3 Source + */ +#define WM2200_AIF1TX5MIX_SRC3_MASK 0x007F /* AIF1TX5MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX5MIX_SRC3_SHIFT 0 /* AIF1TX5MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX5MIX_SRC3_WIDTH 7 /* AIF1TX5MIX_SRC3 - [6:0] */ + +/* + * R1605 (0x645) - AIF1TX5MIX Input 3 Volume + */ +#define WM2200_AIF1TX5MIX_VOL3_MASK 0x00FE /* AIF1TX5MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX5MIX_VOL3_SHIFT 1 /* AIF1TX5MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX5MIX_VOL3_WIDTH 7 /* AIF1TX5MIX_VOL3 - [7:1] */ + +/* + * R1606 (0x646) - AIF1TX5MIX Input 4 Source + */ +#define WM2200_AIF1TX5MIX_SRC4_MASK 0x007F /* AIF1TX5MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX5MIX_SRC4_SHIFT 0 /* AIF1TX5MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX5MIX_SRC4_WIDTH 7 /* AIF1TX5MIX_SRC4 - [6:0] */ + +/* + * R1607 (0x647) - AIF1TX5MIX Input 4 Volume + */ +#define WM2200_AIF1TX5MIX_VOL4_MASK 0x00FE /* AIF1TX5MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX5MIX_VOL4_SHIFT 1 /* AIF1TX5MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX5MIX_VOL4_WIDTH 7 /* AIF1TX5MIX_VOL4 - [7:1] */ + +/* + * R1608 (0x648) - AIF1TX6MIX Input 1 Source + */ +#define WM2200_AIF1TX6MIX_SRC1_MASK 0x007F /* AIF1TX6MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX6MIX_SRC1_SHIFT 0 /* AIF1TX6MIX_SRC1 - [6:0] */ +#define WM2200_AIF1TX6MIX_SRC1_WIDTH 7 /* AIF1TX6MIX_SRC1 - [6:0] */ + +/* + * R1609 (0x649) - AIF1TX6MIX Input 1 Volume + */ +#define WM2200_AIF1TX6MIX_VOL1_MASK 0x00FE /* AIF1TX6MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX6MIX_VOL1_SHIFT 1 /* AIF1TX6MIX_VOL1 - [7:1] */ +#define WM2200_AIF1TX6MIX_VOL1_WIDTH 7 /* AIF1TX6MIX_VOL1 - [7:1] */ + +/* + * R1610 (0x64A) - AIF1TX6MIX Input 2 Source + */ +#define WM2200_AIF1TX6MIX_SRC2_MASK 0x007F /* AIF1TX6MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX6MIX_SRC2_SHIFT 0 /* AIF1TX6MIX_SRC2 - [6:0] */ +#define WM2200_AIF1TX6MIX_SRC2_WIDTH 7 /* AIF1TX6MIX_SRC2 - [6:0] */ + +/* + * R1611 (0x64B) - AIF1TX6MIX Input 2 Volume + */ +#define WM2200_AIF1TX6MIX_VOL2_MASK 0x00FE /* AIF1TX6MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX6MIX_VOL2_SHIFT 1 /* AIF1TX6MIX_VOL2 - [7:1] */ +#define WM2200_AIF1TX6MIX_VOL2_WIDTH 7 /* AIF1TX6MIX_VOL2 - [7:1] */ + +/* + * R1612 (0x64C) - AIF1TX6MIX Input 3 Source + */ +#define WM2200_AIF1TX6MIX_SRC3_MASK 0x007F /* AIF1TX6MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX6MIX_SRC3_SHIFT 0 /* AIF1TX6MIX_SRC3 - [6:0] */ +#define WM2200_AIF1TX6MIX_SRC3_WIDTH 7 /* AIF1TX6MIX_SRC3 - [6:0] */ + +/* + * R1613 (0x64D) - AIF1TX6MIX Input 3 Volume + */ +#define WM2200_AIF1TX6MIX_VOL3_MASK 0x00FE /* AIF1TX6MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX6MIX_VOL3_SHIFT 1 /* AIF1TX6MIX_VOL3 - [7:1] */ +#define WM2200_AIF1TX6MIX_VOL3_WIDTH 7 /* AIF1TX6MIX_VOL3 - [7:1] */ + +/* + * R1614 (0x64E) - AIF1TX6MIX Input 4 Source + */ +#define WM2200_AIF1TX6MIX_SRC4_MASK 0x007F /* AIF1TX6MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX6MIX_SRC4_SHIFT 0 /* AIF1TX6MIX_SRC4 - [6:0] */ +#define WM2200_AIF1TX6MIX_SRC4_WIDTH 7 /* AIF1TX6MIX_SRC4 - [6:0] */ + +/* + * R1615 (0x64F) - AIF1TX6MIX Input 4 Volume + */ +#define WM2200_AIF1TX6MIX_VOL4_MASK 0x00FE /* AIF1TX6MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX6MIX_VOL4_SHIFT 1 /* AIF1TX6MIX_VOL4 - [7:1] */ +#define WM2200_AIF1TX6MIX_VOL4_WIDTH 7 /* AIF1TX6MIX_VOL4 - [7:1] */ + +/* + * R1616 (0x650) - EQLMIX Input 1 Source + */ +#define WM2200_EQLMIX_SRC1_MASK 0x007F /* EQLMIX_SRC1 - [6:0] */ +#define WM2200_EQLMIX_SRC1_SHIFT 0 /* EQLMIX_SRC1 - [6:0] */ +#define WM2200_EQLMIX_SRC1_WIDTH 7 /* EQLMIX_SRC1 - [6:0] */ + +/* + * R1617 (0x651) - EQLMIX Input 1 Volume + */ +#define WM2200_EQLMIX_VOL1_MASK 0x00FE /* EQLMIX_VOL1 - [7:1] */ +#define WM2200_EQLMIX_VOL1_SHIFT 1 /* EQLMIX_VOL1 - [7:1] */ +#define WM2200_EQLMIX_VOL1_WIDTH 7 /* EQLMIX_VOL1 - [7:1] */ + +/* + * R1618 (0x652) - EQLMIX Input 2 Source + */ +#define WM2200_EQLMIX_SRC2_MASK 0x007F /* EQLMIX_SRC2 - [6:0] */ +#define WM2200_EQLMIX_SRC2_SHIFT 0 /* EQLMIX_SRC2 - [6:0] */ +#define WM2200_EQLMIX_SRC2_WIDTH 7 /* EQLMIX_SRC2 - [6:0] */ + +/* + * R1619 (0x653) - EQLMIX Input 2 Volume + */ +#define WM2200_EQLMIX_VOL2_MASK 0x00FE /* EQLMIX_VOL2 - [7:1] */ +#define WM2200_EQLMIX_VOL2_SHIFT 1 /* EQLMIX_VOL2 - [7:1] */ +#define WM2200_EQLMIX_VOL2_WIDTH 7 /* EQLMIX_VOL2 - [7:1] */ + +/* + * R1620 (0x654) - EQLMIX Input 3 Source + */ +#define WM2200_EQLMIX_SRC3_MASK 0x007F /* EQLMIX_SRC3 - [6:0] */ +#define WM2200_EQLMIX_SRC3_SHIFT 0 /* EQLMIX_SRC3 - [6:0] */ +#define WM2200_EQLMIX_SRC3_WIDTH 7 /* EQLMIX_SRC3 - [6:0] */ + +/* + * R1621 (0x655) - EQLMIX Input 3 Volume + */ +#define WM2200_EQLMIX_VOL3_MASK 0x00FE /* EQLMIX_VOL3 - [7:1] */ +#define WM2200_EQLMIX_VOL3_SHIFT 1 /* EQLMIX_VOL3 - [7:1] */ +#define WM2200_EQLMIX_VOL3_WIDTH 7 /* EQLMIX_VOL3 - [7:1] */ + +/* + * R1622 (0x656) - EQLMIX Input 4 Source + */ +#define WM2200_EQLMIX_SRC4_MASK 0x007F /* EQLMIX_SRC4 - [6:0] */ +#define WM2200_EQLMIX_SRC4_SHIFT 0 /* EQLMIX_SRC4 - [6:0] */ +#define WM2200_EQLMIX_SRC4_WIDTH 7 /* EQLMIX_SRC4 - [6:0] */ + +/* + * R1623 (0x657) - EQLMIX Input 4 Volume + */ +#define WM2200_EQLMIX_VOL4_MASK 0x00FE /* EQLMIX_VOL4 - [7:1] */ +#define WM2200_EQLMIX_VOL4_SHIFT 1 /* EQLMIX_VOL4 - [7:1] */ +#define WM2200_EQLMIX_VOL4_WIDTH 7 /* EQLMIX_VOL4 - [7:1] */ + +/* + * R1624 (0x658) - EQRMIX Input 1 Source + */ +#define WM2200_EQRMIX_SRC1_MASK 0x007F /* EQRMIX_SRC1 - [6:0] */ +#define WM2200_EQRMIX_SRC1_SHIFT 0 /* EQRMIX_SRC1 - [6:0] */ +#define WM2200_EQRMIX_SRC1_WIDTH 7 /* EQRMIX_SRC1 - [6:0] */ + +/* + * R1625 (0x659) - EQRMIX Input 1 Volume + */ +#define WM2200_EQRMIX_VOL1_MASK 0x00FE /* EQRMIX_VOL1 - [7:1] */ +#define WM2200_EQRMIX_VOL1_SHIFT 1 /* EQRMIX_VOL1 - [7:1] */ +#define WM2200_EQRMIX_VOL1_WIDTH 7 /* EQRMIX_VOL1 - [7:1] */ + +/* + * R1626 (0x65A) - EQRMIX Input 2 Source + */ +#define WM2200_EQRMIX_SRC2_MASK 0x007F /* EQRMIX_SRC2 - [6:0] */ +#define WM2200_EQRMIX_SRC2_SHIFT 0 /* EQRMIX_SRC2 - [6:0] */ +#define WM2200_EQRMIX_SRC2_WIDTH 7 /* EQRMIX_SRC2 - [6:0] */ + +/* + * R1627 (0x65B) - EQRMIX Input 2 Volume + */ +#define WM2200_EQRMIX_VOL2_MASK 0x00FE /* EQRMIX_VOL2 - [7:1] */ +#define WM2200_EQRMIX_VOL2_SHIFT 1 /* EQRMIX_VOL2 - [7:1] */ +#define WM2200_EQRMIX_VOL2_WIDTH 7 /* EQRMIX_VOL2 - [7:1] */ + +/* + * R1628 (0x65C) - EQRMIX Input 3 Source + */ +#define WM2200_EQRMIX_SRC3_MASK 0x007F /* EQRMIX_SRC3 - [6:0] */ +#define WM2200_EQRMIX_SRC3_SHIFT 0 /* EQRMIX_SRC3 - [6:0] */ +#define WM2200_EQRMIX_SRC3_WIDTH 7 /* EQRMIX_SRC3 - [6:0] */ + +/* + * R1629 (0x65D) - EQRMIX Input 3 Volume + */ +#define WM2200_EQRMIX_VOL3_MASK 0x00FE /* EQRMIX_VOL3 - [7:1] */ +#define WM2200_EQRMIX_VOL3_SHIFT 1 /* EQRMIX_VOL3 - [7:1] */ +#define WM2200_EQRMIX_VOL3_WIDTH 7 /* EQRMIX_VOL3 - [7:1] */ + +/* + * R1630 (0x65E) - EQRMIX Input 4 Source + */ +#define WM2200_EQRMIX_SRC4_MASK 0x007F /* EQRMIX_SRC4 - [6:0] */ +#define WM2200_EQRMIX_SRC4_SHIFT 0 /* EQRMIX_SRC4 - [6:0] */ +#define WM2200_EQRMIX_SRC4_WIDTH 7 /* EQRMIX_SRC4 - [6:0] */ + +/* + * R1631 (0x65F) - EQRMIX Input 4 Volume + */ +#define WM2200_EQRMIX_VOL4_MASK 0x00FE /* EQRMIX_VOL4 - [7:1] */ +#define WM2200_EQRMIX_VOL4_SHIFT 1 /* EQRMIX_VOL4 - [7:1] */ +#define WM2200_EQRMIX_VOL4_WIDTH 7 /* EQRMIX_VOL4 - [7:1] */ + +/* + * R1632 (0x660) - LHPF1MIX Input 1 Source + */ +#define WM2200_LHPF1MIX_SRC1_MASK 0x007F /* LHPF1MIX_SRC1 - [6:0] */ +#define WM2200_LHPF1MIX_SRC1_SHIFT 0 /* LHPF1MIX_SRC1 - [6:0] */ +#define WM2200_LHPF1MIX_SRC1_WIDTH 7 /* LHPF1MIX_SRC1 - [6:0] */ + +/* + * R1633 (0x661) - LHPF1MIX Input 1 Volume + */ +#define WM2200_LHPF1MIX_VOL1_MASK 0x00FE /* LHPF1MIX_VOL1 - [7:1] */ +#define WM2200_LHPF1MIX_VOL1_SHIFT 1 /* LHPF1MIX_VOL1 - [7:1] */ +#define WM2200_LHPF1MIX_VOL1_WIDTH 7 /* LHPF1MIX_VOL1 - [7:1] */ + +/* + * R1634 (0x662) - LHPF1MIX Input 2 Source + */ +#define WM2200_LHPF1MIX_SRC2_MASK 0x007F /* LHPF1MIX_SRC2 - [6:0] */ +#define WM2200_LHPF1MIX_SRC2_SHIFT 0 /* LHPF1MIX_SRC2 - [6:0] */ +#define WM2200_LHPF1MIX_SRC2_WIDTH 7 /* LHPF1MIX_SRC2 - [6:0] */ + +/* + * R1635 (0x663) - LHPF1MIX Input 2 Volume + */ +#define WM2200_LHPF1MIX_VOL2_MASK 0x00FE /* LHPF1MIX_VOL2 - [7:1] */ +#define WM2200_LHPF1MIX_VOL2_SHIFT 1 /* LHPF1MIX_VOL2 - [7:1] */ +#define WM2200_LHPF1MIX_VOL2_WIDTH 7 /* LHPF1MIX_VOL2 - [7:1] */ + +/* + * R1636 (0x664) - LHPF1MIX Input 3 Source + */ +#define WM2200_LHPF1MIX_SRC3_MASK 0x007F /* LHPF1MIX_SRC3 - [6:0] */ +#define WM2200_LHPF1MIX_SRC3_SHIFT 0 /* LHPF1MIX_SRC3 - [6:0] */ +#define WM2200_LHPF1MIX_SRC3_WIDTH 7 /* LHPF1MIX_SRC3 - [6:0] */ + +/* + * R1637 (0x665) - LHPF1MIX Input 3 Volume + */ +#define WM2200_LHPF1MIX_VOL3_MASK 0x00FE /* LHPF1MIX_VOL3 - [7:1] */ +#define WM2200_LHPF1MIX_VOL3_SHIFT 1 /* LHPF1MIX_VOL3 - [7:1] */ +#define WM2200_LHPF1MIX_VOL3_WIDTH 7 /* LHPF1MIX_VOL3 - [7:1] */ + +/* + * R1638 (0x666) - LHPF1MIX Input 4 Source + */ +#define WM2200_LHPF1MIX_SRC4_MASK 0x007F /* LHPF1MIX_SRC4 - [6:0] */ +#define WM2200_LHPF1MIX_SRC4_SHIFT 0 /* LHPF1MIX_SRC4 - [6:0] */ +#define WM2200_LHPF1MIX_SRC4_WIDTH 7 /* LHPF1MIX_SRC4 - [6:0] */ + +/* + * R1639 (0x667) - LHPF1MIX Input 4 Volume + */ +#define WM2200_LHPF1MIX_VOL4_MASK 0x00FE /* LHPF1MIX_VOL4 - [7:1] */ +#define WM2200_LHPF1MIX_VOL4_SHIFT 1 /* LHPF1MIX_VOL4 - [7:1] */ +#define WM2200_LHPF1MIX_VOL4_WIDTH 7 /* LHPF1MIX_VOL4 - [7:1] */ + +/* + * R1640 (0x668) - LHPF2MIX Input 1 Source + */ +#define WM2200_LHPF2MIX_SRC1_MASK 0x007F /* LHPF2MIX_SRC1 - [6:0] */ +#define WM2200_LHPF2MIX_SRC1_SHIFT 0 /* LHPF2MIX_SRC1 - [6:0] */ +#define WM2200_LHPF2MIX_SRC1_WIDTH 7 /* LHPF2MIX_SRC1 - [6:0] */ + +/* + * R1641 (0x669) - LHPF2MIX Input 1 Volume + */ +#define WM2200_LHPF2MIX_VOL1_MASK 0x00FE /* LHPF2MIX_VOL1 - [7:1] */ +#define WM2200_LHPF2MIX_VOL1_SHIFT 1 /* LHPF2MIX_VOL1 - [7:1] */ +#define WM2200_LHPF2MIX_VOL1_WIDTH 7 /* LHPF2MIX_VOL1 - [7:1] */ + +/* + * R1642 (0x66A) - LHPF2MIX Input 2 Source + */ +#define WM2200_LHPF2MIX_SRC2_MASK 0x007F /* LHPF2MIX_SRC2 - [6:0] */ +#define WM2200_LHPF2MIX_SRC2_SHIFT 0 /* LHPF2MIX_SRC2 - [6:0] */ +#define WM2200_LHPF2MIX_SRC2_WIDTH 7 /* LHPF2MIX_SRC2 - [6:0] */ + +/* + * R1643 (0x66B) - LHPF2MIX Input 2 Volume + */ +#define WM2200_LHPF2MIX_VOL2_MASK 0x00FE /* LHPF2MIX_VOL2 - [7:1] */ +#define WM2200_LHPF2MIX_VOL2_SHIFT 1 /* LHPF2MIX_VOL2 - [7:1] */ +#define WM2200_LHPF2MIX_VOL2_WIDTH 7 /* LHPF2MIX_VOL2 - [7:1] */ + +/* + * R1644 (0x66C) - LHPF2MIX Input 3 Source + */ +#define WM2200_LHPF2MIX_SRC3_MASK 0x007F /* LHPF2MIX_SRC3 - [6:0] */ +#define WM2200_LHPF2MIX_SRC3_SHIFT 0 /* LHPF2MIX_SRC3 - [6:0] */ +#define WM2200_LHPF2MIX_SRC3_WIDTH 7 /* LHPF2MIX_SRC3 - [6:0] */ + +/* + * R1645 (0x66D) - LHPF2MIX Input 3 Volume + */ +#define WM2200_LHPF2MIX_VOL3_MASK 0x00FE /* LHPF2MIX_VOL3 - [7:1] */ +#define WM2200_LHPF2MIX_VOL3_SHIFT 1 /* LHPF2MIX_VOL3 - [7:1] */ +#define WM2200_LHPF2MIX_VOL3_WIDTH 7 /* LHPF2MIX_VOL3 - [7:1] */ + +/* + * R1646 (0x66E) - LHPF2MIX Input 4 Source + */ +#define WM2200_LHPF2MIX_SRC4_MASK 0x007F /* LHPF2MIX_SRC4 - [6:0] */ +#define WM2200_LHPF2MIX_SRC4_SHIFT 0 /* LHPF2MIX_SRC4 - [6:0] */ +#define WM2200_LHPF2MIX_SRC4_WIDTH 7 /* LHPF2MIX_SRC4 - [6:0] */ + +/* + * R1647 (0x66F) - LHPF2MIX Input 4 Volume + */ +#define WM2200_LHPF2MIX_VOL4_MASK 0x00FE /* LHPF2MIX_VOL4 - [7:1] */ +#define WM2200_LHPF2MIX_VOL4_SHIFT 1 /* LHPF2MIX_VOL4 - [7:1] */ +#define WM2200_LHPF2MIX_VOL4_WIDTH 7 /* LHPF2MIX_VOL4 - [7:1] */ + +/* + * R1648 (0x670) - DSP1LMIX Input 1 Source + */ +#define WM2200_DSP1LMIX_SRC1_MASK 0x007F /* DSP1LMIX_SRC1 - [6:0] */ +#define WM2200_DSP1LMIX_SRC1_SHIFT 0 /* DSP1LMIX_SRC1 - [6:0] */ +#define WM2200_DSP1LMIX_SRC1_WIDTH 7 /* DSP1LMIX_SRC1 - [6:0] */ + +/* + * R1649 (0x671) - DSP1LMIX Input 1 Volume + */ +#define WM2200_DSP1LMIX_VOL1_MASK 0x00FE /* DSP1LMIX_VOL1 - [7:1] */ +#define WM2200_DSP1LMIX_VOL1_SHIFT 1 /* DSP1LMIX_VOL1 - [7:1] */ +#define WM2200_DSP1LMIX_VOL1_WIDTH 7 /* DSP1LMIX_VOL1 - [7:1] */ + +/* + * R1650 (0x672) - DSP1LMIX Input 2 Source + */ +#define WM2200_DSP1LMIX_SRC2_MASK 0x007F /* DSP1LMIX_SRC2 - [6:0] */ +#define WM2200_DSP1LMIX_SRC2_SHIFT 0 /* DSP1LMIX_SRC2 - [6:0] */ +#define WM2200_DSP1LMIX_SRC2_WIDTH 7 /* DSP1LMIX_SRC2 - [6:0] */ + +/* + * R1651 (0x673) - DSP1LMIX Input 2 Volume + */ +#define WM2200_DSP1LMIX_VOL2_MASK 0x00FE /* DSP1LMIX_VOL2 - [7:1] */ +#define WM2200_DSP1LMIX_VOL2_SHIFT 1 /* DSP1LMIX_VOL2 - [7:1] */ +#define WM2200_DSP1LMIX_VOL2_WIDTH 7 /* DSP1LMIX_VOL2 - [7:1] */ + +/* + * R1652 (0x674) - DSP1LMIX Input 3 Source + */ +#define WM2200_DSP1LMIX_SRC3_MASK 0x007F /* DSP1LMIX_SRC3 - [6:0] */ +#define WM2200_DSP1LMIX_SRC3_SHIFT 0 /* DSP1LMIX_SRC3 - [6:0] */ +#define WM2200_DSP1LMIX_SRC3_WIDTH 7 /* DSP1LMIX_SRC3 - [6:0] */ + +/* + * R1653 (0x675) - DSP1LMIX Input 3 Volume + */ +#define WM2200_DSP1LMIX_VOL3_MASK 0x00FE /* DSP1LMIX_VOL3 - [7:1] */ +#define WM2200_DSP1LMIX_VOL3_SHIFT 1 /* DSP1LMIX_VOL3 - [7:1] */ +#define WM2200_DSP1LMIX_VOL3_WIDTH 7 /* DSP1LMIX_VOL3 - [7:1] */ + +/* + * R1654 (0x676) - DSP1LMIX Input 4 Source + */ +#define WM2200_DSP1LMIX_SRC4_MASK 0x007F /* DSP1LMIX_SRC4 - [6:0] */ +#define WM2200_DSP1LMIX_SRC4_SHIFT 0 /* DSP1LMIX_SRC4 - [6:0] */ +#define WM2200_DSP1LMIX_SRC4_WIDTH 7 /* DSP1LMIX_SRC4 - [6:0] */ + +/* + * R1655 (0x677) - DSP1LMIX Input 4 Volume + */ +#define WM2200_DSP1LMIX_VOL4_MASK 0x00FE /* DSP1LMIX_VOL4 - [7:1] */ +#define WM2200_DSP1LMIX_VOL4_SHIFT 1 /* DSP1LMIX_VOL4 - [7:1] */ +#define WM2200_DSP1LMIX_VOL4_WIDTH 7 /* DSP1LMIX_VOL4 - [7:1] */ + +/* + * R1656 (0x678) - DSP1RMIX Input 1 Source + */ +#define WM2200_DSP1RMIX_SRC1_MASK 0x007F /* DSP1RMIX_SRC1 - [6:0] */ +#define WM2200_DSP1RMIX_SRC1_SHIFT 0 /* DSP1RMIX_SRC1 - [6:0] */ +#define WM2200_DSP1RMIX_SRC1_WIDTH 7 /* DSP1RMIX_SRC1 - [6:0] */ + +/* + * R1657 (0x679) - DSP1RMIX Input 1 Volume + */ +#define WM2200_DSP1RMIX_VOL1_MASK 0x00FE /* DSP1RMIX_VOL1 - [7:1] */ +#define WM2200_DSP1RMIX_VOL1_SHIFT 1 /* DSP1RMIX_VOL1 - [7:1] */ +#define WM2200_DSP1RMIX_VOL1_WIDTH 7 /* DSP1RMIX_VOL1 - [7:1] */ + +/* + * R1658 (0x67A) - DSP1RMIX Input 2 Source + */ +#define WM2200_DSP1RMIX_SRC2_MASK 0x007F /* DSP1RMIX_SRC2 - [6:0] */ +#define WM2200_DSP1RMIX_SRC2_SHIFT 0 /* DSP1RMIX_SRC2 - [6:0] */ +#define WM2200_DSP1RMIX_SRC2_WIDTH 7 /* DSP1RMIX_SRC2 - [6:0] */ + +/* + * R1659 (0x67B) - DSP1RMIX Input 2 Volume + */ +#define WM2200_DSP1RMIX_VOL2_MASK 0x00FE /* DSP1RMIX_VOL2 - [7:1] */ +#define WM2200_DSP1RMIX_VOL2_SHIFT 1 /* DSP1RMIX_VOL2 - [7:1] */ +#define WM2200_DSP1RMIX_VOL2_WIDTH 7 /* DSP1RMIX_VOL2 - [7:1] */ + +/* + * R1660 (0x67C) - DSP1RMIX Input 3 Source + */ +#define WM2200_DSP1RMIX_SRC3_MASK 0x007F /* DSP1RMIX_SRC3 - [6:0] */ +#define WM2200_DSP1RMIX_SRC3_SHIFT 0 /* DSP1RMIX_SRC3 - [6:0] */ +#define WM2200_DSP1RMIX_SRC3_WIDTH 7 /* DSP1RMIX_SRC3 - [6:0] */ + +/* + * R1661 (0x67D) - DSP1RMIX Input 3 Volume + */ +#define WM2200_DSP1RMIX_VOL3_MASK 0x00FE /* DSP1RMIX_VOL3 - [7:1] */ +#define WM2200_DSP1RMIX_VOL3_SHIFT 1 /* DSP1RMIX_VOL3 - [7:1] */ +#define WM2200_DSP1RMIX_VOL3_WIDTH 7 /* DSP1RMIX_VOL3 - [7:1] */ + +/* + * R1662 (0x67E) - DSP1RMIX Input 4 Source + */ +#define WM2200_DSP1RMIX_SRC4_MASK 0x007F /* DSP1RMIX_SRC4 - [6:0] */ +#define WM2200_DSP1RMIX_SRC4_SHIFT 0 /* DSP1RMIX_SRC4 - [6:0] */ +#define WM2200_DSP1RMIX_SRC4_WIDTH 7 /* DSP1RMIX_SRC4 - [6:0] */ + +/* + * R1663 (0x67F) - DSP1RMIX Input 4 Volume + */ +#define WM2200_DSP1RMIX_VOL4_MASK 0x00FE /* DSP1RMIX_VOL4 - [7:1] */ +#define WM2200_DSP1RMIX_VOL4_SHIFT 1 /* DSP1RMIX_VOL4 - [7:1] */ +#define WM2200_DSP1RMIX_VOL4_WIDTH 7 /* DSP1RMIX_VOL4 - [7:1] */ + +/* + * R1664 (0x680) - DSP1AUX1MIX Input 1 Source + */ +#define WM2200_DSP1AUX1MIX_SRC1_MASK 0x007F /* DSP1AUX1MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX1MIX_SRC1_SHIFT 0 /* DSP1AUX1MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX1MIX_SRC1_WIDTH 7 /* DSP1AUX1MIX_SRC1 - [6:0] */ + +/* + * R1665 (0x681) - DSP1AUX2MIX Input 1 Source + */ +#define WM2200_DSP1AUX2MIX_SRC1_MASK 0x007F /* DSP1AUX2MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX2MIX_SRC1_SHIFT 0 /* DSP1AUX2MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX2MIX_SRC1_WIDTH 7 /* DSP1AUX2MIX_SRC1 - [6:0] */ + +/* + * R1666 (0x682) - DSP1AUX3MIX Input 1 Source + */ +#define WM2200_DSP1AUX3MIX_SRC1_MASK 0x007F /* DSP1AUX3MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX3MIX_SRC1_SHIFT 0 /* DSP1AUX3MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX3MIX_SRC1_WIDTH 7 /* DSP1AUX3MIX_SRC1 - [6:0] */ + +/* + * R1667 (0x683) - DSP1AUX4MIX Input 1 Source + */ +#define WM2200_DSP1AUX4MIX_SRC1_MASK 0x007F /* DSP1AUX4MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX4MIX_SRC1_SHIFT 0 /* DSP1AUX4MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX4MIX_SRC1_WIDTH 7 /* DSP1AUX4MIX_SRC1 - [6:0] */ + +/* + * R1668 (0x684) - DSP1AUX5MIX Input 1 Source + */ +#define WM2200_DSP1AUX5MIX_SRC1_MASK 0x007F /* DSP1AUX5MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX5MIX_SRC1_SHIFT 0 /* DSP1AUX5MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX5MIX_SRC1_WIDTH 7 /* DSP1AUX5MIX_SRC1 - [6:0] */ + +/* + * R1669 (0x685) - DSP1AUX6MIX Input 1 Source + */ +#define WM2200_DSP1AUX6MIX_SRC1_MASK 0x007F /* DSP1AUX6MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX6MIX_SRC1_SHIFT 0 /* DSP1AUX6MIX_SRC1 - [6:0] */ +#define WM2200_DSP1AUX6MIX_SRC1_WIDTH 7 /* DSP1AUX6MIX_SRC1 - [6:0] */ + +/* + * R1670 (0x686) - DSP2LMIX Input 1 Source + */ +#define WM2200_DSP2LMIX_SRC1_MASK 0x007F /* DSP2LMIX_SRC1 - [6:0] */ +#define WM2200_DSP2LMIX_SRC1_SHIFT 0 /* DSP2LMIX_SRC1 - [6:0] */ +#define WM2200_DSP2LMIX_SRC1_WIDTH 7 /* DSP2LMIX_SRC1 - [6:0] */ + +/* + * R1671 (0x687) - DSP2LMIX Input 1 Volume + */ +#define WM2200_DSP2LMIX_VOL1_MASK 0x00FE /* DSP2LMIX_VOL1 - [7:1] */ +#define WM2200_DSP2LMIX_VOL1_SHIFT 1 /* DSP2LMIX_VOL1 - [7:1] */ +#define WM2200_DSP2LMIX_VOL1_WIDTH 7 /* DSP2LMIX_VOL1 - [7:1] */ + +/* + * R1672 (0x688) - DSP2LMIX Input 2 Source + */ +#define WM2200_DSP2LMIX_SRC2_MASK 0x007F /* DSP2LMIX_SRC2 - [6:0] */ +#define WM2200_DSP2LMIX_SRC2_SHIFT 0 /* DSP2LMIX_SRC2 - [6:0] */ +#define WM2200_DSP2LMIX_SRC2_WIDTH 7 /* DSP2LMIX_SRC2 - [6:0] */ + +/* + * R1673 (0x689) - DSP2LMIX Input 2 Volume + */ +#define WM2200_DSP2LMIX_VOL2_MASK 0x00FE /* DSP2LMIX_VOL2 - [7:1] */ +#define WM2200_DSP2LMIX_VOL2_SHIFT 1 /* DSP2LMIX_VOL2 - [7:1] */ +#define WM2200_DSP2LMIX_VOL2_WIDTH 7 /* DSP2LMIX_VOL2 - [7:1] */ + +/* + * R1674 (0x68A) - DSP2LMIX Input 3 Source + */ +#define WM2200_DSP2LMIX_SRC3_MASK 0x007F /* DSP2LMIX_SRC3 - [6:0] */ +#define WM2200_DSP2LMIX_SRC3_SHIFT 0 /* DSP2LMIX_SRC3 - [6:0] */ +#define WM2200_DSP2LMIX_SRC3_WIDTH 7 /* DSP2LMIX_SRC3 - [6:0] */ + +/* + * R1675 (0x68B) - DSP2LMIX Input 3 Volume + */ +#define WM2200_DSP2LMIX_VOL3_MASK 0x00FE /* DSP2LMIX_VOL3 - [7:1] */ +#define WM2200_DSP2LMIX_VOL3_SHIFT 1 /* DSP2LMIX_VOL3 - [7:1] */ +#define WM2200_DSP2LMIX_VOL3_WIDTH 7 /* DSP2LMIX_VOL3 - [7:1] */ + +/* + * R1676 (0x68C) - DSP2LMIX Input 4 Source + */ +#define WM2200_DSP2LMIX_SRC4_MASK 0x007F /* DSP2LMIX_SRC4 - [6:0] */ +#define WM2200_DSP2LMIX_SRC4_SHIFT 0 /* DSP2LMIX_SRC4 - [6:0] */ +#define WM2200_DSP2LMIX_SRC4_WIDTH 7 /* DSP2LMIX_SRC4 - [6:0] */ + +/* + * R1677 (0x68D) - DSP2LMIX Input 4 Volume + */ +#define WM2200_DSP2LMIX_VOL4_MASK 0x00FE /* DSP2LMIX_VOL4 - [7:1] */ +#define WM2200_DSP2LMIX_VOL4_SHIFT 1 /* DSP2LMIX_VOL4 - [7:1] */ +#define WM2200_DSP2LMIX_VOL4_WIDTH 7 /* DSP2LMIX_VOL4 - [7:1] */ + +/* + * R1678 (0x68E) - DSP2RMIX Input 1 Source + */ +#define WM2200_DSP2RMIX_SRC1_MASK 0x007F /* DSP2RMIX_SRC1 - [6:0] */ +#define WM2200_DSP2RMIX_SRC1_SHIFT 0 /* DSP2RMIX_SRC1 - [6:0] */ +#define WM2200_DSP2RMIX_SRC1_WIDTH 7 /* DSP2RMIX_SRC1 - [6:0] */ + +/* + * R1679 (0x68F) - DSP2RMIX Input 1 Volume + */ +#define WM2200_DSP2RMIX_VOL1_MASK 0x00FE /* DSP2RMIX_VOL1 - [7:1] */ +#define WM2200_DSP2RMIX_VOL1_SHIFT 1 /* DSP2RMIX_VOL1 - [7:1] */ +#define WM2200_DSP2RMIX_VOL1_WIDTH 7 /* DSP2RMIX_VOL1 - [7:1] */ + +/* + * R1680 (0x690) - DSP2RMIX Input 2 Source + */ +#define WM2200_DSP2RMIX_SRC2_MASK 0x007F /* DSP2RMIX_SRC2 - [6:0] */ +#define WM2200_DSP2RMIX_SRC2_SHIFT 0 /* DSP2RMIX_SRC2 - [6:0] */ +#define WM2200_DSP2RMIX_SRC2_WIDTH 7 /* DSP2RMIX_SRC2 - [6:0] */ + +/* + * R1681 (0x691) - DSP2RMIX Input 2 Volume + */ +#define WM2200_DSP2RMIX_VOL2_MASK 0x00FE /* DSP2RMIX_VOL2 - [7:1] */ +#define WM2200_DSP2RMIX_VOL2_SHIFT 1 /* DSP2RMIX_VOL2 - [7:1] */ +#define WM2200_DSP2RMIX_VOL2_WIDTH 7 /* DSP2RMIX_VOL2 - [7:1] */ + +/* + * R1682 (0x692) - DSP2RMIX Input 3 Source + */ +#define WM2200_DSP2RMIX_SRC3_MASK 0x007F /* DSP2RMIX_SRC3 - [6:0] */ +#define WM2200_DSP2RMIX_SRC3_SHIFT 0 /* DSP2RMIX_SRC3 - [6:0] */ +#define WM2200_DSP2RMIX_SRC3_WIDTH 7 /* DSP2RMIX_SRC3 - [6:0] */ + +/* + * R1683 (0x693) - DSP2RMIX Input 3 Volume + */ +#define WM2200_DSP2RMIX_VOL3_MASK 0x00FE /* DSP2RMIX_VOL3 - [7:1] */ +#define WM2200_DSP2RMIX_VOL3_SHIFT 1 /* DSP2RMIX_VOL3 - [7:1] */ +#define WM2200_DSP2RMIX_VOL3_WIDTH 7 /* DSP2RMIX_VOL3 - [7:1] */ + +/* + * R1684 (0x694) - DSP2RMIX Input 4 Source + */ +#define WM2200_DSP2RMIX_SRC4_MASK 0x007F /* DSP2RMIX_SRC4 - [6:0] */ +#define WM2200_DSP2RMIX_SRC4_SHIFT 0 /* DSP2RMIX_SRC4 - [6:0] */ +#define WM2200_DSP2RMIX_SRC4_WIDTH 7 /* DSP2RMIX_SRC4 - [6:0] */ + +/* + * R1685 (0x695) - DSP2RMIX Input 4 Volume + */ +#define WM2200_DSP2RMIX_VOL4_MASK 0x00FE /* DSP2RMIX_VOL4 - [7:1] */ +#define WM2200_DSP2RMIX_VOL4_SHIFT 1 /* DSP2RMIX_VOL4 - [7:1] */ +#define WM2200_DSP2RMIX_VOL4_WIDTH 7 /* DSP2RMIX_VOL4 - [7:1] */ + +/* + * R1686 (0x696) - DSP2AUX1MIX Input 1 Source + */ +#define WM2200_DSP2AUX1MIX_SRC1_MASK 0x007F /* DSP2AUX1MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX1MIX_SRC1_SHIFT 0 /* DSP2AUX1MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX1MIX_SRC1_WIDTH 7 /* DSP2AUX1MIX_SRC1 - [6:0] */ + +/* + * R1687 (0x697) - DSP2AUX2MIX Input 1 Source + */ +#define WM2200_DSP2AUX2MIX_SRC1_MASK 0x007F /* DSP2AUX2MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX2MIX_SRC1_SHIFT 0 /* DSP2AUX2MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX2MIX_SRC1_WIDTH 7 /* DSP2AUX2MIX_SRC1 - [6:0] */ + +/* + * R1688 (0x698) - DSP2AUX3MIX Input 1 Source + */ +#define WM2200_DSP2AUX3MIX_SRC1_MASK 0x007F /* DSP2AUX3MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX3MIX_SRC1_SHIFT 0 /* DSP2AUX3MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX3MIX_SRC1_WIDTH 7 /* DSP2AUX3MIX_SRC1 - [6:0] */ + +/* + * R1689 (0x699) - DSP2AUX4MIX Input 1 Source + */ +#define WM2200_DSP2AUX4MIX_SRC1_MASK 0x007F /* DSP2AUX4MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX4MIX_SRC1_SHIFT 0 /* DSP2AUX4MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX4MIX_SRC1_WIDTH 7 /* DSP2AUX4MIX_SRC1 - [6:0] */ + +/* + * R1690 (0x69A) - DSP2AUX5MIX Input 1 Source + */ +#define WM2200_DSP2AUX5MIX_SRC1_MASK 0x007F /* DSP2AUX5MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX5MIX_SRC1_SHIFT 0 /* DSP2AUX5MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX5MIX_SRC1_WIDTH 7 /* DSP2AUX5MIX_SRC1 - [6:0] */ + +/* + * R1691 (0x69B) - DSP2AUX6MIX Input 1 Source + */ +#define WM2200_DSP2AUX6MIX_SRC1_MASK 0x007F /* DSP2AUX6MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX6MIX_SRC1_SHIFT 0 /* DSP2AUX6MIX_SRC1 - [6:0] */ +#define WM2200_DSP2AUX6MIX_SRC1_WIDTH 7 /* DSP2AUX6MIX_SRC1 - [6:0] */ + +/* + * R1792 (0x700) - GPIO CTRL 1 + */ +#define WM2200_GP1_DIR 0x8000 /* GP1_DIR */ +#define WM2200_GP1_DIR_MASK 0x8000 /* GP1_DIR */ +#define WM2200_GP1_DIR_SHIFT 15 /* GP1_DIR */ +#define WM2200_GP1_DIR_WIDTH 1 /* GP1_DIR */ +#define WM2200_GP1_PU 0x4000 /* GP1_PU */ +#define WM2200_GP1_PU_MASK 0x4000 /* GP1_PU */ +#define WM2200_GP1_PU_SHIFT 14 /* GP1_PU */ +#define WM2200_GP1_PU_WIDTH 1 /* GP1_PU */ +#define WM2200_GP1_PD 0x2000 /* GP1_PD */ +#define WM2200_GP1_PD_MASK 0x2000 /* GP1_PD */ +#define WM2200_GP1_PD_SHIFT 13 /* GP1_PD */ +#define WM2200_GP1_PD_WIDTH 1 /* GP1_PD */ +#define WM2200_GP1_POL 0x0400 /* GP1_POL */ +#define WM2200_GP1_POL_MASK 0x0400 /* GP1_POL */ +#define WM2200_GP1_POL_SHIFT 10 /* GP1_POL */ +#define WM2200_GP1_POL_WIDTH 1 /* GP1_POL */ +#define WM2200_GP1_OP_CFG 0x0200 /* GP1_OP_CFG */ +#define WM2200_GP1_OP_CFG_MASK 0x0200 /* GP1_OP_CFG */ +#define WM2200_GP1_OP_CFG_SHIFT 9 /* GP1_OP_CFG */ +#define WM2200_GP1_OP_CFG_WIDTH 1 /* GP1_OP_CFG */ +#define WM2200_GP1_DB 0x0100 /* GP1_DB */ +#define WM2200_GP1_DB_MASK 0x0100 /* GP1_DB */ +#define WM2200_GP1_DB_SHIFT 8 /* GP1_DB */ +#define WM2200_GP1_DB_WIDTH 1 /* GP1_DB */ +#define WM2200_GP1_LVL 0x0040 /* GP1_LVL */ +#define WM2200_GP1_LVL_MASK 0x0040 /* GP1_LVL */ +#define WM2200_GP1_LVL_SHIFT 6 /* GP1_LVL */ +#define WM2200_GP1_LVL_WIDTH 1 /* GP1_LVL */ +#define WM2200_GP1_FN_MASK 0x003F /* GP1_FN - [5:0] */ +#define WM2200_GP1_FN_SHIFT 0 /* GP1_FN - [5:0] */ +#define WM2200_GP1_FN_WIDTH 6 /* GP1_FN - [5:0] */ + +/* + * R1793 (0x701) - GPIO CTRL 2 + */ +#define WM2200_GP2_DIR 0x8000 /* GP2_DIR */ +#define WM2200_GP2_DIR_MASK 0x8000 /* GP2_DIR */ +#define WM2200_GP2_DIR_SHIFT 15 /* GP2_DIR */ +#define WM2200_GP2_DIR_WIDTH 1 /* GP2_DIR */ +#define WM2200_GP2_PU 0x4000 /* GP2_PU */ +#define WM2200_GP2_PU_MASK 0x4000 /* GP2_PU */ +#define WM2200_GP2_PU_SHIFT 14 /* GP2_PU */ +#define WM2200_GP2_PU_WIDTH 1 /* GP2_PU */ +#define WM2200_GP2_PD 0x2000 /* GP2_PD */ +#define WM2200_GP2_PD_MASK 0x2000 /* GP2_PD */ +#define WM2200_GP2_PD_SHIFT 13 /* GP2_PD */ +#define WM2200_GP2_PD_WIDTH 1 /* GP2_PD */ +#define WM2200_GP2_POL 0x0400 /* GP2_POL */ +#define WM2200_GP2_POL_MASK 0x0400 /* GP2_POL */ +#define WM2200_GP2_POL_SHIFT 10 /* GP2_POL */ +#define WM2200_GP2_POL_WIDTH 1 /* GP2_POL */ +#define WM2200_GP2_OP_CFG 0x0200 /* GP2_OP_CFG */ +#define WM2200_GP2_OP_CFG_MASK 0x0200 /* GP2_OP_CFG */ +#define WM2200_GP2_OP_CFG_SHIFT 9 /* GP2_OP_CFG */ +#define WM2200_GP2_OP_CFG_WIDTH 1 /* GP2_OP_CFG */ +#define WM2200_GP2_DB 0x0100 /* GP2_DB */ +#define WM2200_GP2_DB_MASK 0x0100 /* GP2_DB */ +#define WM2200_GP2_DB_SHIFT 8 /* GP2_DB */ +#define WM2200_GP2_DB_WIDTH 1 /* GP2_DB */ +#define WM2200_GP2_LVL 0x0040 /* GP2_LVL */ +#define WM2200_GP2_LVL_MASK 0x0040 /* GP2_LVL */ +#define WM2200_GP2_LVL_SHIFT 6 /* GP2_LVL */ +#define WM2200_GP2_LVL_WIDTH 1 /* GP2_LVL */ +#define WM2200_GP2_FN_MASK 0x003F /* GP2_FN - [5:0] */ +#define WM2200_GP2_FN_SHIFT 0 /* GP2_FN - [5:0] */ +#define WM2200_GP2_FN_WIDTH 6 /* GP2_FN - [5:0] */ + +/* + * R1794 (0x702) - GPIO CTRL 3 + */ +#define WM2200_GP3_DIR 0x8000 /* GP3_DIR */ +#define WM2200_GP3_DIR_MASK 0x8000 /* GP3_DIR */ +#define WM2200_GP3_DIR_SHIFT 15 /* GP3_DIR */ +#define WM2200_GP3_DIR_WIDTH 1 /* GP3_DIR */ +#define WM2200_GP3_PU 0x4000 /* GP3_PU */ +#define WM2200_GP3_PU_MASK 0x4000 /* GP3_PU */ +#define WM2200_GP3_PU_SHIFT 14 /* GP3_PU */ +#define WM2200_GP3_PU_WIDTH 1 /* GP3_PU */ +#define WM2200_GP3_PD 0x2000 /* GP3_PD */ +#define WM2200_GP3_PD_MASK 0x2000 /* GP3_PD */ +#define WM2200_GP3_PD_SHIFT 13 /* GP3_PD */ +#define WM2200_GP3_PD_WIDTH 1 /* GP3_PD */ +#define WM2200_GP3_POL 0x0400 /* GP3_POL */ +#define WM2200_GP3_POL_MASK 0x0400 /* GP3_POL */ +#define WM2200_GP3_POL_SHIFT 10 /* GP3_POL */ +#define WM2200_GP3_POL_WIDTH 1 /* GP3_POL */ +#define WM2200_GP3_OP_CFG 0x0200 /* GP3_OP_CFG */ +#define WM2200_GP3_OP_CFG_MASK 0x0200 /* GP3_OP_CFG */ +#define WM2200_GP3_OP_CFG_SHIFT 9 /* GP3_OP_CFG */ +#define WM2200_GP3_OP_CFG_WIDTH 1 /* GP3_OP_CFG */ +#define WM2200_GP3_DB 0x0100 /* GP3_DB */ +#define WM2200_GP3_DB_MASK 0x0100 /* GP3_DB */ +#define WM2200_GP3_DB_SHIFT 8 /* GP3_DB */ +#define WM2200_GP3_DB_WIDTH 1 /* GP3_DB */ +#define WM2200_GP3_LVL 0x0040 /* GP3_LVL */ +#define WM2200_GP3_LVL_MASK 0x0040 /* GP3_LVL */ +#define WM2200_GP3_LVL_SHIFT 6 /* GP3_LVL */ +#define WM2200_GP3_LVL_WIDTH 1 /* GP3_LVL */ +#define WM2200_GP3_FN_MASK 0x003F /* GP3_FN - [5:0] */ +#define WM2200_GP3_FN_SHIFT 0 /* GP3_FN - [5:0] */ +#define WM2200_GP3_FN_WIDTH 6 /* GP3_FN - [5:0] */ + +/* + * R1795 (0x703) - GPIO CTRL 4 + */ +#define WM2200_GP4_DIR 0x8000 /* GP4_DIR */ +#define WM2200_GP4_DIR_MASK 0x8000 /* GP4_DIR */ +#define WM2200_GP4_DIR_SHIFT 15 /* GP4_DIR */ +#define WM2200_GP4_DIR_WIDTH 1 /* GP4_DIR */ +#define WM2200_GP4_PU 0x4000 /* GP4_PU */ +#define WM2200_GP4_PU_MASK 0x4000 /* GP4_PU */ +#define WM2200_GP4_PU_SHIFT 14 /* GP4_PU */ +#define WM2200_GP4_PU_WIDTH 1 /* GP4_PU */ +#define WM2200_GP4_PD 0x2000 /* GP4_PD */ +#define WM2200_GP4_PD_MASK 0x2000 /* GP4_PD */ +#define WM2200_GP4_PD_SHIFT 13 /* GP4_PD */ +#define WM2200_GP4_PD_WIDTH 1 /* GP4_PD */ +#define WM2200_GP4_POL 0x0400 /* GP4_POL */ +#define WM2200_GP4_POL_MASK 0x0400 /* GP4_POL */ +#define WM2200_GP4_POL_SHIFT 10 /* GP4_POL */ +#define WM2200_GP4_POL_WIDTH 1 /* GP4_POL */ +#define WM2200_GP4_OP_CFG 0x0200 /* GP4_OP_CFG */ +#define WM2200_GP4_OP_CFG_MASK 0x0200 /* GP4_OP_CFG */ +#define WM2200_GP4_OP_CFG_SHIFT 9 /* GP4_OP_CFG */ +#define WM2200_GP4_OP_CFG_WIDTH 1 /* GP4_OP_CFG */ +#define WM2200_GP4_DB 0x0100 /* GP4_DB */ +#define WM2200_GP4_DB_MASK 0x0100 /* GP4_DB */ +#define WM2200_GP4_DB_SHIFT 8 /* GP4_DB */ +#define WM2200_GP4_DB_WIDTH 1 /* GP4_DB */ +#define WM2200_GP4_LVL 0x0040 /* GP4_LVL */ +#define WM2200_GP4_LVL_MASK 0x0040 /* GP4_LVL */ +#define WM2200_GP4_LVL_SHIFT 6 /* GP4_LVL */ +#define WM2200_GP4_LVL_WIDTH 1 /* GP4_LVL */ +#define WM2200_GP4_FN_MASK 0x003F /* GP4_FN - [5:0] */ +#define WM2200_GP4_FN_SHIFT 0 /* GP4_FN - [5:0] */ +#define WM2200_GP4_FN_WIDTH 6 /* GP4_FN - [5:0] */ + +/* + * R1799 (0x707) - ADPS1 IRQ0 + */ +#define WM2200_DSP_IRQ1 0x0002 /* DSP_IRQ1 */ +#define WM2200_DSP_IRQ1_MASK 0x0002 /* DSP_IRQ1 */ +#define WM2200_DSP_IRQ1_SHIFT 1 /* DSP_IRQ1 */ +#define WM2200_DSP_IRQ1_WIDTH 1 /* DSP_IRQ1 */ +#define WM2200_DSP_IRQ0 0x0001 /* DSP_IRQ0 */ +#define WM2200_DSP_IRQ0_MASK 0x0001 /* DSP_IRQ0 */ +#define WM2200_DSP_IRQ0_SHIFT 0 /* DSP_IRQ0 */ +#define WM2200_DSP_IRQ0_WIDTH 1 /* DSP_IRQ0 */ + +/* + * R1800 (0x708) - ADPS1 IRQ1 + */ +#define WM2200_DSP_IRQ3 0x0002 /* DSP_IRQ3 */ +#define WM2200_DSP_IRQ3_MASK 0x0002 /* DSP_IRQ3 */ +#define WM2200_DSP_IRQ3_SHIFT 1 /* DSP_IRQ3 */ +#define WM2200_DSP_IRQ3_WIDTH 1 /* DSP_IRQ3 */ +#define WM2200_DSP_IRQ2 0x0001 /* DSP_IRQ2 */ +#define WM2200_DSP_IRQ2_MASK 0x0001 /* DSP_IRQ2 */ +#define WM2200_DSP_IRQ2_SHIFT 0 /* DSP_IRQ2 */ +#define WM2200_DSP_IRQ2_WIDTH 1 /* DSP_IRQ2 */ + +/* + * R1801 (0x709) - Misc Pad Ctrl 1 + */ +#define WM2200_LDO1ENA_PD 0x8000 /* LDO1ENA_PD */ +#define WM2200_LDO1ENA_PD_MASK 0x8000 /* LDO1ENA_PD */ +#define WM2200_LDO1ENA_PD_SHIFT 15 /* LDO1ENA_PD */ +#define WM2200_LDO1ENA_PD_WIDTH 1 /* LDO1ENA_PD */ +#define WM2200_MCLK2_PD 0x2000 /* MCLK2_PD */ +#define WM2200_MCLK2_PD_MASK 0x2000 /* MCLK2_PD */ +#define WM2200_MCLK2_PD_SHIFT 13 /* MCLK2_PD */ +#define WM2200_MCLK2_PD_WIDTH 1 /* MCLK2_PD */ +#define WM2200_MCLK1_PD 0x1000 /* MCLK1_PD */ +#define WM2200_MCLK1_PD_MASK 0x1000 /* MCLK1_PD */ +#define WM2200_MCLK1_PD_SHIFT 12 /* MCLK1_PD */ +#define WM2200_MCLK1_PD_WIDTH 1 /* MCLK1_PD */ +#define WM2200_DACLRCLK1_PU 0x0400 /* DACLRCLK1_PU */ +#define WM2200_DACLRCLK1_PU_MASK 0x0400 /* DACLRCLK1_PU */ +#define WM2200_DACLRCLK1_PU_SHIFT 10 /* DACLRCLK1_PU */ +#define WM2200_DACLRCLK1_PU_WIDTH 1 /* DACLRCLK1_PU */ +#define WM2200_DACLRCLK1_PD 0x0200 /* DACLRCLK1_PD */ +#define WM2200_DACLRCLK1_PD_MASK 0x0200 /* DACLRCLK1_PD */ +#define WM2200_DACLRCLK1_PD_SHIFT 9 /* DACLRCLK1_PD */ +#define WM2200_DACLRCLK1_PD_WIDTH 1 /* DACLRCLK1_PD */ +#define WM2200_BCLK1_PU 0x0100 /* BCLK1_PU */ +#define WM2200_BCLK1_PU_MASK 0x0100 /* BCLK1_PU */ +#define WM2200_BCLK1_PU_SHIFT 8 /* BCLK1_PU */ +#define WM2200_BCLK1_PU_WIDTH 1 /* BCLK1_PU */ +#define WM2200_BCLK1_PD 0x0080 /* BCLK1_PD */ +#define WM2200_BCLK1_PD_MASK 0x0080 /* BCLK1_PD */ +#define WM2200_BCLK1_PD_SHIFT 7 /* BCLK1_PD */ +#define WM2200_BCLK1_PD_WIDTH 1 /* BCLK1_PD */ +#define WM2200_DACDAT1_PU 0x0040 /* DACDAT1_PU */ +#define WM2200_DACDAT1_PU_MASK 0x0040 /* DACDAT1_PU */ +#define WM2200_DACDAT1_PU_SHIFT 6 /* DACDAT1_PU */ +#define WM2200_DACDAT1_PU_WIDTH 1 /* DACDAT1_PU */ +#define WM2200_DACDAT1_PD 0x0020 /* DACDAT1_PD */ +#define WM2200_DACDAT1_PD_MASK 0x0020 /* DACDAT1_PD */ +#define WM2200_DACDAT1_PD_SHIFT 5 /* DACDAT1_PD */ +#define WM2200_DACDAT1_PD_WIDTH 1 /* DACDAT1_PD */ +#define WM2200_DMICDAT3_PD 0x0010 /* DMICDAT3_PD */ +#define WM2200_DMICDAT3_PD_MASK 0x0010 /* DMICDAT3_PD */ +#define WM2200_DMICDAT3_PD_SHIFT 4 /* DMICDAT3_PD */ +#define WM2200_DMICDAT3_PD_WIDTH 1 /* DMICDAT3_PD */ +#define WM2200_DMICDAT2_PD 0x0008 /* DMICDAT2_PD */ +#define WM2200_DMICDAT2_PD_MASK 0x0008 /* DMICDAT2_PD */ +#define WM2200_DMICDAT2_PD_SHIFT 3 /* DMICDAT2_PD */ +#define WM2200_DMICDAT2_PD_WIDTH 1 /* DMICDAT2_PD */ +#define WM2200_DMICDAT1_PD 0x0004 /* DMICDAT1_PD */ +#define WM2200_DMICDAT1_PD_MASK 0x0004 /* DMICDAT1_PD */ +#define WM2200_DMICDAT1_PD_SHIFT 2 /* DMICDAT1_PD */ +#define WM2200_DMICDAT1_PD_WIDTH 1 /* DMICDAT1_PD */ +#define WM2200_RSTB_PU 0x0002 /* RSTB_PU */ +#define WM2200_RSTB_PU_MASK 0x0002 /* RSTB_PU */ +#define WM2200_RSTB_PU_SHIFT 1 /* RSTB_PU */ +#define WM2200_RSTB_PU_WIDTH 1 /* RSTB_PU */ +#define WM2200_ADDR_PD 0x0001 /* ADDR_PD */ +#define WM2200_ADDR_PD_MASK 0x0001 /* ADDR_PD */ +#define WM2200_ADDR_PD_SHIFT 0 /* ADDR_PD */ +#define WM2200_ADDR_PD_WIDTH 1 /* ADDR_PD */ + +/* + * R2048 (0x800) - Interrupt Status 1 + */ +#define WM2200_DSP_IRQ0_EINT 0x0080 /* DSP_IRQ0_EINT */ +#define WM2200_DSP_IRQ0_EINT_MASK 0x0080 /* DSP_IRQ0_EINT */ +#define WM2200_DSP_IRQ0_EINT_SHIFT 7 /* DSP_IRQ0_EINT */ +#define WM2200_DSP_IRQ0_EINT_WIDTH 1 /* DSP_IRQ0_EINT */ +#define WM2200_DSP_IRQ1_EINT 0x0040 /* DSP_IRQ1_EINT */ +#define WM2200_DSP_IRQ1_EINT_MASK 0x0040 /* DSP_IRQ1_EINT */ +#define WM2200_DSP_IRQ1_EINT_SHIFT 6 /* DSP_IRQ1_EINT */ +#define WM2200_DSP_IRQ1_EINT_WIDTH 1 /* DSP_IRQ1_EINT */ +#define WM2200_DSP_IRQ2_EINT 0x0020 /* DSP_IRQ2_EINT */ +#define WM2200_DSP_IRQ2_EINT_MASK 0x0020 /* DSP_IRQ2_EINT */ +#define WM2200_DSP_IRQ2_EINT_SHIFT 5 /* DSP_IRQ2_EINT */ +#define WM2200_DSP_IRQ2_EINT_WIDTH 1 /* DSP_IRQ2_EINT */ +#define WM2200_DSP_IRQ3_EINT 0x0010 /* DSP_IRQ3_EINT */ +#define WM2200_DSP_IRQ3_EINT_MASK 0x0010 /* DSP_IRQ3_EINT */ +#define WM2200_DSP_IRQ3_EINT_SHIFT 4 /* DSP_IRQ3_EINT */ +#define WM2200_DSP_IRQ3_EINT_WIDTH 1 /* DSP_IRQ3_EINT */ +#define WM2200_GP4_EINT 0x0008 /* GP4_EINT */ +#define WM2200_GP4_EINT_MASK 0x0008 /* GP4_EINT */ +#define WM2200_GP4_EINT_SHIFT 3 /* GP4_EINT */ +#define WM2200_GP4_EINT_WIDTH 1 /* GP4_EINT */ +#define WM2200_GP3_EINT 0x0004 /* GP3_EINT */ +#define WM2200_GP3_EINT_MASK 0x0004 /* GP3_EINT */ +#define WM2200_GP3_EINT_SHIFT 2 /* GP3_EINT */ +#define WM2200_GP3_EINT_WIDTH 1 /* GP3_EINT */ +#define WM2200_GP2_EINT 0x0002 /* GP2_EINT */ +#define WM2200_GP2_EINT_MASK 0x0002 /* GP2_EINT */ +#define WM2200_GP2_EINT_SHIFT 1 /* GP2_EINT */ +#define WM2200_GP2_EINT_WIDTH 1 /* GP2_EINT */ +#define WM2200_GP1_EINT 0x0001 /* GP1_EINT */ +#define WM2200_GP1_EINT_MASK 0x0001 /* GP1_EINT */ +#define WM2200_GP1_EINT_SHIFT 0 /* GP1_EINT */ +#define WM2200_GP1_EINT_WIDTH 1 /* GP1_EINT */ + +/* + * R2049 (0x801) - Interrupt Status 1 Mask + */ +#define WM2200_IM_DSP_IRQ0_EINT 0x0080 /* IM_DSP_IRQ0_EINT */ +#define WM2200_IM_DSP_IRQ0_EINT_MASK 0x0080 /* IM_DSP_IRQ0_EINT */ +#define WM2200_IM_DSP_IRQ0_EINT_SHIFT 7 /* IM_DSP_IRQ0_EINT */ +#define WM2200_IM_DSP_IRQ0_EINT_WIDTH 1 /* IM_DSP_IRQ0_EINT */ +#define WM2200_IM_DSP_IRQ1_EINT 0x0040 /* IM_DSP_IRQ1_EINT */ +#define WM2200_IM_DSP_IRQ1_EINT_MASK 0x0040 /* IM_DSP_IRQ1_EINT */ +#define WM2200_IM_DSP_IRQ1_EINT_SHIFT 6 /* IM_DSP_IRQ1_EINT */ +#define WM2200_IM_DSP_IRQ1_EINT_WIDTH 1 /* IM_DSP_IRQ1_EINT */ +#define WM2200_IM_DSP_IRQ2_EINT 0x0020 /* IM_DSP_IRQ2_EINT */ +#define WM2200_IM_DSP_IRQ2_EINT_MASK 0x0020 /* IM_DSP_IRQ2_EINT */ +#define WM2200_IM_DSP_IRQ2_EINT_SHIFT 5 /* IM_DSP_IRQ2_EINT */ +#define WM2200_IM_DSP_IRQ2_EINT_WIDTH 1 /* IM_DSP_IRQ2_EINT */ +#define WM2200_IM_DSP_IRQ3_EINT 0x0010 /* IM_DSP_IRQ3_EINT */ +#define WM2200_IM_DSP_IRQ3_EINT_MASK 0x0010 /* IM_DSP_IRQ3_EINT */ +#define WM2200_IM_DSP_IRQ3_EINT_SHIFT 4 /* IM_DSP_IRQ3_EINT */ +#define WM2200_IM_DSP_IRQ3_EINT_WIDTH 1 /* IM_DSP_IRQ3_EINT */ +#define WM2200_IM_GP4_EINT 0x0008 /* IM_GP4_EINT */ +#define WM2200_IM_GP4_EINT_MASK 0x0008 /* IM_GP4_EINT */ +#define WM2200_IM_GP4_EINT_SHIFT 3 /* IM_GP4_EINT */ +#define WM2200_IM_GP4_EINT_WIDTH 1 /* IM_GP4_EINT */ +#define WM2200_IM_GP3_EINT 0x0004 /* IM_GP3_EINT */ +#define WM2200_IM_GP3_EINT_MASK 0x0004 /* IM_GP3_EINT */ +#define WM2200_IM_GP3_EINT_SHIFT 2 /* IM_GP3_EINT */ +#define WM2200_IM_GP3_EINT_WIDTH 1 /* IM_GP3_EINT */ +#define WM2200_IM_GP2_EINT 0x0002 /* IM_GP2_EINT */ +#define WM2200_IM_GP2_EINT_MASK 0x0002 /* IM_GP2_EINT */ +#define WM2200_IM_GP2_EINT_SHIFT 1 /* IM_GP2_EINT */ +#define WM2200_IM_GP2_EINT_WIDTH 1 /* IM_GP2_EINT */ +#define WM2200_IM_GP1_EINT 0x0001 /* IM_GP1_EINT */ +#define WM2200_IM_GP1_EINT_MASK 0x0001 /* IM_GP1_EINT */ +#define WM2200_IM_GP1_EINT_SHIFT 0 /* IM_GP1_EINT */ +#define WM2200_IM_GP1_EINT_WIDTH 1 /* IM_GP1_EINT */ + +/* + * R2050 (0x802) - Interrupt Status 2 + */ +#define WM2200_WSEQ_BUSY_EINT 0x0100 /* WSEQ_BUSY_EINT */ +#define WM2200_WSEQ_BUSY_EINT_MASK 0x0100 /* WSEQ_BUSY_EINT */ +#define WM2200_WSEQ_BUSY_EINT_SHIFT 8 /* WSEQ_BUSY_EINT */ +#define WM2200_WSEQ_BUSY_EINT_WIDTH 1 /* WSEQ_BUSY_EINT */ +#define WM2200_FLL_LOCK_EINT 0x0002 /* FLL_LOCK_EINT */ +#define WM2200_FLL_LOCK_EINT_MASK 0x0002 /* FLL_LOCK_EINT */ +#define WM2200_FLL_LOCK_EINT_SHIFT 1 /* FLL_LOCK_EINT */ +#define WM2200_FLL_LOCK_EINT_WIDTH 1 /* FLL_LOCK_EINT */ +#define WM2200_CLKGEN_EINT 0x0001 /* CLKGEN_EINT */ +#define WM2200_CLKGEN_EINT_MASK 0x0001 /* CLKGEN_EINT */ +#define WM2200_CLKGEN_EINT_SHIFT 0 /* CLKGEN_EINT */ +#define WM2200_CLKGEN_EINT_WIDTH 1 /* CLKGEN_EINT */ + +/* + * R2051 (0x803) - Interrupt Raw Status 2 + */ +#define WM2200_WSEQ_BUSY_STS 0x0100 /* WSEQ_BUSY_STS */ +#define WM2200_WSEQ_BUSY_STS_MASK 0x0100 /* WSEQ_BUSY_STS */ +#define WM2200_WSEQ_BUSY_STS_SHIFT 8 /* WSEQ_BUSY_STS */ +#define WM2200_WSEQ_BUSY_STS_WIDTH 1 /* WSEQ_BUSY_STS */ +#define WM2200_FLL_LOCK_STS 0x0002 /* FLL_LOCK_STS */ +#define WM2200_FLL_LOCK_STS_MASK 0x0002 /* FLL_LOCK_STS */ +#define WM2200_FLL_LOCK_STS_SHIFT 1 /* FLL_LOCK_STS */ +#define WM2200_FLL_LOCK_STS_WIDTH 1 /* FLL_LOCK_STS */ +#define WM2200_CLKGEN_STS 0x0001 /* CLKGEN_STS */ +#define WM2200_CLKGEN_STS_MASK 0x0001 /* CLKGEN_STS */ +#define WM2200_CLKGEN_STS_SHIFT 0 /* CLKGEN_STS */ +#define WM2200_CLKGEN_STS_WIDTH 1 /* CLKGEN_STS */ + +/* + * R2052 (0x804) - Interrupt Status 2 Mask + */ +#define WM2200_IM_WSEQ_BUSY_EINT 0x0100 /* IM_WSEQ_BUSY_EINT */ +#define WM2200_IM_WSEQ_BUSY_EINT_MASK 0x0100 /* IM_WSEQ_BUSY_EINT */ +#define WM2200_IM_WSEQ_BUSY_EINT_SHIFT 8 /* IM_WSEQ_BUSY_EINT */ +#define WM2200_IM_WSEQ_BUSY_EINT_WIDTH 1 /* IM_WSEQ_BUSY_EINT */ +#define WM2200_IM_FLL_LOCK_EINT 0x0002 /* IM_FLL_LOCK_EINT */ +#define WM2200_IM_FLL_LOCK_EINT_MASK 0x0002 /* IM_FLL_LOCK_EINT */ +#define WM2200_IM_FLL_LOCK_EINT_SHIFT 1 /* IM_FLL_LOCK_EINT */ +#define WM2200_IM_FLL_LOCK_EINT_WIDTH 1 /* IM_FLL_LOCK_EINT */ +#define WM2200_IM_CLKGEN_EINT 0x0001 /* IM_CLKGEN_EINT */ +#define WM2200_IM_CLKGEN_EINT_MASK 0x0001 /* IM_CLKGEN_EINT */ +#define WM2200_IM_CLKGEN_EINT_SHIFT 0 /* IM_CLKGEN_EINT */ +#define WM2200_IM_CLKGEN_EINT_WIDTH 1 /* IM_CLKGEN_EINT */ + +/* + * R2056 (0x808) - Interrupt Control + */ +#define WM2200_IM_IRQ 0x0001 /* IM_IRQ */ +#define WM2200_IM_IRQ_MASK 0x0001 /* IM_IRQ */ +#define WM2200_IM_IRQ_SHIFT 0 /* IM_IRQ */ +#define WM2200_IM_IRQ_WIDTH 1 /* IM_IRQ */ + +/* + * R2304 (0x900) - EQL_1 + */ +#define WM2200_EQL_B1_GAIN_MASK 0xF800 /* EQL_B1_GAIN - [15:11] */ +#define WM2200_EQL_B1_GAIN_SHIFT 11 /* EQL_B1_GAIN - [15:11] */ +#define WM2200_EQL_B1_GAIN_WIDTH 5 /* EQL_B1_GAIN - [15:11] */ +#define WM2200_EQL_B2_GAIN_MASK 0x07C0 /* EQL_B2_GAIN - [10:6] */ +#define WM2200_EQL_B2_GAIN_SHIFT 6 /* EQL_B2_GAIN - [10:6] */ +#define WM2200_EQL_B2_GAIN_WIDTH 5 /* EQL_B2_GAIN - [10:6] */ +#define WM2200_EQL_B3_GAIN_MASK 0x003E /* EQL_B3_GAIN - [5:1] */ +#define WM2200_EQL_B3_GAIN_SHIFT 1 /* EQL_B3_GAIN - [5:1] */ +#define WM2200_EQL_B3_GAIN_WIDTH 5 /* EQL_B3_GAIN - [5:1] */ +#define WM2200_EQL_ENA 0x0001 /* EQL_ENA */ +#define WM2200_EQL_ENA_MASK 0x0001 /* EQL_ENA */ +#define WM2200_EQL_ENA_SHIFT 0 /* EQL_ENA */ +#define WM2200_EQL_ENA_WIDTH 1 /* EQL_ENA */ + +/* + * R2305 (0x901) - EQL_2 + */ +#define WM2200_EQL_B4_GAIN_MASK 0xF800 /* EQL_B4_GAIN - [15:11] */ +#define WM2200_EQL_B4_GAIN_SHIFT 11 /* EQL_B4_GAIN - [15:11] */ +#define WM2200_EQL_B4_GAIN_WIDTH 5 /* EQL_B4_GAIN - [15:11] */ +#define WM2200_EQL_B5_GAIN_MASK 0x07C0 /* EQL_B5_GAIN - [10:6] */ +#define WM2200_EQL_B5_GAIN_SHIFT 6 /* EQL_B5_GAIN - [10:6] */ +#define WM2200_EQL_B5_GAIN_WIDTH 5 /* EQL_B5_GAIN - [10:6] */ + +/* + * R2306 (0x902) - EQL_3 + */ +#define WM2200_EQL_B1_A_MASK 0xFFFF /* EQL_B1_A - [15:0] */ +#define WM2200_EQL_B1_A_SHIFT 0 /* EQL_B1_A - [15:0] */ +#define WM2200_EQL_B1_A_WIDTH 16 /* EQL_B1_A - [15:0] */ + +/* + * R2307 (0x903) - EQL_4 + */ +#define WM2200_EQL_B1_B_MASK 0xFFFF /* EQL_B1_B - [15:0] */ +#define WM2200_EQL_B1_B_SHIFT 0 /* EQL_B1_B - [15:0] */ +#define WM2200_EQL_B1_B_WIDTH 16 /* EQL_B1_B - [15:0] */ + +/* + * R2308 (0x904) - EQL_5 + */ +#define WM2200_EQL_B1_PG_MASK 0xFFFF /* EQL_B1_PG - [15:0] */ +#define WM2200_EQL_B1_PG_SHIFT 0 /* EQL_B1_PG - [15:0] */ +#define WM2200_EQL_B1_PG_WIDTH 16 /* EQL_B1_PG - [15:0] */ + +/* + * R2309 (0x905) - EQL_6 + */ +#define WM2200_EQL_B2_A_MASK 0xFFFF /* EQL_B2_A - [15:0] */ +#define WM2200_EQL_B2_A_SHIFT 0 /* EQL_B2_A - [15:0] */ +#define WM2200_EQL_B2_A_WIDTH 16 /* EQL_B2_A - [15:0] */ + +/* + * R2310 (0x906) - EQL_7 + */ +#define WM2200_EQL_B2_B_MASK 0xFFFF /* EQL_B2_B - [15:0] */ +#define WM2200_EQL_B2_B_SHIFT 0 /* EQL_B2_B - [15:0] */ +#define WM2200_EQL_B2_B_WIDTH 16 /* EQL_B2_B - [15:0] */ + +/* + * R2311 (0x907) - EQL_8 + */ +#define WM2200_EQL_B2_C_MASK 0xFFFF /* EQL_B2_C - [15:0] */ +#define WM2200_EQL_B2_C_SHIFT 0 /* EQL_B2_C - [15:0] */ +#define WM2200_EQL_B2_C_WIDTH 16 /* EQL_B2_C - [15:0] */ + +/* + * R2312 (0x908) - EQL_9 + */ +#define WM2200_EQL_B2_PG_MASK 0xFFFF /* EQL_B2_PG - [15:0] */ +#define WM2200_EQL_B2_PG_SHIFT 0 /* EQL_B2_PG - [15:0] */ +#define WM2200_EQL_B2_PG_WIDTH 16 /* EQL_B2_PG - [15:0] */ + +/* + * R2313 (0x909) - EQL_10 + */ +#define WM2200_EQL_B3_A_MASK 0xFFFF /* EQL_B3_A - [15:0] */ +#define WM2200_EQL_B3_A_SHIFT 0 /* EQL_B3_A - [15:0] */ +#define WM2200_EQL_B3_A_WIDTH 16 /* EQL_B3_A - [15:0] */ + +/* + * R2314 (0x90A) - EQL_11 + */ +#define WM2200_EQL_B3_B_MASK 0xFFFF /* EQL_B3_B - [15:0] */ +#define WM2200_EQL_B3_B_SHIFT 0 /* EQL_B3_B - [15:0] */ +#define WM2200_EQL_B3_B_WIDTH 16 /* EQL_B3_B - [15:0] */ + +/* + * R2315 (0x90B) - EQL_12 + */ +#define WM2200_EQL_B3_C_MASK 0xFFFF /* EQL_B3_C - [15:0] */ +#define WM2200_EQL_B3_C_SHIFT 0 /* EQL_B3_C - [15:0] */ +#define WM2200_EQL_B3_C_WIDTH 16 /* EQL_B3_C - [15:0] */ + +/* + * R2316 (0x90C) - EQL_13 + */ +#define WM2200_EQL_B3_PG_MASK 0xFFFF /* EQL_B3_PG - [15:0] */ +#define WM2200_EQL_B3_PG_SHIFT 0 /* EQL_B3_PG - [15:0] */ +#define WM2200_EQL_B3_PG_WIDTH 16 /* EQL_B3_PG - [15:0] */ + +/* + * R2317 (0x90D) - EQL_14 + */ +#define WM2200_EQL_B4_A_MASK 0xFFFF /* EQL_B4_A - [15:0] */ +#define WM2200_EQL_B4_A_SHIFT 0 /* EQL_B4_A - [15:0] */ +#define WM2200_EQL_B4_A_WIDTH 16 /* EQL_B4_A - [15:0] */ + +/* + * R2318 (0x90E) - EQL_15 + */ +#define WM2200_EQL_B4_B_MASK 0xFFFF /* EQL_B4_B - [15:0] */ +#define WM2200_EQL_B4_B_SHIFT 0 /* EQL_B4_B - [15:0] */ +#define WM2200_EQL_B4_B_WIDTH 16 /* EQL_B4_B - [15:0] */ + +/* + * R2319 (0x90F) - EQL_16 + */ +#define WM2200_EQL_B4_C_MASK 0xFFFF /* EQL_B4_C - [15:0] */ +#define WM2200_EQL_B4_C_SHIFT 0 /* EQL_B4_C - [15:0] */ +#define WM2200_EQL_B4_C_WIDTH 16 /* EQL_B4_C - [15:0] */ + +/* + * R2320 (0x910) - EQL_17 + */ +#define WM2200_EQL_B4_PG_MASK 0xFFFF /* EQL_B4_PG - [15:0] */ +#define WM2200_EQL_B4_PG_SHIFT 0 /* EQL_B4_PG - [15:0] */ +#define WM2200_EQL_B4_PG_WIDTH 16 /* EQL_B4_PG - [15:0] */ + +/* + * R2321 (0x911) - EQL_18 + */ +#define WM2200_EQL_B5_A_MASK 0xFFFF /* EQL_B5_A - [15:0] */ +#define WM2200_EQL_B5_A_SHIFT 0 /* EQL_B5_A - [15:0] */ +#define WM2200_EQL_B5_A_WIDTH 16 /* EQL_B5_A - [15:0] */ + +/* + * R2322 (0x912) - EQL_19 + */ +#define WM2200_EQL_B5_B_MASK 0xFFFF /* EQL_B5_B - [15:0] */ +#define WM2200_EQL_B5_B_SHIFT 0 /* EQL_B5_B - [15:0] */ +#define WM2200_EQL_B5_B_WIDTH 16 /* EQL_B5_B - [15:0] */ + +/* + * R2323 (0x913) - EQL_20 + */ +#define WM2200_EQL_B5_PG_MASK 0xFFFF /* EQL_B5_PG - [15:0] */ +#define WM2200_EQL_B5_PG_SHIFT 0 /* EQL_B5_PG - [15:0] */ +#define WM2200_EQL_B5_PG_WIDTH 16 /* EQL_B5_PG - [15:0] */ + +/* + * R2326 (0x916) - EQR_1 + */ +#define WM2200_EQR_B1_GAIN_MASK 0xF800 /* EQR_B1_GAIN - [15:11] */ +#define WM2200_EQR_B1_GAIN_SHIFT 11 /* EQR_B1_GAIN - [15:11] */ +#define WM2200_EQR_B1_GAIN_WIDTH 5 /* EQR_B1_GAIN - [15:11] */ +#define WM2200_EQR_B2_GAIN_MASK 0x07C0 /* EQR_B2_GAIN - [10:6] */ +#define WM2200_EQR_B2_GAIN_SHIFT 6 /* EQR_B2_GAIN - [10:6] */ +#define WM2200_EQR_B2_GAIN_WIDTH 5 /* EQR_B2_GAIN - [10:6] */ +#define WM2200_EQR_B3_GAIN_MASK 0x003E /* EQR_B3_GAIN - [5:1] */ +#define WM2200_EQR_B3_GAIN_SHIFT 1 /* EQR_B3_GAIN - [5:1] */ +#define WM2200_EQR_B3_GAIN_WIDTH 5 /* EQR_B3_GAIN - [5:1] */ +#define WM2200_EQR_ENA 0x0001 /* EQR_ENA */ +#define WM2200_EQR_ENA_MASK 0x0001 /* EQR_ENA */ +#define WM2200_EQR_ENA_SHIFT 0 /* EQR_ENA */ +#define WM2200_EQR_ENA_WIDTH 1 /* EQR_ENA */ + +/* + * R2327 (0x917) - EQR_2 + */ +#define WM2200_EQR_B4_GAIN_MASK 0xF800 /* EQR_B4_GAIN - [15:11] */ +#define WM2200_EQR_B4_GAIN_SHIFT 11 /* EQR_B4_GAIN - [15:11] */ +#define WM2200_EQR_B4_GAIN_WIDTH 5 /* EQR_B4_GAIN - [15:11] */ +#define WM2200_EQR_B5_GAIN_MASK 0x07C0 /* EQR_B5_GAIN - [10:6] */ +#define WM2200_EQR_B5_GAIN_SHIFT 6 /* EQR_B5_GAIN - [10:6] */ +#define WM2200_EQR_B5_GAIN_WIDTH 5 /* EQR_B5_GAIN - [10:6] */ + +/* + * R2328 (0x918) - EQR_3 + */ +#define WM2200_EQR_B1_A_MASK 0xFFFF /* EQR_B1_A - [15:0] */ +#define WM2200_EQR_B1_A_SHIFT 0 /* EQR_B1_A - [15:0] */ +#define WM2200_EQR_B1_A_WIDTH 16 /* EQR_B1_A - [15:0] */ + +/* + * R2329 (0x919) - EQR_4 + */ +#define WM2200_EQR_B1_B_MASK 0xFFFF /* EQR_B1_B - [15:0] */ +#define WM2200_EQR_B1_B_SHIFT 0 /* EQR_B1_B - [15:0] */ +#define WM2200_EQR_B1_B_WIDTH 16 /* EQR_B1_B - [15:0] */ + +/* + * R2330 (0x91A) - EQR_5 + */ +#define WM2200_EQR_B1_PG_MASK 0xFFFF /* EQR_B1_PG - [15:0] */ +#define WM2200_EQR_B1_PG_SHIFT 0 /* EQR_B1_PG - [15:0] */ +#define WM2200_EQR_B1_PG_WIDTH 16 /* EQR_B1_PG - [15:0] */ + +/* + * R2331 (0x91B) - EQR_6 + */ +#define WM2200_EQR_B2_A_MASK 0xFFFF /* EQR_B2_A - [15:0] */ +#define WM2200_EQR_B2_A_SHIFT 0 /* EQR_B2_A - [15:0] */ +#define WM2200_EQR_B2_A_WIDTH 16 /* EQR_B2_A - [15:0] */ + +/* + * R2332 (0x91C) - EQR_7 + */ +#define WM2200_EQR_B2_B_MASK 0xFFFF /* EQR_B2_B - [15:0] */ +#define WM2200_EQR_B2_B_SHIFT 0 /* EQR_B2_B - [15:0] */ +#define WM2200_EQR_B2_B_WIDTH 16 /* EQR_B2_B - [15:0] */ + +/* + * R2333 (0x91D) - EQR_8 + */ +#define WM2200_EQR_B2_C_MASK 0xFFFF /* EQR_B2_C - [15:0] */ +#define WM2200_EQR_B2_C_SHIFT 0 /* EQR_B2_C - [15:0] */ +#define WM2200_EQR_B2_C_WIDTH 16 /* EQR_B2_C - [15:0] */ + +/* + * R2334 (0x91E) - EQR_9 + */ +#define WM2200_EQR_B2_PG_MASK 0xFFFF /* EQR_B2_PG - [15:0] */ +#define WM2200_EQR_B2_PG_SHIFT 0 /* EQR_B2_PG - [15:0] */ +#define WM2200_EQR_B2_PG_WIDTH 16 /* EQR_B2_PG - [15:0] */ + +/* + * R2335 (0x91F) - EQR_10 + */ +#define WM2200_EQR_B3_A_MASK 0xFFFF /* EQR_B3_A - [15:0] */ +#define WM2200_EQR_B3_A_SHIFT 0 /* EQR_B3_A - [15:0] */ +#define WM2200_EQR_B3_A_WIDTH 16 /* EQR_B3_A - [15:0] */ + +/* + * R2336 (0x920) - EQR_11 + */ +#define WM2200_EQR_B3_B_MASK 0xFFFF /* EQR_B3_B - [15:0] */ +#define WM2200_EQR_B3_B_SHIFT 0 /* EQR_B3_B - [15:0] */ +#define WM2200_EQR_B3_B_WIDTH 16 /* EQR_B3_B - [15:0] */ + +/* + * R2337 (0x921) - EQR_12 + */ +#define WM2200_EQR_B3_C_MASK 0xFFFF /* EQR_B3_C - [15:0] */ +#define WM2200_EQR_B3_C_SHIFT 0 /* EQR_B3_C - [15:0] */ +#define WM2200_EQR_B3_C_WIDTH 16 /* EQR_B3_C - [15:0] */ + +/* + * R2338 (0x922) - EQR_13 + */ +#define WM2200_EQR_B3_PG_MASK 0xFFFF /* EQR_B3_PG - [15:0] */ +#define WM2200_EQR_B3_PG_SHIFT 0 /* EQR_B3_PG - [15:0] */ +#define WM2200_EQR_B3_PG_WIDTH 16 /* EQR_B3_PG - [15:0] */ + +/* + * R2339 (0x923) - EQR_14 + */ +#define WM2200_EQR_B4_A_MASK 0xFFFF /* EQR_B4_A - [15:0] */ +#define WM2200_EQR_B4_A_SHIFT 0 /* EQR_B4_A - [15:0] */ +#define WM2200_EQR_B4_A_WIDTH 16 /* EQR_B4_A - [15:0] */ + +/* + * R2340 (0x924) - EQR_15 + */ +#define WM2200_EQR_B4_B_MASK 0xFFFF /* EQR_B4_B - [15:0] */ +#define WM2200_EQR_B4_B_SHIFT 0 /* EQR_B4_B - [15:0] */ +#define WM2200_EQR_B4_B_WIDTH 16 /* EQR_B4_B - [15:0] */ + +/* + * R2341 (0x925) - EQR_16 + */ +#define WM2200_EQR_B4_C_MASK 0xFFFF /* EQR_B4_C - [15:0] */ +#define WM2200_EQR_B4_C_SHIFT 0 /* EQR_B4_C - [15:0] */ +#define WM2200_EQR_B4_C_WIDTH 16 /* EQR_B4_C - [15:0] */ + +/* + * R2342 (0x926) - EQR_17 + */ +#define WM2200_EQR_B4_PG_MASK 0xFFFF /* EQR_B4_PG - [15:0] */ +#define WM2200_EQR_B4_PG_SHIFT 0 /* EQR_B4_PG - [15:0] */ +#define WM2200_EQR_B4_PG_WIDTH 16 /* EQR_B4_PG - [15:0] */ + +/* + * R2343 (0x927) - EQR_18 + */ +#define WM2200_EQR_B5_A_MASK 0xFFFF /* EQR_B5_A - [15:0] */ +#define WM2200_EQR_B5_A_SHIFT 0 /* EQR_B5_A - [15:0] */ +#define WM2200_EQR_B5_A_WIDTH 16 /* EQR_B5_A - [15:0] */ + +/* + * R2344 (0x928) - EQR_19 + */ +#define WM2200_EQR_B5_B_MASK 0xFFFF /* EQR_B5_B - [15:0] */ +#define WM2200_EQR_B5_B_SHIFT 0 /* EQR_B5_B - [15:0] */ +#define WM2200_EQR_B5_B_WIDTH 16 /* EQR_B5_B - [15:0] */ + +/* + * R2345 (0x929) - EQR_20 + */ +#define WM2200_EQR_B5_PG_MASK 0xFFFF /* EQR_B5_PG - [15:0] */ +#define WM2200_EQR_B5_PG_SHIFT 0 /* EQR_B5_PG - [15:0] */ +#define WM2200_EQR_B5_PG_WIDTH 16 /* EQR_B5_PG - [15:0] */ + +/* + * R2366 (0x93E) - HPLPF1_1 + */ +#define WM2200_LHPF1_MODE 0x0002 /* LHPF1_MODE */ +#define WM2200_LHPF1_MODE_MASK 0x0002 /* LHPF1_MODE */ +#define WM2200_LHPF1_MODE_SHIFT 1 /* LHPF1_MODE */ +#define WM2200_LHPF1_MODE_WIDTH 1 /* LHPF1_MODE */ +#define WM2200_LHPF1_ENA 0x0001 /* LHPF1_ENA */ +#define WM2200_LHPF1_ENA_MASK 0x0001 /* LHPF1_ENA */ +#define WM2200_LHPF1_ENA_SHIFT 0 /* LHPF1_ENA */ +#define WM2200_LHPF1_ENA_WIDTH 1 /* LHPF1_ENA */ + +/* + * R2367 (0x93F) - HPLPF1_2 + */ +#define WM2200_LHPF1_COEFF_MASK 0xFFFF /* LHPF1_COEFF - [15:0] */ +#define WM2200_LHPF1_COEFF_SHIFT 0 /* LHPF1_COEFF - [15:0] */ +#define WM2200_LHPF1_COEFF_WIDTH 16 /* LHPF1_COEFF - [15:0] */ + +/* + * R2370 (0x942) - HPLPF2_1 + */ +#define WM2200_LHPF2_MODE 0x0002 /* LHPF2_MODE */ +#define WM2200_LHPF2_MODE_MASK 0x0002 /* LHPF2_MODE */ +#define WM2200_LHPF2_MODE_SHIFT 1 /* LHPF2_MODE */ +#define WM2200_LHPF2_MODE_WIDTH 1 /* LHPF2_MODE */ +#define WM2200_LHPF2_ENA 0x0001 /* LHPF2_ENA */ +#define WM2200_LHPF2_ENA_MASK 0x0001 /* LHPF2_ENA */ +#define WM2200_LHPF2_ENA_SHIFT 0 /* LHPF2_ENA */ +#define WM2200_LHPF2_ENA_WIDTH 1 /* LHPF2_ENA */ + +/* + * R2371 (0x943) - HPLPF2_2 + */ +#define WM2200_LHPF2_COEFF_MASK 0xFFFF /* LHPF2_COEFF - [15:0] */ +#define WM2200_LHPF2_COEFF_SHIFT 0 /* LHPF2_COEFF - [15:0] */ +#define WM2200_LHPF2_COEFF_WIDTH 16 /* LHPF2_COEFF - [15:0] */ + +/* + * R2560 (0xA00) - DSP1 Control 1 + */ +#define WM2200_DSP1_RW_SEQUENCE_ENA 0x0001 /* DSP1_RW_SEQUENCE_ENA */ +#define WM2200_DSP1_RW_SEQUENCE_ENA_MASK 0x0001 /* DSP1_RW_SEQUENCE_ENA */ +#define WM2200_DSP1_RW_SEQUENCE_ENA_SHIFT 0 /* DSP1_RW_SEQUENCE_ENA */ +#define WM2200_DSP1_RW_SEQUENCE_ENA_WIDTH 1 /* DSP1_RW_SEQUENCE_ENA */ + +/* + * R2562 (0xA02) - DSP1 Control 2 + */ +#define WM2200_DSP1_PAGE_BASE_PM_0_MASK 0xFF00 /* DSP1_PAGE_BASE_PM - [15:8] */ +#define WM2200_DSP1_PAGE_BASE_PM_0_SHIFT 8 /* DSP1_PAGE_BASE_PM - [15:8] */ +#define WM2200_DSP1_PAGE_BASE_PM_0_WIDTH 8 /* DSP1_PAGE_BASE_PM - [15:8] */ + +/* + * R2563 (0xA03) - DSP1 Control 3 + */ +#define WM2200_DSP1_PAGE_BASE_DM_0_MASK 0xFF00 /* DSP1_PAGE_BASE_DM - [15:8] */ +#define WM2200_DSP1_PAGE_BASE_DM_0_SHIFT 8 /* DSP1_PAGE_BASE_DM - [15:8] */ +#define WM2200_DSP1_PAGE_BASE_DM_0_WIDTH 8 /* DSP1_PAGE_BASE_DM - [15:8] */ + +/* + * R2564 (0xA04) - DSP1 Control 4 + */ +#define WM2200_DSP1_PAGE_BASE_ZM_0_MASK 0xFF00 /* DSP1_PAGE_BASE_ZM - [15:8] */ +#define WM2200_DSP1_PAGE_BASE_ZM_0_SHIFT 8 /* DSP1_PAGE_BASE_ZM - [15:8] */ +#define WM2200_DSP1_PAGE_BASE_ZM_0_WIDTH 8 /* DSP1_PAGE_BASE_ZM - [15:8] */ + +/* + * R2566 (0xA06) - DSP1 Control 5 + */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_0_MASK 0x3FFF /* DSP1_START_ADDRESS_WDMA_BUFFER_0 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_0_SHIFT 0 /* DSP1_START_ADDRESS_WDMA_BUFFER_0 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_0_WIDTH 14 /* DSP1_START_ADDRESS_WDMA_BUFFER_0 - [13:0] */ + +/* + * R2567 (0xA07) - DSP1 Control 6 + */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_1_MASK 0x3FFF /* DSP1_START_ADDRESS_WDMA_BUFFER_1 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_1_SHIFT 0 /* DSP1_START_ADDRESS_WDMA_BUFFER_1 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_1_WIDTH 14 /* DSP1_START_ADDRESS_WDMA_BUFFER_1 - [13:0] */ + +/* + * R2568 (0xA08) - DSP1 Control 7 + */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_2_MASK 0x3FFF /* DSP1_START_ADDRESS_WDMA_BUFFER_2 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_2_SHIFT 0 /* DSP1_START_ADDRESS_WDMA_BUFFER_2 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_2_WIDTH 14 /* DSP1_START_ADDRESS_WDMA_BUFFER_2 - [13:0] */ + +/* + * R2569 (0xA09) - DSP1 Control 8 + */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_3_MASK 0x3FFF /* DSP1_START_ADDRESS_WDMA_BUFFER_3 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_3_SHIFT 0 /* DSP1_START_ADDRESS_WDMA_BUFFER_3 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_3_WIDTH 14 /* DSP1_START_ADDRESS_WDMA_BUFFER_3 - [13:0] */ + +/* + * R2570 (0xA0A) - DSP1 Control 9 + */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_4_MASK 0x3FFF /* DSP1_START_ADDRESS_WDMA_BUFFER_4 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_4_SHIFT 0 /* DSP1_START_ADDRESS_WDMA_BUFFER_4 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_4_WIDTH 14 /* DSP1_START_ADDRESS_WDMA_BUFFER_4 - [13:0] */ + +/* + * R2571 (0xA0B) - DSP1 Control 10 + */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_5_MASK 0x3FFF /* DSP1_START_ADDRESS_WDMA_BUFFER_5 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_5_SHIFT 0 /* DSP1_START_ADDRESS_WDMA_BUFFER_5 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_5_WIDTH 14 /* DSP1_START_ADDRESS_WDMA_BUFFER_5 - [13:0] */ + +/* + * R2572 (0xA0C) - DSP1 Control 11 + */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_6_MASK 0x3FFF /* DSP1_START_ADDRESS_WDMA_BUFFER_6 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_6_SHIFT 0 /* DSP1_START_ADDRESS_WDMA_BUFFER_6 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_6_WIDTH 14 /* DSP1_START_ADDRESS_WDMA_BUFFER_6 - [13:0] */ + +/* + * R2573 (0xA0D) - DSP1 Control 12 + */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_7_MASK 0x3FFF /* DSP1_START_ADDRESS_WDMA_BUFFER_7 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_7_SHIFT 0 /* DSP1_START_ADDRESS_WDMA_BUFFER_7 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_WDMA_BUFFER_7_WIDTH 14 /* DSP1_START_ADDRESS_WDMA_BUFFER_7 - [13:0] */ + +/* + * R2575 (0xA0F) - DSP1 Control 13 + */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_0_MASK 0x3FFF /* DSP1_START_ADDRESS_RDMA_BUFFER_0 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_0_SHIFT 0 /* DSP1_START_ADDRESS_RDMA_BUFFER_0 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_0_WIDTH 14 /* DSP1_START_ADDRESS_RDMA_BUFFER_0 - [13:0] */ + +/* + * R2576 (0xA10) - DSP1 Control 14 + */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_1_MASK 0x3FFF /* DSP1_START_ADDRESS_RDMA_BUFFER_1 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_1_SHIFT 0 /* DSP1_START_ADDRESS_RDMA_BUFFER_1 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_1_WIDTH 14 /* DSP1_START_ADDRESS_RDMA_BUFFER_1 - [13:0] */ + +/* + * R2577 (0xA11) - DSP1 Control 15 + */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_2_MASK 0x3FFF /* DSP1_START_ADDRESS_RDMA_BUFFER_2 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_2_SHIFT 0 /* DSP1_START_ADDRESS_RDMA_BUFFER_2 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_2_WIDTH 14 /* DSP1_START_ADDRESS_RDMA_BUFFER_2 - [13:0] */ + +/* + * R2578 (0xA12) - DSP1 Control 16 + */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_3_MASK 0x3FFF /* DSP1_START_ADDRESS_RDMA_BUFFER_3 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_3_SHIFT 0 /* DSP1_START_ADDRESS_RDMA_BUFFER_3 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_3_WIDTH 14 /* DSP1_START_ADDRESS_RDMA_BUFFER_3 - [13:0] */ + +/* + * R2579 (0xA13) - DSP1 Control 17 + */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_4_MASK 0x3FFF /* DSP1_START_ADDRESS_RDMA_BUFFER_4 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_4_SHIFT 0 /* DSP1_START_ADDRESS_RDMA_BUFFER_4 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_4_WIDTH 14 /* DSP1_START_ADDRESS_RDMA_BUFFER_4 - [13:0] */ + +/* + * R2580 (0xA14) - DSP1 Control 18 + */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_5_MASK 0x3FFF /* DSP1_START_ADDRESS_RDMA_BUFFER_5 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_5_SHIFT 0 /* DSP1_START_ADDRESS_RDMA_BUFFER_5 - [13:0] */ +#define WM2200_DSP1_START_ADDRESS_RDMA_BUFFER_5_WIDTH 14 /* DSP1_START_ADDRESS_RDMA_BUFFER_5 - [13:0] */ + +/* + * R2582 (0xA16) - DSP1 Control 19 + */ +#define WM2200_DSP1_WDMA_BUFFER_LENGTH_MASK 0x00FF /* DSP1_WDMA_BUFFER_LENGTH - [7:0] */ +#define WM2200_DSP1_WDMA_BUFFER_LENGTH_SHIFT 0 /* DSP1_WDMA_BUFFER_LENGTH - [7:0] */ +#define WM2200_DSP1_WDMA_BUFFER_LENGTH_WIDTH 8 /* DSP1_WDMA_BUFFER_LENGTH - [7:0] */ + +/* + * R2583 (0xA17) - DSP1 Control 20 + */ +#define WM2200_DSP1_WDMA_CHANNEL_ENABLE_MASK 0x00FF /* DSP1_WDMA_CHANNEL_ENABLE - [7:0] */ +#define WM2200_DSP1_WDMA_CHANNEL_ENABLE_SHIFT 0 /* DSP1_WDMA_CHANNEL_ENABLE - [7:0] */ +#define WM2200_DSP1_WDMA_CHANNEL_ENABLE_WIDTH 8 /* DSP1_WDMA_CHANNEL_ENABLE - [7:0] */ + +/* + * R2584 (0xA18) - DSP1 Control 21 + */ +#define WM2200_DSP1_RDMA_CHANNEL_ENABLE_MASK 0x003F /* DSP1_RDMA_CHANNEL_ENABLE - [5:0] */ +#define WM2200_DSP1_RDMA_CHANNEL_ENABLE_SHIFT 0 /* DSP1_RDMA_CHANNEL_ENABLE - [5:0] */ +#define WM2200_DSP1_RDMA_CHANNEL_ENABLE_WIDTH 6 /* DSP1_RDMA_CHANNEL_ENABLE - [5:0] */ + +/* + * R2586 (0xA1A) - DSP1 Control 22 + */ +#define WM2200_DSP1_DM_SIZE_MASK 0xFFFF /* DSP1_DM_SIZE - [15:0] */ +#define WM2200_DSP1_DM_SIZE_SHIFT 0 /* DSP1_DM_SIZE - [15:0] */ +#define WM2200_DSP1_DM_SIZE_WIDTH 16 /* DSP1_DM_SIZE - [15:0] */ + +/* + * R2587 (0xA1B) - DSP1 Control 23 + */ +#define WM2200_DSP1_PM_SIZE_MASK 0xFFFF /* DSP1_PM_SIZE - [15:0] */ +#define WM2200_DSP1_PM_SIZE_SHIFT 0 /* DSP1_PM_SIZE - [15:0] */ +#define WM2200_DSP1_PM_SIZE_WIDTH 16 /* DSP1_PM_SIZE - [15:0] */ + +/* + * R2588 (0xA1C) - DSP1 Control 24 + */ +#define WM2200_DSP1_ZM_SIZE_MASK 0xFFFF /* DSP1_ZM_SIZE - [15:0] */ +#define WM2200_DSP1_ZM_SIZE_SHIFT 0 /* DSP1_ZM_SIZE - [15:0] */ +#define WM2200_DSP1_ZM_SIZE_WIDTH 16 /* DSP1_ZM_SIZE - [15:0] */ + +/* + * R2590 (0xA1E) - DSP1 Control 25 + */ +#define WM2200_DSP1_PING_FULL 0x8000 /* DSP1_PING_FULL */ +#define WM2200_DSP1_PING_FULL_MASK 0x8000 /* DSP1_PING_FULL */ +#define WM2200_DSP1_PING_FULL_SHIFT 15 /* DSP1_PING_FULL */ +#define WM2200_DSP1_PING_FULL_WIDTH 1 /* DSP1_PING_FULL */ +#define WM2200_DSP1_PONG_FULL 0x4000 /* DSP1_PONG_FULL */ +#define WM2200_DSP1_PONG_FULL_MASK 0x4000 /* DSP1_PONG_FULL */ +#define WM2200_DSP1_PONG_FULL_SHIFT 14 /* DSP1_PONG_FULL */ +#define WM2200_DSP1_PONG_FULL_WIDTH 1 /* DSP1_PONG_FULL */ +#define WM2200_DSP1_WDMA_ACTIVE_CHANNELS_MASK 0x00FF /* DSP1_WDMA_ACTIVE_CHANNELS - [7:0] */ +#define WM2200_DSP1_WDMA_ACTIVE_CHANNELS_SHIFT 0 /* DSP1_WDMA_ACTIVE_CHANNELS - [7:0] */ +#define WM2200_DSP1_WDMA_ACTIVE_CHANNELS_WIDTH 8 /* DSP1_WDMA_ACTIVE_CHANNELS - [7:0] */ + +/* + * R2592 (0xA20) - DSP1 Control 26 + */ +#define WM2200_DSP1_SCRATCH_0_MASK 0xFFFF /* DSP1_SCRATCH_0 - [15:0] */ +#define WM2200_DSP1_SCRATCH_0_SHIFT 0 /* DSP1_SCRATCH_0 - [15:0] */ +#define WM2200_DSP1_SCRATCH_0_WIDTH 16 /* DSP1_SCRATCH_0 - [15:0] */ + +/* + * R2593 (0xA21) - DSP1 Control 27 + */ +#define WM2200_DSP1_SCRATCH_1_MASK 0xFFFF /* DSP1_SCRATCH_1 - [15:0] */ +#define WM2200_DSP1_SCRATCH_1_SHIFT 0 /* DSP1_SCRATCH_1 - [15:0] */ +#define WM2200_DSP1_SCRATCH_1_WIDTH 16 /* DSP1_SCRATCH_1 - [15:0] */ + +/* + * R2594 (0xA22) - DSP1 Control 28 + */ +#define WM2200_DSP1_SCRATCH_2_MASK 0xFFFF /* DSP1_SCRATCH_2 - [15:0] */ +#define WM2200_DSP1_SCRATCH_2_SHIFT 0 /* DSP1_SCRATCH_2 - [15:0] */ +#define WM2200_DSP1_SCRATCH_2_WIDTH 16 /* DSP1_SCRATCH_2 - [15:0] */ + +/* + * R2595 (0xA23) - DSP1 Control 29 + */ +#define WM2200_DSP1_SCRATCH_3_MASK 0xFFFF /* DSP1_SCRATCH_3 - [15:0] */ +#define WM2200_DSP1_SCRATCH_3_SHIFT 0 /* DSP1_SCRATCH_3 - [15:0] */ +#define WM2200_DSP1_SCRATCH_3_WIDTH 16 /* DSP1_SCRATCH_3 - [15:0] */ + +/* + * R2596 (0xA24) - DSP1 Control 30 + */ +#define WM2200_DSP1_DBG_CLK_ENA 0x0008 /* DSP1_DBG_CLK_ENA */ +#define WM2200_DSP1_DBG_CLK_ENA_MASK 0x0008 /* DSP1_DBG_CLK_ENA */ +#define WM2200_DSP1_DBG_CLK_ENA_SHIFT 3 /* DSP1_DBG_CLK_ENA */ +#define WM2200_DSP1_DBG_CLK_ENA_WIDTH 1 /* DSP1_DBG_CLK_ENA */ +#define WM2200_DSP1_SYS_ENA 0x0004 /* DSP1_SYS_ENA */ +#define WM2200_DSP1_SYS_ENA_MASK 0x0004 /* DSP1_SYS_ENA */ +#define WM2200_DSP1_SYS_ENA_SHIFT 2 /* DSP1_SYS_ENA */ +#define WM2200_DSP1_SYS_ENA_WIDTH 1 /* DSP1_SYS_ENA */ +#define WM2200_DSP1_CORE_ENA 0x0002 /* DSP1_CORE_ENA */ +#define WM2200_DSP1_CORE_ENA_MASK 0x0002 /* DSP1_CORE_ENA */ +#define WM2200_DSP1_CORE_ENA_SHIFT 1 /* DSP1_CORE_ENA */ +#define WM2200_DSP1_CORE_ENA_WIDTH 1 /* DSP1_CORE_ENA */ +#define WM2200_DSP1_START 0x0001 /* DSP1_START */ +#define WM2200_DSP1_START_MASK 0x0001 /* DSP1_START */ +#define WM2200_DSP1_START_SHIFT 0 /* DSP1_START */ +#define WM2200_DSP1_START_WIDTH 1 /* DSP1_START */ + +/* + * R2598 (0xA26) - DSP1 Control 31 + */ +#define WM2200_DSP1_CLK_RATE_MASK 0x0018 /* DSP1_CLK_RATE - [4:3] */ +#define WM2200_DSP1_CLK_RATE_SHIFT 3 /* DSP1_CLK_RATE - [4:3] */ +#define WM2200_DSP1_CLK_RATE_WIDTH 2 /* DSP1_CLK_RATE - [4:3] */ +#define WM2200_DSP1_CLK_AVAIL 0x0004 /* DSP1_CLK_AVAIL */ +#define WM2200_DSP1_CLK_AVAIL_MASK 0x0004 /* DSP1_CLK_AVAIL */ +#define WM2200_DSP1_CLK_AVAIL_SHIFT 2 /* DSP1_CLK_AVAIL */ +#define WM2200_DSP1_CLK_AVAIL_WIDTH 1 /* DSP1_CLK_AVAIL */ +#define WM2200_DSP1_CLK_REQ_MASK 0x0003 /* DSP1_CLK_REQ - [1:0] */ +#define WM2200_DSP1_CLK_REQ_SHIFT 0 /* DSP1_CLK_REQ - [1:0] */ +#define WM2200_DSP1_CLK_REQ_WIDTH 2 /* DSP1_CLK_REQ - [1:0] */ + +/* + * R2816 (0xB00) - DSP2 Control 1 + */ +#define WM2200_DSP2_RW_SEQUENCE_ENA 0x0001 /* DSP2_RW_SEQUENCE_ENA */ +#define WM2200_DSP2_RW_SEQUENCE_ENA_MASK 0x0001 /* DSP2_RW_SEQUENCE_ENA */ +#define WM2200_DSP2_RW_SEQUENCE_ENA_SHIFT 0 /* DSP2_RW_SEQUENCE_ENA */ +#define WM2200_DSP2_RW_SEQUENCE_ENA_WIDTH 1 /* DSP2_RW_SEQUENCE_ENA */ + +/* + * R2818 (0xB02) - DSP2 Control 2 + */ +#define WM2200_DSP2_PAGE_BASE_PM_0_MASK 0xFF00 /* DSP2_PAGE_BASE_PM - [15:8] */ +#define WM2200_DSP2_PAGE_BASE_PM_0_SHIFT 8 /* DSP2_PAGE_BASE_PM - [15:8] */ +#define WM2200_DSP2_PAGE_BASE_PM_0_WIDTH 8 /* DSP2_PAGE_BASE_PM - [15:8] */ + +/* + * R2819 (0xB03) - DSP2 Control 3 + */ +#define WM2200_DSP2_PAGE_BASE_DM_0_MASK 0xFF00 /* DSP2_PAGE_BASE_DM - [15:8] */ +#define WM2200_DSP2_PAGE_BASE_DM_0_SHIFT 8 /* DSP2_PAGE_BASE_DM - [15:8] */ +#define WM2200_DSP2_PAGE_BASE_DM_0_WIDTH 8 /* DSP2_PAGE_BASE_DM - [15:8] */ + +/* + * R2820 (0xB04) - DSP2 Control 4 + */ +#define WM2200_DSP2_PAGE_BASE_ZM_0_MASK 0xFF00 /* DSP2_PAGE_BASE_ZM - [15:8] */ +#define WM2200_DSP2_PAGE_BASE_ZM_0_SHIFT 8 /* DSP2_PAGE_BASE_ZM - [15:8] */ +#define WM2200_DSP2_PAGE_BASE_ZM_0_WIDTH 8 /* DSP2_PAGE_BASE_ZM - [15:8] */ + +/* + * R2822 (0xB06) - DSP2 Control 5 + */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_0_MASK 0x3FFF /* DSP2_START_ADDRESS_WDMA_BUFFER_0 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_0_SHIFT 0 /* DSP2_START_ADDRESS_WDMA_BUFFER_0 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_0_WIDTH 14 /* DSP2_START_ADDRESS_WDMA_BUFFER_0 - [13:0] */ + +/* + * R2823 (0xB07) - DSP2 Control 6 + */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_1_MASK 0x3FFF /* DSP2_START_ADDRESS_WDMA_BUFFER_1 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_1_SHIFT 0 /* DSP2_START_ADDRESS_WDMA_BUFFER_1 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_1_WIDTH 14 /* DSP2_START_ADDRESS_WDMA_BUFFER_1 - [13:0] */ + +/* + * R2824 (0xB08) - DSP2 Control 7 + */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_2_MASK 0x3FFF /* DSP2_START_ADDRESS_WDMA_BUFFER_2 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_2_SHIFT 0 /* DSP2_START_ADDRESS_WDMA_BUFFER_2 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_2_WIDTH 14 /* DSP2_START_ADDRESS_WDMA_BUFFER_2 - [13:0] */ + +/* + * R2825 (0xB09) - DSP2 Control 8 + */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_3_MASK 0x3FFF /* DSP2_START_ADDRESS_WDMA_BUFFER_3 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_3_SHIFT 0 /* DSP2_START_ADDRESS_WDMA_BUFFER_3 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_3_WIDTH 14 /* DSP2_START_ADDRESS_WDMA_BUFFER_3 - [13:0] */ + +/* + * R2826 (0xB0A) - DSP2 Control 9 + */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_4_MASK 0x3FFF /* DSP2_START_ADDRESS_WDMA_BUFFER_4 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_4_SHIFT 0 /* DSP2_START_ADDRESS_WDMA_BUFFER_4 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_4_WIDTH 14 /* DSP2_START_ADDRESS_WDMA_BUFFER_4 - [13:0] */ + +/* + * R2827 (0xB0B) - DSP2 Control 10 + */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_5_MASK 0x3FFF /* DSP2_START_ADDRESS_WDMA_BUFFER_5 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_5_SHIFT 0 /* DSP2_START_ADDRESS_WDMA_BUFFER_5 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_5_WIDTH 14 /* DSP2_START_ADDRESS_WDMA_BUFFER_5 - [13:0] */ + +/* + * R2828 (0xB0C) - DSP2 Control 11 + */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_6_MASK 0x3FFF /* DSP2_START_ADDRESS_WDMA_BUFFER_6 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_6_SHIFT 0 /* DSP2_START_ADDRESS_WDMA_BUFFER_6 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_6_WIDTH 14 /* DSP2_START_ADDRESS_WDMA_BUFFER_6 - [13:0] */ + +/* + * R2829 (0xB0D) - DSP2 Control 12 + */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_7_MASK 0x3FFF /* DSP2_START_ADDRESS_WDMA_BUFFER_7 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_7_SHIFT 0 /* DSP2_START_ADDRESS_WDMA_BUFFER_7 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_WDMA_BUFFER_7_WIDTH 14 /* DSP2_START_ADDRESS_WDMA_BUFFER_7 - [13:0] */ + +/* + * R2831 (0xB0F) - DSP2 Control 13 + */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_0_MASK 0x3FFF /* DSP2_START_ADDRESS_RDMA_BUFFER_0 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_0_SHIFT 0 /* DSP2_START_ADDRESS_RDMA_BUFFER_0 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_0_WIDTH 14 /* DSP2_START_ADDRESS_RDMA_BUFFER_0 - [13:0] */ + +/* + * R2832 (0xB10) - DSP2 Control 14 + */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_1_MASK 0x3FFF /* DSP2_START_ADDRESS_RDMA_BUFFER_1 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_1_SHIFT 0 /* DSP2_START_ADDRESS_RDMA_BUFFER_1 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_1_WIDTH 14 /* DSP2_START_ADDRESS_RDMA_BUFFER_1 - [13:0] */ + +/* + * R2833 (0xB11) - DSP2 Control 15 + */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_2_MASK 0x3FFF /* DSP2_START_ADDRESS_RDMA_BUFFER_2 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_2_SHIFT 0 /* DSP2_START_ADDRESS_RDMA_BUFFER_2 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_2_WIDTH 14 /* DSP2_START_ADDRESS_RDMA_BUFFER_2 - [13:0] */ + +/* + * R2834 (0xB12) - DSP2 Control 16 + */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_3_MASK 0x3FFF /* DSP2_START_ADDRESS_RDMA_BUFFER_3 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_3_SHIFT 0 /* DSP2_START_ADDRESS_RDMA_BUFFER_3 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_3_WIDTH 14 /* DSP2_START_ADDRESS_RDMA_BUFFER_3 - [13:0] */ + +/* + * R2835 (0xB13) - DSP2 Control 17 + */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_4_MASK 0x3FFF /* DSP2_START_ADDRESS_RDMA_BUFFER_4 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_4_SHIFT 0 /* DSP2_START_ADDRESS_RDMA_BUFFER_4 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_4_WIDTH 14 /* DSP2_START_ADDRESS_RDMA_BUFFER_4 - [13:0] */ + +/* + * R2836 (0xB14) - DSP2 Control 18 + */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_5_MASK 0x3FFF /* DSP2_START_ADDRESS_RDMA_BUFFER_5 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_5_SHIFT 0 /* DSP2_START_ADDRESS_RDMA_BUFFER_5 - [13:0] */ +#define WM2200_DSP2_START_ADDRESS_RDMA_BUFFER_5_WIDTH 14 /* DSP2_START_ADDRESS_RDMA_BUFFER_5 - [13:0] */ + +/* + * R2838 (0xB16) - DSP2 Control 19 + */ +#define WM2200_DSP2_WDMA_BUFFER_LENGTH_MASK 0x00FF /* DSP2_WDMA_BUFFER_LENGTH - [7:0] */ +#define WM2200_DSP2_WDMA_BUFFER_LENGTH_SHIFT 0 /* DSP2_WDMA_BUFFER_LENGTH - [7:0] */ +#define WM2200_DSP2_WDMA_BUFFER_LENGTH_WIDTH 8 /* DSP2_WDMA_BUFFER_LENGTH - [7:0] */ + +/* + * R2839 (0xB17) - DSP2 Control 20 + */ +#define WM2200_DSP2_WDMA_CHANNEL_ENABLE_MASK 0x00FF /* DSP2_WDMA_CHANNEL_ENABLE - [7:0] */ +#define WM2200_DSP2_WDMA_CHANNEL_ENABLE_SHIFT 0 /* DSP2_WDMA_CHANNEL_ENABLE - [7:0] */ +#define WM2200_DSP2_WDMA_CHANNEL_ENABLE_WIDTH 8 /* DSP2_WDMA_CHANNEL_ENABLE - [7:0] */ + +/* + * R2840 (0xB18) - DSP2 Control 21 + */ +#define WM2200_DSP2_RDMA_CHANNEL_ENABLE_MASK 0x003F /* DSP2_RDMA_CHANNEL_ENABLE - [5:0] */ +#define WM2200_DSP2_RDMA_CHANNEL_ENABLE_SHIFT 0 /* DSP2_RDMA_CHANNEL_ENABLE - [5:0] */ +#define WM2200_DSP2_RDMA_CHANNEL_ENABLE_WIDTH 6 /* DSP2_RDMA_CHANNEL_ENABLE - [5:0] */ + +/* + * R2842 (0xB1A) - DSP2 Control 22 + */ +#define WM2200_DSP2_DM_SIZE_MASK 0xFFFF /* DSP2_DM_SIZE - [15:0] */ +#define WM2200_DSP2_DM_SIZE_SHIFT 0 /* DSP2_DM_SIZE - [15:0] */ +#define WM2200_DSP2_DM_SIZE_WIDTH 16 /* DSP2_DM_SIZE - [15:0] */ + +/* + * R2843 (0xB1B) - DSP2 Control 23 + */ +#define WM2200_DSP2_PM_SIZE_MASK 0xFFFF /* DSP2_PM_SIZE - [15:0] */ +#define WM2200_DSP2_PM_SIZE_SHIFT 0 /* DSP2_PM_SIZE - [15:0] */ +#define WM2200_DSP2_PM_SIZE_WIDTH 16 /* DSP2_PM_SIZE - [15:0] */ + +/* + * R2844 (0xB1C) - DSP2 Control 24 + */ +#define WM2200_DSP2_ZM_SIZE_MASK 0xFFFF /* DSP2_ZM_SIZE - [15:0] */ +#define WM2200_DSP2_ZM_SIZE_SHIFT 0 /* DSP2_ZM_SIZE - [15:0] */ +#define WM2200_DSP2_ZM_SIZE_WIDTH 16 /* DSP2_ZM_SIZE - [15:0] */ + +/* + * R2846 (0xB1E) - DSP2 Control 25 + */ +#define WM2200_DSP2_PING_FULL 0x8000 /* DSP2_PING_FULL */ +#define WM2200_DSP2_PING_FULL_MASK 0x8000 /* DSP2_PING_FULL */ +#define WM2200_DSP2_PING_FULL_SHIFT 15 /* DSP2_PING_FULL */ +#define WM2200_DSP2_PING_FULL_WIDTH 1 /* DSP2_PING_FULL */ +#define WM2200_DSP2_PONG_FULL 0x4000 /* DSP2_PONG_FULL */ +#define WM2200_DSP2_PONG_FULL_MASK 0x4000 /* DSP2_PONG_FULL */ +#define WM2200_DSP2_PONG_FULL_SHIFT 14 /* DSP2_PONG_FULL */ +#define WM2200_DSP2_PONG_FULL_WIDTH 1 /* DSP2_PONG_FULL */ +#define WM2200_DSP2_WDMA_ACTIVE_CHANNELS_MASK 0x00FF /* DSP2_WDMA_ACTIVE_CHANNELS - [7:0] */ +#define WM2200_DSP2_WDMA_ACTIVE_CHANNELS_SHIFT 0 /* DSP2_WDMA_ACTIVE_CHANNELS - [7:0] */ +#define WM2200_DSP2_WDMA_ACTIVE_CHANNELS_WIDTH 8 /* DSP2_WDMA_ACTIVE_CHANNELS - [7:0] */ + +/* + * R2848 (0xB20) - DSP2 Control 26 + */ +#define WM2200_DSP2_SCRATCH_0_MASK 0xFFFF /* DSP2_SCRATCH_0 - [15:0] */ +#define WM2200_DSP2_SCRATCH_0_SHIFT 0 /* DSP2_SCRATCH_0 - [15:0] */ +#define WM2200_DSP2_SCRATCH_0_WIDTH 16 /* DSP2_SCRATCH_0 - [15:0] */ + +/* + * R2849 (0xB21) - DSP2 Control 27 + */ +#define WM2200_DSP2_SCRATCH_1_MASK 0xFFFF /* DSP2_SCRATCH_1 - [15:0] */ +#define WM2200_DSP2_SCRATCH_1_SHIFT 0 /* DSP2_SCRATCH_1 - [15:0] */ +#define WM2200_DSP2_SCRATCH_1_WIDTH 16 /* DSP2_SCRATCH_1 - [15:0] */ + +/* + * R2850 (0xB22) - DSP2 Control 28 + */ +#define WM2200_DSP2_SCRATCH_2_MASK 0xFFFF /* DSP2_SCRATCH_2 - [15:0] */ +#define WM2200_DSP2_SCRATCH_2_SHIFT 0 /* DSP2_SCRATCH_2 - [15:0] */ +#define WM2200_DSP2_SCRATCH_2_WIDTH 16 /* DSP2_SCRATCH_2 - [15:0] */ + +/* + * R2851 (0xB23) - DSP2 Control 29 + */ +#define WM2200_DSP2_SCRATCH_3_MASK 0xFFFF /* DSP2_SCRATCH_3 - [15:0] */ +#define WM2200_DSP2_SCRATCH_3_SHIFT 0 /* DSP2_SCRATCH_3 - [15:0] */ +#define WM2200_DSP2_SCRATCH_3_WIDTH 16 /* DSP2_SCRATCH_3 - [15:0] */ + +/* + * R2852 (0xB24) - DSP2 Control 30 + */ +#define WM2200_DSP2_DBG_CLK_ENA 0x0008 /* DSP2_DBG_CLK_ENA */ +#define WM2200_DSP2_DBG_CLK_ENA_MASK 0x0008 /* DSP2_DBG_CLK_ENA */ +#define WM2200_DSP2_DBG_CLK_ENA_SHIFT 3 /* DSP2_DBG_CLK_ENA */ +#define WM2200_DSP2_DBG_CLK_ENA_WIDTH 1 /* DSP2_DBG_CLK_ENA */ +#define WM2200_DSP2_SYS_ENA 0x0004 /* DSP2_SYS_ENA */ +#define WM2200_DSP2_SYS_ENA_MASK 0x0004 /* DSP2_SYS_ENA */ +#define WM2200_DSP2_SYS_ENA_SHIFT 2 /* DSP2_SYS_ENA */ +#define WM2200_DSP2_SYS_ENA_WIDTH 1 /* DSP2_SYS_ENA */ +#define WM2200_DSP2_CORE_ENA 0x0002 /* DSP2_CORE_ENA */ +#define WM2200_DSP2_CORE_ENA_MASK 0x0002 /* DSP2_CORE_ENA */ +#define WM2200_DSP2_CORE_ENA_SHIFT 1 /* DSP2_CORE_ENA */ +#define WM2200_DSP2_CORE_ENA_WIDTH 1 /* DSP2_CORE_ENA */ +#define WM2200_DSP2_START 0x0001 /* DSP2_START */ +#define WM2200_DSP2_START_MASK 0x0001 /* DSP2_START */ +#define WM2200_DSP2_START_SHIFT 0 /* DSP2_START */ +#define WM2200_DSP2_START_WIDTH 1 /* DSP2_START */ + +/* + * R2854 (0xB26) - DSP2 Control 31 + */ +#define WM2200_DSP2_CLK_RATE_MASK 0x0018 /* DSP2_CLK_RATE - [4:3] */ +#define WM2200_DSP2_CLK_RATE_SHIFT 3 /* DSP2_CLK_RATE - [4:3] */ +#define WM2200_DSP2_CLK_RATE_WIDTH 2 /* DSP2_CLK_RATE - [4:3] */ +#define WM2200_DSP2_CLK_AVAIL 0x0004 /* DSP2_CLK_AVAIL */ +#define WM2200_DSP2_CLK_AVAIL_MASK 0x0004 /* DSP2_CLK_AVAIL */ +#define WM2200_DSP2_CLK_AVAIL_SHIFT 2 /* DSP2_CLK_AVAIL */ +#define WM2200_DSP2_CLK_AVAIL_WIDTH 1 /* DSP2_CLK_AVAIL */ +#define WM2200_DSP2_CLK_REQ_MASK 0x0003 /* DSP2_CLK_REQ - [1:0] */ +#define WM2200_DSP2_CLK_REQ_SHIFT 0 /* DSP2_CLK_REQ - [1:0] */ +#define WM2200_DSP2_CLK_REQ_WIDTH 2 /* DSP2_CLK_REQ - [1:0] */ + +#endif -- cgit v1.2.3-18-g5258 From fb644e9ce02a6965a2419325a03a9ea531840bcd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 19:53:58 +0000 Subject: ASoC: dapm: Drop runtime PM references asynchronously We don't really care if any action is taken immediately so let the PM core defer things if it wants to. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 30f9b5c71ee..4973545f2a3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1285,7 +1285,7 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) dev_err(d->dev, "Failed to turn off bias: %d\n", ret); if (d->dev) - pm_runtime_put_sync(d->dev); + pm_runtime_put(d->dev); } /* If we just powered up then move to active bias */ -- cgit v1.2.3-18-g5258 From afe62367e02fd7b83a6a824a20ad432fa5b00040 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 19:55:22 +0000 Subject: ASoC: dapm: Ignore isolated signal generators for power purposes A signal generator has no power control itself and so shouldn't cause a power up of the device. Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4973545f2a3..ec58a314656 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1435,9 +1435,13 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) /* Supplies and micbiases only bring the * context up to STANDBY as unless something * else is active and passing audio they - * generally don't require full power. + * generally don't require full power. Signal + * generators are virtual pins and have no + * power impact themselves. */ switch (w->id) { + case snd_soc_dapm_siggen: + break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: case snd_soc_dapm_micbias: -- cgit v1.2.3-18-g5258 From 17e3e57b65720628754e9afc6919e30776c0c822 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 23 Jan 2012 15:00:17 +0000 Subject: ASoC: wm5100: Convert to devm_regulator_bulk_get() Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index c1c8bdb7bb0..39de946bf25 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2504,8 +2504,9 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(wm5100->core_supplies); i++) wm5100->core_supplies[i].supply = wm5100_core_supply_names[i]; - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); + ret = devm_regulator_bulk_get(&i2c->dev, + ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); @@ -2517,7 +2518,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", ret); - goto err_core; + goto err_regmap; } if (wm5100->pdata.ldo_ena) { @@ -2686,9 +2687,6 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -err_core: - regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); err_regmap: regmap_exit(wm5100->regmap); err: @@ -2711,8 +2709,6 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } - regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); regmap_exit(wm5100->regmap); return 0; -- cgit v1.2.3-18-g5258 From 62c1c40127e351de7fbc1d8c782f7508f8314aab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Jan 2012 17:28:49 +0000 Subject: ASoC: wm5100: Use pm_runtime for powerdown managment Using pm_runtime to decide if the device should go into full power down has the dual advantage of allowing easier integration with non-DAPM reasons to power on the device (like the FLL) and allowing userspace to control the final power down which is useful for tuning retention of DSP firmware. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 116 ++++++++++++++++++++++------------------------ 1 file changed, 56 insertions(+), 60 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 39de946bf25..c6c382197fe 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -1261,54 +1262,6 @@ static const __devinitdata struct reg_default wm5100_reva_patches[] = { { WM5100_AUDIO_IF_3_19, 1 }, }; -static int wm5100_set_bias_level(struct snd_soc_codec *codec, - enum snd_soc_bias_level level) -{ - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - int ret; - - switch (level) { - case SND_SOC_BIAS_ON: - break; - - case SND_SOC_BIAS_PREPARE: - break; - - case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - if (ret != 0) { - dev_err(codec->dev, - "Failed to enable supplies: %d\n", - ret); - return ret; - } - - if (wm5100->pdata.ldo_ena) { - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, - 1); - msleep(2); - } - - regcache_cache_only(wm5100->regmap, false); - regcache_sync(wm5100->regmap); - } - break; - - case SND_SOC_BIAS_OFF: - regcache_cache_only(wm5100->regmap, true); - if (wm5100->pdata.ldo_ena) - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - break; - } - codec->dapm.bias_level = level; - - return 0; -} - static int wm5100_dai_to_base(struct snd_soc_dai *dai) { switch (dai->id) { @@ -1836,6 +1789,8 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (!Fout) { dev_dbg(codec->dev, "FLL%d disabled", fll_id); + if (fll->fout) + pm_runtime_put(codec->dev); fll->fout = 0; snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, 0); return 0; @@ -1880,6 +1835,8 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Clear any pending completions */ try_wait_for_completion(&fll->lock); + pm_runtime_get_sync(codec->dev); + snd_soc_update_bits(codec, base + 1, WM5100_FLL1_ENA, WM5100_FLL1_ENA); if (i2c->irq) @@ -1914,6 +1871,7 @@ static int wm5100_set_fll(struct snd_soc_codec *codec, int fll_id, int source, } if (i == timeout) { dev_err(codec->dev, "FLL%d lock timed out\n", fll_id); + pm_runtime_put(codec->dev); return -ETIMEDOUT; } @@ -2377,9 +2335,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) return ret; } - regcache_cache_only(wm5100->regmap, true); - - for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, WM5100_OUT_VU); @@ -2405,14 +2360,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) } } - /* We'll get woken up again when the system has something useful - * for us to do. - */ - if (wm5100->pdata.ldo_ena) - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), - wm5100->core_supplies); - return 0; err_gpio: @@ -2444,7 +2391,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .set_sysclk = wm5100_set_sysclk, .set_pll = wm5100_set_fll, - .set_bias_level = wm5100_set_bias_level, .idle_bias_off = 1, .reg_cache_size = WM5100_MAX_REGISTER, .volatile_register = wm5100_soc_volatile, @@ -2661,6 +2607,10 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, } } + pm_runtime_set_active(&i2c->dev); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm5100, wm5100_dai, ARRAY_SIZE(wm5100_dai)); @@ -2714,6 +2664,51 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *i2c) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm5100_runtime_suspend(struct device *dev) +{ + struct wm5100_priv *wm5100 = dev_get_drvdata(dev); + + regcache_cache_only(wm5100->regmap, true); + regcache_mark_dirty(wm5100->regmap); + if (wm5100->pdata.ldo_ena) + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + + return 0; +} + +static int wm5100_runtime_resume(struct device *dev) +{ + struct wm5100_priv *wm5100 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), + wm5100->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (wm5100->pdata.ldo_ena) { + gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 1); + msleep(2); + } + + regcache_cache_only(wm5100->regmap, false); + regcache_sync(wm5100->regmap); + + return 0; +} +#endif + +static struct dev_pm_ops wm5100_pm = { + SET_RUNTIME_PM_OPS(wm5100_runtime_suspend, wm5100_runtime_resume, + NULL) +}; + static const struct i2c_device_id wm5100_i2c_id[] = { { "wm5100", 0 }, { } @@ -2724,6 +2719,7 @@ static struct i2c_driver wm5100_i2c_driver = { .driver = { .name = "wm5100", .owner = THIS_MODULE, + .pm = &wm5100_pm, }, .probe = wm5100_i2c_probe, .remove = __devexit_p(wm5100_i2c_remove), -- cgit v1.2.3-18-g5258 From d7b3557077ee1620fbc290e5577bbb7d65063ab1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Jan 2012 18:00:42 +0000 Subject: ASoC: wm8996: Handle failures to determine accessory polarity If we get an indeterminate impedance with both headset polarities then give up and report the accessory as a headphone rather than continually retrying. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 38 ++++++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8e8f8d1fef9..cde11ca9d9e 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -90,6 +90,7 @@ struct wm8996_priv { struct snd_soc_jack *jack; bool detecting; bool jack_mic; + int jack_flips; wm8996_polarity_fn polarity_cb; #ifdef CONFIG_GPIOLIB @@ -2437,6 +2438,7 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8996->jack = jack; wm8996->detecting = true; wm8996->polarity_cb = polarity_cb; + wm8996->jack_flips = 0; if (wm8996->polarity_cb) wm8996->polarity_cb(codec, 0); @@ -2552,6 +2554,19 @@ static void wm8996_hpdet_start(struct snd_soc_codec *codec) WM8996_HP_POLL, WM8996_HP_POLL); } +static void wm8996_report_headphone(struct snd_soc_codec *codec) +{ + dev_dbg(codec->dev, "Headphone detected\n"); + wm8996_hpdet_start(codec); + + /* Increase the detection rate a bit for responsiveness. */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, + WM8996_MICD_RATE_MASK | + WM8996_MICD_BIAS_STARTTIME_MASK, + 7 << WM8996_MICD_RATE_SHIFT | + 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); +} + static void wm8996_micd(struct snd_soc_codec *codec) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); @@ -2571,6 +2586,7 @@ static void wm8996_micd(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Jack removal detected\n"); wm8996->jack_mic = false; wm8996->detecting = true; + wm8996->jack_flips = 0; snd_soc_jack_report(wm8996->jack, 0, SND_JACK_LINEOUT | SND_JACK_HEADSET | SND_JACK_BTN_0); @@ -2611,9 +2627,17 @@ static void wm8996_micd(struct snd_soc_codec *codec) /* If we detected a lower impedence during initial startup * then we probably have the wrong polarity, flip it. Don't * do this for the lowest impedences to speed up detection of - * plain headphones. + * plain headphones. If both polarities report a low + * impedence then give up and report headphones. */ if (wm8996->detecting && (val & 0x3f0)) { + wm8996->jack_flips++; + + if (wm8996->jack_flips > 1) { + wm8996_report_headphone(codec); + return; + } + reg = snd_soc_read(codec, WM8996_ACCESSORY_DETECT_MODE_2); reg ^= WM8996_HPOUT1FB_SRC | WM8996_MICD_SRC | WM8996_MICD_BIAS_SRC; @@ -2640,17 +2664,7 @@ static void wm8996_micd(struct snd_soc_codec *codec) snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); } else if (wm8996->detecting) { - dev_dbg(codec->dev, "Headphone detected\n"); - wm8996_hpdet_start(codec); - - /* Increase the detection rate a bit for - * responsiveness. - */ - snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK | - WM8996_MICD_BIAS_STARTTIME_MASK, - 7 << WM8996_MICD_RATE_SHIFT | - 7 << WM8996_MICD_BIAS_STARTTIME_SHIFT); + wm8996_report_headphone(codec); } } } -- cgit v1.2.3-18-g5258 From 7c08b51f2fbb76b768d78ca6b0e13155d2c1e811 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Jan 2012 18:33:24 +0000 Subject: ASoC: wm8994: Report any low impedance accessory as a headphone Report any accessory with a low impedance as a headphone, previously anything other than a short or microphone would not be reported at all. The most likely reason is a microphone with incorrect polarity. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b047bfada70..11ca19b72d7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3089,7 +3089,7 @@ static void wm8958_default_micdet(u16 status, void *data) } - if (wm8994->mic_detecting && status & 0x4) { + if (wm8994->mic_detecting && status & 0xfc) { dev_dbg(codec->dev, "Detected headphone\n"); wm8994->mic_detecting = false; -- cgit v1.2.3-18-g5258 From eb3032f8b9c06b3ff06a318aa5842c5e14e1fa95 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 27 Jan 2012 18:02:09 +0800 Subject: ASoC: Set idle_bias_off flag in snd_soc_codec_driver Since commit 33c5f969 "ASoC: Allow idle_bias_off to be specified in CODEC drivers", now we can set idle_bias_off flag in struct snd_soc_codec_driver for devices can unconditionally support idle_bias_off. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 3 +-- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/sn95031.c | 3 +-- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/wm8770.c | 3 +-- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 4 +--- 11 files changed, 11 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 971ba452917..facda33db1c 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1244,8 +1244,6 @@ static int adau1373_probe(struct snd_soc_codec *codec) return ret; } - codec->dapm.idle_bias_off = true; - if (pdata) { if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) return -EINVAL; @@ -1340,6 +1338,7 @@ static struct snd_soc_codec_driver adau1373_codec_driver = { .suspend = adau1373_suspend, .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), .reg_cache_default = adau1373_default_regs, .reg_word_size = sizeof(uint8_t), diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 6b325ea0386..78e9ce48bb9 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -457,7 +457,6 @@ static int adau1701_probe(struct snd_soc_codec *codec) { int ret; - codec->dapm.idle_bias_off = 1; codec->control_data = to_i2c_client(codec->dev); ret = adau1701_load_firmware(codec); @@ -473,6 +472,7 @@ static int adau1701_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver adau1701_codec_drv = { .probe = adau1701_probe, .set_bias_level = adau1701_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ADAU1701_NUM_REGS, .reg_word_size = sizeof(u16), diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index f99baa0b8c3..aa0392360da 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -827,8 +827,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - codec->dapm.idle_bias_off = 1; - /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) @@ -891,6 +889,7 @@ struct snd_soc_codec_driver sn95031_codec = { .read = sn95031_read, .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, + .idle_bias_off = true, .dapm_widgets = sn95031_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets), .dapm_routes = sn95031_audio_map, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 492f22f8a4d..285b7a22dc1 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1377,7 +1377,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - codec->dapm.idle_bias_off = 1; ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); if (ret != 0) { @@ -1471,6 +1470,7 @@ static int aic3x_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .set_bias_level = aic3x_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(aic3x_reg), .reg_word_size = sizeof(u8), .reg_cache_default = aic3x_reg, diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index c06c3e4b912..2c957c84570 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1395,7 +1395,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) codec->control_data = dac33->control_data; codec->hw_write = (hw_write_t) i2c_master_send; - codec->dapm.idle_bias_off = 1; dac33->codec = codec; /* Read the tlv320dac33 ID registers */ @@ -1476,6 +1475,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = { .read = dac33_read_reg_cache, .write = dac33_write_locked, .set_bias_level = dac33_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(dac33_reg), .reg_word_size = sizeof(u8), .reg_cache_default = dac33_reg, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index a193f5fa4b3..3039ba209d1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2227,7 +2227,6 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, twl4030); /* Set the defaults, and power up the codec */ twl4030->sysclk = twl4030_audio_get_mclk() / 1000; - codec->dapm.idle_bias_off = 1; twl4030_init_chip(codec); @@ -2253,6 +2252,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = { .read = twl4030_read_reg_cache, .write = twl4030_write, .set_bias_level = twl4030_set_bias_level, + .idle_bias_off = true, .reg_cache_size = sizeof(twl4030_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl4030_reg, diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 19374a9e5ba..bd60f847762 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -580,8 +580,6 @@ static int wm8770_probe(struct snd_soc_codec *codec) wm8770 = snd_soc_codec_get_drvdata(codec); wm8770->codec = codec; - codec->dapm.idle_bias_off = 1; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8770->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); @@ -679,6 +677,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8770 = { .suspend = wm8770_suspend, .resume = wm8770_resume, .set_bias_level = wm8770_set_bias_level, + .idle_bias_off = true, .reg_cache_size = ARRAY_SIZE(wm8770_reg_defs), .reg_word_size = sizeof (u16), .reg_cache_default = wm8770_reg_defs diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 8abe3757a97..7ee8dcf1fe3 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -549,7 +549,6 @@ static int wm8804_probe(struct snd_soc_codec *codec) wm8804 = snd_soc_codec_get_drvdata(codec); - codec->dapm.idle_bias_off = 1; codec->control_data = wm8804->regmap; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); @@ -678,6 +677,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8804 = { .suspend = wm8804_suspend, .resume = wm8804_resume, .set_bias_level = wm8804_set_bias_level, + .idle_bias_off = true, .controls = wm8804_snd_controls, .num_controls = ARRAY_SIZE(wm8804_snd_controls), diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 14afc119334..37079eace41 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2088,7 +2088,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) int ret, i; codec->cache_sync = 1; - codec->dapm.idle_bias_off = 1; codec->control_data = wm8904->regmap; switch (wm8904->devtype) { @@ -2237,6 +2236,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8904 = { .suspend = wm8904_suspend, .resume = wm8904_resume, .set_bias_level = wm8904_set_bias_level, + .idle_bias_off = true, }; static const struct regmap_config wm8904_regmap = { diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index c8aada597d7..89a864287c1 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2047,7 +2047,6 @@ static int wm8995_probe(struct snd_soc_codec *codec) int i; int ret; - codec->dapm.idle_bias_off = 1; wm8995 = snd_soc_codec_get_drvdata(codec); wm8995->codec = codec; @@ -2241,6 +2240,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8995 = { .suspend = wm8995_suspend, .resume = wm8995_resume, .set_bias_level = wm8995_set_bias_level, + .idle_bias_off = true, }; static struct regmap_config wm8995_regmap = { diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index cde11ca9d9e..7f7e914f5a9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2804,7 +2804,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) int ret; struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - struct snd_soc_dapm_context *dapm = &codec->dapm; int i, irq_flags; wm8996->codec = codec; @@ -2812,8 +2811,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) init_completion(&wm8996->dcs_done); init_completion(&wm8996->fll_lock); - dapm->idle_bias_off = true; - codec->control_data = wm8996->regmap; ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); @@ -3067,6 +3064,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .probe = wm8996_probe, .remove = wm8996_remove, .set_bias_level = wm8996_set_bias_level, + .idle_bias_off = true, .seq_notifier = wm8996_seq_notifier, .controls = wm8996_snd_controls, .num_controls = ARRAY_SIZE(wm8996_snd_controls), -- cgit v1.2.3-18-g5258 From 98654d3fa2e6983378e3510131c5c45be97c4906 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 27 Jan 2012 15:23:51 +0800 Subject: ALSA: aoa: Convert onyx and tas codec drivers to module_i2c_driver This patch converts onyx and tas codec drivers to use the module_i2c_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin Signed-off-by: Takashi Iwai --- sound/aoa/codecs/onyx.c | 13 +------------ sound/aoa/codecs/tas.c | 13 +------------ 2 files changed, 2 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 762af68c899..270790d384e 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1132,15 +1132,4 @@ static struct i2c_driver onyx_driver = { .id_table = onyx_i2c_id, }; -static int __init onyx_init(void) -{ - return i2c_add_driver(&onyx_driver); -} - -static void __exit onyx_exit(void) -{ - i2c_del_driver(&onyx_driver); -} - -module_init(onyx_init); -module_exit(onyx_exit); +module_i2c_driver(onyx_driver); diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index fd2188c3df2..8e63d1f35ce 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -1026,15 +1026,4 @@ static struct i2c_driver tas_driver = { .id_table = tas_i2c_id, }; -static int __init tas_init(void) -{ - return i2c_add_driver(&tas_driver); -} - -static void __exit tas_exit(void) -{ - i2c_del_driver(&tas_driver); -} - -module_init(tas_init); -module_exit(tas_exit); +module_i2c_driver(tas_driver); -- cgit v1.2.3-18-g5258 From f443ac935a2fd80f177c6b5a580cc54ef18c552d Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 27 Jan 2012 15:29:13 +0800 Subject: ALSA: Convert at73c213 to module_spi_driver This patch converts at73c213 to use the module_spi_driver() macro which makes the code smaller and a bit simpler. Signed-off-by: Axel Lin Signed-off-by: Takashi Iwai --- sound/spi/at73c213.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4dd051bdf4f..c6500d00053 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1112,17 +1112,7 @@ static struct spi_driver at73c213_driver = { .remove = __devexit_p(snd_at73c213_remove), }; -static int __init at73c213_init(void) -{ - return spi_register_driver(&at73c213_driver); -} -module_init(at73c213_init); - -static void __exit at73c213_exit(void) -{ - spi_unregister_driver(&at73c213_driver); -} -module_exit(at73c213_exit); +module_spi_driver(at73c213_driver); MODULE_AUTHOR("Hans-Christian Egtvedt "); MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); -- cgit v1.2.3-18-g5258 From 2693efd660e3385b3bf02261f773939a037fe560 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 27 Jan 2012 19:36:45 +0000 Subject: ASoC: wm8962: Enable idle_bias_off The WM8962 has ground referenced and class D outputs so can happily go down to BIAS_OFF without a large startup time. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2a654fd42d1..b6fcdcc4341 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3636,6 +3636,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .set_pll = wm8962_set_fll, .reg_cache_size = WM8962_MAX_REGISTER, .volatile_register = wm8962_soc_volatile, + .idle_bias_off = true, }; /* Improve power consumption for IN4 DC measurement mode */ -- cgit v1.2.3-18-g5258 From ffa8d9df55206b72c94b33138e1feaafcaeaffdb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 29 Jan 2012 21:45:31 +0000 Subject: ASoC: wm2200: Remove trailing whitespace Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 678 +++++++++++++++++++++++----------------------- 1 file changed, 339 insertions(+), 339 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 956490d320a..0db24758c46 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -65,344 +65,344 @@ struct wm2200_priv { }; static struct reg_default wm2200_reg_defaults[] = { - { 0x000B, 0x0000 }, /* R11 - Tone Generator 1 */ - { 0x0102, 0x0000 }, /* R258 - Clocking 3 */ - { 0x0103, 0x0011 }, /* R259 - Clocking 4 */ - { 0x0111, 0x0000 }, /* R273 - FLL Control 1 */ - { 0x0112, 0x0000 }, /* R274 - FLL Control 2 */ - { 0x0113, 0x0000 }, /* R275 - FLL Control 3 */ - { 0x0114, 0x0000 }, /* R276 - FLL Control 4 */ - { 0x0116, 0x0177 }, /* R278 - FLL Control 6 */ - { 0x0117, 0x0004 }, /* R279 - FLL Control 7 */ - { 0x0119, 0x0000 }, /* R281 - FLL EFS 1 */ - { 0x011A, 0x0002 }, /* R282 - FLL EFS 2 */ - { 0x0200, 0x0000 }, /* R512 - Mic Charge Pump 1 */ - { 0x0201, 0x03FF }, /* R513 - Mic Charge Pump 2 */ - { 0x0202, 0x9BDE }, /* R514 - DM Charge Pump 1 */ - { 0x020C, 0x0000 }, /* R524 - Mic Bias Ctrl 1 */ - { 0x020D, 0x0000 }, /* R525 - Mic Bias Ctrl 2 */ - { 0x020F, 0x0000 }, /* R527 - Ear Piece Ctrl 1 */ - { 0x0210, 0x0000 }, /* R528 - Ear Piece Ctrl 2 */ - { 0x0301, 0x0000 }, /* R769 - Input Enables */ - { 0x0302, 0x2240 }, /* R770 - IN1L Control */ - { 0x0303, 0x0040 }, /* R771 - IN1R Control */ - { 0x0304, 0x2240 }, /* R772 - IN2L Control */ - { 0x0305, 0x0040 }, /* R773 - IN2R Control */ - { 0x0306, 0x2240 }, /* R774 - IN3L Control */ - { 0x0307, 0x0040 }, /* R775 - IN3R Control */ - { 0x030A, 0x0000 }, /* R778 - RXANC_SRC */ - { 0x030B, 0x0022 }, /* R779 - Input Volume Ramp */ - { 0x030C, 0x0180 }, /* R780 - ADC Digital Volume 1L */ - { 0x030D, 0x0180 }, /* R781 - ADC Digital Volume 1R */ - { 0x030E, 0x0180 }, /* R782 - ADC Digital Volume 2L */ - { 0x030F, 0x0180 }, /* R783 - ADC Digital Volume 2R */ - { 0x0310, 0x0180 }, /* R784 - ADC Digital Volume 3L */ - { 0x0311, 0x0180 }, /* R785 - ADC Digital Volume 3R */ - { 0x0400, 0x0000 }, /* R1024 - Output Enables */ - { 0x0401, 0x0000 }, /* R1025 - DAC Volume Limit 1L */ - { 0x0402, 0x0000 }, /* R1026 - DAC Volume Limit 1R */ - { 0x0403, 0x0000 }, /* R1027 - DAC Volume Limit 2L */ - { 0x0404, 0x0000 }, /* R1028 - DAC Volume Limit 2R */ - { 0x0409, 0x0000 }, /* R1033 - DAC AEC Control 1 */ - { 0x040A, 0x0022 }, /* R1034 - Output Volume Ramp */ - { 0x040B, 0x0180 }, /* R1035 - DAC Digital Volume 1L */ - { 0x040C, 0x0180 }, /* R1036 - DAC Digital Volume 1R */ - { 0x040D, 0x0180 }, /* R1037 - DAC Digital Volume 2L */ - { 0x040E, 0x0180 }, /* R1038 - DAC Digital Volume 2R */ - { 0x0417, 0x0069 }, /* R1047 - PDM 1 */ - { 0x0418, 0x0000 }, /* R1048 - PDM 2 */ - { 0x0500, 0x0000 }, /* R1280 - Audio IF 1_1 */ - { 0x0501, 0x0008 }, /* R1281 - Audio IF 1_2 */ - { 0x0502, 0x0000 }, /* R1282 - Audio IF 1_3 */ - { 0x0503, 0x0000 }, /* R1283 - Audio IF 1_4 */ - { 0x0504, 0x0000 }, /* R1284 - Audio IF 1_5 */ - { 0x0505, 0x0001 }, /* R1285 - Audio IF 1_6 */ - { 0x0506, 0x0001 }, /* R1286 - Audio IF 1_7 */ - { 0x0507, 0x0000 }, /* R1287 - Audio IF 1_8 */ - { 0x0508, 0x0000 }, /* R1288 - Audio IF 1_9 */ - { 0x0509, 0x0000 }, /* R1289 - Audio IF 1_10 */ - { 0x050A, 0x0000 }, /* R1290 - Audio IF 1_11 */ - { 0x050B, 0x0000 }, /* R1291 - Audio IF 1_12 */ - { 0x050C, 0x0000 }, /* R1292 - Audio IF 1_13 */ - { 0x050D, 0x0000 }, /* R1293 - Audio IF 1_14 */ - { 0x050E, 0x0000 }, /* R1294 - Audio IF 1_15 */ - { 0x050F, 0x0000 }, /* R1295 - Audio IF 1_16 */ - { 0x0510, 0x0000 }, /* R1296 - Audio IF 1_17 */ - { 0x0511, 0x0000 }, /* R1297 - Audio IF 1_18 */ - { 0x0512, 0x0000 }, /* R1298 - Audio IF 1_19 */ - { 0x0513, 0x0000 }, /* R1299 - Audio IF 1_20 */ - { 0x0514, 0x0000 }, /* R1300 - Audio IF 1_21 */ - { 0x0515, 0x0001 }, /* R1301 - Audio IF 1_22 */ - { 0x0600, 0x0000 }, /* R1536 - OUT1LMIX Input 1 Source */ - { 0x0601, 0x0080 }, /* R1537 - OUT1LMIX Input 1 Volume */ - { 0x0602, 0x0000 }, /* R1538 - OUT1LMIX Input 2 Source */ - { 0x0603, 0x0080 }, /* R1539 - OUT1LMIX Input 2 Volume */ - { 0x0604, 0x0000 }, /* R1540 - OUT1LMIX Input 3 Source */ - { 0x0605, 0x0080 }, /* R1541 - OUT1LMIX Input 3 Volume */ - { 0x0606, 0x0000 }, /* R1542 - OUT1LMIX Input 4 Source */ - { 0x0607, 0x0080 }, /* R1543 - OUT1LMIX Input 4 Volume */ - { 0x0608, 0x0000 }, /* R1544 - OUT1RMIX Input 1 Source */ - { 0x0609, 0x0080 }, /* R1545 - OUT1RMIX Input 1 Volume */ - { 0x060A, 0x0000 }, /* R1546 - OUT1RMIX Input 2 Source */ - { 0x060B, 0x0080 }, /* R1547 - OUT1RMIX Input 2 Volume */ - { 0x060C, 0x0000 }, /* R1548 - OUT1RMIX Input 3 Source */ - { 0x060D, 0x0080 }, /* R1549 - OUT1RMIX Input 3 Volume */ - { 0x060E, 0x0000 }, /* R1550 - OUT1RMIX Input 4 Source */ - { 0x060F, 0x0080 }, /* R1551 - OUT1RMIX Input 4 Volume */ - { 0x0610, 0x0000 }, /* R1552 - OUT2LMIX Input 1 Source */ - { 0x0611, 0x0080 }, /* R1553 - OUT2LMIX Input 1 Volume */ - { 0x0612, 0x0000 }, /* R1554 - OUT2LMIX Input 2 Source */ - { 0x0613, 0x0080 }, /* R1555 - OUT2LMIX Input 2 Volume */ - { 0x0614, 0x0000 }, /* R1556 - OUT2LMIX Input 3 Source */ - { 0x0615, 0x0080 }, /* R1557 - OUT2LMIX Input 3 Volume */ - { 0x0616, 0x0000 }, /* R1558 - OUT2LMIX Input 4 Source */ - { 0x0617, 0x0080 }, /* R1559 - OUT2LMIX Input 4 Volume */ - { 0x0618, 0x0000 }, /* R1560 - OUT2RMIX Input 1 Source */ - { 0x0619, 0x0080 }, /* R1561 - OUT2RMIX Input 1 Volume */ - { 0x061A, 0x0000 }, /* R1562 - OUT2RMIX Input 2 Source */ - { 0x061B, 0x0080 }, /* R1563 - OUT2RMIX Input 2 Volume */ - { 0x061C, 0x0000 }, /* R1564 - OUT2RMIX Input 3 Source */ - { 0x061D, 0x0080 }, /* R1565 - OUT2RMIX Input 3 Volume */ - { 0x061E, 0x0000 }, /* R1566 - OUT2RMIX Input 4 Source */ - { 0x061F, 0x0080 }, /* R1567 - OUT2RMIX Input 4 Volume */ - { 0x0620, 0x0000 }, /* R1568 - AIF1TX1MIX Input 1 Source */ - { 0x0621, 0x0080 }, /* R1569 - AIF1TX1MIX Input 1 Volume */ - { 0x0622, 0x0000 }, /* R1570 - AIF1TX1MIX Input 2 Source */ - { 0x0623, 0x0080 }, /* R1571 - AIF1TX1MIX Input 2 Volume */ - { 0x0624, 0x0000 }, /* R1572 - AIF1TX1MIX Input 3 Source */ - { 0x0625, 0x0080 }, /* R1573 - AIF1TX1MIX Input 3 Volume */ - { 0x0626, 0x0000 }, /* R1574 - AIF1TX1MIX Input 4 Source */ - { 0x0627, 0x0080 }, /* R1575 - AIF1TX1MIX Input 4 Volume */ - { 0x0628, 0x0000 }, /* R1576 - AIF1TX2MIX Input 1 Source */ - { 0x0629, 0x0080 }, /* R1577 - AIF1TX2MIX Input 1 Volume */ - { 0x062A, 0x0000 }, /* R1578 - AIF1TX2MIX Input 2 Source */ - { 0x062B, 0x0080 }, /* R1579 - AIF1TX2MIX Input 2 Volume */ - { 0x062C, 0x0000 }, /* R1580 - AIF1TX2MIX Input 3 Source */ - { 0x062D, 0x0080 }, /* R1581 - AIF1TX2MIX Input 3 Volume */ - { 0x062E, 0x0000 }, /* R1582 - AIF1TX2MIX Input 4 Source */ - { 0x062F, 0x0080 }, /* R1583 - AIF1TX2MIX Input 4 Volume */ - { 0x0630, 0x0000 }, /* R1584 - AIF1TX3MIX Input 1 Source */ - { 0x0631, 0x0080 }, /* R1585 - AIF1TX3MIX Input 1 Volume */ - { 0x0632, 0x0000 }, /* R1586 - AIF1TX3MIX Input 2 Source */ - { 0x0633, 0x0080 }, /* R1587 - AIF1TX3MIX Input 2 Volume */ - { 0x0634, 0x0000 }, /* R1588 - AIF1TX3MIX Input 3 Source */ - { 0x0635, 0x0080 }, /* R1589 - AIF1TX3MIX Input 3 Volume */ - { 0x0636, 0x0000 }, /* R1590 - AIF1TX3MIX Input 4 Source */ - { 0x0637, 0x0080 }, /* R1591 - AIF1TX3MIX Input 4 Volume */ - { 0x0638, 0x0000 }, /* R1592 - AIF1TX4MIX Input 1 Source */ - { 0x0639, 0x0080 }, /* R1593 - AIF1TX4MIX Input 1 Volume */ - { 0x063A, 0x0000 }, /* R1594 - AIF1TX4MIX Input 2 Source */ - { 0x063B, 0x0080 }, /* R1595 - AIF1TX4MIX Input 2 Volume */ - { 0x063C, 0x0000 }, /* R1596 - AIF1TX4MIX Input 3 Source */ - { 0x063D, 0x0080 }, /* R1597 - AIF1TX4MIX Input 3 Volume */ - { 0x063E, 0x0000 }, /* R1598 - AIF1TX4MIX Input 4 Source */ - { 0x063F, 0x0080 }, /* R1599 - AIF1TX4MIX Input 4 Volume */ - { 0x0640, 0x0000 }, /* R1600 - AIF1TX5MIX Input 1 Source */ - { 0x0641, 0x0080 }, /* R1601 - AIF1TX5MIX Input 1 Volume */ - { 0x0642, 0x0000 }, /* R1602 - AIF1TX5MIX Input 2 Source */ - { 0x0643, 0x0080 }, /* R1603 - AIF1TX5MIX Input 2 Volume */ - { 0x0644, 0x0000 }, /* R1604 - AIF1TX5MIX Input 3 Source */ - { 0x0645, 0x0080 }, /* R1605 - AIF1TX5MIX Input 3 Volume */ - { 0x0646, 0x0000 }, /* R1606 - AIF1TX5MIX Input 4 Source */ - { 0x0647, 0x0080 }, /* R1607 - AIF1TX5MIX Input 4 Volume */ - { 0x0648, 0x0000 }, /* R1608 - AIF1TX6MIX Input 1 Source */ - { 0x0649, 0x0080 }, /* R1609 - AIF1TX6MIX Input 1 Volume */ - { 0x064A, 0x0000 }, /* R1610 - AIF1TX6MIX Input 2 Source */ - { 0x064B, 0x0080 }, /* R1611 - AIF1TX6MIX Input 2 Volume */ - { 0x064C, 0x0000 }, /* R1612 - AIF1TX6MIX Input 3 Source */ - { 0x064D, 0x0080 }, /* R1613 - AIF1TX6MIX Input 3 Volume */ - { 0x064E, 0x0000 }, /* R1614 - AIF1TX6MIX Input 4 Source */ - { 0x064F, 0x0080 }, /* R1615 - AIF1TX6MIX Input 4 Volume */ - { 0x0650, 0x0000 }, /* R1616 - EQLMIX Input 1 Source */ - { 0x0651, 0x0080 }, /* R1617 - EQLMIX Input 1 Volume */ - { 0x0652, 0x0000 }, /* R1618 - EQLMIX Input 2 Source */ - { 0x0653, 0x0080 }, /* R1619 - EQLMIX Input 2 Volume */ - { 0x0654, 0x0000 }, /* R1620 - EQLMIX Input 3 Source */ - { 0x0655, 0x0080 }, /* R1621 - EQLMIX Input 3 Volume */ - { 0x0656, 0x0000 }, /* R1622 - EQLMIX Input 4 Source */ - { 0x0657, 0x0080 }, /* R1623 - EQLMIX Input 4 Volume */ - { 0x0658, 0x0000 }, /* R1624 - EQRMIX Input 1 Source */ - { 0x0659, 0x0080 }, /* R1625 - EQRMIX Input 1 Volume */ - { 0x065A, 0x0000 }, /* R1626 - EQRMIX Input 2 Source */ - { 0x065B, 0x0080 }, /* R1627 - EQRMIX Input 2 Volume */ - { 0x065C, 0x0000 }, /* R1628 - EQRMIX Input 3 Source */ - { 0x065D, 0x0080 }, /* R1629 - EQRMIX Input 3 Volume */ - { 0x065E, 0x0000 }, /* R1630 - EQRMIX Input 4 Source */ - { 0x065F, 0x0080 }, /* R1631 - EQRMIX Input 4 Volume */ - { 0x0660, 0x0000 }, /* R1632 - LHPF1MIX Input 1 Source */ - { 0x0661, 0x0080 }, /* R1633 - LHPF1MIX Input 1 Volume */ - { 0x0662, 0x0000 }, /* R1634 - LHPF1MIX Input 2 Source */ - { 0x0663, 0x0080 }, /* R1635 - LHPF1MIX Input 2 Volume */ - { 0x0664, 0x0000 }, /* R1636 - LHPF1MIX Input 3 Source */ - { 0x0665, 0x0080 }, /* R1637 - LHPF1MIX Input 3 Volume */ - { 0x0666, 0x0000 }, /* R1638 - LHPF1MIX Input 4 Source */ - { 0x0667, 0x0080 }, /* R1639 - LHPF1MIX Input 4 Volume */ - { 0x0668, 0x0000 }, /* R1640 - LHPF2MIX Input 1 Source */ - { 0x0669, 0x0080 }, /* R1641 - LHPF2MIX Input 1 Volume */ - { 0x066A, 0x0000 }, /* R1642 - LHPF2MIX Input 2 Source */ - { 0x066B, 0x0080 }, /* R1643 - LHPF2MIX Input 2 Volume */ - { 0x066C, 0x0000 }, /* R1644 - LHPF2MIX Input 3 Source */ - { 0x066D, 0x0080 }, /* R1645 - LHPF2MIX Input 3 Volume */ - { 0x066E, 0x0000 }, /* R1646 - LHPF2MIX Input 4 Source */ - { 0x066F, 0x0080 }, /* R1647 - LHPF2MIX Input 4 Volume */ - { 0x0670, 0x0000 }, /* R1648 - DSP1LMIX Input 1 Source */ - { 0x0671, 0x0080 }, /* R1649 - DSP1LMIX Input 1 Volume */ - { 0x0672, 0x0000 }, /* R1650 - DSP1LMIX Input 2 Source */ - { 0x0673, 0x0080 }, /* R1651 - DSP1LMIX Input 2 Volume */ - { 0x0674, 0x0000 }, /* R1652 - DSP1LMIX Input 3 Source */ - { 0x0675, 0x0080 }, /* R1653 - DSP1LMIX Input 3 Volume */ - { 0x0676, 0x0000 }, /* R1654 - DSP1LMIX Input 4 Source */ - { 0x0677, 0x0080 }, /* R1655 - DSP1LMIX Input 4 Volume */ - { 0x0678, 0x0000 }, /* R1656 - DSP1RMIX Input 1 Source */ - { 0x0679, 0x0080 }, /* R1657 - DSP1RMIX Input 1 Volume */ - { 0x067A, 0x0000 }, /* R1658 - DSP1RMIX Input 2 Source */ - { 0x067B, 0x0080 }, /* R1659 - DSP1RMIX Input 2 Volume */ - { 0x067C, 0x0000 }, /* R1660 - DSP1RMIX Input 3 Source */ - { 0x067D, 0x0080 }, /* R1661 - DSP1RMIX Input 3 Volume */ - { 0x067E, 0x0000 }, /* R1662 - DSP1RMIX Input 4 Source */ - { 0x067F, 0x0080 }, /* R1663 - DSP1RMIX Input 4 Volume */ - { 0x0680, 0x0000 }, /* R1664 - DSP1AUX1MIX Input 1 Source */ - { 0x0681, 0x0000 }, /* R1665 - DSP1AUX2MIX Input 1 Source */ - { 0x0682, 0x0000 }, /* R1666 - DSP1AUX3MIX Input 1 Source */ - { 0x0683, 0x0000 }, /* R1667 - DSP1AUX4MIX Input 1 Source */ - { 0x0684, 0x0000 }, /* R1668 - DSP1AUX5MIX Input 1 Source */ - { 0x0685, 0x0000 }, /* R1669 - DSP1AUX6MIX Input 1 Source */ - { 0x0686, 0x0000 }, /* R1670 - DSP2LMIX Input 1 Source */ - { 0x0687, 0x0080 }, /* R1671 - DSP2LMIX Input 1 Volume */ - { 0x0688, 0x0000 }, /* R1672 - DSP2LMIX Input 2 Source */ - { 0x0689, 0x0080 }, /* R1673 - DSP2LMIX Input 2 Volume */ - { 0x068A, 0x0000 }, /* R1674 - DSP2LMIX Input 3 Source */ - { 0x068B, 0x0080 }, /* R1675 - DSP2LMIX Input 3 Volume */ - { 0x068C, 0x0000 }, /* R1676 - DSP2LMIX Input 4 Source */ - { 0x068D, 0x0080 }, /* R1677 - DSP2LMIX Input 4 Volume */ - { 0x068E, 0x0000 }, /* R1678 - DSP2RMIX Input 1 Source */ - { 0x068F, 0x0080 }, /* R1679 - DSP2RMIX Input 1 Volume */ - { 0x0690, 0x0000 }, /* R1680 - DSP2RMIX Input 2 Source */ - { 0x0691, 0x0080 }, /* R1681 - DSP2RMIX Input 2 Volume */ - { 0x0692, 0x0000 }, /* R1682 - DSP2RMIX Input 3 Source */ - { 0x0693, 0x0080 }, /* R1683 - DSP2RMIX Input 3 Volume */ - { 0x0694, 0x0000 }, /* R1684 - DSP2RMIX Input 4 Source */ - { 0x0695, 0x0080 }, /* R1685 - DSP2RMIX Input 4 Volume */ - { 0x0696, 0x0000 }, /* R1686 - DSP2AUX1MIX Input 1 Source */ - { 0x0697, 0x0000 }, /* R1687 - DSP2AUX2MIX Input 1 Source */ - { 0x0698, 0x0000 }, /* R1688 - DSP2AUX3MIX Input 1 Source */ - { 0x0699, 0x0000 }, /* R1689 - DSP2AUX4MIX Input 1 Source */ - { 0x069A, 0x0000 }, /* R1690 - DSP2AUX5MIX Input 1 Source */ - { 0x069B, 0x0000 }, /* R1691 - DSP2AUX6MIX Input 1 Source */ - { 0x0700, 0xA101 }, /* R1792 - GPIO CTRL 1 */ - { 0x0701, 0xA101 }, /* R1793 - GPIO CTRL 2 */ - { 0x0702, 0xA101 }, /* R1794 - GPIO CTRL 3 */ - { 0x0703, 0xA101 }, /* R1795 - GPIO CTRL 4 */ - { 0x0709, 0x0000 }, /* R1801 - Misc Pad Ctrl 1 */ - { 0x0801, 0x00FF }, /* R2049 - Interrupt Status 1 Mask */ - { 0x0804, 0xFFFF }, /* R2052 - Interrupt Status 2 Mask */ - { 0x0808, 0x0000 }, /* R2056 - Interrupt Control */ - { 0x0900, 0x0000 }, /* R2304 - EQL_1 */ - { 0x0901, 0x0000 }, /* R2305 - EQL_2 */ - { 0x0902, 0x0000 }, /* R2306 - EQL_3 */ - { 0x0903, 0x0000 }, /* R2307 - EQL_4 */ - { 0x0904, 0x0000 }, /* R2308 - EQL_5 */ - { 0x0905, 0x0000 }, /* R2309 - EQL_6 */ - { 0x0906, 0x0000 }, /* R2310 - EQL_7 */ - { 0x0907, 0x0000 }, /* R2311 - EQL_8 */ - { 0x0908, 0x0000 }, /* R2312 - EQL_9 */ - { 0x0909, 0x0000 }, /* R2313 - EQL_10 */ - { 0x090A, 0x0000 }, /* R2314 - EQL_11 */ - { 0x090B, 0x0000 }, /* R2315 - EQL_12 */ - { 0x090C, 0x0000 }, /* R2316 - EQL_13 */ - { 0x090D, 0x0000 }, /* R2317 - EQL_14 */ - { 0x090E, 0x0000 }, /* R2318 - EQL_15 */ - { 0x090F, 0x0000 }, /* R2319 - EQL_16 */ - { 0x0910, 0x0000 }, /* R2320 - EQL_17 */ - { 0x0911, 0x0000 }, /* R2321 - EQL_18 */ - { 0x0912, 0x0000 }, /* R2322 - EQL_19 */ - { 0x0913, 0x0000 }, /* R2323 - EQL_20 */ - { 0x0916, 0x0000 }, /* R2326 - EQR_1 */ - { 0x0917, 0x0000 }, /* R2327 - EQR_2 */ - { 0x0918, 0x0000 }, /* R2328 - EQR_3 */ - { 0x0919, 0x0000 }, /* R2329 - EQR_4 */ - { 0x091A, 0x0000 }, /* R2330 - EQR_5 */ - { 0x091B, 0x0000 }, /* R2331 - EQR_6 */ - { 0x091C, 0x0000 }, /* R2332 - EQR_7 */ - { 0x091D, 0x0000 }, /* R2333 - EQR_8 */ - { 0x091E, 0x0000 }, /* R2334 - EQR_9 */ - { 0x091F, 0x0000 }, /* R2335 - EQR_10 */ - { 0x0920, 0x0000 }, /* R2336 - EQR_11 */ - { 0x0921, 0x0000 }, /* R2337 - EQR_12 */ - { 0x0922, 0x0000 }, /* R2338 - EQR_13 */ - { 0x0923, 0x0000 }, /* R2339 - EQR_14 */ - { 0x0924, 0x0000 }, /* R2340 - EQR_15 */ - { 0x0925, 0x0000 }, /* R2341 - EQR_16 */ - { 0x0926, 0x0000 }, /* R2342 - EQR_17 */ - { 0x0927, 0x0000 }, /* R2343 - EQR_18 */ - { 0x0928, 0x0000 }, /* R2344 - EQR_19 */ - { 0x0929, 0x0000 }, /* R2345 - EQR_20 */ - { 0x093E, 0x0000 }, /* R2366 - HPLPF1_1 */ - { 0x093F, 0x0000 }, /* R2367 - HPLPF1_2 */ - { 0x0942, 0x0000 }, /* R2370 - HPLPF2_1 */ - { 0x0943, 0x0000 }, /* R2371 - HPLPF2_2 */ - { 0x0A00, 0x0000 }, /* R2560 - DSP1 Control 1 */ - { 0x0A02, 0x0000 }, /* R2562 - DSP1 Control 2 */ - { 0x0A03, 0x0000 }, /* R2563 - DSP1 Control 3 */ - { 0x0A04, 0x0000 }, /* R2564 - DSP1 Control 4 */ - { 0x0A06, 0x0000 }, /* R2566 - DSP1 Control 5 */ - { 0x0A07, 0x0000 }, /* R2567 - DSP1 Control 6 */ - { 0x0A08, 0x0000 }, /* R2568 - DSP1 Control 7 */ - { 0x0A09, 0x0000 }, /* R2569 - DSP1 Control 8 */ - { 0x0A0A, 0x0000 }, /* R2570 - DSP1 Control 9 */ - { 0x0A0B, 0x0000 }, /* R2571 - DSP1 Control 10 */ - { 0x0A0C, 0x0000 }, /* R2572 - DSP1 Control 11 */ - { 0x0A0D, 0x0000 }, /* R2573 - DSP1 Control 12 */ - { 0x0A0F, 0x0000 }, /* R2575 - DSP1 Control 13 */ - { 0x0A10, 0x0000 }, /* R2576 - DSP1 Control 14 */ - { 0x0A11, 0x0000 }, /* R2577 - DSP1 Control 15 */ - { 0x0A12, 0x0000 }, /* R2578 - DSP1 Control 16 */ - { 0x0A13, 0x0000 }, /* R2579 - DSP1 Control 17 */ - { 0x0A14, 0x0000 }, /* R2580 - DSP1 Control 18 */ - { 0x0A16, 0x0000 }, /* R2582 - DSP1 Control 19 */ - { 0x0A17, 0x0000 }, /* R2583 - DSP1 Control 20 */ - { 0x0A18, 0x0000 }, /* R2584 - DSP1 Control 21 */ - { 0x0A1A, 0x1800 }, /* R2586 - DSP1 Control 22 */ - { 0x0A1B, 0x1000 }, /* R2587 - DSP1 Control 23 */ - { 0x0A1C, 0x0400 }, /* R2588 - DSP1 Control 24 */ - { 0x0A1E, 0x0000 }, /* R2590 - DSP1 Control 25 */ - { 0x0A20, 0x0000 }, /* R2592 - DSP1 Control 26 */ - { 0x0A21, 0x0000 }, /* R2593 - DSP1 Control 27 */ - { 0x0A22, 0x0000 }, /* R2594 - DSP1 Control 28 */ - { 0x0A23, 0x0000 }, /* R2595 - DSP1 Control 29 */ - { 0x0A24, 0x0000 }, /* R2596 - DSP1 Control 30 */ - { 0x0A26, 0x0000 }, /* R2598 - DSP1 Control 31 */ - { 0x0B00, 0x0000 }, /* R2816 - DSP2 Control 1 */ - { 0x0B02, 0x0000 }, /* R2818 - DSP2 Control 2 */ - { 0x0B03, 0x0000 }, /* R2819 - DSP2 Control 3 */ - { 0x0B04, 0x0000 }, /* R2820 - DSP2 Control 4 */ - { 0x0B06, 0x0000 }, /* R2822 - DSP2 Control 5 */ - { 0x0B07, 0x0000 }, /* R2823 - DSP2 Control 6 */ - { 0x0B08, 0x0000 }, /* R2824 - DSP2 Control 7 */ - { 0x0B09, 0x0000 }, /* R2825 - DSP2 Control 8 */ - { 0x0B0A, 0x0000 }, /* R2826 - DSP2 Control 9 */ - { 0x0B0B, 0x0000 }, /* R2827 - DSP2 Control 10 */ - { 0x0B0C, 0x0000 }, /* R2828 - DSP2 Control 11 */ - { 0x0B0D, 0x0000 }, /* R2829 - DSP2 Control 12 */ - { 0x0B0F, 0x0000 }, /* R2831 - DSP2 Control 13 */ - { 0x0B10, 0x0000 }, /* R2832 - DSP2 Control 14 */ - { 0x0B11, 0x0000 }, /* R2833 - DSP2 Control 15 */ - { 0x0B12, 0x0000 }, /* R2834 - DSP2 Control 16 */ - { 0x0B13, 0x0000 }, /* R2835 - DSP2 Control 17 */ - { 0x0B14, 0x0000 }, /* R2836 - DSP2 Control 18 */ - { 0x0B16, 0x0000 }, /* R2838 - DSP2 Control 19 */ - { 0x0B17, 0x0000 }, /* R2839 - DSP2 Control 20 */ - { 0x0B18, 0x0000 }, /* R2840 - DSP2 Control 21 */ - { 0x0B1A, 0x0800 }, /* R2842 - DSP2 Control 22 */ - { 0x0B1B, 0x1000 }, /* R2843 - DSP2 Control 23 */ - { 0x0B1C, 0x0400 }, /* R2844 - DSP2 Control 24 */ - { 0x0B1E, 0x0000 }, /* R2846 - DSP2 Control 25 */ - { 0x0B20, 0x0000 }, /* R2848 - DSP2 Control 26 */ - { 0x0B21, 0x0000 }, /* R2849 - DSP2 Control 27 */ - { 0x0B22, 0x0000 }, /* R2850 - DSP2 Control 28 */ - { 0x0B23, 0x0000 }, /* R2851 - DSP2 Control 29 */ - { 0x0B24, 0x0000 }, /* R2852 - DSP2 Control 30 */ - { 0x0B26, 0x0000 }, /* R2854 - DSP2 Control 31 */ + { 0x000B, 0x0000 }, /* R11 - Tone Generator 1 */ + { 0x0102, 0x0000 }, /* R258 - Clocking 3 */ + { 0x0103, 0x0011 }, /* R259 - Clocking 4 */ + { 0x0111, 0x0000 }, /* R273 - FLL Control 1 */ + { 0x0112, 0x0000 }, /* R274 - FLL Control 2 */ + { 0x0113, 0x0000 }, /* R275 - FLL Control 3 */ + { 0x0114, 0x0000 }, /* R276 - FLL Control 4 */ + { 0x0116, 0x0177 }, /* R278 - FLL Control 6 */ + { 0x0117, 0x0004 }, /* R279 - FLL Control 7 */ + { 0x0119, 0x0000 }, /* R281 - FLL EFS 1 */ + { 0x011A, 0x0002 }, /* R282 - FLL EFS 2 */ + { 0x0200, 0x0000 }, /* R512 - Mic Charge Pump 1 */ + { 0x0201, 0x03FF }, /* R513 - Mic Charge Pump 2 */ + { 0x0202, 0x9BDE }, /* R514 - DM Charge Pump 1 */ + { 0x020C, 0x0000 }, /* R524 - Mic Bias Ctrl 1 */ + { 0x020D, 0x0000 }, /* R525 - Mic Bias Ctrl 2 */ + { 0x020F, 0x0000 }, /* R527 - Ear Piece Ctrl 1 */ + { 0x0210, 0x0000 }, /* R528 - Ear Piece Ctrl 2 */ + { 0x0301, 0x0000 }, /* R769 - Input Enables */ + { 0x0302, 0x2240 }, /* R770 - IN1L Control */ + { 0x0303, 0x0040 }, /* R771 - IN1R Control */ + { 0x0304, 0x2240 }, /* R772 - IN2L Control */ + { 0x0305, 0x0040 }, /* R773 - IN2R Control */ + { 0x0306, 0x2240 }, /* R774 - IN3L Control */ + { 0x0307, 0x0040 }, /* R775 - IN3R Control */ + { 0x030A, 0x0000 }, /* R778 - RXANC_SRC */ + { 0x030B, 0x0022 }, /* R779 - Input Volume Ramp */ + { 0x030C, 0x0180 }, /* R780 - ADC Digital Volume 1L */ + { 0x030D, 0x0180 }, /* R781 - ADC Digital Volume 1R */ + { 0x030E, 0x0180 }, /* R782 - ADC Digital Volume 2L */ + { 0x030F, 0x0180 }, /* R783 - ADC Digital Volume 2R */ + { 0x0310, 0x0180 }, /* R784 - ADC Digital Volume 3L */ + { 0x0311, 0x0180 }, /* R785 - ADC Digital Volume 3R */ + { 0x0400, 0x0000 }, /* R1024 - Output Enables */ + { 0x0401, 0x0000 }, /* R1025 - DAC Volume Limit 1L */ + { 0x0402, 0x0000 }, /* R1026 - DAC Volume Limit 1R */ + { 0x0403, 0x0000 }, /* R1027 - DAC Volume Limit 2L */ + { 0x0404, 0x0000 }, /* R1028 - DAC Volume Limit 2R */ + { 0x0409, 0x0000 }, /* R1033 - DAC AEC Control 1 */ + { 0x040A, 0x0022 }, /* R1034 - Output Volume Ramp */ + { 0x040B, 0x0180 }, /* R1035 - DAC Digital Volume 1L */ + { 0x040C, 0x0180 }, /* R1036 - DAC Digital Volume 1R */ + { 0x040D, 0x0180 }, /* R1037 - DAC Digital Volume 2L */ + { 0x040E, 0x0180 }, /* R1038 - DAC Digital Volume 2R */ + { 0x0417, 0x0069 }, /* R1047 - PDM 1 */ + { 0x0418, 0x0000 }, /* R1048 - PDM 2 */ + { 0x0500, 0x0000 }, /* R1280 - Audio IF 1_1 */ + { 0x0501, 0x0008 }, /* R1281 - Audio IF 1_2 */ + { 0x0502, 0x0000 }, /* R1282 - Audio IF 1_3 */ + { 0x0503, 0x0000 }, /* R1283 - Audio IF 1_4 */ + { 0x0504, 0x0000 }, /* R1284 - Audio IF 1_5 */ + { 0x0505, 0x0001 }, /* R1285 - Audio IF 1_6 */ + { 0x0506, 0x0001 }, /* R1286 - Audio IF 1_7 */ + { 0x0507, 0x0000 }, /* R1287 - Audio IF 1_8 */ + { 0x0508, 0x0000 }, /* R1288 - Audio IF 1_9 */ + { 0x0509, 0x0000 }, /* R1289 - Audio IF 1_10 */ + { 0x050A, 0x0000 }, /* R1290 - Audio IF 1_11 */ + { 0x050B, 0x0000 }, /* R1291 - Audio IF 1_12 */ + { 0x050C, 0x0000 }, /* R1292 - Audio IF 1_13 */ + { 0x050D, 0x0000 }, /* R1293 - Audio IF 1_14 */ + { 0x050E, 0x0000 }, /* R1294 - Audio IF 1_15 */ + { 0x050F, 0x0000 }, /* R1295 - Audio IF 1_16 */ + { 0x0510, 0x0000 }, /* R1296 - Audio IF 1_17 */ + { 0x0511, 0x0000 }, /* R1297 - Audio IF 1_18 */ + { 0x0512, 0x0000 }, /* R1298 - Audio IF 1_19 */ + { 0x0513, 0x0000 }, /* R1299 - Audio IF 1_20 */ + { 0x0514, 0x0000 }, /* R1300 - Audio IF 1_21 */ + { 0x0515, 0x0001 }, /* R1301 - Audio IF 1_22 */ + { 0x0600, 0x0000 }, /* R1536 - OUT1LMIX Input 1 Source */ + { 0x0601, 0x0080 }, /* R1537 - OUT1LMIX Input 1 Volume */ + { 0x0602, 0x0000 }, /* R1538 - OUT1LMIX Input 2 Source */ + { 0x0603, 0x0080 }, /* R1539 - OUT1LMIX Input 2 Volume */ + { 0x0604, 0x0000 }, /* R1540 - OUT1LMIX Input 3 Source */ + { 0x0605, 0x0080 }, /* R1541 - OUT1LMIX Input 3 Volume */ + { 0x0606, 0x0000 }, /* R1542 - OUT1LMIX Input 4 Source */ + { 0x0607, 0x0080 }, /* R1543 - OUT1LMIX Input 4 Volume */ + { 0x0608, 0x0000 }, /* R1544 - OUT1RMIX Input 1 Source */ + { 0x0609, 0x0080 }, /* R1545 - OUT1RMIX Input 1 Volume */ + { 0x060A, 0x0000 }, /* R1546 - OUT1RMIX Input 2 Source */ + { 0x060B, 0x0080 }, /* R1547 - OUT1RMIX Input 2 Volume */ + { 0x060C, 0x0000 }, /* R1548 - OUT1RMIX Input 3 Source */ + { 0x060D, 0x0080 }, /* R1549 - OUT1RMIX Input 3 Volume */ + { 0x060E, 0x0000 }, /* R1550 - OUT1RMIX Input 4 Source */ + { 0x060F, 0x0080 }, /* R1551 - OUT1RMIX Input 4 Volume */ + { 0x0610, 0x0000 }, /* R1552 - OUT2LMIX Input 1 Source */ + { 0x0611, 0x0080 }, /* R1553 - OUT2LMIX Input 1 Volume */ + { 0x0612, 0x0000 }, /* R1554 - OUT2LMIX Input 2 Source */ + { 0x0613, 0x0080 }, /* R1555 - OUT2LMIX Input 2 Volume */ + { 0x0614, 0x0000 }, /* R1556 - OUT2LMIX Input 3 Source */ + { 0x0615, 0x0080 }, /* R1557 - OUT2LMIX Input 3 Volume */ + { 0x0616, 0x0000 }, /* R1558 - OUT2LMIX Input 4 Source */ + { 0x0617, 0x0080 }, /* R1559 - OUT2LMIX Input 4 Volume */ + { 0x0618, 0x0000 }, /* R1560 - OUT2RMIX Input 1 Source */ + { 0x0619, 0x0080 }, /* R1561 - OUT2RMIX Input 1 Volume */ + { 0x061A, 0x0000 }, /* R1562 - OUT2RMIX Input 2 Source */ + { 0x061B, 0x0080 }, /* R1563 - OUT2RMIX Input 2 Volume */ + { 0x061C, 0x0000 }, /* R1564 - OUT2RMIX Input 3 Source */ + { 0x061D, 0x0080 }, /* R1565 - OUT2RMIX Input 3 Volume */ + { 0x061E, 0x0000 }, /* R1566 - OUT2RMIX Input 4 Source */ + { 0x061F, 0x0080 }, /* R1567 - OUT2RMIX Input 4 Volume */ + { 0x0620, 0x0000 }, /* R1568 - AIF1TX1MIX Input 1 Source */ + { 0x0621, 0x0080 }, /* R1569 - AIF1TX1MIX Input 1 Volume */ + { 0x0622, 0x0000 }, /* R1570 - AIF1TX1MIX Input 2 Source */ + { 0x0623, 0x0080 }, /* R1571 - AIF1TX1MIX Input 2 Volume */ + { 0x0624, 0x0000 }, /* R1572 - AIF1TX1MIX Input 3 Source */ + { 0x0625, 0x0080 }, /* R1573 - AIF1TX1MIX Input 3 Volume */ + { 0x0626, 0x0000 }, /* R1574 - AIF1TX1MIX Input 4 Source */ + { 0x0627, 0x0080 }, /* R1575 - AIF1TX1MIX Input 4 Volume */ + { 0x0628, 0x0000 }, /* R1576 - AIF1TX2MIX Input 1 Source */ + { 0x0629, 0x0080 }, /* R1577 - AIF1TX2MIX Input 1 Volume */ + { 0x062A, 0x0000 }, /* R1578 - AIF1TX2MIX Input 2 Source */ + { 0x062B, 0x0080 }, /* R1579 - AIF1TX2MIX Input 2 Volume */ + { 0x062C, 0x0000 }, /* R1580 - AIF1TX2MIX Input 3 Source */ + { 0x062D, 0x0080 }, /* R1581 - AIF1TX2MIX Input 3 Volume */ + { 0x062E, 0x0000 }, /* R1582 - AIF1TX2MIX Input 4 Source */ + { 0x062F, 0x0080 }, /* R1583 - AIF1TX2MIX Input 4 Volume */ + { 0x0630, 0x0000 }, /* R1584 - AIF1TX3MIX Input 1 Source */ + { 0x0631, 0x0080 }, /* R1585 - AIF1TX3MIX Input 1 Volume */ + { 0x0632, 0x0000 }, /* R1586 - AIF1TX3MIX Input 2 Source */ + { 0x0633, 0x0080 }, /* R1587 - AIF1TX3MIX Input 2 Volume */ + { 0x0634, 0x0000 }, /* R1588 - AIF1TX3MIX Input 3 Source */ + { 0x0635, 0x0080 }, /* R1589 - AIF1TX3MIX Input 3 Volume */ + { 0x0636, 0x0000 }, /* R1590 - AIF1TX3MIX Input 4 Source */ + { 0x0637, 0x0080 }, /* R1591 - AIF1TX3MIX Input 4 Volume */ + { 0x0638, 0x0000 }, /* R1592 - AIF1TX4MIX Input 1 Source */ + { 0x0639, 0x0080 }, /* R1593 - AIF1TX4MIX Input 1 Volume */ + { 0x063A, 0x0000 }, /* R1594 - AIF1TX4MIX Input 2 Source */ + { 0x063B, 0x0080 }, /* R1595 - AIF1TX4MIX Input 2 Volume */ + { 0x063C, 0x0000 }, /* R1596 - AIF1TX4MIX Input 3 Source */ + { 0x063D, 0x0080 }, /* R1597 - AIF1TX4MIX Input 3 Volume */ + { 0x063E, 0x0000 }, /* R1598 - AIF1TX4MIX Input 4 Source */ + { 0x063F, 0x0080 }, /* R1599 - AIF1TX4MIX Input 4 Volume */ + { 0x0640, 0x0000 }, /* R1600 - AIF1TX5MIX Input 1 Source */ + { 0x0641, 0x0080 }, /* R1601 - AIF1TX5MIX Input 1 Volume */ + { 0x0642, 0x0000 }, /* R1602 - AIF1TX5MIX Input 2 Source */ + { 0x0643, 0x0080 }, /* R1603 - AIF1TX5MIX Input 2 Volume */ + { 0x0644, 0x0000 }, /* R1604 - AIF1TX5MIX Input 3 Source */ + { 0x0645, 0x0080 }, /* R1605 - AIF1TX5MIX Input 3 Volume */ + { 0x0646, 0x0000 }, /* R1606 - AIF1TX5MIX Input 4 Source */ + { 0x0647, 0x0080 }, /* R1607 - AIF1TX5MIX Input 4 Volume */ + { 0x0648, 0x0000 }, /* R1608 - AIF1TX6MIX Input 1 Source */ + { 0x0649, 0x0080 }, /* R1609 - AIF1TX6MIX Input 1 Volume */ + { 0x064A, 0x0000 }, /* R1610 - AIF1TX6MIX Input 2 Source */ + { 0x064B, 0x0080 }, /* R1611 - AIF1TX6MIX Input 2 Volume */ + { 0x064C, 0x0000 }, /* R1612 - AIF1TX6MIX Input 3 Source */ + { 0x064D, 0x0080 }, /* R1613 - AIF1TX6MIX Input 3 Volume */ + { 0x064E, 0x0000 }, /* R1614 - AIF1TX6MIX Input 4 Source */ + { 0x064F, 0x0080 }, /* R1615 - AIF1TX6MIX Input 4 Volume */ + { 0x0650, 0x0000 }, /* R1616 - EQLMIX Input 1 Source */ + { 0x0651, 0x0080 }, /* R1617 - EQLMIX Input 1 Volume */ + { 0x0652, 0x0000 }, /* R1618 - EQLMIX Input 2 Source */ + { 0x0653, 0x0080 }, /* R1619 - EQLMIX Input 2 Volume */ + { 0x0654, 0x0000 }, /* R1620 - EQLMIX Input 3 Source */ + { 0x0655, 0x0080 }, /* R1621 - EQLMIX Input 3 Volume */ + { 0x0656, 0x0000 }, /* R1622 - EQLMIX Input 4 Source */ + { 0x0657, 0x0080 }, /* R1623 - EQLMIX Input 4 Volume */ + { 0x0658, 0x0000 }, /* R1624 - EQRMIX Input 1 Source */ + { 0x0659, 0x0080 }, /* R1625 - EQRMIX Input 1 Volume */ + { 0x065A, 0x0000 }, /* R1626 - EQRMIX Input 2 Source */ + { 0x065B, 0x0080 }, /* R1627 - EQRMIX Input 2 Volume */ + { 0x065C, 0x0000 }, /* R1628 - EQRMIX Input 3 Source */ + { 0x065D, 0x0080 }, /* R1629 - EQRMIX Input 3 Volume */ + { 0x065E, 0x0000 }, /* R1630 - EQRMIX Input 4 Source */ + { 0x065F, 0x0080 }, /* R1631 - EQRMIX Input 4 Volume */ + { 0x0660, 0x0000 }, /* R1632 - LHPF1MIX Input 1 Source */ + { 0x0661, 0x0080 }, /* R1633 - LHPF1MIX Input 1 Volume */ + { 0x0662, 0x0000 }, /* R1634 - LHPF1MIX Input 2 Source */ + { 0x0663, 0x0080 }, /* R1635 - LHPF1MIX Input 2 Volume */ + { 0x0664, 0x0000 }, /* R1636 - LHPF1MIX Input 3 Source */ + { 0x0665, 0x0080 }, /* R1637 - LHPF1MIX Input 3 Volume */ + { 0x0666, 0x0000 }, /* R1638 - LHPF1MIX Input 4 Source */ + { 0x0667, 0x0080 }, /* R1639 - LHPF1MIX Input 4 Volume */ + { 0x0668, 0x0000 }, /* R1640 - LHPF2MIX Input 1 Source */ + { 0x0669, 0x0080 }, /* R1641 - LHPF2MIX Input 1 Volume */ + { 0x066A, 0x0000 }, /* R1642 - LHPF2MIX Input 2 Source */ + { 0x066B, 0x0080 }, /* R1643 - LHPF2MIX Input 2 Volume */ + { 0x066C, 0x0000 }, /* R1644 - LHPF2MIX Input 3 Source */ + { 0x066D, 0x0080 }, /* R1645 - LHPF2MIX Input 3 Volume */ + { 0x066E, 0x0000 }, /* R1646 - LHPF2MIX Input 4 Source */ + { 0x066F, 0x0080 }, /* R1647 - LHPF2MIX Input 4 Volume */ + { 0x0670, 0x0000 }, /* R1648 - DSP1LMIX Input 1 Source */ + { 0x0671, 0x0080 }, /* R1649 - DSP1LMIX Input 1 Volume */ + { 0x0672, 0x0000 }, /* R1650 - DSP1LMIX Input 2 Source */ + { 0x0673, 0x0080 }, /* R1651 - DSP1LMIX Input 2 Volume */ + { 0x0674, 0x0000 }, /* R1652 - DSP1LMIX Input 3 Source */ + { 0x0675, 0x0080 }, /* R1653 - DSP1LMIX Input 3 Volume */ + { 0x0676, 0x0000 }, /* R1654 - DSP1LMIX Input 4 Source */ + { 0x0677, 0x0080 }, /* R1655 - DSP1LMIX Input 4 Volume */ + { 0x0678, 0x0000 }, /* R1656 - DSP1RMIX Input 1 Source */ + { 0x0679, 0x0080 }, /* R1657 - DSP1RMIX Input 1 Volume */ + { 0x067A, 0x0000 }, /* R1658 - DSP1RMIX Input 2 Source */ + { 0x067B, 0x0080 }, /* R1659 - DSP1RMIX Input 2 Volume */ + { 0x067C, 0x0000 }, /* R1660 - DSP1RMIX Input 3 Source */ + { 0x067D, 0x0080 }, /* R1661 - DSP1RMIX Input 3 Volume */ + { 0x067E, 0x0000 }, /* R1662 - DSP1RMIX Input 4 Source */ + { 0x067F, 0x0080 }, /* R1663 - DSP1RMIX Input 4 Volume */ + { 0x0680, 0x0000 }, /* R1664 - DSP1AUX1MIX Input 1 Source */ + { 0x0681, 0x0000 }, /* R1665 - DSP1AUX2MIX Input 1 Source */ + { 0x0682, 0x0000 }, /* R1666 - DSP1AUX3MIX Input 1 Source */ + { 0x0683, 0x0000 }, /* R1667 - DSP1AUX4MIX Input 1 Source */ + { 0x0684, 0x0000 }, /* R1668 - DSP1AUX5MIX Input 1 Source */ + { 0x0685, 0x0000 }, /* R1669 - DSP1AUX6MIX Input 1 Source */ + { 0x0686, 0x0000 }, /* R1670 - DSP2LMIX Input 1 Source */ + { 0x0687, 0x0080 }, /* R1671 - DSP2LMIX Input 1 Volume */ + { 0x0688, 0x0000 }, /* R1672 - DSP2LMIX Input 2 Source */ + { 0x0689, 0x0080 }, /* R1673 - DSP2LMIX Input 2 Volume */ + { 0x068A, 0x0000 }, /* R1674 - DSP2LMIX Input 3 Source */ + { 0x068B, 0x0080 }, /* R1675 - DSP2LMIX Input 3 Volume */ + { 0x068C, 0x0000 }, /* R1676 - DSP2LMIX Input 4 Source */ + { 0x068D, 0x0080 }, /* R1677 - DSP2LMIX Input 4 Volume */ + { 0x068E, 0x0000 }, /* R1678 - DSP2RMIX Input 1 Source */ + { 0x068F, 0x0080 }, /* R1679 - DSP2RMIX Input 1 Volume */ + { 0x0690, 0x0000 }, /* R1680 - DSP2RMIX Input 2 Source */ + { 0x0691, 0x0080 }, /* R1681 - DSP2RMIX Input 2 Volume */ + { 0x0692, 0x0000 }, /* R1682 - DSP2RMIX Input 3 Source */ + { 0x0693, 0x0080 }, /* R1683 - DSP2RMIX Input 3 Volume */ + { 0x0694, 0x0000 }, /* R1684 - DSP2RMIX Input 4 Source */ + { 0x0695, 0x0080 }, /* R1685 - DSP2RMIX Input 4 Volume */ + { 0x0696, 0x0000 }, /* R1686 - DSP2AUX1MIX Input 1 Source */ + { 0x0697, 0x0000 }, /* R1687 - DSP2AUX2MIX Input 1 Source */ + { 0x0698, 0x0000 }, /* R1688 - DSP2AUX3MIX Input 1 Source */ + { 0x0699, 0x0000 }, /* R1689 - DSP2AUX4MIX Input 1 Source */ + { 0x069A, 0x0000 }, /* R1690 - DSP2AUX5MIX Input 1 Source */ + { 0x069B, 0x0000 }, /* R1691 - DSP2AUX6MIX Input 1 Source */ + { 0x0700, 0xA101 }, /* R1792 - GPIO CTRL 1 */ + { 0x0701, 0xA101 }, /* R1793 - GPIO CTRL 2 */ + { 0x0702, 0xA101 }, /* R1794 - GPIO CTRL 3 */ + { 0x0703, 0xA101 }, /* R1795 - GPIO CTRL 4 */ + { 0x0709, 0x0000 }, /* R1801 - Misc Pad Ctrl 1 */ + { 0x0801, 0x00FF }, /* R2049 - Interrupt Status 1 Mask */ + { 0x0804, 0xFFFF }, /* R2052 - Interrupt Status 2 Mask */ + { 0x0808, 0x0000 }, /* R2056 - Interrupt Control */ + { 0x0900, 0x0000 }, /* R2304 - EQL_1 */ + { 0x0901, 0x0000 }, /* R2305 - EQL_2 */ + { 0x0902, 0x0000 }, /* R2306 - EQL_3 */ + { 0x0903, 0x0000 }, /* R2307 - EQL_4 */ + { 0x0904, 0x0000 }, /* R2308 - EQL_5 */ + { 0x0905, 0x0000 }, /* R2309 - EQL_6 */ + { 0x0906, 0x0000 }, /* R2310 - EQL_7 */ + { 0x0907, 0x0000 }, /* R2311 - EQL_8 */ + { 0x0908, 0x0000 }, /* R2312 - EQL_9 */ + { 0x0909, 0x0000 }, /* R2313 - EQL_10 */ + { 0x090A, 0x0000 }, /* R2314 - EQL_11 */ + { 0x090B, 0x0000 }, /* R2315 - EQL_12 */ + { 0x090C, 0x0000 }, /* R2316 - EQL_13 */ + { 0x090D, 0x0000 }, /* R2317 - EQL_14 */ + { 0x090E, 0x0000 }, /* R2318 - EQL_15 */ + { 0x090F, 0x0000 }, /* R2319 - EQL_16 */ + { 0x0910, 0x0000 }, /* R2320 - EQL_17 */ + { 0x0911, 0x0000 }, /* R2321 - EQL_18 */ + { 0x0912, 0x0000 }, /* R2322 - EQL_19 */ + { 0x0913, 0x0000 }, /* R2323 - EQL_20 */ + { 0x0916, 0x0000 }, /* R2326 - EQR_1 */ + { 0x0917, 0x0000 }, /* R2327 - EQR_2 */ + { 0x0918, 0x0000 }, /* R2328 - EQR_3 */ + { 0x0919, 0x0000 }, /* R2329 - EQR_4 */ + { 0x091A, 0x0000 }, /* R2330 - EQR_5 */ + { 0x091B, 0x0000 }, /* R2331 - EQR_6 */ + { 0x091C, 0x0000 }, /* R2332 - EQR_7 */ + { 0x091D, 0x0000 }, /* R2333 - EQR_8 */ + { 0x091E, 0x0000 }, /* R2334 - EQR_9 */ + { 0x091F, 0x0000 }, /* R2335 - EQR_10 */ + { 0x0920, 0x0000 }, /* R2336 - EQR_11 */ + { 0x0921, 0x0000 }, /* R2337 - EQR_12 */ + { 0x0922, 0x0000 }, /* R2338 - EQR_13 */ + { 0x0923, 0x0000 }, /* R2339 - EQR_14 */ + { 0x0924, 0x0000 }, /* R2340 - EQR_15 */ + { 0x0925, 0x0000 }, /* R2341 - EQR_16 */ + { 0x0926, 0x0000 }, /* R2342 - EQR_17 */ + { 0x0927, 0x0000 }, /* R2343 - EQR_18 */ + { 0x0928, 0x0000 }, /* R2344 - EQR_19 */ + { 0x0929, 0x0000 }, /* R2345 - EQR_20 */ + { 0x093E, 0x0000 }, /* R2366 - HPLPF1_1 */ + { 0x093F, 0x0000 }, /* R2367 - HPLPF1_2 */ + { 0x0942, 0x0000 }, /* R2370 - HPLPF2_1 */ + { 0x0943, 0x0000 }, /* R2371 - HPLPF2_2 */ + { 0x0A00, 0x0000 }, /* R2560 - DSP1 Control 1 */ + { 0x0A02, 0x0000 }, /* R2562 - DSP1 Control 2 */ + { 0x0A03, 0x0000 }, /* R2563 - DSP1 Control 3 */ + { 0x0A04, 0x0000 }, /* R2564 - DSP1 Control 4 */ + { 0x0A06, 0x0000 }, /* R2566 - DSP1 Control 5 */ + { 0x0A07, 0x0000 }, /* R2567 - DSP1 Control 6 */ + { 0x0A08, 0x0000 }, /* R2568 - DSP1 Control 7 */ + { 0x0A09, 0x0000 }, /* R2569 - DSP1 Control 8 */ + { 0x0A0A, 0x0000 }, /* R2570 - DSP1 Control 9 */ + { 0x0A0B, 0x0000 }, /* R2571 - DSP1 Control 10 */ + { 0x0A0C, 0x0000 }, /* R2572 - DSP1 Control 11 */ + { 0x0A0D, 0x0000 }, /* R2573 - DSP1 Control 12 */ + { 0x0A0F, 0x0000 }, /* R2575 - DSP1 Control 13 */ + { 0x0A10, 0x0000 }, /* R2576 - DSP1 Control 14 */ + { 0x0A11, 0x0000 }, /* R2577 - DSP1 Control 15 */ + { 0x0A12, 0x0000 }, /* R2578 - DSP1 Control 16 */ + { 0x0A13, 0x0000 }, /* R2579 - DSP1 Control 17 */ + { 0x0A14, 0x0000 }, /* R2580 - DSP1 Control 18 */ + { 0x0A16, 0x0000 }, /* R2582 - DSP1 Control 19 */ + { 0x0A17, 0x0000 }, /* R2583 - DSP1 Control 20 */ + { 0x0A18, 0x0000 }, /* R2584 - DSP1 Control 21 */ + { 0x0A1A, 0x1800 }, /* R2586 - DSP1 Control 22 */ + { 0x0A1B, 0x1000 }, /* R2587 - DSP1 Control 23 */ + { 0x0A1C, 0x0400 }, /* R2588 - DSP1 Control 24 */ + { 0x0A1E, 0x0000 }, /* R2590 - DSP1 Control 25 */ + { 0x0A20, 0x0000 }, /* R2592 - DSP1 Control 26 */ + { 0x0A21, 0x0000 }, /* R2593 - DSP1 Control 27 */ + { 0x0A22, 0x0000 }, /* R2594 - DSP1 Control 28 */ + { 0x0A23, 0x0000 }, /* R2595 - DSP1 Control 29 */ + { 0x0A24, 0x0000 }, /* R2596 - DSP1 Control 30 */ + { 0x0A26, 0x0000 }, /* R2598 - DSP1 Control 31 */ + { 0x0B00, 0x0000 }, /* R2816 - DSP2 Control 1 */ + { 0x0B02, 0x0000 }, /* R2818 - DSP2 Control 2 */ + { 0x0B03, 0x0000 }, /* R2819 - DSP2 Control 3 */ + { 0x0B04, 0x0000 }, /* R2820 - DSP2 Control 4 */ + { 0x0B06, 0x0000 }, /* R2822 - DSP2 Control 5 */ + { 0x0B07, 0x0000 }, /* R2823 - DSP2 Control 6 */ + { 0x0B08, 0x0000 }, /* R2824 - DSP2 Control 7 */ + { 0x0B09, 0x0000 }, /* R2825 - DSP2 Control 8 */ + { 0x0B0A, 0x0000 }, /* R2826 - DSP2 Control 9 */ + { 0x0B0B, 0x0000 }, /* R2827 - DSP2 Control 10 */ + { 0x0B0C, 0x0000 }, /* R2828 - DSP2 Control 11 */ + { 0x0B0D, 0x0000 }, /* R2829 - DSP2 Control 12 */ + { 0x0B0F, 0x0000 }, /* R2831 - DSP2 Control 13 */ + { 0x0B10, 0x0000 }, /* R2832 - DSP2 Control 14 */ + { 0x0B11, 0x0000 }, /* R2833 - DSP2 Control 15 */ + { 0x0B12, 0x0000 }, /* R2834 - DSP2 Control 16 */ + { 0x0B13, 0x0000 }, /* R2835 - DSP2 Control 17 */ + { 0x0B14, 0x0000 }, /* R2836 - DSP2 Control 18 */ + { 0x0B16, 0x0000 }, /* R2838 - DSP2 Control 19 */ + { 0x0B17, 0x0000 }, /* R2839 - DSP2 Control 20 */ + { 0x0B18, 0x0000 }, /* R2840 - DSP2 Control 21 */ + { 0x0B1A, 0x0800 }, /* R2842 - DSP2 Control 22 */ + { 0x0B1B, 0x1000 }, /* R2843 - DSP2 Control 23 */ + { 0x0B1C, 0x0400 }, /* R2844 - DSP2 Control 24 */ + { 0x0B1E, 0x0000 }, /* R2846 - DSP2 Control 25 */ + { 0x0B20, 0x0000 }, /* R2848 - DSP2 Control 26 */ + { 0x0B21, 0x0000 }, /* R2849 - DSP2 Control 27 */ + { 0x0B22, 0x0000 }, /* R2850 - DSP2 Control 28 */ + { 0x0B23, 0x0000 }, /* R2851 - DSP2 Control 29 */ + { 0x0B24, 0x0000 }, /* R2852 - DSP2 Control 30 */ + { 0x0B26, 0x0000 }, /* R2854 - DSP2 Control 31 */ }; static bool wm2200_volatile_register(struct device *dev, unsigned int reg) @@ -974,7 +974,7 @@ static int wm2200_mixer_values[] = { static WM2200_MUX_CTL_DECL(name##_in1); \ static WM2200_MUX_CTL_DECL(name##_in2); \ static WM2200_MUX_CTL_DECL(name##_in3); \ - static WM2200_MUX_CTL_DECL(name##_in4) + static WM2200_MUX_CTL_DECL(name##_in4) static const struct snd_kcontrol_new wm2200_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL, -- cgit v1.2.3-18-g5258 From 394d2bbae3dbde0972b90415fc4b5628bbfd700f Mon Sep 17 00:00:00 2001 From: Wolfram Sang Date: Fri, 27 Jan 2012 16:10:23 +0100 Subject: ASoC: max9768: add driver for max9768 amplifier Add a driver supporting the volume control and the mute pin. Shdn pin and DAPM are not taken care of yet. Signed-off-by: Wolfram Sang Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9768.c | 253 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 259 insertions(+) create mode 100644 sound/soc/codecs/max9768.c (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 39aec3c12cf..6508e8b790b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C + select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C @@ -429,6 +430,9 @@ config SND_SOC_WM9713 config SND_SOC_LM4857 tristate +config SND_SOC_MAX9768 + tristate + config SND_SOC_MAX9877 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 716c5842bfd..6662eb0cdcc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -25,6 +25,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o +snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o @@ -130,6 +131,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c new file mode 100644 index 00000000000..63489952584 --- /dev/null +++ b/sound/soc/codecs/max9768.c @@ -0,0 +1,253 @@ +/* + * MAX9768 AMP driver + * + * Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 of the License. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include + +/* "Registers" */ +#define MAX9768_VOL 0 +#define MAX9768_CTRL 3 + +/* Commands */ +#define MAX9768_CTRL_PWM 0x15 +#define MAX9768_CTRL_FILTERLESS 0x16 + +struct max9768 { + struct regmap *regmap; + int mute_gpio; + int shdn_gpio; + u32 flags; +}; + +static struct reg_default max9768_default_regs[] = { + { 0, 0 }, + { 3, MAX9768_CTRL_FILTERLESS}, +}; + +static int max9768_get_gpio(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); + int val = gpio_get_value_cansleep(max9768->mute_gpio); + + ucontrol->value.integer.value[0] = !val; + + return 0; +} + +static int max9768_set_gpio(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); + + gpio_set_value_cansleep(max9768->mute_gpio, !ucontrol->value.integer.value[0]); + + return 0; +} + +static const unsigned int volume_tlv[] = { + TLV_DB_RANGE_HEAD(43), + 0, 0, TLV_DB_SCALE_ITEM(-16150, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-9280, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-9030, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(-8680, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(-8430, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(-8080, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(-7830, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(-7470, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(-7220, 0, 0), + 9, 9, TLV_DB_SCALE_ITEM(-6870, 0, 0), + 10, 10, TLV_DB_SCALE_ITEM(-6620, 0, 0), + 11, 11, TLV_DB_SCALE_ITEM(-6270, 0, 0), + 12, 12, TLV_DB_SCALE_ITEM(-6020, 0, 0), + 13, 13, TLV_DB_SCALE_ITEM(-5670, 0, 0), + 14, 14, TLV_DB_SCALE_ITEM(-5420, 0, 0), + 15, 17, TLV_DB_SCALE_ITEM(-5060, 250, 0), + 18, 18, TLV_DB_SCALE_ITEM(-4370, 0, 0), + 19, 19, TLV_DB_SCALE_ITEM(-4210, 0, 0), + 20, 20, TLV_DB_SCALE_ITEM(-3960, 0, 0), + 21, 21, TLV_DB_SCALE_ITEM(-3760, 0, 0), + 22, 22, TLV_DB_SCALE_ITEM(-3600, 0, 0), + 23, 23, TLV_DB_SCALE_ITEM(-3340, 0, 0), + 24, 24, TLV_DB_SCALE_ITEM(-3150, 0, 0), + 25, 25, TLV_DB_SCALE_ITEM(-2980, 0, 0), + 26, 26, TLV_DB_SCALE_ITEM(-2720, 0, 0), + 27, 27, TLV_DB_SCALE_ITEM(-2520, 0, 0), + 28, 30, TLV_DB_SCALE_ITEM(-2350, 190, 0), + 31, 31, TLV_DB_SCALE_ITEM(-1750, 0, 0), + 32, 34, TLV_DB_SCALE_ITEM(-1640, 100, 0), + 35, 37, TLV_DB_SCALE_ITEM(-1310, 110, 0), + 38, 39, TLV_DB_SCALE_ITEM(-990, 100, 0), + 40, 40, TLV_DB_SCALE_ITEM(-710, 0, 0), + 41, 41, TLV_DB_SCALE_ITEM(-600, 0, 0), + 42, 42, TLV_DB_SCALE_ITEM(-500, 0, 0), + 43, 43, TLV_DB_SCALE_ITEM(-340, 0, 0), + 44, 44, TLV_DB_SCALE_ITEM(-190, 0, 0), + 45, 45, TLV_DB_SCALE_ITEM(-50, 0, 0), + 46, 46, TLV_DB_SCALE_ITEM(50, 0, 0), + 47, 50, TLV_DB_SCALE_ITEM(120, 40, 0), + 51, 57, TLV_DB_SCALE_ITEM(290, 50, 0), + 58, 58, TLV_DB_SCALE_ITEM(650, 0, 0), + 59, 62, TLV_DB_SCALE_ITEM(700, 60, 0), + 63, 63, TLV_DB_SCALE_ITEM(950, 0, 0), +}; + +static const struct snd_kcontrol_new max9768_volume[] = { + SOC_SINGLE_TLV("Playback Volume", MAX9768_VOL, 0, 63, 0, volume_tlv), +}; + +static const struct snd_kcontrol_new max9768_mute[] = { + SOC_SINGLE_BOOL_EXT("Mute Switch", 0, max9768_get_gpio, max9768_set_gpio), +}; + +static int max9768_probe(struct snd_soc_codec *codec) +{ + struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = max9768->regmap; + ret = snd_soc_codec_set_cache_io(codec, 2, 6, SND_SOC_REGMAP); + if (ret) + return ret; + + if (max9768->flags & MAX9768_FLAG_CLASSIC_PWM) { + ret = snd_soc_write(codec, MAX9768_CTRL, MAX9768_CTRL_PWM); + if (ret) + return ret; + } + + if (gpio_is_valid(max9768->mute_gpio)) { + ret = snd_soc_add_controls(codec, max9768_mute, + ARRAY_SIZE(max9768_mute)); + if (ret) + return ret; + } + + return 0; +} + +static struct snd_soc_codec_driver max9768_codec_driver = { + .probe = max9768_probe, + .controls = max9768_volume, + .num_controls = ARRAY_SIZE(max9768_volume), +}; + +static bool max9768_always_false(struct device *dev, unsigned int reg) +{ + return false; +} + +static const struct regmap_config max9768_i2c_regmap_config = { + .reg_bits = 2, + .val_bits = 6, + .max_register = 3, + .reg_defaults = max9768_default_regs, + .num_reg_defaults = ARRAY_SIZE(max9768_default_regs), + .volatile_reg = max9768_always_false, + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit max9768_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct max9768 *max9768; + struct max9768_pdata *pdata = client->dev.platform_data; + int err; + + max9768 = devm_kzalloc(&client->dev, sizeof(*max9768), GFP_KERNEL); + if (!max9768) + return -ENOMEM; + + if (pdata) { + /* Mute on powerup to avoid clicks */ + err = gpio_request_one(pdata->mute_gpio, GPIOF_INIT_HIGH, "MAX9768 Mute"); + max9768->mute_gpio = err ?: pdata->mute_gpio; + + /* Activate chip by releasing shutdown, enables I2C */ + err = gpio_request_one(pdata->shdn_gpio, GPIOF_INIT_HIGH, "MAX9768 Shutdown"); + max9768->shdn_gpio = err ?: pdata->shdn_gpio; + + max9768->flags = pdata->flags; + } else { + max9768->shdn_gpio = -EINVAL; + max9768->mute_gpio = -EINVAL; + } + + i2c_set_clientdata(client, max9768); + + max9768->regmap = regmap_init_i2c(client, &max9768_i2c_regmap_config); + if (IS_ERR(max9768->regmap)) { + err = PTR_ERR(max9768->regmap); + goto err_gpio_free; + } + + err = snd_soc_register_codec(&client->dev, &max9768_codec_driver, NULL, 0); + if (err) + goto err_regmap_free; + + return 0; + + err_regmap_free: + regmap_exit(max9768->regmap); + err_gpio_free: + if (gpio_is_valid(max9768->shdn_gpio)) + gpio_free(max9768->shdn_gpio); + if (gpio_is_valid(max9768->mute_gpio)) + gpio_free(max9768->mute_gpio); + + return err; +} + +static int __devexit max9768_i2c_remove(struct i2c_client *client) +{ + struct max9768 *max9768 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + regmap_exit(max9768->regmap); + + if (gpio_is_valid(max9768->shdn_gpio)) + gpio_free(max9768->shdn_gpio); + if (gpio_is_valid(max9768->mute_gpio)) + gpio_free(max9768->mute_gpio); + + return 0; +} + +static const struct i2c_device_id max9768_i2c_id[] = { + { "max9768", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9768_i2c_id); + +static struct i2c_driver max9768_i2c_driver = { + .driver = { + .name = "max9768", + .owner = THIS_MODULE, + }, + .probe = max9768_i2c_probe, + .remove = __devexit_p(max9768_i2c_remove), + .id_table = max9768_i2c_id, +}; +module_i2c_driver(max9768_i2c_driver); + +MODULE_AUTHOR("Wolfram Sang "); +MODULE_DESCRIPTION("ASoC MAX9768 amplifier driver"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3-18-g5258 From 69edea8a50fb88edd0f23eecfb89ef513b68eaee Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Jan 2012 21:51:45 +0000 Subject: ASoC: max9768: Fix up review comments in max9768 driver The Mute Switch should be called Playback Switch to match the volume and the regmap core has been updated so we don't need to mark all the registers as non-volatile. Signed-off-by: Mark Brown Acked-by: Wolfram Sang --- sound/soc/codecs/max9768.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 63489952584..79e99018591 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -115,7 +115,7 @@ static const struct snd_kcontrol_new max9768_volume[] = { }; static const struct snd_kcontrol_new max9768_mute[] = { - SOC_SINGLE_BOOL_EXT("Mute Switch", 0, max9768_get_gpio, max9768_set_gpio), + SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio), }; static int max9768_probe(struct snd_soc_codec *codec) @@ -150,18 +150,12 @@ static struct snd_soc_codec_driver max9768_codec_driver = { .num_controls = ARRAY_SIZE(max9768_volume), }; -static bool max9768_always_false(struct device *dev, unsigned int reg) -{ - return false; -} - static const struct regmap_config max9768_i2c_regmap_config = { .reg_bits = 2, .val_bits = 6, .max_register = 3, .reg_defaults = max9768_default_regs, .num_reg_defaults = ARRAY_SIZE(max9768_default_regs), - .volatile_reg = max9768_always_false, .cache_type = REGCACHE_RBTREE, }; -- cgit v1.2.3-18-g5258 From bc122e34469de6ec4b7ca96d3a41724f9e4b1cf4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 25 Jan 2012 19:55:56 +0000 Subject: ASoC: fsi: Remove unneeded empty runtime PM callbacks The runtime PM core no longer requires any callbacks so don't provide empty ones for it any more. Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 14 -------------- 1 file changed, 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index db6c89a28bd..3241e5bdd54 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1408,23 +1408,9 @@ static int fsi_resume(struct device *dev) return 0; } -static int fsi_runtime_nop(struct device *dev) -{ - /* Runtime PM callback shared between ->runtime_suspend() - * and ->runtime_resume(). Simply returns success. - * - * This driver re-initializes all registers after - * pm_runtime_get_sync() anyway so there is no need - * to save and restore registers here. - */ - return 0; -} - static struct dev_pm_ops fsi_pm_ops = { .suspend = fsi_suspend, .resume = fsi_resume, - .runtime_suspend = fsi_runtime_nop, - .runtime_resume = fsi_runtime_nop, }; static struct fsi_core fsi1_core = { -- cgit v1.2.3-18-g5258 From 489773c22397c001f54ba4f30856f24407d8d091 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jan 2012 15:20:24 +0000 Subject: ASoC: wm8993: Convert to use a regmap patch Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index dd687c3a84f..474dc72b4d8 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1077,11 +1077,6 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(wm8993->regmap, false); regcache_sync(wm8993->regmap); - /* Tune DC servo configuration */ - snd_soc_write(codec, 0x44, 3); - snd_soc_write(codec, 0x56, 3); - snd_soc_write(codec, 0x44, 0); - /* Bring up VMID with fast soft start */ snd_soc_update_bits(codec, WM8993_ANTIPOP2, WM8993_STARTUP_BIAS_ENA | @@ -1691,6 +1686,13 @@ static int wm8993_resume(struct snd_soc_codec *codec) #define wm8993_resume NULL #endif +/* Tune DC servo configuration */ +static struct reg_default wm8993_regmap_patch[] = { + { 0x44, 3 }, + { 0x56, 3 }, + { 0x44, 0 }, +}; + static const struct regmap_config wm8993_regmap = { .reg_bits = 8, .val_bits = 16, @@ -1769,6 +1771,12 @@ static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_enable; + ret = regmap_register_patch(wm8993->regmap, wm8993_regmap_patch, + ARRAY_SIZE(wm8993_regmap_patch)); + if (ret != 0) + dev_warn(wm8993->dev, "Failed to apply regmap patch: %d\n", + ret); + if (i2c->irq) { /* Put GPIO1 into interrupt mode (only GPIO1 can output IRQ) */ ret = regmap_update_bits(wm8993->regmap, WM8993_GPIO1, -- cgit v1.2.3-18-g5258 From 5aa44b132eca1d08f5bb9c5c5cb46eec612db686 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jan 2012 15:30:28 +0000 Subject: ASoC: core: Support suspend to disk Use the same pm_ops for all system suspend and resume paths. This isn't ideal for suspend to disk with older CODECs as we'll suspend and then resume the CODEC before powering off all of which takes a long time due to VMID ramps but it's very simple to implement and for modern CODECs the overhead should be minimal. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 35a1e639d7f..091d5f37ae6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1663,8 +1663,7 @@ int snd_soc_poweroff(struct device *dev) EXPORT_SYMBOL_GPL(snd_soc_poweroff); const struct dev_pm_ops snd_soc_pm_ops = { - .suspend = snd_soc_suspend, - .resume = snd_soc_resume, + SET_SYSTEM_SLEEP_PM_OPS(snd_soc_suspend, snd_soc_resume) .poweroff = snd_soc_poweroff, }; EXPORT_SYMBOL_GPL(snd_soc_pm_ops); -- cgit v1.2.3-18-g5258 From f959dee9c7b5e36a139e1e8fcfedbddfea65d00d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 31 Jan 2012 16:16:47 +0000 Subject: ASoC: wm_hubs: Push check for idle_bias_off out into drivers For later wm_hubs devices we have much less need to keep the biases up even when using single ended line outputs so flag idle_bias_off for everything except the WM8993 and WM8994. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 6 ++++++ sound/soc/codecs/wm8994.c | 8 ++++++++ sound/soc/codecs/wm_hubs.c | 6 ------ 3 files changed, 14 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 474dc72b4d8..db51007a6a4 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1622,6 +1622,12 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff, wm8993->pdata.lineout2_diff); + /* If the line outputs are differential then we aren't presenting + * VMID as an output and can disable it. + */ + if (wm8993->pdata.lineout1_diff && wm8993->pdata.lineout2_diff) + codec->dapm.idle_bias_off = 1; + return 0; } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 11ca19b72d7..c26291844e5 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3396,10 +3396,18 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) pm_runtime_enable(codec->dev); pm_runtime_resume(codec->dev); + /* By default use idle_bias_off, will override for WM8994 */ + codec->dapm.idle_bias_off = 1; + /* Set revision-specific configuration */ wm8994->revision = snd_soc_read(codec, WM8994_CHIP_REVISION); switch (control->type) { case WM8994: + /* Single ended line outputs should have VMID on. */ + if (!wm8994->pdata->lineout1_diff || + !wm8994->pdata->lineout2_diff) + codec->dapm.idle_bias_off = 0; + switch (wm8994->revision) { case 2: case 3: diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2a61094075f..ec7d49033d4 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -952,12 +952,6 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, WM8993_LINEOUT2_MODE, WM8993_LINEOUT2_MODE); - /* If the line outputs are differential then we aren't presenting - * VMID as an output and can disable it. - */ - if (lineout1_diff && lineout2_diff) - codec->dapm.idle_bias_off = 1; - if (lineout1fb) snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); -- cgit v1.2.3-18-g5258 From b4dc0a75afb93f317eb3ad0a4d91f7ccfd01cd15 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Tue, 31 Jan 2012 09:26:59 +0200 Subject: ASoC: Tegra+ALC5632: Implement device tree support in board file This patch implements device tree support for Tegra boards with ALC5632 codec. Signed-off-by: Leon Romanovsky Acked-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 78 ++++++++++++++++++++++++++++++++++++----- 1 file changed, 69 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 4a0e805c4ed..c0ba1e42019 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -36,6 +36,7 @@ struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; + struct platform_device *pcm_dev; }; static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, @@ -128,9 +129,7 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_alc5632_dai = { .name = "ALC5632", .stream_name = "ALC5632 PCM", - .codec_name = "alc5632.0-001e", .platform_name = "tegra-pcm-audio", - .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "alc5632-hifi", .init = tegra_alc5632_asoc_init, .ops = &tegra_alc5632_asoc_ops, @@ -163,26 +162,78 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) sizeof(struct tegra_alc5632), GFP_KERNEL); if (!alc5632) { dev_err(&pdev->dev, "Can't allocate tegra_alc5632\n"); - return -ENOMEM; + ret = -ENOMEM; + goto err; } - ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); - if (ret) - return ret; - card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, alc5632); + alc5632->pcm_dev = ERR_PTR(-EINVAL); + + if (!(pdev->dev.of_node)) { + dev_err(&pdev->dev, "Must be instantiated using device tree\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); + if (ret) + goto err; + + ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing"); + if (ret) + goto err; + + tegra_alc5632_dai.codec_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,audio-codec", 0); + + if (!tegra_alc5632_dai.codec_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,audio-codec' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + tegra_alc5632_dai.cpu_dai_of_node = of_parse_phandle( + pdev->dev.of_node, "nvidia,i2s-controller", 0); + if (!tegra_alc5632_dai.cpu_dai_of_node) { + dev_err(&pdev->dev, + "Property 'nvidia,i2s-controller' missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + alc5632->pcm_dev = platform_device_register_simple( + "tegra-pcm-audio", -1, NULL, 0); + if (IS_ERR(alc5632->pcm_dev)) { + dev_err(&pdev->dev, + "Can't instantiate tegra-pcm-audio\n"); + ret = PTR_ERR(alc5632->pcm_dev); + goto err; + } + + ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); + if (ret) + goto err_unregister; + ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - tegra_asoc_utils_fini(&alc5632->util_data); - return ret; + goto err_fini_utils; } return 0; + +err_fini_utils: + tegra_asoc_utils_fini(&alc5632->util_data); +err_unregister: + if (!IS_ERR(alc5632->pcm_dev)) + platform_device_unregister(alc5632->pcm_dev); +err: + return ret; } static int __devexit tegra_alc5632_remove(struct platform_device *pdev) @@ -193,15 +244,23 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev) snd_soc_unregister_card(card); tegra_asoc_utils_fini(&alc5632->util_data); + if (!IS_ERR(alc5632->pcm_dev)) + platform_device_unregister(alc5632->pcm_dev); return 0; } +static const struct of_device_id tegra_alc5632_of_match[] __devinitconst = { + { .compatible = "nvidia,tegra-audio-alc5632", }, + {}, +}; + static struct platform_driver tegra_alc5632_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, + .of_match_table = tegra_alc5632_of_match, }, .probe = tegra_alc5632_probe, .remove = __devexit_p(tegra_alc5632_remove), @@ -212,3 +271,4 @@ MODULE_AUTHOR("Leon Romanovsky "); MODULE_DESCRIPTION("Tegra+ALC5632 machine ASoC driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, tegra_alc5632_of_match); -- cgit v1.2.3-18-g5258 From 4a086e4cc423c5f89ea7b2e25c29800057477b58 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 21:50:00 +0000 Subject: ASoC: wm8996: Switch to using common code for managing CPVDD supply Nice code saving. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 32 +++++--------------------------- 1 file changed, 5 insertions(+), 27 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 3fc30c29f5f..42af0a39683 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -73,7 +73,6 @@ struct wm8996_priv { struct regulator_bulk_data supplies[WM8996_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8996_NUM_SUPPLIES]; - struct regulator *cpvdd; int bg_ena; struct wm8996_pdata pdata; @@ -793,29 +792,18 @@ static int bg_event(struct snd_soc_dapm_widget *w, static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int ret = 0; switch (event) { - case SND_SOC_DAPM_PRE_PMU: - ret = regulator_enable(wm8996->cpvdd); - if (ret != 0) - dev_err(codec->dev, "Failed to enable CPVDD: %d\n", - ret); - break; case SND_SOC_DAPM_POST_PMU: msleep(5); break; - case SND_SOC_DAPM_POST_PMD: - regulator_disable_deferred(wm8996->cpvdd, 20); - break; default: BUG(); ret = -EINVAL; } - return ret; + return 0; } static int rmv_short_event(struct snd_soc_dapm_widget *w, @@ -1117,12 +1105,12 @@ SND_SOC_DAPM_INPUT("IN2RP"), SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), @@ -1281,6 +1269,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "AIFCLK", NULL, "SYSCLK" }, { "SYSDSPCLK", NULL, "SYSCLK" }, { "Charge Pump", NULL, "SYSCLK" }, + { "Charge Pump", NULL, "CPVDD" }, { "MICB1", NULL, "LDO2" }, { "MICB1", NULL, "MICB1 Audio" }, @@ -3049,7 +3038,6 @@ static int wm8996_remove(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) regulator_unregister_notifier(wm8996->supplies[i].consumer, &wm8996->disable_nb[i]); - regulator_put(wm8996->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); return 0; @@ -3172,18 +3160,11 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, goto err_gpio; } - wm8996->cpvdd = regulator_get(&i2c->dev, "CPVDD"); - if (IS_ERR(wm8996->cpvdd)) { - ret = PTR_ERR(wm8996->cpvdd); - dev_err(&i2c->dev, "Failed to get CPVDD: %d\n", ret); - goto err_get; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); - goto err_cpvdd; + goto err_get; } if (wm8996->pdata.ldo_ena > 0) { @@ -3245,8 +3226,6 @@ err_enable: if (wm8996->pdata.ldo_ena > 0) gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); -err_cpvdd: - regulator_put(wm8996->cpvdd); err_get: regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); err_gpio: @@ -3263,7 +3242,6 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); wm8996_free_gpio(wm8996); - regulator_put(wm8996->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); regmap_exit(wm8996->regmap); if (wm8996->pdata.ldo_ena > 0) { -- cgit v1.2.3-18-g5258 From 24e0c57b8ed2b3a9fe07c07edc1c0062df5be8bf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 21 Jan 2012 22:18:52 +0000 Subject: ASoC: wm8996: Use devm_regulator_bulk_get() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 42af0a39683..10f41c88888 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3153,8 +3153,8 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) wm8996->supplies[i].supply = wm8996_supply_names[i]; - ret = regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8996->supplies), - wm8996->supplies); + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8996->supplies), + wm8996->supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); goto err_gpio; @@ -3164,7 +3164,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, wm8996->supplies); if (ret != 0) { dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; + goto err_gpio; } if (wm8996->pdata.ldo_ena > 0) { @@ -3226,8 +3226,6 @@ err_enable: if (wm8996->pdata.ldo_ena > 0) gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); err_gpio: if (wm8996->pdata.ldo_ena > 0) gpio_free(wm8996->pdata.ldo_ena); @@ -3242,7 +3240,6 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); wm8996_free_gpio(wm8996); - regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); regmap_exit(wm8996->regmap); if (wm8996->pdata.ldo_ena > 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); -- cgit v1.2.3-18-g5258 From 2633f736470e803ac9f5372a0d83ba108345a80a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 26 Jan 2012 17:43:09 +0000 Subject: ASoC: wm5100: Handle failures to determine accessory polarity If we get an indeterminate impedance with both headset polarities then give up and report the accessory as a headphone rather than continually retrying. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 37 +++++++++++++++++++++++++------------ 1 file changed, 25 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index c6c382197fe..f6b6ea89802 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -72,6 +72,7 @@ struct wm5100_priv { bool jack_detecting; bool jack_mic; int jack_mode; + int jack_flips; struct wm5100_fll fll[2]; @@ -1996,6 +1997,19 @@ static void wm5100_set_detect_mode(struct wm5100_priv *wm5100, int the_mode) wm5100->jack_mode); } +static void wm5100_report_headphone(struct wm5100_priv *wm5100) +{ + dev_dbg(wm5100->dev, "Headphone detected\n"); + wm5100->jack_detecting = false; + snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, + SND_JACK_HEADPHONE); + + /* Increase the detection rate a bit for responsiveness. */ + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 7 << WM5100_ACCDET_RATE_SHIFT); +} + static void wm5100_micd_irq(struct wm5100_priv *wm5100) { unsigned int val; @@ -2020,6 +2034,7 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100) dev_dbg(wm5100->dev, "Jack removal detected\n"); wm5100->jack_mic = false; wm5100->jack_detecting = true; + wm5100->jack_flips = 0; snd_soc_jack_report(wm5100->jack, 0, SND_JACK_LINEOUT | SND_JACK_HEADSET | SND_JACK_BTN_0); @@ -2058,10 +2073,16 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100) /* If we detected a lower impedence during initial startup * then we probably have the wrong polarity, flip it. Don't * do this for the lowest impedences to speed up detection of - * plain headphones. + * plain headphones and give up if neither polarity looks + * sensible. */ if (wm5100->jack_detecting && (val & 0x3f8)) { - wm5100_set_detect_mode(wm5100, !wm5100->jack_mode); + wm5100->jack_flips++; + + if (wm5100->jack_flips > 1) + wm5100_report_headphone(wm5100); + else + wm5100_set_detect_mode(wm5100, !wm5100->jack_mode); return; } @@ -2075,16 +2096,7 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100) snd_soc_jack_report(wm5100->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); } else if (wm5100->jack_detecting) { - dev_dbg(wm5100->dev, "Headphone detected\n"); - snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, - SND_JACK_HEADPHONE); - - /* Increase the detection rate a bit for - * responsiveness. - */ - regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - 7 << WM5100_ACCDET_RATE_SHIFT); + wm5100_report_headphone(wm5100); } } } @@ -2096,6 +2108,7 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) if (jack) { wm5100->jack = jack; wm5100->jack_detecting = true; + wm5100->jack_flips = 0; wm5100_set_detect_mode(wm5100, 0); -- cgit v1.2.3-18-g5258 From f3bafaa096b64dabbaea3e3a2be213a9628ebe4d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Feb 2012 12:42:39 +0000 Subject: ASoC: wm9712: Remove driver specific version I don't think it's been updated since the driver was merged. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index b7b31f84c10..247f19a31ba 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -22,8 +22,6 @@ #include #include "wm9712.h" -#define WM9712_VERSION "0.4" - static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg); static int ac97_write(struct snd_soc_codec *codec, @@ -619,8 +617,6 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) { int ret = 0; - printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION); - ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); -- cgit v1.2.3-18-g5258 From 471280b715769be8d368f9d6bbd28558d7661f5f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Feb 2012 12:42:59 +0000 Subject: ASoC: wm9712: Add TLV information for microphone input Signed-off-by: Mark Brown --- sound/soc/codecs/wm9712.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 247f19a31ba..7291eabb0eb 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -20,6 +20,7 @@ #include #include #include +#include #include "wm9712.h" static unsigned int ac97_read(struct snd_soc_codec *codec, @@ -69,6 +70,9 @@ static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer", static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"}; static const char *wm9712_diff_sel[] = {"Mic", "Line"}; +static const DECLARE_TLV_DB_SCALE(main_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 2000, 0); + static const struct soc_enum wm9712_enum[] = { SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select), SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux), @@ -147,9 +151,9 @@ SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), -SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), -SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), -SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), +SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), +SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv), +SOC_SINGLE_TLV("Mic Boost Volume", AC97_MIC, 7, 1, 0, boost_tlv), }; /* We have to create a fake left and right HP mixers because -- cgit v1.2.3-18-g5258 From 980b0bc69270a9650ffb94f08dc87740ee1fb9b4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 1 Feb 2012 19:24:21 +0100 Subject: ASoC: blackfin: Use dai_fmt Use the dai_link's dai_fmt attribute to setup the DAI format instead of doing this manually. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ad1836.c | 17 +++++------------ sound/soc/blackfin/bf5xx-ad193x.c | 17 +++++------------ sound/soc/blackfin/bf5xx-ad73311.c | 29 ++++------------------------- sound/soc/blackfin/bf5xx-ssm2602.c | 20 ++++---------------- sound/soc/blackfin/bfin-eval-adau1373.c | 13 ++----------- sound/soc/blackfin/bfin-eval-adau1701.c | 16 +++++----------- sound/soc/blackfin/bfin-eval-adav80x.c | 13 ++----------- 7 files changed, 27 insertions(+), 98 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 60962ce6cd4..d542d406377 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -40,20 +40,8 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; /* set cpu DAI channel mapping */ ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), @@ -68,6 +56,9 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { .hw_params = bf5xx_ad1836_hw_params, }; +#define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { { .name = "ad1836", @@ -77,6 +68,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }, { .name = "ad1836", @@ -86,6 +78,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 2d8d82dbc15..0e55e9f2a51 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -60,18 +60,6 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, break; } - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, SND_SOC_CLOCK_IN); @@ -92,6 +80,9 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, return 0; } +#define BF5XX_AD193X_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_ops bf5xx_ad193x_ops = { .hw_params = bf5xx_ad193x_hw_params, }; @@ -105,6 +96,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, + .dai_fmt = BF5XX_AD193X_DAIFMT, }, { .name = "ad193x", @@ -114,6 +106,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .platform_name = "bfin-tdm-pcm-audio", .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, + .dai_fmt = BF5XX_AD193X_DAIFMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 8e49508596d..61cc91d4a02 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -145,29 +145,8 @@ static int bf5xx_probe(struct snd_soc_card *card) return 0; } -static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret = 0; - - pr_debug("%s rate %d format %x\n", __func__, params_rate(params), - params_format(params)); - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - - return 0; -} - - -static struct snd_soc_ops bf5xx_ad73311_ops = { - .hw_params = bf5xx_ad73311_hw_params, -}; +#define BF5XX_AD7311_DAI_FMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { { @@ -177,7 +156,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ad73311", - .ops = &bf5xx_ad73311_ops, + .dai_fmt = BF5XX_AD7311_DAI_FMT, }, { .name = "ad73311", @@ -186,7 +165,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ad73311", - .ops = &bf5xx_ad73311_ops, + .dai_fmt = BF5XX_AD7311_DAI_FMT, }, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index 03030323804..df3ac73f877 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -49,7 +49,6 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int clk = 0; int ret = 0; @@ -75,21 +74,6 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream, break; } - /* - * CODEC is master for BCLK and LRC in this configuration. - */ - - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) @@ -102,6 +86,10 @@ static struct snd_soc_ops bf5xx_ssm2602_ops = { .hw_params = bf5xx_ssm2602_hw_params, }; +/* CODEC is master for BCLK and LRC in this configuration. */ +#define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { { .name = "ssm2602", diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 26b271c62ef..f3adbdbdd5e 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -67,21 +67,10 @@ static int bfin_eval_adau1373_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; int pll_rate; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - switch (params_rate(params)) { case 48000: case 8000: @@ -143,6 +132,8 @@ static struct snd_soc_dai_link bfin_eval_adau1373_dai = { .codec_name = "adau1373.0-001a", .ops = &bfin_eval_adau1373_ops, .init = bfin_eval_adau1373_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }; static struct snd_soc_card bfin_eval_adau1373 = { diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index c0064fa1dca..b0531fc9d81 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -37,20 +37,9 @@ static int bfin_eval_adau1701_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000, SND_SOC_CLOCK_IN); @@ -61,6 +50,9 @@ static struct snd_soc_ops bfin_eval_adau1701_ops = { .hw_params = bfin_eval_adau1701_hw_params, }; +#define BFIN_EVAL_ADAU1701_DAI_FMT (SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM) + static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { { .name = "adau1701", @@ -70,6 +62,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "adau1701.0-0034", .ops = &bfin_eval_adau1701_ops, + .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, }, { .name = "adau1701", @@ -79,6 +72,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "adau1701.0-0034", .ops = &bfin_eval_adau1701_ops, + .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, }, }; diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 4ef079f95e2..84b09987b7f 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -34,20 +34,9 @@ static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) - return ret; - ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL, 27000000, params_rate(params) * 256); if (ret) @@ -88,6 +77,8 @@ static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = { .platform_name = "bfin-i2s-pcm-audio", .init = bfin_eval_adav80x_codec_init, .ops = &bfin_eval_adav80x_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, }, }; -- cgit v1.2.3-18-g5258 From 5813db970d7db40db5979f2f74f42935450e8e9c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 29 Jan 2012 16:52:02 +0800 Subject: ASoC: Use dai_fmt in afeb9260 machine driver Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/atmel/snd-soc-afeb9260.c | 23 ++--------------------- 1 file changed, 2 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index 4ca667d477f..cb0130cf8f1 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -46,29 +46,8 @@ static int afeb9260_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; - /* Set codec DAI configuration */ - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S| - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set codec DAI configuration\n"); - return err; - } - - /* Set cpu DAI configuration */ - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM); - if (err < 0) { - printk(KERN_ERR "can't set cpu DAI configuration\n"); - return err; - } - /* Set the codec system clock for DAC and ADC */ err = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); @@ -129,6 +108,8 @@ static struct snd_soc_dai_link afeb9260_dai = { .platform_name = "atmel_pcm-audio", .codec_name = "tlv320aic23-codec.0-001a", .init = afeb9260_tlv320aic23_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &afeb9260_ops, }; -- cgit v1.2.3-18-g5258 From bc6c117ef0d8c09d643ab3da8b4976e32e2fcdab Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 29 Jan 2012 16:52:51 +0800 Subject: ASoC: Convert afeb9260 to table based DAPM init Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/atmel/snd-soc-afeb9260.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index cb0130cf8f1..f65f08beac3 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -70,7 +70,7 @@ static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { SND_SOC_DAPM_MIC("Mic Jack", NULL), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route afeb9260_audio_map[] = { {"Headphone Jack", NULL, "LHPOUT"}, {"Headphone Jack", NULL, "RHPOUT"}, @@ -85,13 +85,6 @@ static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - /* Add afeb9260 specific widgets */ - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - /* Set up afeb9260 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Line In"); snd_soc_dapm_enable_pin(dapm, "Mic Jack"); @@ -119,6 +112,11 @@ static struct snd_soc_card snd_soc_machine_afeb9260 = { .owner = THIS_MODULE, .dai_link = &afeb9260_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = afeb9260_audio_map, + .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map), }; static struct platform_device *afeb9260_snd_device; -- cgit v1.2.3-18-g5258 From 4f2864a49bf058184e85c9f5a2f4578f11992c7d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 25 Jan 2012 11:54:19 +0100 Subject: ALSA: hda - Enable sync_write and reset for Conexant codecs This is an attempt to fix S3-resume problems reported for a few laptops with different Conexant codecs. They show the communication stalls at some time in S3, and the driver falls back into the single-cmd mode. This leads to the silent output or the lack of auto-mute feature. As a workaround, here enables the sync_write and the bus-reset flags to make the communication more stable. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=740115 Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738397 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a7a5733aa4d..117ae4c22be 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4414,6 +4414,18 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->patch_ops = cx_auto_patch_ops; if (spec->beep_amp) snd_hda_attach_beep_device(codec, spec->beep_amp); + + /* Some laptops with Conexant chips show stalls in S3 resume, + * which falls into the single-cmd mode. + * Better to make reset, then. + */ + if (!codec->bus->sync_write) { + snd_printd("hda_codec: " + "Enable sync_write for stable communication\n"); + codec->bus->sync_write = 1; + codec->bus->allow_bus_reset = 1; + } + return 0; } -- cgit v1.2.3-18-g5258 From 25bfe662e8c42f84851f79ed6ada5ef96a2ff329 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Feb 2012 21:30:32 +0000 Subject: ASoC: pcm: Improve error logging Use the standard logging macros and use dev_ variants where we can, also reporting error codes whenever we report an error. These changes (the error codes in particular) make it noticeably easier to figure out what went wrong just from the basic dmesg output. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-pcm.c | 39 ++++++++++++++++++++++----------------- 1 file changed, 22 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 93be95b7864..121318defea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -123,8 +123,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->startup) { ret = cpu_dai->driver->ops->startup(substream, cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: can't open interface %s\n", - cpu_dai->name); + dev_err(cpu_dai->dev, "can't open interface %s: %d\n", + cpu_dai->name, ret); goto out; } } @@ -132,7 +132,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (platform->driver->ops && platform->driver->ops->open) { ret = platform->driver->ops->open(substream); if (ret < 0) { - printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); + dev_err(platform->dev, "can't open platform %s: %d\n", + platform->name, ret); goto platform_err; } } @@ -140,8 +141,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (codec_dai->driver->ops->startup) { ret = codec_dai->driver->ops->startup(substream, codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: can't open codec %s\n", - codec_dai->name); + dev_err(codec_dai->dev, "can't open codec %s: %d\n", + codec_dai->name, ret); goto codec_dai_err; } } @@ -149,7 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (rtd->dai_link->ops && rtd->dai_link->ops->startup) { ret = rtd->dai_link->ops->startup(substream); if (ret < 0) { - printk(KERN_ERR "asoc: %s startup failed\n", rtd->dai_link->name); + pr_err("asoc: %s startup failed: %d\n", + rtd->dai_link->name, ret); goto machine_err; } } @@ -413,7 +415,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); if (ret < 0) { - printk(KERN_ERR "asoc: machine prepare error\n"); + pr_err("asoc: machine prepare error: %d\n", ret); goto out; } } @@ -421,7 +423,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (platform->driver->ops && platform->driver->ops->prepare) { ret = platform->driver->ops->prepare(substream); if (ret < 0) { - printk(KERN_ERR "asoc: platform prepare error\n"); + dev_err(platform->dev, "platform prepare error: %d\n", + ret); goto out; } } @@ -429,7 +432,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (codec_dai->driver->ops->prepare) { ret = codec_dai->driver->ops->prepare(substream, codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: codec DAI prepare error\n"); + dev_err(codec_dai->dev, "DAI prepare error: %d\n", + ret); goto out; } } @@ -437,7 +441,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: cpu DAI prepare error\n"); + dev_err(cpu_dai->dev, "DAI prepare error: %d\n", + ret); goto out; } } @@ -484,7 +489,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: machine hw_params failed\n"); + pr_err("asoc: machine hw_params failed: %d\n", ret); goto out; } } @@ -492,8 +497,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (codec_dai->driver->ops->hw_params) { ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: can't set codec %s hw params\n", - codec_dai->name); + dev_err(codec_dai->dev, "can't set %s hw params: %d\n", + codec_dai->name, ret); goto codec_err; } } @@ -501,8 +506,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (cpu_dai->driver->ops->hw_params) { ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: interface %s hw params failed\n", - cpu_dai->name); + dev_err(cpu_dai->dev, "%s hw params failed: %d\n", + cpu_dai->name, ret); goto interface_err; } } @@ -510,8 +515,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, if (platform->driver->ops && platform->driver->ops->hw_params) { ret = platform->driver->ops->hw_params(substream, params); if (ret < 0) { - printk(KERN_ERR "asoc: platform %s hw params failed\n", - platform->name); + dev_err(platform->dev, "%s hw params failed: %d\n", + platform->name, ret); goto platform_err; } } -- cgit v1.2.3-18-g5258 From d559f1e5ad2d839f0a2192526c857cd0b24bf420 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Thu, 2 Feb 2012 22:13:37 +0200 Subject: ASoC: Tegra+ALC5632: Enable headset autodetection on PAZ00 board. This patch is adding device tree support of headset autodetection on PAZ00 board. Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 39 +++++++++++++++++++++++++++++++++++---- 1 file changed, 35 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index c0ba1e42019..17941392fd7 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include @@ -34,9 +35,13 @@ #define DRV_NAME "tegra-alc5632" +#define GPIO_HP_DET BIT(0) + struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; struct platform_device *pcm_dev; + int gpio_requested; + int gpio_hp_det; }; static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, @@ -86,6 +91,13 @@ static struct snd_soc_jack_pin tegra_alc5632_hs_jack_pins[] = { }, }; +static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { + .name = "Headset detection", + .report = SND_JACK_HEADSET, + .debounce_time = 150, + .invert = 1, +}; + static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { SND_SOC_DAPM_SPK("Int Spk", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), @@ -114,6 +126,9 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; + struct device_node *np = codec->card->dev->of_node; + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); + int ret; snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &tegra_alc5632_hs_jack); @@ -121,6 +136,16 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(tegra_alc5632_hs_jack_pins), tegra_alc5632_hs_jack_pins); + machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + + if (gpio_is_valid(machine->gpio_hp_det)) { + tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; + snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, + 1, + &tegra_alc5632_hp_jack_gpio); + machine->gpio_requested |= GPIO_HP_DET; + } + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); return 0; @@ -239,13 +264,19 @@ err: static int __devexit tegra_alc5632_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); + struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); + + if (machine->gpio_requested & GPIO_HP_DET) + snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, + 1, + &tegra_alc5632_hp_jack_gpio); + machine->gpio_requested = 0; snd_soc_unregister_card(card); - tegra_asoc_utils_fini(&alc5632->util_data); - if (!IS_ERR(alc5632->pcm_dev)) - platform_device_unregister(alc5632->pcm_dev); + tegra_asoc_utils_fini(&machine->util_data); + if (!IS_ERR(machine->pcm_dev)) + platform_device_unregister(machine->pcm_dev); return 0; } -- cgit v1.2.3-18-g5258 From 9f71770b88d1dafa46d4f3c3b359d1791e23eecf Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Thu, 2 Feb 2012 22:13:38 +0200 Subject: ASoC: tegra: Remove unused DAPM route structure. All DAPM routes are configured via device tree, and there is no need in DAPM route structures in board file. Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 16 ---------------- 1 file changed, 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 17941392fd7..c000f51c2ff 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -104,20 +104,6 @@ static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), }; -static const struct snd_soc_dapm_route tegra_alc5632_audio_map[] = { - /* Internal Speaker */ - {"Int Spk", NULL, "SPKOUT"}, - {"Int Spk", NULL, "SPKOUTN"}, - - /* Headset Mic */ - {"MIC1", NULL, "MICBIAS1"}, - {"MICBIAS1", NULL, "Headset Mic"}, - - /* Headset Stereophone */ - {"Headset Stereophone", NULL, "HPR"}, - {"Headset Stereophone", NULL, "HPL"}, -}; - static const struct snd_kcontrol_new tegra_alc5632_controls[] = { SOC_DAPM_PIN_SWITCH("Int Spk"), }; @@ -172,8 +158,6 @@ static struct snd_soc_card snd_soc_tegra_alc5632 = { .num_controls = ARRAY_SIZE(tegra_alc5632_controls), .dapm_widgets = tegra_alc5632_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tegra_alc5632_dapm_widgets), - .dapm_routes = tegra_alc5632_audio_map, - .num_dapm_routes = ARRAY_SIZE(tegra_alc5632_audio_map), .fully_routed = true, }; -- cgit v1.2.3-18-g5258 From 356268bde2efc8aa36364d3f3113a7cf92e079a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Feb 2012 10:18:48 +0100 Subject: ALSA: hda - Remove fallback to model=ideapad for Lenovo with cx5066 The Lenovo laptops with cx5066 chips seem to work better with model=auto. Let's get rid of the fallback to the wrong model. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738397 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 117ae4c22be..0eb526c672b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3034,7 +3034,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; -- cgit v1.2.3-18-g5258 From c1e6f10ea94715f00cce4c9aaf7fc91fb34ec52d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:50:09 -0800 Subject: ASoC: fsi: reduce runtime calculation by using pre-setting Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3241e5bdd54..0d78740d0a6 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -167,6 +167,7 @@ struct fsi_stream { int buff_sample_pos; /* sample position of ALSA buffer */ int period_samples; /* sample number / 1 period */ int period_pos; /* current period position */ + int sample_width; /* sample width */ int uerr_num; int oerr_num; @@ -406,6 +407,7 @@ static void fsi_stream_push(struct fsi_priv *fsi, io->buff_sample_pos = 0; io->period_samples = fsi_frame2sample(fsi, runtime->period_size); io->period_pos = 0; + io->sample_width = samples_to_bytes(runtime, 1); io->oerr_num = -1; /* ignore 1st err */ io->uerr_num = -1; /* ignore 1st err */ spin_unlock_irqrestore(&master->lock, flags); @@ -431,6 +433,7 @@ static void fsi_stream_pop(struct fsi_priv *fsi, int is_play) io->buff_sample_pos = 0; io->period_samples = 0; io->period_pos = 0; + io->sample_width = 0; io->oerr_num = 0; io->uerr_num = 0; spin_unlock_irqrestore(&master->lock, flags); @@ -752,7 +755,6 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) int is_play = fsi_stream_is_play(stream); struct fsi_stream *io = fsi_get_stream(fsi, is_play); int sample_residues; - int sample_width; int samples; int samples_max; int over_period; @@ -780,9 +782,6 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) io->buff_sample_pos = 0; } - /* get 1 sample data width */ - sample_width = samples_to_bytes(runtime, 1); - /* get number of residue samples */ sample_residues = io->buff_sample_capa - io->buff_sample_pos; @@ -798,7 +797,7 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) samples = sample_residues; - switch (sample_width) { + switch (io->sample_width) { case 2: fn = fsi_dma_soft_push16; break; @@ -818,7 +817,7 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) samples_max = sample_residues; samples = fsi_get_current_fifo_samples(fsi, is_play); - switch (sample_width) { + switch (io->sample_width) { case 2: fn = fsi_dma_soft_pop16; break; -- cgit v1.2.3-18-g5258 From 4e62d84d9da5190c303d6408180fbfee414d25bc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:50:35 -0800 Subject: ASoC: fsi: tidyup: fsi_stream_xx() functions were gathered This patch gathered fsi_stream_xxx() functions in order to make it readable. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 115 ++++++++++++++++++++++++++--------------------------- 1 file changed, 57 insertions(+), 58 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 0d78740d0a6..162416e3e6b 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -338,22 +338,6 @@ static u32 fsi_get_info_flags(struct fsi_priv *fsi) master->info->portb_flags; } -static inline int fsi_stream_is_play(int stream) -{ - return stream == SNDRV_PCM_STREAM_PLAYBACK; -} - -static inline int fsi_is_play(struct snd_pcm_substream *substream) -{ - return fsi_stream_is_play(substream->stream); -} - -static inline struct fsi_stream *fsi_get_stream(struct fsi_priv *fsi, - int is_play) -{ - return is_play ? &fsi->playback : &fsi->capture; -} - static u32 fsi_get_port_shift(struct fsi_priv *fsi, int is_play) { int is_porta = fsi_is_port_a(fsi); @@ -377,10 +361,60 @@ static int fsi_sample2frame(struct fsi_priv *fsi, int samples) return samples / fsi->chan_num; } +static int fsi_get_current_fifo_samples(struct fsi_priv *fsi, int is_play) +{ + u32 status; + int frames; + + status = is_play ? + fsi_reg_read(fsi, DOFF_ST) : + fsi_reg_read(fsi, DIFF_ST); + + frames = 0x1ff & (status >> 8); + + return fsi_frame2sample(fsi, frames); +} + +static void fsi_count_fifo_err(struct fsi_priv *fsi) +{ + u32 ostatus = fsi_reg_read(fsi, DOFF_ST); + u32 istatus = fsi_reg_read(fsi, DIFF_ST); + + if (ostatus & ERR_OVER) + fsi->playback.oerr_num++; + + if (ostatus & ERR_UNDER) + fsi->playback.uerr_num++; + + if (istatus & ERR_OVER) + fsi->capture.oerr_num++; + + if (istatus & ERR_UNDER) + fsi->capture.uerr_num++; + + fsi_reg_write(fsi, DOFF_ST, 0); + fsi_reg_write(fsi, DIFF_ST, 0); +} + +/* + * fsi_stream_xx() function + */ +#define fsi_is_play(substream) fsi_stream_is_play(substream->stream) +static inline int fsi_stream_is_play(int stream) +{ + return stream == SNDRV_PCM_STREAM_PLAYBACK; +} + +static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi, + int is_play) +{ + return is_play ? &fsi->playback : &fsi->capture; +} + static int fsi_stream_is_working(struct fsi_priv *fsi, int is_play) { - struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct fsi_stream *io = fsi_stream_get(fsi, is_play); struct fsi_master *master = fsi_get_master(fsi); unsigned long flags; int ret; @@ -396,7 +430,7 @@ static void fsi_stream_push(struct fsi_priv *fsi, int is_play, struct snd_pcm_substream *substream) { - struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct fsi_stream *io = fsi_stream_get(fsi, is_play); struct snd_pcm_runtime *runtime = substream->runtime; struct fsi_master *master = fsi_get_master(fsi); unsigned long flags; @@ -415,7 +449,7 @@ static void fsi_stream_push(struct fsi_priv *fsi, static void fsi_stream_pop(struct fsi_priv *fsi, int is_play) { - struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct fsi_stream *io = fsi_stream_get(fsi, is_play); struct snd_soc_dai *dai = fsi_get_dai(io->substream); struct fsi_master *master = fsi_get_master(fsi); unsigned long flags; @@ -439,41 +473,6 @@ static void fsi_stream_pop(struct fsi_priv *fsi, int is_play) spin_unlock_irqrestore(&master->lock, flags); } -static int fsi_get_current_fifo_samples(struct fsi_priv *fsi, int is_play) -{ - u32 status; - int frames; - - status = is_play ? - fsi_reg_read(fsi, DOFF_ST) : - fsi_reg_read(fsi, DIFF_ST); - - frames = 0x1ff & (status >> 8); - - return fsi_frame2sample(fsi, frames); -} - -static void fsi_count_fifo_err(struct fsi_priv *fsi) -{ - u32 ostatus = fsi_reg_read(fsi, DOFF_ST); - u32 istatus = fsi_reg_read(fsi, DIFF_ST); - - if (ostatus & ERR_OVER) - fsi->playback.oerr_num++; - - if (ostatus & ERR_UNDER) - fsi->playback.uerr_num++; - - if (istatus & ERR_OVER) - fsi->capture.oerr_num++; - - if (istatus & ERR_UNDER) - fsi->capture.uerr_num++; - - fsi_reg_write(fsi, DOFF_ST, 0); - fsi_reg_write(fsi, DIFF_ST, 0); -} - /* * dma function */ @@ -481,7 +480,7 @@ static void fsi_count_fifo_err(struct fsi_priv *fsi) static u8 *fsi_dma_get_area(struct fsi_priv *fsi, int stream) { int is_play = fsi_stream_is_play(stream); - struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct fsi_stream *io = fsi_stream_get(fsi, is_play); struct snd_pcm_runtime *runtime = io->substream->runtime; return runtime->dma_area + @@ -698,7 +697,7 @@ static void fsi_fifo_init(struct fsi_priv *fsi, struct device *dev) { struct fsi_master *master = fsi_get_master(fsi); - struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct fsi_stream *io = fsi_stream_get(fsi, is_play); u32 shift, i; int frame_capa; @@ -753,7 +752,7 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream = NULL; int is_play = fsi_stream_is_play(stream); - struct fsi_stream *io = fsi_get_stream(fsi, is_play); + struct fsi_stream *io = fsi_stream_get(fsi, is_play); int sample_residues; int samples; int samples_max; @@ -1150,7 +1149,7 @@ static int fsi_hw_free(struct snd_pcm_substream *substream) static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct fsi_priv *fsi = fsi_get_priv(substream); - struct fsi_stream *io = fsi_get_stream(fsi, fsi_is_play(substream)); + struct fsi_stream *io = fsi_stream_get(fsi, fsi_is_play(substream)); int samples_pos = io->buff_sample_pos - 1; if (samples_pos < 0) -- cgit v1.2.3-18-g5258 From 376cf38a90507f82d22b951b7776557aefe6109c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:50:59 -0800 Subject: ASoC: fsi: data push/pop calculation part was divided Next transfer data size of "push" and "pop" had calculated on shared function. But it was not readable code. This patch divided it into for push, and for pop. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 104 +++++++++++++++++++++++------------------------------ 1 file changed, 45 insertions(+), 59 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 162416e3e6b..cbb5643794b 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -747,17 +747,14 @@ static void fsi_fifo_init(struct fsi_priv *fsi, } } -static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) +static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, struct fsi_stream *io, + void (*run16)(struct fsi_priv *fsi, int size), + void (*run32)(struct fsi_priv *fsi, int size), + int samples) { struct snd_pcm_runtime *runtime; - struct snd_pcm_substream *substream = NULL; - int is_play = fsi_stream_is_play(stream); - struct fsi_stream *io = fsi_stream_get(fsi, is_play); - int sample_residues; - int samples; - int samples_max; + struct snd_pcm_substream *substream; int over_period; - void (*fn)(struct fsi_priv *fsi, int size); if (!fsi || !io->substream || @@ -781,57 +778,17 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) io->buff_sample_pos = 0; } - /* get number of residue samples */ - sample_residues = io->buff_sample_capa - io->buff_sample_pos; - - if (is_play) { - /* - * for play-back - * - * samples_max : number of FSI fifo free samples space - * samples : number of ALSA residue samples - */ - samples_max = io->fifo_sample_capa; - samples_max -= fsi_get_current_fifo_samples(fsi, is_play); - - samples = sample_residues; - - switch (io->sample_width) { - case 2: - fn = fsi_dma_soft_push16; - break; - case 4: - fn = fsi_dma_soft_push32; - break; - default: - return -EINVAL; - } - } else { - /* - * for capture - * - * samples_max : number of ALSA free samples space - * samples : number of samples in FSI fifo - */ - samples_max = sample_residues; - samples = fsi_get_current_fifo_samples(fsi, is_play); - - switch (io->sample_width) { - case 2: - fn = fsi_dma_soft_pop16; - break; - case 4: - fn = fsi_dma_soft_pop32; - break; - default: - return -EINVAL; - } + switch (io->sample_width) { + case 2: + run16(fsi, samples); + break; + case 4: + run32(fsi, samples); + break; + default: + return -EINVAL; } - samples = min(samples, samples_max); - - fn(fsi, samples); - /* update buff_sample_pos */ io->buff_sample_pos += samples; @@ -843,12 +800,41 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, int stream) static int fsi_data_pop(struct fsi_priv *fsi) { - return fsi_fifo_data_ctrl(fsi, SNDRV_PCM_STREAM_CAPTURE); + int is_play = fsi_stream_is_play(SNDRV_PCM_STREAM_CAPTURE); + int sample_residues; /* samples in FSI fifo */ + int sample_space; /* ALSA free samples space */ + int samples; + struct fsi_stream *io = fsi_stream_get(fsi, is_play); + + sample_residues = fsi_get_current_fifo_samples(fsi, is_play); + sample_space = io->buff_sample_capa - io->buff_sample_pos; + + samples = min(sample_residues, sample_space); + + return fsi_fifo_data_ctrl(fsi, io, + fsi_dma_soft_pop16, + fsi_dma_soft_pop32, + samples); } static int fsi_data_push(struct fsi_priv *fsi) { - return fsi_fifo_data_ctrl(fsi, SNDRV_PCM_STREAM_PLAYBACK); + int is_play = fsi_stream_is_play(SNDRV_PCM_STREAM_PLAYBACK); + int sample_residues; /* ALSA residue samples */ + int sample_space; /* FSI fifo free samples space */ + int samples; + struct fsi_stream *io = fsi_stream_get(fsi, is_play); + + sample_residues = io->buff_sample_capa - io->buff_sample_pos; + sample_space = io->fifo_sample_capa - + fsi_get_current_fifo_samples(fsi, is_play); + + samples = min(sample_residues, sample_space); + + return fsi_fifo_data_ctrl(fsi, io, + fsi_dma_soft_push16, + fsi_dma_soft_push32, + samples); } static irqreturn_t fsi_interrupt(int irq, void *data) -- cgit v1.2.3-18-g5258 From d78629e2a4457149bd21fdb0cdbbb1c3ec019d96 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:51:14 -0800 Subject: ASoC: fsi: rename fsi_dma_soft_xxx() to fsi_pio_xxx() This is preparation for DMAEngine support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index cbb5643794b..05307acb2bd 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -474,10 +474,10 @@ static void fsi_stream_pop(struct fsi_priv *fsi, int is_play) } /* - * dma function + * pio function */ -static u8 *fsi_dma_get_area(struct fsi_priv *fsi, int stream) +static u8 *fsi_pio_get_area(struct fsi_priv *fsi, int stream) { int is_play = fsi_stream_is_play(stream); struct fsi_stream *io = fsi_stream_get(fsi, is_play); @@ -487,47 +487,47 @@ static u8 *fsi_dma_get_area(struct fsi_priv *fsi, int stream) samples_to_bytes(runtime, io->buff_sample_pos); } -static void fsi_dma_soft_push16(struct fsi_priv *fsi, int num) +static void fsi_pio_push16(struct fsi_priv *fsi, int num) { u16 *start; int i; - start = (u16 *)fsi_dma_get_area(fsi, SNDRV_PCM_STREAM_PLAYBACK); + start = (u16 *)fsi_pio_get_area(fsi, SNDRV_PCM_STREAM_PLAYBACK); for (i = 0; i < num; i++) fsi_reg_write(fsi, DODT, ((u32)*(start + i) << 8)); } -static void fsi_dma_soft_pop16(struct fsi_priv *fsi, int num) +static void fsi_pio_pop16(struct fsi_priv *fsi, int num) { u16 *start; int i; - start = (u16 *)fsi_dma_get_area(fsi, SNDRV_PCM_STREAM_CAPTURE); + start = (u16 *)fsi_pio_get_area(fsi, SNDRV_PCM_STREAM_CAPTURE); for (i = 0; i < num; i++) *(start + i) = (u16)(fsi_reg_read(fsi, DIDT) >> 8); } -static void fsi_dma_soft_push32(struct fsi_priv *fsi, int num) +static void fsi_pio_push32(struct fsi_priv *fsi, int num) { u32 *start; int i; - start = (u32 *)fsi_dma_get_area(fsi, SNDRV_PCM_STREAM_PLAYBACK); + start = (u32 *)fsi_pio_get_area(fsi, SNDRV_PCM_STREAM_PLAYBACK); for (i = 0; i < num; i++) fsi_reg_write(fsi, DODT, *(start + i)); } -static void fsi_dma_soft_pop32(struct fsi_priv *fsi, int num) +static void fsi_pio_pop32(struct fsi_priv *fsi, int num) { u32 *start; int i; - start = (u32 *)fsi_dma_get_area(fsi, SNDRV_PCM_STREAM_CAPTURE); + start = (u32 *)fsi_pio_get_area(fsi, SNDRV_PCM_STREAM_CAPTURE); for (i = 0; i < num; i++) *(start + i) = fsi_reg_read(fsi, DIDT); @@ -812,8 +812,8 @@ static int fsi_data_pop(struct fsi_priv *fsi) samples = min(sample_residues, sample_space); return fsi_fifo_data_ctrl(fsi, io, - fsi_dma_soft_pop16, - fsi_dma_soft_pop32, + fsi_pio_pop16, + fsi_pio_pop32, samples); } @@ -832,8 +832,8 @@ static int fsi_data_push(struct fsi_priv *fsi) samples = min(sample_residues, sample_space); return fsi_fifo_data_ctrl(fsi, io, - fsi_dma_soft_push16, - fsi_dma_soft_push32, + fsi_pio_push16, + fsi_pio_push32, samples); } -- cgit v1.2.3-18-g5258 From b49e8027810b674dc0bf0ba3d629c5fae52d78f3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:51:29 -0800 Subject: ASoC: fsi: tidyup: move fsi_fifo_init() onto fsi_hw_startup() fsi_fifo_init() is called only from fsi_hw_startup() Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 109 ++++++++++++++++++++++++++--------------------------- 1 file changed, 54 insertions(+), 55 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 05307acb2bd..79485ed9fd5 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -692,61 +692,6 @@ static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int is_play, int enable) /* * ctrl function */ -static void fsi_fifo_init(struct fsi_priv *fsi, - int is_play, - struct device *dev) -{ - struct fsi_master *master = fsi_get_master(fsi); - struct fsi_stream *io = fsi_stream_get(fsi, is_play); - u32 shift, i; - int frame_capa; - - /* get on-chip RAM capacity */ - shift = fsi_master_read(master, FIFO_SZ); - shift >>= fsi_get_port_shift(fsi, is_play); - shift &= FIFO_SZ_MASK; - frame_capa = 256 << shift; - dev_dbg(dev, "fifo = %d words\n", frame_capa); - - /* - * The maximum number of sample data varies depending - * on the number of channels selected for the format. - * - * FIFOs are used in 4-channel units in 3-channel mode - * and in 8-channel units in 5- to 7-channel mode - * meaning that more FIFOs than the required size of DPRAM - * are used. - * - * ex) if 256 words of DP-RAM is connected - * 1 channel: 256 (256 x 1 = 256) - * 2 channels: 128 (128 x 2 = 256) - * 3 channels: 64 ( 64 x 3 = 192) - * 4 channels: 64 ( 64 x 4 = 256) - * 5 channels: 32 ( 32 x 5 = 160) - * 6 channels: 32 ( 32 x 6 = 192) - * 7 channels: 32 ( 32 x 7 = 224) - * 8 channels: 32 ( 32 x 8 = 256) - */ - for (i = 1; i < fsi->chan_num; i <<= 1) - frame_capa >>= 1; - dev_dbg(dev, "%d channel %d store\n", - fsi->chan_num, frame_capa); - - io->fifo_sample_capa = fsi_frame2sample(fsi, frame_capa); - - /* - * set interrupt generation factor - * clear FIFO - */ - if (is_play) { - fsi_reg_write(fsi, DOFF_CTL, IRQ_HALF); - fsi_reg_mask_set(fsi, DOFF_CTL, FIFO_CLR, FIFO_CLR); - } else { - fsi_reg_write(fsi, DIFF_CTL, IRQ_HALF); - fsi_reg_mask_set(fsi, DIFF_CTL, FIFO_CLR, FIFO_CLR); - } -} - static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, struct fsi_stream *io, void (*run16)(struct fsi_priv *fsi, int size), void (*run32)(struct fsi_priv *fsi, int size), @@ -867,6 +812,60 @@ static irqreturn_t fsi_interrupt(int irq, void *data) /* * dai ops */ +static void fsi_fifo_init(struct fsi_priv *fsi, + int is_play, + struct device *dev) +{ + struct fsi_master *master = fsi_get_master(fsi); + struct fsi_stream *io = fsi_stream_get(fsi, is_play); + u32 shift, i; + int frame_capa; + + /* get on-chip RAM capacity */ + shift = fsi_master_read(master, FIFO_SZ); + shift >>= fsi_get_port_shift(fsi, is_play); + shift &= FIFO_SZ_MASK; + frame_capa = 256 << shift; + dev_dbg(dev, "fifo = %d words\n", frame_capa); + + /* + * The maximum number of sample data varies depending + * on the number of channels selected for the format. + * + * FIFOs are used in 4-channel units in 3-channel mode + * and in 8-channel units in 5- to 7-channel mode + * meaning that more FIFOs than the required size of DPRAM + * are used. + * + * ex) if 256 words of DP-RAM is connected + * 1 channel: 256 (256 x 1 = 256) + * 2 channels: 128 (128 x 2 = 256) + * 3 channels: 64 ( 64 x 3 = 192) + * 4 channels: 64 ( 64 x 4 = 256) + * 5 channels: 32 ( 32 x 5 = 160) + * 6 channels: 32 ( 32 x 6 = 192) + * 7 channels: 32 ( 32 x 7 = 224) + * 8 channels: 32 ( 32 x 8 = 256) + */ + for (i = 1; i < fsi->chan_num; i <<= 1) + frame_capa >>= 1; + dev_dbg(dev, "%d channel %d store\n", + fsi->chan_num, frame_capa); + + io->fifo_sample_capa = fsi_frame2sample(fsi, frame_capa); + + /* + * set interrupt generation factor + * clear FIFO + */ + if (is_play) { + fsi_reg_write(fsi, DOFF_CTL, IRQ_HALF); + fsi_reg_mask_set(fsi, DOFF_CTL, FIFO_CLR, FIFO_CLR); + } else { + fsi_reg_write(fsi, DIFF_CTL, IRQ_HALF); + fsi_reg_mask_set(fsi, DIFF_CTL, FIFO_CLR, FIFO_CLR); + } +} static int fsi_hw_startup(struct fsi_priv *fsi, int is_play, -- cgit v1.2.3-18-g5258 From 41bba151939e21e21d18f7df005ce3a06714a69a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:51:53 -0800 Subject: ASoC: fsi: remove unnecessary parameter from fsi_hw_shutdown() This is preparation for DMAEngine support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 79485ed9fd5..a0a9c367148 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -927,7 +927,6 @@ static int fsi_hw_startup(struct fsi_priv *fsi, } static void fsi_hw_shutdown(struct fsi_priv *fsi, - int is_play, struct device *dev) { if (fsi_is_clk_master(fsi)) @@ -947,9 +946,8 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); - int is_play = fsi_is_play(substream); - fsi_hw_shutdown(fsi, is_play, dai->dev); + fsi_hw_shutdown(fsi, dai->dev); fsi->rate = 0; } @@ -1342,7 +1340,7 @@ static void __fsi_suspend(struct fsi_priv *fsi, return; fsi_port_stop(fsi, is_play); - fsi_hw_shutdown(fsi, is_play, dev); + fsi_hw_shutdown(fsi, dev); } static void __fsi_resume(struct fsi_priv *fsi, -- cgit v1.2.3-18-g5258 From 8c4152951cab90b52406afc72b62e9590bbe2d85 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:52:07 -0800 Subject: ASoC: fsi: rename fsi_stream_push/pop() to fsi_stream_init/quit() This is preparation for DMAEngine support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index a0a9c367148..2e2663bb224 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -426,7 +426,7 @@ static int fsi_stream_is_working(struct fsi_priv *fsi, return ret; } -static void fsi_stream_push(struct fsi_priv *fsi, +static void fsi_stream_init(struct fsi_priv *fsi, int is_play, struct snd_pcm_substream *substream) { @@ -447,7 +447,7 @@ static void fsi_stream_push(struct fsi_priv *fsi, spin_unlock_irqrestore(&master->lock, flags); } -static void fsi_stream_pop(struct fsi_priv *fsi, int is_play) +static void fsi_stream_quit(struct fsi_priv *fsi, int is_play) { struct fsi_stream *io = fsi_stream_get(fsi, is_play); struct snd_soc_dai *dai = fsi_get_dai(io->substream); @@ -960,13 +960,13 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - fsi_stream_push(fsi, is_play, substream); + fsi_stream_init(fsi, is_play, substream); ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); fsi_port_start(fsi, is_play); break; case SNDRV_PCM_TRIGGER_STOP: fsi_port_stop(fsi, is_play); - fsi_stream_pop(fsi, is_play); + fsi_stream_quit(fsi, is_play); break; } -- cgit v1.2.3-18-g5258 From 95b0cf05976b7d0571e283b1fcd4c32095018cd6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:52:38 -0800 Subject: ASoC: fsi: modify fsi_pio_get_area() parameter and using position This patch modify fsi_pio_get_area() parameter to use struct fsi_stream, and used it on fsi_fifo_data_ctrl(). This is just prepare cleanup for DMAEngine support. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 43 +++++++++++++++++-------------------------- 1 file changed, 17 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 2e2663bb224..c814d8a7cec 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -477,58 +477,46 @@ static void fsi_stream_quit(struct fsi_priv *fsi, int is_play) * pio function */ -static u8 *fsi_pio_get_area(struct fsi_priv *fsi, int stream) +static u8 *fsi_pio_get_area(struct fsi_priv *fsi, struct fsi_stream *io) { - int is_play = fsi_stream_is_play(stream); - struct fsi_stream *io = fsi_stream_get(fsi, is_play); struct snd_pcm_runtime *runtime = io->substream->runtime; return runtime->dma_area + samples_to_bytes(runtime, io->buff_sample_pos); } -static void fsi_pio_push16(struct fsi_priv *fsi, int num) +static void fsi_pio_push16(struct fsi_priv *fsi, u8 *_buf, int num) { - u16 *start; + u16 *start = (u16 *)_buf; int i; - start = (u16 *)fsi_pio_get_area(fsi, SNDRV_PCM_STREAM_PLAYBACK); - for (i = 0; i < num; i++) fsi_reg_write(fsi, DODT, ((u32)*(start + i) << 8)); } -static void fsi_pio_pop16(struct fsi_priv *fsi, int num) +static void fsi_pio_pop16(struct fsi_priv *fsi, u8 *_buf, int num) { - u16 *start; + u16 *start = (u16 *)_buf; int i; - start = (u16 *)fsi_pio_get_area(fsi, SNDRV_PCM_STREAM_CAPTURE); - - for (i = 0; i < num; i++) *(start + i) = (u16)(fsi_reg_read(fsi, DIDT) >> 8); } -static void fsi_pio_push32(struct fsi_priv *fsi, int num) +static void fsi_pio_push32(struct fsi_priv *fsi, u8 *_buf, int num) { - u32 *start; + u32 *start = (u32 *)_buf; int i; - start = (u32 *)fsi_pio_get_area(fsi, SNDRV_PCM_STREAM_PLAYBACK); - - for (i = 0; i < num; i++) fsi_reg_write(fsi, DODT, *(start + i)); } -static void fsi_pio_pop32(struct fsi_priv *fsi, int num) +static void fsi_pio_pop32(struct fsi_priv *fsi, u8 *_buf, int num) { - u32 *start; + u32 *start = (u32 *)_buf; int i; - start = (u32 *)fsi_pio_get_area(fsi, SNDRV_PCM_STREAM_CAPTURE); - for (i = 0; i < num; i++) *(start + i) = fsi_reg_read(fsi, DIDT); } @@ -693,12 +681,13 @@ static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int is_play, int enable) * ctrl function */ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, struct fsi_stream *io, - void (*run16)(struct fsi_priv *fsi, int size), - void (*run32)(struct fsi_priv *fsi, int size), - int samples) + void (*run16)(struct fsi_priv *fsi, u8 *buf, int samples), + void (*run32)(struct fsi_priv *fsi, u8 *buf, int samples), + int samples) { struct snd_pcm_runtime *runtime; struct snd_pcm_substream *substream; + u8 *buf; int over_period; if (!fsi || @@ -723,12 +712,14 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, struct fsi_stream *io, io->buff_sample_pos = 0; } + buf = fsi_pio_get_area(fsi, io); + switch (io->sample_width) { case 2: - run16(fsi, samples); + run16(fsi, buf, samples); break; case 4: - run32(fsi, samples); + run32(fsi, buf, samples); break; default: return -EINVAL; -- cgit v1.2.3-18-g5258 From a449e46754616a13e1bee649e37bcdf10d1b794a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:54:02 -0800 Subject: ASoC: fsi: re-define fsi_is_play() and fsi_stream_is_play() This patch re-define fsi_is_play() and fsi_stream_is_play(). fsi_data_pop/push() function keeps direct value of "is_play" at this point, but it will be removed soon. This is just prepare cleanup for DMAEngine support. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index c814d8a7cec..1cbe474046f 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -296,6 +296,11 @@ static int fsi_is_spdif(struct fsi_priv *fsi) return fsi->spdif; } +static int fsi_is_play(struct snd_pcm_substream *substream) +{ + return substream->stream == SNDRV_PCM_STREAM_PLAYBACK; +} + static struct snd_soc_dai *fsi_get_dai(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -399,10 +404,10 @@ static void fsi_count_fifo_err(struct fsi_priv *fsi) /* * fsi_stream_xx() function */ -#define fsi_is_play(substream) fsi_stream_is_play(substream->stream) -static inline int fsi_stream_is_play(int stream) +static inline int fsi_stream_is_play(struct fsi_priv *fsi, + struct fsi_stream *io) { - return stream == SNDRV_PCM_STREAM_PLAYBACK; + return &fsi->playback == io; } static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi, @@ -736,7 +741,7 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, struct fsi_stream *io, static int fsi_data_pop(struct fsi_priv *fsi) { - int is_play = fsi_stream_is_play(SNDRV_PCM_STREAM_CAPTURE); + int is_play = 0; int sample_residues; /* samples in FSI fifo */ int sample_space; /* ALSA free samples space */ int samples; @@ -755,7 +760,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) static int fsi_data_push(struct fsi_priv *fsi) { - int is_play = fsi_stream_is_play(SNDRV_PCM_STREAM_PLAYBACK); + int is_play = 1; int sample_residues; /* ALSA residue samples */ int sample_space; /* FSI fifo free samples space */ int samples; -- cgit v1.2.3-18-g5258 From 7b1b3331e65e47b6abb32be0a3db46bcf423145a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:55:26 -0800 Subject: ASoC: fsi: use fsi_stream in fsi_get_current_fifo_samples() parameter fsi_get_current_fifo_samples() uses fsi_stream instead of is_play. This is just prepare cleanup for DMAEngine support. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 1cbe474046f..24dbe165eda 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -210,6 +210,8 @@ struct fsi_master { spinlock_t lock; }; +static int fsi_stream_is_play(struct fsi_priv *fsi, struct fsi_stream *io); + /* * basic read write function */ @@ -366,8 +368,10 @@ static int fsi_sample2frame(struct fsi_priv *fsi, int samples) return samples / fsi->chan_num; } -static int fsi_get_current_fifo_samples(struct fsi_priv *fsi, int is_play) +static int fsi_get_current_fifo_samples(struct fsi_priv *fsi, + struct fsi_stream *io) { + int is_play = fsi_stream_is_play(fsi, io); u32 status; int frames; @@ -747,7 +751,7 @@ static int fsi_data_pop(struct fsi_priv *fsi) int samples; struct fsi_stream *io = fsi_stream_get(fsi, is_play); - sample_residues = fsi_get_current_fifo_samples(fsi, is_play); + sample_residues = fsi_get_current_fifo_samples(fsi, io); sample_space = io->buff_sample_capa - io->buff_sample_pos; samples = min(sample_residues, sample_space); @@ -768,7 +772,7 @@ static int fsi_data_push(struct fsi_priv *fsi) sample_residues = io->buff_sample_capa - io->buff_sample_pos; sample_space = io->fifo_sample_capa - - fsi_get_current_fifo_samples(fsi, is_play); + fsi_get_current_fifo_samples(fsi, io); samples = min(sample_residues, sample_space); -- cgit v1.2.3-18-g5258 From 5e97313ac483f03a9af661aada356980fe310e0d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:55:55 -0800 Subject: ASoC: fsi: add fsi_stream_handler and PIO handler This patch adds struct fsi_stream_handler and defined fsi_pio_push/pop_handler. these are controled by fsi_steam_xxx() function. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 133 +++++++++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 118 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 24dbe165eda..b02886ad6f8 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -159,18 +159,27 @@ typedef int (*set_rate_func)(struct device *dev, int is_porta, int rate, int ena * struct */ +struct fsi_stream_handler; struct fsi_stream { - struct snd_pcm_substream *substream; + /* + * these are initialized by fsi_stream_init() + */ + struct snd_pcm_substream *substream; int fifo_sample_capa; /* sample capacity of FSI FIFO */ int buff_sample_capa; /* sample capacity of ALSA buffer */ int buff_sample_pos; /* sample position of ALSA buffer */ int period_samples; /* sample number / 1 period */ int period_pos; /* current period position */ int sample_width; /* sample width */ - int uerr_num; int oerr_num; + + /* + * thse are initialized by fsi_handler_init() + */ + struct fsi_stream_handler *handler; + struct fsi_priv *priv; }; struct fsi_priv { @@ -190,6 +199,16 @@ struct fsi_priv { long rate; }; +struct fsi_stream_handler { + int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io); + int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); + int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); +}; +#define fsi_stream_handler_call(io, func, args...) \ + (!(io) ? -ENODEV : \ + !((io)->handler->func) ? 0 : \ + (io)->handler->func(args)) + struct fsi_core { int ver; @@ -435,6 +454,11 @@ static int fsi_stream_is_working(struct fsi_priv *fsi, return ret; } +static struct fsi_priv *fsi_stream_to_priv(struct fsi_stream *io) +{ + return io->priv; +} + static void fsi_stream_init(struct fsi_priv *fsi, int is_play, struct snd_pcm_substream *substream) @@ -482,6 +506,53 @@ static void fsi_stream_quit(struct fsi_priv *fsi, int is_play) spin_unlock_irqrestore(&master->lock, flags); } +static int fsi_stream_transfer(struct fsi_stream *io) +{ + struct fsi_priv *fsi = fsi_stream_to_priv(io); + if (!fsi) + return -EIO; + + return fsi_stream_handler_call(io, transfer, fsi, io); +} + +static int fsi_stream_probe(struct fsi_priv *fsi) +{ + struct fsi_stream *io; + int ret1, ret2; + + io = &fsi->playback; + ret1 = fsi_stream_handler_call(io, probe, fsi, io); + + io = &fsi->capture; + ret2 = fsi_stream_handler_call(io, probe, fsi, io); + + if (ret1 < 0) + return ret1; + if (ret2 < 0) + return ret2; + + return 0; +} + +static int fsi_stream_remove(struct fsi_priv *fsi) +{ + struct fsi_stream *io; + int ret1, ret2; + + io = &fsi->playback; + ret1 = fsi_stream_handler_call(io, remove, fsi, io); + + io = &fsi->capture; + ret2 = fsi_stream_handler_call(io, remove, fsi, io); + + if (ret1 < 0) + return ret1; + if (ret2 < 0) + return ret2; + + return 0; +} + /* * pio function */ @@ -743,13 +814,11 @@ static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, struct fsi_stream *io, return 0; } -static int fsi_data_pop(struct fsi_priv *fsi) +static int fsi_pio_pop(struct fsi_priv *fsi, struct fsi_stream *io) { - int is_play = 0; int sample_residues; /* samples in FSI fifo */ int sample_space; /* ALSA free samples space */ int samples; - struct fsi_stream *io = fsi_stream_get(fsi, is_play); sample_residues = fsi_get_current_fifo_samples(fsi, io); sample_space = io->buff_sample_capa - io->buff_sample_pos; @@ -762,13 +831,11 @@ static int fsi_data_pop(struct fsi_priv *fsi) samples); } -static int fsi_data_push(struct fsi_priv *fsi) +static int fsi_pio_push(struct fsi_priv *fsi, struct fsi_stream *io) { - int is_play = 1; int sample_residues; /* ALSA residue samples */ int sample_space; /* FSI fifo free samples space */ int samples; - struct fsi_stream *io = fsi_stream_get(fsi, is_play); sample_residues = io->buff_sample_capa - io->buff_sample_pos; sample_space = io->fifo_sample_capa - @@ -782,6 +849,14 @@ static int fsi_data_push(struct fsi_priv *fsi) samples); } +static struct fsi_stream_handler fsi_pio_push_handler = { + .transfer = fsi_pio_push, +}; + +static struct fsi_stream_handler fsi_pio_pop_handler = { + .transfer = fsi_pio_pop, +}; + static irqreturn_t fsi_interrupt(int irq, void *data) { struct fsi_master *master = data; @@ -792,13 +867,13 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_master_mask_set(master, SOFT_RST, IR, IR); if (int_st & AB_IO(1, AO_SHIFT)) - fsi_data_push(&master->fsia); + fsi_stream_transfer(&master->fsia.playback); if (int_st & AB_IO(1, BO_SHIFT)) - fsi_data_push(&master->fsib); + fsi_stream_transfer(&master->fsib.playback); if (int_st & AB_IO(1, AI_SHIFT)) - fsi_data_pop(&master->fsia); + fsi_stream_transfer(&master->fsia.capture); if (int_st & AB_IO(1, BI_SHIFT)) - fsi_data_pop(&master->fsib); + fsi_stream_transfer(&master->fsib.capture); fsi_count_fifo_err(&master->fsia); fsi_count_fifo_err(&master->fsib); @@ -955,14 +1030,16 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); + struct fsi_stream *io = fsi_stream_get(fsi, fsi_is_play(substream)); int is_play = fsi_is_play(substream); int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: fsi_stream_init(fsi, is_play, substream); - ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); - fsi_port_start(fsi, is_play); + ret = fsi_stream_transfer(io); + if (0 == ret) + fsi_port_start(fsi, is_play); break; case SNDRV_PCM_TRIGGER_STOP: fsi_port_stop(fsi, is_play); @@ -1224,6 +1301,13 @@ static struct snd_soc_platform_driver fsi_soc_platform = { /* * platform function */ +static void fsi_handler_init(struct fsi_priv *fsi) +{ + fsi->playback.handler = &fsi_pio_push_handler; /* default PIO */ + fsi->playback.priv = fsi; + fsi->capture.handler = &fsi_pio_pop_handler; /* default PIO */ + fsi->capture.priv = fsi; +} static int fsi_probe(struct platform_device *pdev) { @@ -1270,10 +1354,22 @@ static int fsi_probe(struct platform_device *pdev) /* FSI A setting */ master->fsia.base = master->base; master->fsia.master = master; + fsi_handler_init(&master->fsia); + ret = fsi_stream_probe(&master->fsia); + if (ret < 0) { + dev_err(&pdev->dev, "FSIA stream probe failed\n"); + goto exit_iounmap; + } /* FSI B setting */ master->fsib.base = master->base + 0x40; master->fsib.master = master; + fsi_handler_init(&master->fsib); + ret = fsi_stream_probe(&master->fsib); + if (ret < 0) { + dev_err(&pdev->dev, "FSIB stream probe failed\n"); + goto exit_fsia; + } pm_runtime_enable(&pdev->dev); dev_set_drvdata(&pdev->dev, master); @@ -1282,7 +1378,7 @@ static int fsi_probe(struct platform_device *pdev) id_entry->name, master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); - goto exit_iounmap; + goto exit_fsib; } ret = snd_soc_register_platform(&pdev->dev, &fsi_soc_platform); @@ -1304,6 +1400,10 @@ exit_snd_soc: snd_soc_unregister_platform(&pdev->dev); exit_free_irq: free_irq(irq, master); +exit_fsib: + fsi_stream_remove(&master->fsib); +exit_fsia: + fsi_stream_remove(&master->fsia); exit_iounmap: iounmap(master->base); pm_runtime_disable(&pdev->dev); @@ -1326,6 +1426,9 @@ static int fsi_remove(struct platform_device *pdev) snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&pdev->dev); + fsi_stream_remove(&master->fsia); + fsi_stream_remove(&master->fsib); + iounmap(master->base); kfree(master); -- cgit v1.2.3-18-g5258 From 1b0ca1a0c056c7c97b18e363f939f0635ca093af Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:56:27 -0800 Subject: ASoC: fsi: tidyup: fsi_pio_xxx() are gathered Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 101 ++++++++++++++++++++++++++--------------------------- 1 file changed, 49 insertions(+), 52 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index b02886ad6f8..7c93b7c2fdb 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -553,54 +553,6 @@ static int fsi_stream_remove(struct fsi_priv *fsi) return 0; } -/* - * pio function - */ - -static u8 *fsi_pio_get_area(struct fsi_priv *fsi, struct fsi_stream *io) -{ - struct snd_pcm_runtime *runtime = io->substream->runtime; - - return runtime->dma_area + - samples_to_bytes(runtime, io->buff_sample_pos); -} - -static void fsi_pio_push16(struct fsi_priv *fsi, u8 *_buf, int num) -{ - u16 *start = (u16 *)_buf; - int i; - - for (i = 0; i < num; i++) - fsi_reg_write(fsi, DODT, ((u32)*(start + i) << 8)); -} - -static void fsi_pio_pop16(struct fsi_priv *fsi, u8 *_buf, int num) -{ - u16 *start = (u16 *)_buf; - int i; - - for (i = 0; i < num; i++) - *(start + i) = (u16)(fsi_reg_read(fsi, DIDT) >> 8); -} - -static void fsi_pio_push32(struct fsi_priv *fsi, u8 *_buf, int num) -{ - u32 *start = (u32 *)_buf; - int i; - - for (i = 0; i < num; i++) - fsi_reg_write(fsi, DODT, *(start + i)); -} - -static void fsi_pio_pop32(struct fsi_priv *fsi, u8 *_buf, int num) -{ - u32 *start = (u32 *)_buf; - int i; - - for (i = 0; i < num; i++) - *(start + i) = fsi_reg_read(fsi, DIDT); -} - /* * irq function */ @@ -757,10 +709,55 @@ static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int is_play, int enable) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } + /* - * ctrl function + * pio data transfer handler */ -static int fsi_fifo_data_ctrl(struct fsi_priv *fsi, struct fsi_stream *io, +static void fsi_pio_push16(struct fsi_priv *fsi, u8 *_buf, int samples) +{ + u16 *buf = (u16 *)_buf; + int i; + + for (i = 0; i < samples; i++) + fsi_reg_write(fsi, DODT, ((u32)*(buf + i) << 8)); +} + +static void fsi_pio_pop16(struct fsi_priv *fsi, u8 *_buf, int samples) +{ + u16 *buf = (u16 *)_buf; + int i; + + for (i = 0; i < samples; i++) + *(buf + i) = (u16)(fsi_reg_read(fsi, DIDT) >> 8); +} + +static void fsi_pio_push32(struct fsi_priv *fsi, u8 *_buf, int samples) +{ + u32 *buf = (u32 *)_buf; + int i; + + for (i = 0; i < samples; i++) + fsi_reg_write(fsi, DODT, *(buf + i)); +} + +static void fsi_pio_pop32(struct fsi_priv *fsi, u8 *_buf, int samples) +{ + u32 *buf = (u32 *)_buf; + int i; + + for (i = 0; i < samples; i++) + *(buf + i) = fsi_reg_read(fsi, DIDT); +} + +static u8 *fsi_pio_get_area(struct fsi_priv *fsi, struct fsi_stream *io) +{ + struct snd_pcm_runtime *runtime = io->substream->runtime; + + return runtime->dma_area + + samples_to_bytes(runtime, io->buff_sample_pos); +} + +static int fsi_pio_transfer(struct fsi_priv *fsi, struct fsi_stream *io, void (*run16)(struct fsi_priv *fsi, u8 *buf, int samples), void (*run32)(struct fsi_priv *fsi, u8 *buf, int samples), int samples) @@ -825,7 +822,7 @@ static int fsi_pio_pop(struct fsi_priv *fsi, struct fsi_stream *io) samples = min(sample_residues, sample_space); - return fsi_fifo_data_ctrl(fsi, io, + return fsi_pio_transfer(fsi, io, fsi_pio_pop16, fsi_pio_pop32, samples); @@ -843,7 +840,7 @@ static int fsi_pio_push(struct fsi_priv *fsi, struct fsi_stream *io) samples = min(sample_residues, sample_space); - return fsi_fifo_data_ctrl(fsi, io, + return fsi_pio_transfer(fsi, io, fsi_pio_push16, fsi_pio_push32, samples); -- cgit v1.2.3-18-g5258 From 938e2a8da5b2c1cb21c200e97736259948a3d12c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:56:57 -0800 Subject: ASoC: fsi: don't use is_play as a parameter of fsi functions is_play should be kept as local valuable. it prepare cleanup for DMAEngine support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 94 ++++++++++++++++++++++++++---------------------------- 1 file changed, 45 insertions(+), 49 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 7c93b7c2fdb..7dec144b846 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -364,8 +364,9 @@ static u32 fsi_get_info_flags(struct fsi_priv *fsi) master->info->portb_flags; } -static u32 fsi_get_port_shift(struct fsi_priv *fsi, int is_play) +static u32 fsi_get_port_shift(struct fsi_priv *fsi, struct fsi_stream *io) { + int is_play = fsi_stream_is_play(fsi, io); int is_porta = fsi_is_port_a(fsi); u32 shift; @@ -434,15 +435,14 @@ static inline int fsi_stream_is_play(struct fsi_priv *fsi, } static inline struct fsi_stream *fsi_stream_get(struct fsi_priv *fsi, - int is_play) + struct snd_pcm_substream *substream) { - return is_play ? &fsi->playback : &fsi->capture; + return fsi_is_play(substream) ? &fsi->playback : &fsi->capture; } static int fsi_stream_is_working(struct fsi_priv *fsi, - int is_play) + struct fsi_stream *io) { - struct fsi_stream *io = fsi_stream_get(fsi, is_play); struct fsi_master *master = fsi_get_master(fsi); unsigned long flags; int ret; @@ -460,10 +460,9 @@ static struct fsi_priv *fsi_stream_to_priv(struct fsi_stream *io) } static void fsi_stream_init(struct fsi_priv *fsi, - int is_play, + struct fsi_stream *io, struct snd_pcm_substream *substream) { - struct fsi_stream *io = fsi_stream_get(fsi, is_play); struct snd_pcm_runtime *runtime = substream->runtime; struct fsi_master *master = fsi_get_master(fsi); unsigned long flags; @@ -480,9 +479,8 @@ static void fsi_stream_init(struct fsi_priv *fsi, spin_unlock_irqrestore(&master->lock, flags); } -static void fsi_stream_quit(struct fsi_priv *fsi, int is_play) +static void fsi_stream_quit(struct fsi_priv *fsi, struct fsi_stream *io) { - struct fsi_stream *io = fsi_stream_get(fsi, is_play); struct snd_soc_dai *dai = fsi_get_dai(io->substream); struct fsi_master *master = fsi_get_master(fsi); unsigned long flags; @@ -557,18 +555,18 @@ static int fsi_stream_remove(struct fsi_priv *fsi) * irq function */ -static void fsi_irq_enable(struct fsi_priv *fsi, int is_play) +static void fsi_irq_enable(struct fsi_priv *fsi, struct fsi_stream *io) { - u32 data = AB_IO(1, fsi_get_port_shift(fsi, is_play)); + u32 data = AB_IO(1, fsi_get_port_shift(fsi, io)); struct fsi_master *master = fsi_get_master(fsi); fsi_core_mask_set(master, imsk, data, data); fsi_core_mask_set(master, iemsk, data, data); } -static void fsi_irq_disable(struct fsi_priv *fsi, int is_play) +static void fsi_irq_disable(struct fsi_priv *fsi, struct fsi_stream *io) { - u32 data = AB_IO(1, fsi_get_port_shift(fsi, is_play)); + u32 data = AB_IO(1, fsi_get_port_shift(fsi, io)); struct fsi_master *master = fsi_get_master(fsi); fsi_core_mask_set(master, imsk, data, 0); @@ -585,8 +583,8 @@ static void fsi_irq_clear_status(struct fsi_priv *fsi) u32 data = 0; struct fsi_master *master = fsi_get_master(fsi); - data |= AB_IO(1, fsi_get_port_shift(fsi, 0)); - data |= AB_IO(1, fsi_get_port_shift(fsi, 1)); + data |= AB_IO(1, fsi_get_port_shift(fsi, &fsi->playback)); + data |= AB_IO(1, fsi_get_port_shift(fsi, &fsi->capture)); /* clear interrupt factor */ fsi_core_mask_set(master, int_st, data, 0); @@ -695,15 +693,16 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, #define fsi_port_start(f, i) __fsi_port_clk_ctrl(f, i, 1) #define fsi_port_stop(f, i) __fsi_port_clk_ctrl(f, i, 0) -static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, int is_play, int enable) +static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, struct fsi_stream *io, + int enable) { struct fsi_master *master = fsi_get_master(fsi); u32 clk = fsi_is_port_a(fsi) ? CRA : CRB; if (enable) - fsi_irq_enable(fsi, is_play); + fsi_irq_enable(fsi, io); else - fsi_irq_disable(fsi, is_play); + fsi_irq_disable(fsi, io); if (fsi_is_clk_master(fsi)) fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); @@ -885,17 +884,17 @@ static irqreturn_t fsi_interrupt(int irq, void *data) * dai ops */ static void fsi_fifo_init(struct fsi_priv *fsi, - int is_play, + struct fsi_stream *io, struct device *dev) { struct fsi_master *master = fsi_get_master(fsi); - struct fsi_stream *io = fsi_stream_get(fsi, is_play); + int is_play = fsi_stream_is_play(fsi, io); u32 shift, i; int frame_capa; /* get on-chip RAM capacity */ shift = fsi_master_read(master, FIFO_SZ); - shift >>= fsi_get_port_shift(fsi, is_play); + shift >>= fsi_get_port_shift(fsi, io); shift &= FIFO_SZ_MASK; frame_capa = 256 << shift; dev_dbg(dev, "fifo = %d words\n", frame_capa); @@ -940,7 +939,7 @@ static void fsi_fifo_init(struct fsi_priv *fsi, } static int fsi_hw_startup(struct fsi_priv *fsi, - int is_play, + struct fsi_stream *io, struct device *dev) { struct fsi_master *master = fsi_get_master(fsi); @@ -989,11 +988,11 @@ static int fsi_hw_startup(struct fsi_priv *fsi, } /* irq clear */ - fsi_irq_disable(fsi, is_play); + fsi_irq_disable(fsi, io); fsi_irq_clear_status(fsi); /* fifo init */ - fsi_fifo_init(fsi, is_play, dev); + fsi_fifo_init(fsi, io, dev); return 0; } @@ -1009,9 +1008,8 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); - int is_play = fsi_is_play(substream); - return fsi_hw_startup(fsi, is_play, dai->dev); + return fsi_hw_startup(fsi, fsi_stream_get(fsi, substream), dai->dev); } static void fsi_dai_shutdown(struct snd_pcm_substream *substream, @@ -1027,20 +1025,19 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct fsi_priv *fsi = fsi_get_priv(substream); - struct fsi_stream *io = fsi_stream_get(fsi, fsi_is_play(substream)); - int is_play = fsi_is_play(substream); + struct fsi_stream *io = fsi_stream_get(fsi, substream); int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: - fsi_stream_init(fsi, is_play, substream); + fsi_stream_init(fsi, io, substream); ret = fsi_stream_transfer(io); if (0 == ret) - fsi_port_start(fsi, is_play); + fsi_port_start(fsi, io); break; case SNDRV_PCM_TRIGGER_STOP: - fsi_port_stop(fsi, is_play); - fsi_stream_quit(fsi, is_play); + fsi_port_stop(fsi, io); + fsi_stream_quit(fsi, io); break; } @@ -1206,7 +1203,7 @@ static int fsi_hw_free(struct snd_pcm_substream *substream) static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct fsi_priv *fsi = fsi_get_priv(substream); - struct fsi_stream *io = fsi_stream_get(fsi, fsi_is_play(substream)); + struct fsi_stream *io = fsi_stream_get(fsi, substream); int samples_pos = io->buff_sample_pos - 1; if (samples_pos < 0) @@ -1433,30 +1430,29 @@ static int fsi_remove(struct platform_device *pdev) } static void __fsi_suspend(struct fsi_priv *fsi, - int is_play, + struct fsi_stream *io, struct device *dev) { - if (!fsi_stream_is_working(fsi, is_play)) + if (!fsi_stream_is_working(fsi, io)) return; - fsi_port_stop(fsi, is_play); + fsi_port_stop(fsi, io); fsi_hw_shutdown(fsi, dev); } static void __fsi_resume(struct fsi_priv *fsi, - int is_play, + struct fsi_stream *io, struct device *dev) { - if (!fsi_stream_is_working(fsi, is_play)) + if (!fsi_stream_is_working(fsi, io)) return; - fsi_hw_startup(fsi, is_play, dev); + fsi_hw_startup(fsi, io, dev); if (fsi_is_clk_master(fsi) && fsi->rate) fsi_set_master_clk(dev, fsi, fsi->rate, 1); - fsi_port_start(fsi, is_play); - + fsi_port_start(fsi, io); } static int fsi_suspend(struct device *dev) @@ -1465,11 +1461,11 @@ static int fsi_suspend(struct device *dev) struct fsi_priv *fsia = &master->fsia; struct fsi_priv *fsib = &master->fsib; - __fsi_suspend(fsia, 1, dev); - __fsi_suspend(fsia, 0, dev); + __fsi_suspend(fsia, &fsia->playback, dev); + __fsi_suspend(fsia, &fsia->capture, dev); - __fsi_suspend(fsib, 1, dev); - __fsi_suspend(fsib, 0, dev); + __fsi_suspend(fsib, &fsib->playback, dev); + __fsi_suspend(fsib, &fsib->capture, dev); return 0; } @@ -1480,11 +1476,11 @@ static int fsi_resume(struct device *dev) struct fsi_priv *fsia = &master->fsia; struct fsi_priv *fsib = &master->fsib; - __fsi_resume(fsia, 1, dev); - __fsi_resume(fsia, 0, dev); + __fsi_resume(fsia, &fsia->playback, dev); + __fsi_resume(fsia, &fsia->capture, dev); - __fsi_resume(fsib, 1, dev); - __fsi_resume(fsib, 0, dev); + __fsi_resume(fsib, &fsib->playback, dev); + __fsi_resume(fsib, &fsib->capture, dev); return 0; } -- cgit v1.2.3-18-g5258 From 180346ede352b12c72c5aeba2fc806fd32880c16 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:57:25 -0800 Subject: ASoC: fsi: add .start_stop handler to fsi_stream_handler Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 51 +++++++++++++++++++++++++++++---------------------- 1 file changed, 29 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 7dec144b846..8d05e59c883 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -203,6 +203,8 @@ struct fsi_stream_handler { int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); + void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, + int enable); }; #define fsi_stream_handler_call(io, func, args...) \ (!(io) ? -ENODEV : \ @@ -513,6 +515,12 @@ static int fsi_stream_transfer(struct fsi_stream *io) return fsi_stream_handler_call(io, transfer, fsi, io); } +#define fsi_stream_start(fsi, io)\ + fsi_stream_handler_call(io, start_stop, fsi, io, 1) + +#define fsi_stream_stop(fsi, io)\ + fsi_stream_handler_call(io, start_stop, fsi, io, 0) + static int fsi_stream_probe(struct fsi_priv *fsi) { struct fsi_stream *io; @@ -691,24 +699,6 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, return ret; } -#define fsi_port_start(f, i) __fsi_port_clk_ctrl(f, i, 1) -#define fsi_port_stop(f, i) __fsi_port_clk_ctrl(f, i, 0) -static void __fsi_port_clk_ctrl(struct fsi_priv *fsi, struct fsi_stream *io, - int enable) -{ - struct fsi_master *master = fsi_get_master(fsi); - u32 clk = fsi_is_port_a(fsi) ? CRA : CRB; - - if (enable) - fsi_irq_enable(fsi, io); - else - fsi_irq_disable(fsi, io); - - if (fsi_is_clk_master(fsi)) - fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); -} - - /* * pio data transfer handler */ @@ -845,12 +835,29 @@ static int fsi_pio_push(struct fsi_priv *fsi, struct fsi_stream *io) samples); } +static void fsi_pio_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, + int enable) +{ + struct fsi_master *master = fsi_get_master(fsi); + u32 clk = fsi_is_port_a(fsi) ? CRA : CRB; + + if (enable) + fsi_irq_enable(fsi, io); + else + fsi_irq_disable(fsi, io); + + if (fsi_is_clk_master(fsi)) + fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); +} + static struct fsi_stream_handler fsi_pio_push_handler = { .transfer = fsi_pio_push, + .start_stop = fsi_pio_start_stop, }; static struct fsi_stream_handler fsi_pio_pop_handler = { .transfer = fsi_pio_pop, + .start_stop = fsi_pio_start_stop, }; static irqreturn_t fsi_interrupt(int irq, void *data) @@ -1033,10 +1040,10 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, fsi_stream_init(fsi, io, substream); ret = fsi_stream_transfer(io); if (0 == ret) - fsi_port_start(fsi, io); + fsi_stream_start(fsi, io); break; case SNDRV_PCM_TRIGGER_STOP: - fsi_port_stop(fsi, io); + fsi_stream_stop(fsi, io); fsi_stream_quit(fsi, io); break; } @@ -1436,7 +1443,7 @@ static void __fsi_suspend(struct fsi_priv *fsi, if (!fsi_stream_is_working(fsi, io)) return; - fsi_port_stop(fsi, io); + fsi_stream_stop(fsi, io); fsi_hw_shutdown(fsi, dev); } @@ -1452,7 +1459,7 @@ static void __fsi_resume(struct fsi_priv *fsi, if (fsi_is_clk_master(fsi) && fsi->rate) fsi_set_master_clk(dev, fsi, fsi->rate, 1); - fsi_port_start(fsi, io); + fsi_stream_start(fsi, io); } static int fsi_suspend(struct device *dev) -- cgit v1.2.3-18-g5258 From 97df81873e9c1391319dd818bc4b6856517e4939 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:57:40 -0800 Subject: ASoC: fsi: fsi_stream_is_working() care substream->runtime Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 8d05e59c883..1e10184af89 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -450,7 +450,7 @@ static int fsi_stream_is_working(struct fsi_priv *fsi, int ret; spin_lock_irqsave(&master->lock, flags); - ret = !!io->substream; + ret = !!(io->substream && io->substream->runtime); spin_unlock_irqrestore(&master->lock, flags); return ret; @@ -756,9 +756,7 @@ static int fsi_pio_transfer(struct fsi_priv *fsi, struct fsi_stream *io, u8 *buf; int over_period; - if (!fsi || - !io->substream || - !io->substream->runtime) + if (!fsi_stream_is_working(fsi, io)) return -EINVAL; over_period = 0; -- cgit v1.2.3-18-g5258 From fec691e73bf20e1c8e6ecd8e3725e4745bec4e21 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:58:48 -0800 Subject: ASoC: fsi: PortA/B information was controlled by sh_fsi_port_info Current FSI got each PortA/B parameter by porta_flags/portb_flags from platform. And .set_rate function was shared for PortA/B. This structure was not readable and not flexible. This patch adds sh_fsi_port_info, and its own settings was added on each platform. it is preparation for DMAEngine support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 1e10184af89..75d0cda4bad 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -116,7 +116,7 @@ #define FSI_FMTS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE) -typedef int (*set_rate_func)(struct device *dev, int is_porta, int rate, int enable); +typedef int (*set_rate_func)(struct device *dev, int rate, int enable); /* * FSI driver use below type name for variable @@ -185,6 +185,7 @@ struct fsi_stream { struct fsi_priv { void __iomem *base; struct fsi_master *master; + struct sh_fsi_port_info *info; struct fsi_stream playback; struct fsi_stream capture; @@ -227,7 +228,6 @@ struct fsi_master { struct fsi_priv fsia; struct fsi_priv fsib; struct fsi_core *core; - struct sh_fsi_platform_info *info; spinlock_t lock; }; @@ -346,24 +346,20 @@ static struct fsi_priv *fsi_get_priv(struct snd_pcm_substream *substream) return fsi_get_priv_frm_dai(fsi_get_dai(substream)); } -static set_rate_func fsi_get_info_set_rate(struct fsi_master *master) +static set_rate_func fsi_get_info_set_rate(struct fsi_priv *fsi) { - if (!master->info) + if (!fsi->info) return NULL; - return master->info->set_rate; + return fsi->info->set_rate; } static u32 fsi_get_info_flags(struct fsi_priv *fsi) { - int is_porta = fsi_is_port_a(fsi); - struct fsi_master *master = fsi_get_master(fsi); - - if (!master->info) + if (!fsi->info) return 0; - return is_porta ? master->info->porta_flags : - master->info->portb_flags; + return fsi->info->flags; } static u32 fsi_get_port_shift(struct fsi_priv *fsi, struct fsi_stream *io) @@ -628,11 +624,14 @@ static int fsi_set_master_clk(struct device *dev, struct fsi_priv *fsi, long rate, int enable) { struct fsi_master *master = fsi_get_master(fsi); - set_rate_func set_rate = fsi_get_info_set_rate(master); + set_rate_func set_rate = fsi_get_info_set_rate(fsi); int fsi_ver = master->core->ver; int ret; - ret = set_rate(dev, fsi_is_port_a(fsi), rate, enable); + if (!set_rate) + return 0; + + ret = set_rate(dev, rate, enable); if (ret < 0) /* error */ return ret; @@ -1093,8 +1092,7 @@ static int fsi_set_fmt_spdif(struct fsi_priv *fsi) static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct fsi_priv *fsi = fsi_get_priv_frm_dai(dai); - struct fsi_master *master = fsi_get_master(fsi); - set_rate_func set_rate = fsi_get_info_set_rate(master); + set_rate_func set_rate = fsi_get_info_set_rate(fsi); u32 flags = fsi_get_info_flags(fsi); int ret; @@ -1312,6 +1310,7 @@ static int fsi_probe(struct platform_device *pdev) { struct fsi_master *master; const struct platform_device_id *id_entry; + struct sh_fsi_platform_info *info = pdev->dev.platform_data; struct resource *res; unsigned int irq; int ret; @@ -1346,13 +1345,13 @@ static int fsi_probe(struct platform_device *pdev) /* master setting */ master->irq = irq; - master->info = pdev->dev.platform_data; master->core = (struct fsi_core *)id_entry->driver_data; spin_lock_init(&master->lock); /* FSI A setting */ master->fsia.base = master->base; master->fsia.master = master; + master->fsia.info = &info->port_a; fsi_handler_init(&master->fsia); ret = fsi_stream_probe(&master->fsia); if (ret < 0) { @@ -1363,6 +1362,7 @@ static int fsi_probe(struct platform_device *pdev) /* FSI B setting */ master->fsib.base = master->base + 0x40; master->fsib.master = master; + master->fsib.info = &info->port_b; fsi_handler_init(&master->fsib); ret = fsi_stream_probe(&master->fsib); if (ret < 0) { -- cgit v1.2.3-18-g5258 From 83344027cacf1944fe180907fa98ee4116ef33ea Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:59:02 -0800 Subject: ASoC: fsi: add .init/.quit handler support This is preparation for DMAEngine support Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 75d0cda4bad..79a0afb7872 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -201,6 +201,8 @@ struct fsi_priv { }; struct fsi_stream_handler { + int (*init)(struct fsi_priv *fsi, struct fsi_stream *io); + int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io); int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); @@ -474,6 +476,7 @@ static void fsi_stream_init(struct fsi_priv *fsi, io->sample_width = samples_to_bytes(runtime, 1); io->oerr_num = -1; /* ignore 1st err */ io->uerr_num = -1; /* ignore 1st err */ + fsi_stream_handler_call(io, init, fsi, io); spin_unlock_irqrestore(&master->lock, flags); } @@ -491,6 +494,7 @@ static void fsi_stream_quit(struct fsi_priv *fsi, struct fsi_stream *io) if (io->uerr_num > 0) dev_err(dai->dev, "under_run = %d\n", io->uerr_num); + fsi_stream_handler_call(io, quit, fsi, io); io->substream = NULL; io->buff_sample_capa = 0; io->buff_sample_pos = 0; -- cgit v1.2.3-18-g5258 From 9322ca549771f2e84a93ac3f509ade1e4c3cdb35 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Feb 2012 14:28:01 +0100 Subject: ALSA: hda - Add suffix argument to snd_hda_add_vmaster() In most cases, the slave strings for vmaster are identical between volumes and switches except for "xxx Volume" and "xxx Switch" suffix. Now snd_hda_add_vmaster() takes the optional suffix argument so that each string can be composed with the given suffix, and we can share the slave name strings in both volume and switch calls nicely. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 22 ++++++++++---- sound/pci/hda/hda_local.h | 3 +- sound/pci/hda/patch_analog.c | 66 +++++++++--------------------------------- sound/pci/hda/patch_conexant.c | 23 ++++----------- sound/pci/hda/patch_realtek.c | 40 +++++-------------------- sound/pci/hda/patch_sigmatel.c | 29 +++++-------------- sound/pci/hda/patch_via.c | 28 +++++------------- 7 files changed, 59 insertions(+), 152 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c2c65f63bf0..8a2f9dddbf0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2300,7 +2300,7 @@ typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *); /* apply the function to all matching slave ctls in the mixer list */ static int map_slaves(struct hda_codec *codec, const char * const *slaves, - map_slave_func_t func, void *data) + const char *suffix, map_slave_func_t func, void *data) { struct hda_nid_item *items; const char * const *s; @@ -2313,7 +2313,15 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves, sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER) continue; for (s = slaves; *s; s++) { - if (!strcmp(sctl->id.name, *s)) { + char tmpname[sizeof(sctl->id.name)]; + const char *name = *s; + if (suffix) { + snprintf(tmpname, sizeof(tmpname), "%s %s", + name, suffix); + name = tmpname; + } + printk("XXX comparing %s vs %s\n", sctl->id.name, name); + if (!strcmp(sctl->id.name, name)) { err = func(data, sctl); if (err) return err; @@ -2335,6 +2343,7 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) * @name: vmaster control name * @tlv: TLV data (optional) * @slaves: slave control names (optional) + * @suffix: suffix string to each slave name (optional) * * Create a virtual master control with the given name. The TLV data * must be either NULL or a valid data. @@ -2346,12 +2355,13 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) * This function returns zero if successful or a negative error code. */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char * const *slaves) + unsigned int *tlv, const char * const *slaves, + const char *suffix) { struct snd_kcontrol *kctl; int err; - err = map_slaves(codec, slaves, check_slave_present, NULL); + err = map_slaves(codec, slaves, suffix, check_slave_present, NULL); if (err != 1) { snd_printdd("No slave found for %s\n", name); return 0; @@ -2363,8 +2373,8 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, if (err < 0) return err; - err = map_slaves(codec, slaves, (map_slave_func_t)snd_ctl_add_slave, - kctl); + err = map_slaves(codec, slaves, suffix, + (map_slave_func_t)snd_ctl_add_slave, kctl); if (err < 0) return err; return 0; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index aca8d3193b9..6094dea82bc 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -140,7 +140,8 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); int snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char * const *slaves); + unsigned int *tlv, const char * const *slaves, + const char *suffix); int snd_hda_codec_reset(struct hda_codec *codec); /* amp value bits */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9cb14b42dff..9771b070245 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -137,51 +137,17 @@ static int ad198x_init(struct hda_codec *codec) return 0; } -static const char * const ad_slave_vols[] = { - "Front Playback Volume", - "Surround Playback Volume", - "Center Playback Volume", - "LFE Playback Volume", - "Side Playback Volume", - "Headphone Playback Volume", - "Mono Playback Volume", - "Speaker Playback Volume", - "IEC958 Playback Volume", +static const char * const ad_slave_pfxs[] = { + "Front", "Surround", "Center", "LFE", "Side", + "Headphone", "Mono", "Speaker", "IEC958", NULL }; -static const char * const ad_slave_sws[] = { - "Front Playback Switch", - "Surround Playback Switch", - "Center Playback Switch", - "LFE Playback Switch", - "Side Playback Switch", - "Headphone Playback Switch", - "Mono Playback Switch", - "Speaker Playback Switch", - "IEC958 Playback Switch", +static const char * const ad1988_6stack_fp_slave_pfxs[] = { + "Front", "Surround", "Center", "LFE", "Side", "IEC958", NULL }; -static const char * const ad1988_6stack_fp_slave_vols[] = { - "Front Playback Volume", - "Surround Playback Volume", - "Center Playback Volume", - "LFE Playback Volume", - "Side Playback Volume", - "IEC958 Playback Volume", - NULL -}; - -static const char * const ad1988_6stack_fp_slave_sws[] = { - "Front Playback Switch", - "Surround Playback Switch", - "Center Playback Switch", - "LFE Playback Switch", - "Side Playback Switch", - "IEC958 Playback Switch", - NULL -}; static void ad198x_free_kctls(struct hda_codec *codec); #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -260,7 +226,8 @@ static int ad198x_build_controls(struct hda_codec *codec) err = snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, (spec->slave_vols ? - spec->slave_vols : ad_slave_vols)); + spec->slave_vols : ad_slave_pfxs), + "Playback Volume"); if (err < 0) return err; } @@ -268,7 +235,8 @@ static int ad198x_build_controls(struct hda_codec *codec) err = snd_hda_add_vmaster(codec, "Master Playback Switch", NULL, (spec->slave_sws ? - spec->slave_sws : ad_slave_sws)); + spec->slave_sws : ad_slave_pfxs), + "Playback Switch"); if (err < 0) return err; } @@ -3385,8 +3353,8 @@ static int patch_ad1988(struct hda_codec *codec) if (spec->autocfg.hp_pins[0]) { spec->mixers[spec->num_mixers++] = ad1988_hp_mixers; - spec->slave_vols = ad1988_6stack_fp_slave_vols; - spec->slave_sws = ad1988_6stack_fp_slave_sws; + spec->slave_vols = ad1988_6stack_fp_slave_pfxs; + spec->slave_sws = ad1988_6stack_fp_slave_pfxs; spec->alt_dac_nid = ad1988_alt_dac_nid; spec->stream_analog_alt_playback = &ad198x_pcm_analog_alt_playback; @@ -3594,16 +3562,8 @@ static const struct hda_amp_list ad1884_loopbacks[] = { #endif static const char * const ad1884_slave_vols[] = { - "PCM Playback Volume", - "Mic Playback Volume", - "Mono Playback Volume", - "Front Mic Playback Volume", - "Mic Playback Volume", - "CD Playback Volume", - "Internal Mic Playback Volume", - "Docking Mic Playback Volume", - /* "Beep Playback Volume", */ - "IEC958 Playback Volume", + "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD", + "Internal Mic", "Docking Mic", /* "Beep", */ "IEC958", NULL }; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 0eb526c672b..266e5a68baf 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -465,21 +465,8 @@ static const struct snd_kcontrol_new cxt_beep_mixer[] = { }; #endif -static const char * const slave_vols[] = { - "Headphone Playback Volume", - "Speaker Playback Volume", - "Front Playback Volume", - "Surround Playback Volume", - "CLFE Playback Volume", - NULL -}; - -static const char * const slave_sws[] = { - "Headphone Playback Switch", - "Speaker Playback Switch", - "Front Playback Switch", - "Surround Playback Switch", - "CLFE Playback Switch", +static const char * const slave_pfxs[] = { + "Headphone", "Speaker", "Front", "Surround", "CLFE", NULL }; @@ -519,14 +506,16 @@ static int conexant_build_controls(struct hda_codec *codec) snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_vols); + vmaster_tlv, slave_pfxs, + "Playback Volume"); if (err < 0) return err; } if (spec->vmaster_nid && !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_sws); + NULL, slave_pfxs, + "Playback Switch"); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33b6077fcdb..42f18449b82 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1845,36 +1845,10 @@ DEFINE_CAPMIX_NOSRC(3); /* * slave controls for virtual master */ -static const char * const alc_slave_vols[] = { - "Front Playback Volume", - "Surround Playback Volume", - "Center Playback Volume", - "LFE Playback Volume", - "Side Playback Volume", - "Headphone Playback Volume", - "Speaker Playback Volume", - "Mono Playback Volume", - "Line-Out Playback Volume", - "CLFE Playback Volume", - "Bass Speaker Playback Volume", - "PCM Playback Volume", - NULL, -}; - -static const char * const alc_slave_sws[] = { - "Front Playback Switch", - "Surround Playback Switch", - "Center Playback Switch", - "LFE Playback Switch", - "Side Playback Switch", - "Headphone Playback Switch", - "Speaker Playback Switch", - "Mono Playback Switch", - "IEC958 Playback Switch", - "Line-Out Playback Switch", - "CLFE Playback Switch", - "Bass Speaker Playback Switch", - "PCM Playback Switch", +static const char * const alc_slave_pfxs[] = { + "Front", "Surround", "Center", "LFE", "Side", + "Headphone", "Speaker", "Mono", "Line-Out", + "CLFE", "Bass Speaker", "PCM", NULL, }; @@ -1965,14 +1939,16 @@ static int __alc_build_controls(struct hda_codec *codec) snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, alc_slave_vols); + vmaster_tlv, alc_slave_pfxs, + "Playback Volume"); if (err < 0) return err; } if (!spec->no_analog && !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, alc_slave_sws); + NULL, alc_slave_pfxs, + "Playback Switch"); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 948f0be2f4f..de7166a65f8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1060,26 +1060,9 @@ static struct snd_kcontrol_new stac_smux_mixer = { .put = stac92xx_smux_enum_put, }; -static const char * const slave_vols[] = { - "Front Playback Volume", - "Surround Playback Volume", - "Center Playback Volume", - "LFE Playback Volume", - "Side Playback Volume", - "Headphone Playback Volume", - "Speaker Playback Volume", - NULL -}; - -static const char * const slave_sws[] = { - "Front Playback Switch", - "Surround Playback Switch", - "Center Playback Switch", - "LFE Playback Switch", - "Side Playback Switch", - "Headphone Playback Switch", - "Speaker Playback Switch", - "IEC958 Playback Switch", +static const char * const slave_pfxs[] = { + "Front", "Surround", "Center", "LFE", "Side", + "Headphone", "Speaker", "IEC958", NULL }; @@ -1153,13 +1136,15 @@ static int stac92xx_build_controls(struct hda_codec *codec) /* minimum value is actually mute */ vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_vols); + vmaster_tlv, slave_pfxs, + "Playback Volume"); if (err < 0) return err; } if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_sws); + NULL, slave_pfxs, + "Playback Switch"); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 284e311040f..e5842fe1b1e 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1442,25 +1442,9 @@ static const struct hda_pcm_stream via_pcm_digital_capture = { /* * slave controls for virtual master */ -static const char * const via_slave_vols[] = { - "Front Playback Volume", - "Surround Playback Volume", - "Center Playback Volume", - "LFE Playback Volume", - "Side Playback Volume", - "Headphone Playback Volume", - "Speaker Playback Volume", - NULL, -}; - -static const char * const via_slave_sws[] = { - "Front Playback Switch", - "Surround Playback Switch", - "Center Playback Switch", - "LFE Playback Switch", - "Side Playback Switch", - "Headphone Playback Switch", - "Speaker Playback Switch", +static const char * const via_slave_pfxs[] = { + "Front", "Surround", "Center", "LFE", "Side", + "Headphone", "Speaker", NULL, }; @@ -1505,13 +1489,15 @@ static int via_build_controls(struct hda_codec *codec) snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], HDA_OUTPUT, vmaster_tlv); err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, via_slave_vols); + vmaster_tlv, via_slave_pfxs, + "Playback Volume"); if (err < 0) return err; } if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, via_slave_sws); + NULL, via_slave_pfxs, + "Playback Switch"); if (err < 0) return err; } -- cgit v1.2.3-18-g5258 From 022658beab5581ecc1d325d60857f2fc464da22f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 3 Feb 2012 17:43:09 +0000 Subject: ASoC: core: Add support for DAI and machine kcontrols. Currently ASoC can only add kcontrols using codec and platform component device handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for SoC card machine drivers too. This allows the kcontrol to have a direct handle to the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily get it's private data. This change makes snd_soc_add_controls() static and wraps it in the folowing calls (card and dai are new) :- snd_soc_add_card_controls() snd_soc_add_codec_controls() snd_soc_add_dai_controls() snd_soc_add_platform_controls() This patch also does a lot of small mechanical changes in individual codec drivers to replace snd_soc_add_controls() with snd_soc_add_codec_controls(). It also updates the McBSP DAI driver to use snd_soc_add_dai_controls(). Finally, it updates the existing machine drivers that register controls to either :- 1) Use snd_soc_add_card_controls() where no direct codec control is required. 2) Use snd_soc_add_codec_controls() where there is direct codec control. In the case of 1) above we also update the machine drivers to get the correct component data pointers from the kcontrol (rather than getting the machine pointer via the codec pointer). Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 6 +-- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/adau1373.c | 4 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 8 +-- sound/soc/codecs/alc5632.c | 2 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 2 +- sound/soc/codecs/lm4857.c | 2 +- sound/soc/codecs/max98088.c | 4 +- sound/soc/codecs/max98095.c | 6 +-- sound/soc/codecs/max9877.c | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/stac9766.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 4 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/tpa6130a2.c | 4 +- sound/soc/codecs/uda134x.c | 6 +-- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8904.c | 16 +++--- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8958-dsp2.c | 14 ++--- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 6 +-- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 4 +- sound/soc/codecs/wm8994.c | 12 ++--- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 4 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 6 +-- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/mid-x86/mfld_machine.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/n810.c | 15 +++--- sound/soc/omap/omap-mcbsp.c | 8 +-- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/rx51.c | 25 +++++---- sound/soc/pxa/corgi.c | 14 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/poodle.c | 14 +++-- sound/soc/pxa/tosa.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 4 +- sound/soc/samsung/s3c24xx_simtec.c | 6 +-- sound/soc/soc-core.c | 108 ++++++++++++++++++++++++------------- 59 files changed, 198 insertions(+), 172 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 982d201c2e8..12e3b411855 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -277,7 +277,7 @@ static int ad1836_probe(struct snd_soc_codec *codec) if (ad1836->type == AD1836) { /* left/right diff:PGA/MUX */ snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A); - ret = snd_soc_add_controls(codec, ad1836_controls, + ret = snd_soc_add_codec_controls(codec, ad1836_controls, ARRAY_SIZE(ad1836_controls)); if (ret) return ret; @@ -285,11 +285,11 @@ static int ad1836_probe(struct snd_soc_codec *codec) snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00); } - ret = snd_soc_add_controls(codec, ad183x_dac_controls, num_dacs * 2); + ret = snd_soc_add_codec_controls(codec, ad183x_dac_controls, num_dacs * 2); if (ret) return ret; - ret = snd_soc_add_controls(codec, ad183x_adc_controls, num_adcs); + ret = snd_soc_add_codec_controls(codec, ad183x_adc_controls, num_adcs); if (ret) return ret; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 9bba7f84946..8c39dddd7d0 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -228,7 +228,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec) ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); - snd_soc_add_controls(codec, ad1980_snd_ac97_controls, + snd_soc_add_codec_controls(codec, ad1980_snd_ac97_controls, ARRAY_SIZE(ad1980_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index facda33db1c..44f59064d8d 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1257,7 +1257,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) pdata->drc_setting[i]); } - snd_soc_add_controls(codec, adau1373_drc_controls, + snd_soc_add_codec_controls(codec, adau1373_drc_controls, pdata->num_drc); val = 0; @@ -1282,7 +1282,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) } if (!lineout_differential) { - snd_soc_add_controls(codec, adau1373_lineout2_controls, + snd_soc_add_codec_controls(codec, adau1373_lineout2_controls, ARRAY_SIZE(adau1373_lineout2_controls)); } diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 9e809e05d06..dd15516763e 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -382,7 +382,7 @@ static int ak4535_probe(struct snd_soc_codec *codec) /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, ak4535_snd_controls, + snd_soc_add_codec_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); return 0; } diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 5ef70b5d27e..16bd1e7d238 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -476,7 +476,7 @@ static int ak4642_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_add_controls(codec, ak4642_snd_controls, + snd_soc_add_codec_controls(codec, ak4642_snd_controls, ARRAY_SIZE(ak4642_snd_controls)); ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index a53b152e6a0..5fb7c2a80e6 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -628,7 +628,7 @@ static int ak4671_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_add_controls(codec, ak4671_snd_controls, + snd_soc_add_codec_controls(codec, ak4671_snd_controls, ARRAY_SIZE(ak4671_snd_controls)); ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 08f24198c8d..d47b62ddb21 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -925,22 +925,22 @@ static int alc5623_probe(struct snd_soc_codec *codec) switch (alc5623->id) { case 0x21: - snd_soc_add_controls(codec, alc5621_vol_snd_controls, + snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls, ARRAY_SIZE(alc5621_vol_snd_controls)); break; case 0x22: - snd_soc_add_controls(codec, alc5622_vol_snd_controls, + snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls, ARRAY_SIZE(alc5622_vol_snd_controls)); break; case 0x23: - snd_soc_add_controls(codec, alc5623_vol_snd_controls, + snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls, ARRAY_SIZE(alc5623_vol_snd_controls)); break; default: return -EINVAL; } - snd_soc_add_controls(codec, alc5623_snd_controls, + snd_soc_add_codec_controls(codec, alc5623_snd_controls, ARRAY_SIZE(alc5623_snd_controls)); snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets, diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index af9c27ae02f..f69fb426ad0 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -994,7 +994,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) switch (alc5632->id) { case 0x5c: - snd_soc_add_controls(codec, alc5632_vol_snd_controls, + snd_soc_add_codec_controls(codec, alc5632_vol_snd_controls, ARRAY_SIZE(alc5632_vol_snd_controls)); break; default: diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 06d2ea18a54..064cd6a9351 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -157,7 +157,7 @@ static int cq93vc_probe(struct snd_soc_codec *codec) codec->control_data = davinci_vc; /* Set controls */ - snd_soc_add_controls(codec, cq93vc_snd_controls, + snd_soc_add_codec_controls(codec, cq93vc_snd_controls, ARRAY_SIZE(cq93vc_snd_controls)); /* Off, with power on */ diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 055536645da..6baccd285df 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -521,7 +521,7 @@ static int cs4270_probe(struct snd_soc_codec *codec) } /* Add the non-DAPM controls */ - ret = snd_soc_add_controls(codec, cs4270_snd_controls, + ret = snd_soc_add_codec_controls(codec, cs4270_snd_controls, ARRAY_SIZE(cs4270_snd_controls)); if (ret < 0) { dev_err(codec->dev, "failed to add controls\n"); diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f6fe846b6a6..bf7141280a7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -513,7 +513,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) /* Power-up sequence requires 85 uS */ udelay(85); - return snd_soc_add_controls(codec, cs4271_snd_controls, + return snd_soc_add_codec_controls(codec, cs4271_snd_controls, ARRAY_SIZE(cs4271_snd_controls)); } diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 319039240e0..ba4fafb93e5 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -179,7 +179,7 @@ static int lm4857_probe(struct snd_soc_codec *codec) codec->control_data = lm4857->i2c; - ret = snd_soc_add_controls(codec, lm4857_controls, + ret = snd_soc_add_codec_controls(codec, lm4857_controls, ARRAY_SIZE(lm4857_controls)); if (ret) return ret; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 006efcfe6dd..af7324b79dd 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1908,7 +1908,7 @@ static void max98088_handle_eq_pdata(struct snd_soc_codec *codec) max98088->eq_enum.texts = max98088->eq_texts; max98088->eq_enum.max = max98088->eq_textcnt; - ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(codec->dev, "Failed to add EQ control: %d\n", ret); } @@ -2030,7 +2030,7 @@ static int max98088_probe(struct snd_soc_codec *codec) max98088_handle_pdata(codec); - snd_soc_add_controls(codec, max98088_snd_controls, + snd_soc_add_codec_controls(codec, max98088_snd_controls, ARRAY_SIZE(max98088_snd_controls)); err_access: diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index fcfa7497d7b..0bb511a0388 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1284,7 +1284,7 @@ static const struct snd_soc_dapm_route max98095_audio_map[] = { static int max98095_add_widgets(struct snd_soc_codec *codec) { - snd_soc_add_controls(codec, max98095_snd_controls, + snd_soc_add_codec_controls(codec, max98095_snd_controls, ARRAY_SIZE(max98095_snd_controls)); return 0; @@ -1984,7 +1984,7 @@ static void max98095_handle_eq_pdata(struct snd_soc_codec *codec) max98095->eq_enum.texts = max98095->eq_texts; max98095->eq_enum.max = max98095->eq_textcnt; - ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(codec->dev, "Failed to add EQ control: %d\n", ret); } @@ -2139,7 +2139,7 @@ static void max98095_handle_bq_pdata(struct snd_soc_codec *codec) max98095->bq_enum.texts = max98095->bq_texts; max98095->bq_enum.max = max98095->bq_textcnt; - ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(codec->dev, "Failed to add Biquad control: %d\n", ret); } diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index dcf6f2a1600..3a2ba3d8fd6 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -253,7 +253,7 @@ static const struct snd_kcontrol_new max9877_controls[] = { /* This function is called from ASoC machine driver */ int max9877_add_controls(struct snd_soc_codec *codec) { - return snd_soc_add_controls(codec, max9877_controls, + return snd_soc_add_codec_controls(codec, max9877_controls, ARRAY_SIZE(max9877_controls)); } EXPORT_SYMBOL_GPL(max9877_add_controls); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index aa0392360da..50dbdb9357e 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -869,7 +869,7 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) snd_soc_write(codec, SN95031_SSR2, 0x10); snd_soc_write(codec, SN95031_SSR3, 0x40); - snd_soc_add_controls(codec, sn95031_snd_controls, + snd_soc_add_codec_controls(codec, sn95031_snd_controls, ARRAY_SIZE(sn95031_snd_controls)); return 0; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 333dd98af39..de2b20544ce 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -548,7 +548,7 @@ static int ssm2602_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, SSM2602_ROUT1V, ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH); - ret = snd_soc_add_controls(codec, ssm2602_snd_controls, + ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); if (ret) return ret; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index cc0566c22ec..982e437799a 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -355,7 +355,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec) stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, stac9766_snd_ac97_controls, + snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index dfa41a96599..16d55f91a65 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -593,7 +593,7 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); - snd_soc_add_controls(codec, tlv320aic23_snd_controls, + snd_soc_add_codec_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); return 0; diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index a038daec682..802064b5030 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -389,7 +389,7 @@ static int aic26_probe(struct snd_soc_codec *codec) /* register controls */ dev_dbg(codec->dev, "Registering controls\n"); - err = snd_soc_add_controls(codec, aic26_snd_controls, + err = snd_soc_add_codec_controls(codec, aic26_snd_controls, ARRAY_SIZE(aic26_snd_controls)); WARN_ON(err < 0); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 372b0b83bd9..b0a73d37ed5 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -671,7 +671,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, aic32x4_snd_controls, + snd_soc_add_codec_controls(codec, aic32x4_snd_controls, ARRAY_SIZE(aic32x4_snd_controls)); aic32x4_add_widgets(codec); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 285b7a22dc1..0bb7cb8815c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1425,10 +1425,10 @@ static int aic3x_probe(struct snd_soc_codec *codec) (aic3x->setup->gpio_func[1] & 0xf) << 4); } - snd_soc_add_controls(codec, aic3x_snd_controls, + snd_soc_add_codec_controls(codec, aic3x_snd_controls, ARRAY_SIZE(aic3x_snd_controls)); if (aic3x->model == AIC3X_MODEL_3007) - snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 2c957c84570..4587ddd0fbf 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1437,7 +1437,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) /* Only add the FIFO controls, if we have valid IRQ number */ if (dac33->irq >= 0) - snd_soc_add_controls(codec, dac33_mode_snd_controls, + snd_soc_add_codec_controls(codec, dac33_mode_snd_controls, ARRAY_SIZE(dac33_mode_snd_controls)); err_power: diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 363b99dad8e..6fe4aa3ac54 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -351,10 +351,10 @@ int tpa6130a2_add_controls(struct snd_soc_codec *codec) data = i2c_get_clientdata(tpa6130a2_client); if (data->id == TPA6140A2) - return snd_soc_add_controls(codec, tpa6140a2_controls, + return snd_soc_add_codec_controls(codec, tpa6140a2_controls, ARRAY_SIZE(tpa6140a2_controls)); else - return snd_soc_add_controls(codec, tpa6130a2_controls, + return snd_soc_add_codec_controls(codec, tpa6130a2_controls, ARRAY_SIZE(tpa6130a2_controls)); } EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 8f4f469d641..797b0dde2c6 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -531,15 +531,15 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) switch (pd->model) { case UDA134X_UDA1340: case UDA134X_UDA1344: - ret = snd_soc_add_controls(codec, uda1340_snd_controls, + ret = snd_soc_add_codec_controls(codec, uda1340_snd_controls, ARRAY_SIZE(uda1340_snd_controls)); break; case UDA134X_UDA1341: - ret = snd_soc_add_controls(codec, uda1341_snd_controls, + ret = snd_soc_add_codec_controls(codec, uda1341_snd_controls, ARRAY_SIZE(uda1341_snd_controls)); break; case UDA134X_UDA1345: - ret = snd_soc_add_controls(codec, uda1345_snd_controls, + ret = snd_soc_add_codec_controls(codec, uda1345_snd_controls, ARRAY_SIZE(uda1345_snd_controls)); break; default: diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 44aacf927ba..3d868dc4009 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -464,7 +464,7 @@ static int wl1273_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, wl1273); - r = snd_soc_add_controls(codec, wl1273_controls, + r = snd_soc_add_codec_controls(codec, wl1273_controls, ARRAY_SIZE(wl1273_controls)); if (r) kfree(wl1273); diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index ff95e62c56b..4fe9d191e27 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -599,7 +599,7 @@ static int wm8737_probe(struct snd_soc_codec *codec) /* Bias level configuration will have done an extra enable */ regulator_bulk_disable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); - snd_soc_add_controls(codec, wm8737_snd_controls, + snd_soc_add_codec_controls(codec, wm8737_snd_controls, ARRAY_SIZE(wm8737_snd_controls)); wm8737_add_widgets(codec); diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index bd60f847762..a5127b4ff9e 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -641,7 +641,7 @@ static int wm8770_probe(struct snd_soc_codec *codec) /* mute all DACs */ snd_soc_update_bits(codec, WM8770_DACMUTE, 0x10, 0x10); - snd_soc_add_controls(codec, wm8770_snd_controls, + snd_soc_add_codec_controls(codec, wm8770_snd_controls, ARRAY_SIZE(wm8770_snd_controls)); snd_soc_dapm_new_controls(&codec->dapm, wm8770_dapm_widgets, ARRAY_SIZE(wm8770_dapm_widgets)); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 37079eace41..65d525d74c5 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1176,11 +1176,11 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) switch (wm8904->devtype) { case WM8904: - snd_soc_add_controls(codec, wm8904_adc_snd_controls, + snd_soc_add_codec_controls(codec, wm8904_adc_snd_controls, ARRAY_SIZE(wm8904_adc_snd_controls)); - snd_soc_add_controls(codec, wm8904_dac_snd_controls, + snd_soc_add_codec_controls(codec, wm8904_dac_snd_controls, ARRAY_SIZE(wm8904_dac_snd_controls)); - snd_soc_add_controls(codec, wm8904_snd_controls, + snd_soc_add_codec_controls(codec, wm8904_snd_controls, ARRAY_SIZE(wm8904_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8904_adc_dapm_widgets, @@ -1201,7 +1201,7 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) break; case WM8912: - snd_soc_add_controls(codec, wm8904_dac_snd_controls, + snd_soc_add_codec_controls(codec, wm8904_dac_snd_controls, ARRAY_SIZE(wm8904_dac_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets, @@ -2020,7 +2020,7 @@ static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec) wm8904->retune_mobile_enum.max = wm8904->num_retune_mobile_texts; wm8904->retune_mobile_enum.texts = wm8904->retune_mobile_texts; - ret = snd_soc_add_controls(codec, &control, 1); + ret = snd_soc_add_codec_controls(codec, &control, 1); if (ret != 0) dev_err(codec->dev, "Failed to add ReTune Mobile control: %d\n", ret); @@ -2033,7 +2033,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) int ret, i; if (!pdata) { - snd_soc_add_controls(codec, wm8904_eq_controls, + snd_soc_add_codec_controls(codec, wm8904_eq_controls, ARRAY_SIZE(wm8904_eq_controls)); return; } @@ -2061,7 +2061,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) wm8904->drc_enum.max = pdata->num_drc_cfgs; wm8904->drc_enum.texts = wm8904->drc_texts; - ret = snd_soc_add_controls(codec, &control, 1); + ret = snd_soc_add_codec_controls(codec, &control, 1); if (ret != 0) dev_err(codec->dev, "Failed to add DRC mode control: %d\n", ret); @@ -2075,7 +2075,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) if (pdata->num_retune_mobile_cfgs) wm8904_handle_retune_mobile_pdata(codec); else - snd_soc_add_controls(codec, wm8904_eq_controls, + snd_soc_add_codec_controls(codec, wm8904_eq_controls, ARRAY_SIZE(wm8904_eq_controls)); } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index ae1933ed3e0..d2883affea3 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -717,7 +717,7 @@ static int wm8940_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_add_controls(codec, wm8940_snd_controls, + ret = snd_soc_add_codec_controls(codec, wm8940_snd_controls, ARRAY_SIZE(wm8940_snd_controls)); if (ret) return ret; diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 40ac888faf3..1332692ef81 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -920,11 +920,11 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->dsp_active = -1; - snd_soc_add_controls(codec, wm8958_mbc_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_mbc_snd_controls, ARRAY_SIZE(wm8958_mbc_snd_controls)); - snd_soc_add_controls(codec, wm8958_vss_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_vss_snd_controls, ARRAY_SIZE(wm8958_vss_snd_controls)); - snd_soc_add_controls(codec, wm8958_enh_eq_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_enh_eq_snd_controls, ARRAY_SIZE(wm8958_enh_eq_snd_controls)); @@ -958,7 +958,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->mbc_enum.max = pdata->num_mbc_cfgs; wm8994->mbc_enum.texts = wm8994->mbc_texts; - ret = snd_soc_add_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); if (ret != 0) dev_err(wm8994->codec->dev, "Failed to add MBC mode controls: %d\n", ret); @@ -986,7 +986,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_enum.max = pdata->num_vss_cfgs; wm8994->vss_enum.texts = wm8994->vss_texts; - ret = snd_soc_add_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); if (ret != 0) dev_err(wm8994->codec->dev, "Failed to add VSS mode controls: %d\n", ret); @@ -1015,7 +1015,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs; wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts; - ret = snd_soc_add_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); if (ret != 0) dev_err(wm8994->codec->dev, "Failed to add VSS HPFmode controls: %d\n", @@ -1045,7 +1045,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs; wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts; - ret = snd_soc_add_controls(wm8994->codec, control, 1); + ret = snd_soc_add_codec_controls(wm8994->codec, control, 1); if (ret != 0) dev_err(wm8994->codec->dev, "Failed to add enhanced EQ controls: %d\n", diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e5caae32e54..840d72086d0 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -940,7 +940,7 @@ static int wm8960_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8960_LOUT2, 0x100, 0x100); snd_soc_update_bits(codec, WM8960_ROUT2, 0x100, 0x100); - snd_soc_add_controls(codec, wm8960_snd_controls, + snd_soc_add_codec_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); wm8960_add_widgets(codec); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 4f20c72a0f1..05ea7c27409 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1022,7 +1022,7 @@ static int wm8961_probe(struct snd_soc_codec *codec) wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8961_snd_controls, + snd_soc_add_codec_controls(codec, wm8961_snd_controls, ARRAY_SIZE(wm8961_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8961_dapm_widgets, ARRAY_SIZE(wm8961_dapm_widgets)); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b6fcdcc4341..25b6baed3a2 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2405,13 +2405,13 @@ static int wm8962_add_widgets(struct snd_soc_codec *codec) struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_add_controls(codec, wm8962_snd_controls, + snd_soc_add_codec_controls(codec, wm8962_snd_controls, ARRAY_SIZE(wm8962_snd_controls)); if (pdata && pdata->spk_mono) - snd_soc_add_controls(codec, wm8962_spk_mono_controls, + snd_soc_add_codec_controls(codec, wm8962_spk_mono_controls, ARRAY_SIZE(wm8962_spk_mono_controls)); else - snd_soc_add_controls(codec, wm8962_spk_stereo_controls, + snd_soc_add_codec_controls(codec, wm8962_spk_stereo_controls, ARRAY_SIZE(wm8962_spk_stereo_controls)); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index e538edaae1f..9d242351e6e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1356,7 +1356,7 @@ static int wm8990_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_add_controls(codec, wm8990_snd_controls, + snd_soc_add_codec_controls(codec, wm8990_snd_controls, ARRAY_SIZE(wm8990_snd_controls)); wm8990_add_widgets(codec); diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 7ee40da8dbb..9ac31ba9b82 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1297,7 +1297,7 @@ static int wm8991_probe(struct snd_soc_codec *codec) snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8)); snd_soc_write(codec, WM8991_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8)); - snd_soc_add_controls(codec, wm8991_snd_controls, + snd_soc_add_codec_controls(codec, wm8991_snd_controls, ARRAY_SIZE(wm8991_snd_controls)); snd_soc_dapm_new_controls(&codec->dapm, wm8991_dapm_widgets, diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index db51007a6a4..f814d2711b5 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1604,13 +1604,13 @@ static int wm8993_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - snd_soc_add_controls(codec, wm8993_snd_controls, + snd_soc_add_codec_controls(codec, wm8993_snd_controls, ARRAY_SIZE(wm8993_snd_controls)); if (wm8993->pdata.num_retune_configs != 0) { dev_dbg(codec->dev, "Using ReTune Mobile\n"); } else { dev_dbg(codec->dev, "No ReTune Mobile, using normal EQ\n"); - snd_soc_add_controls(codec, wm8993_eq_controls, + snd_soc_add_codec_controls(codec, wm8993_eq_controls, ARRAY_SIZE(wm8993_eq_controls)); } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c26291844e5..8ae6585edbe 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2867,7 +2867,7 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts; wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts; - ret = snd_soc_add_controls(wm8994->codec, controls, + ret = snd_soc_add_codec_controls(wm8994->codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(wm8994->codec->dev, @@ -2920,7 +2920,7 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) wm8994->drc_enum.max = pdata->num_drc_cfgs; wm8994->drc_enum.texts = wm8994->drc_texts; - ret = snd_soc_add_controls(wm8994->codec, controls, + ret = snd_soc_add_codec_controls(wm8994->codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(wm8994->codec->dev, @@ -2936,7 +2936,7 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) if (pdata->num_retune_mobile_cfgs) wm8994_handle_retune_mobile_pdata(wm8994); else - snd_soc_add_controls(wm8994->codec, wm8994_eq_controls, + snd_soc_add_codec_controls(wm8994->codec, wm8994_eq_controls, ARRAY_SIZE(wm8994_eq_controls)); for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) { @@ -3652,7 +3652,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_handle_pdata(wm8994); wm_hubs_add_analogue_controls(codec); - snd_soc_add_controls(codec, wm8994_snd_controls, + snd_soc_add_codec_controls(codec, wm8994_snd_controls, ARRAY_SIZE(wm8994_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets, ARRAY_SIZE(wm8994_dapm_widgets)); @@ -3678,7 +3678,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } break; case WM8958: - snd_soc_add_controls(codec, wm8958_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_snd_controls, ARRAY_SIZE(wm8958_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); @@ -3700,7 +3700,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM1811: - snd_soc_add_controls(codec, wm8958_snd_controls, + snd_soc_add_codec_controls(codec, wm8958_snd_controls, ARRAY_SIZE(wm8958_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 89a864287c1..28c89b094c6 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2136,7 +2136,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995_update_class_w(codec); - snd_soc_add_controls(codec, wm8995_snd_controls, + snd_soc_add_codec_controls(codec, wm8995_snd_controls, ARRAY_SIZE(wm8995_snd_controls)); snd_soc_dapm_new_controls(&codec->dapm, wm8995_dapm_widgets, ARRAY_SIZE(wm8995_dapm_widgets)); diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 10f41c88888..86f449ccf81 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2771,7 +2771,7 @@ static void wm8996_retune_mobile_pdata(struct snd_soc_codec *codec) wm8996->retune_mobile_enum.max = wm8996->num_retune_mobile_texts; wm8996->retune_mobile_enum.texts = wm8996->retune_mobile_texts; - ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls)); + ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) dev_err(codec->dev, "Failed to add ReTune Mobile controls: %d\n", ret); @@ -2966,7 +2966,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) if (wm8996->pdata.num_retune_mobile_cfgs) wm8996_retune_mobile_pdata(codec); else - snd_soc_add_controls(codec, wm8996_eq_controls, + snd_soc_add_codec_controls(codec, wm8996_eq_controls, ARRAY_SIZE(wm8996_eq_controls)); /* If the TX LRCLK pins are not in LRCLK mode configure the diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index a6bab392700..7b09b1f86db 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1287,7 +1287,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); - snd_soc_add_controls(codec, wm9081_eq_controls, + snd_soc_add_codec_controls(codec, wm9081_eq_controls, ARRAY_SIZE(wm9081_eq_controls)); } diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a2b9208a08f..e8280eecd4c 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -433,7 +433,7 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_add_controls(codec, wm9090_controls, + snd_soc_add_codec_controls(codec, wm9090_controls, ARRAY_SIZE(wm9090_controls)); if (wm9090->pdata.lin1_diff) { @@ -442,7 +442,7 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) } else { snd_soc_dapm_add_routes(dapm, audio_map_in1_se, ARRAY_SIZE(audio_map_in1_se)); - snd_soc_add_controls(codec, wm9090_in1_se_controls, + snd_soc_add_codec_controls(codec, wm9090_in1_se_controls, ARRAY_SIZE(wm9090_in1_se_controls)); } @@ -452,7 +452,7 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) } else { snd_soc_dapm_add_routes(dapm, audio_map_in2_se, ARRAY_SIZE(audio_map_in2_se)); - snd_soc_add_controls(codec, wm9090_in2_se_controls, + snd_soc_add_codec_controls(codec, wm9090_in2_se_controls, ARRAY_SIZE(wm9090_in2_se_controls)); } diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 40c92ead85a..cacc6a86b46 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -351,7 +351,7 @@ static int wm9705_soc_probe(struct snd_soc_codec *codec) if (ret) goto reset_err; - snd_soc_add_controls(codec, wm9705_snd_ac97_controls, + snd_soc_add_codec_controls(codec, wm9705_snd_ac97_controls, ARRAY_SIZE(wm9705_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 7291eabb0eb..b342ae50bcd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -637,7 +637,7 @@ static int wm9712_soc_probe(struct snd_soc_codec *codec) ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm9712_snd_ac97_controls, + snd_soc_add_codec_controls(codec, wm9712_snd_ac97_controls, ARRAY_SIZE(wm9712_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2b8479bfcd9..2d22cc70d53 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1216,7 +1216,7 @@ static int wm9713_soc_probe(struct snd_soc_codec *codec) reg = ac97_read(codec, AC97_CD) & 0x7fff; ac97_write(codec, AC97_CD, reg); - snd_soc_add_controls(codec, wm9713_snd_ac97_controls, + snd_soc_add_codec_controls(codec, wm9713_snd_ac97_controls, ARRAY_SIZE(wm9713_snd_ac97_controls)); return 0; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ec7d49033d4..c509911a59f 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -895,7 +895,7 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec) WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU, WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU); - snd_soc_add_controls(codec, analogue_snd_controls, + snd_soc_add_codec_controls(codec, analogue_snd_controls, ARRAY_SIZE(analogue_snd_controls)); snd_soc_dapm_new_controls(dapm, analogue_dapm_widgets, diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 6f77eef0f13..2937e54da49 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -235,7 +235,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_enable_pin(dapm, "Headphones"); snd_soc_dapm_enable_pin(dapm, "Mic"); - ret_val = snd_soc_add_controls(codec, mfld_snd_controls, + ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls, ARRAY_SIZE(mfld_snd_controls)); if (ret_val) { pr_err("soc_add_controls failed %d", ret_val); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index a67f4370bc9..78563bbbbf0 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -570,7 +570,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "AGCOUT"); /* Add virtual switch */ - ret = snd_soc_add_controls(codec, ams_delta_audio_controls, + ret = snd_soc_add_codec_controls(codec, ams_delta_audio_controls, ARRAY_SIZE(ams_delta_audio_controls)); if (ret) dev_warn(card->dev, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 597be412f1e..1490227cd37 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -55,9 +55,8 @@ static int n810_spk_func; static int n810_jack_func; static int n810_dmic_func; -static void n810_ext_control(struct snd_soc_codec *codec) +static void n810_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; int hp = 0, line1l = 0; switch (n810_jack_func) { @@ -102,7 +101,7 @@ static int n810_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); - n810_ext_control(codec); + n810_ext_control(&codec->dapm); return clk_enable(sys_clkout2); } @@ -142,13 +141,13 @@ static int n810_get_spk(struct snd_kcontrol *kcontrol, static int n810_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (n810_spk_func == ucontrol->value.integer.value[0]) return 0; n810_spk_func = ucontrol->value.integer.value[0]; - n810_ext_control(codec); + n810_ext_control(&card->dapm); return 1; } @@ -164,13 +163,13 @@ static int n810_get_jack(struct snd_kcontrol *kcontrol, static int n810_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (n810_jack_func == ucontrol->value.integer.value[0]) return 0; n810_jack_func = ucontrol->value.integer.value[0]; - n810_ext_control(codec); + n810_ext_control(&card->dapm); return 1; } @@ -186,7 +185,7 @@ static int n810_get_input(struct snd_kcontrol *kcontrol, static int n810_set_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (n810_dmic_func == ucontrol->value.integer.value[0]) return 0; diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 017371913ec..1287b870f22 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -744,17 +744,17 @@ static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = { omap_mcbsp3_set_st_ch1_volume), }; -int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id) +int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai) { if (!cpu_is_omap34xx()) return -ENODEV; - switch (mcbsp_id) { + switch (dai->id) { case 1: /* McBSP 2 */ - return snd_soc_add_controls(codec, omap_mcbsp2_st_controls, + return snd_soc_add_dai_controls(dai, omap_mcbsp2_st_controls, ARRAY_SIZE(omap_mcbsp2_st_controls)); case 2: /* McBSP 3 */ - return snd_soc_add_controls(codec, omap_mcbsp3_st_controls, + return snd_soc_add_dai_controls(dai, omap_mcbsp3_st_controls, ARRAY_SIZE(omap_mcbsp3_st_controls)); default: break; diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 65cde9d3807..476fe2add70 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -59,6 +59,6 @@ enum omap_mcbsp_div { #define NUM_LINKS 5 #endif -int omap_mcbsp_st_add_controls(struct snd_soc_codec *codec, int mcbsp_id); +int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai); #endif diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index fada6ef43ee..58936c730a8 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -59,9 +59,8 @@ static int rx51_spk_func; static int rx51_dmic_func; static int rx51_jack_func; -static void rx51_ext_control(struct snd_soc_codec *codec) +static void rx51_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; int hp = 0, hs = 0, tvout = 0; switch (rx51_jack_func) { @@ -102,11 +101,11 @@ static int rx51_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); - rx51_ext_control(codec); + rx51_ext_control(&card->dapm); return 0; } @@ -138,13 +137,13 @@ static int rx51_get_spk(struct snd_kcontrol *kcontrol, static int rx51_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (rx51_spk_func == ucontrol->value.integer.value[0]) return 0; rx51_spk_func = ucontrol->value.integer.value[0]; - rx51_ext_control(codec); + rx51_ext_control(&card->dapm); return 1; } @@ -184,13 +183,13 @@ static int rx51_get_input(struct snd_kcontrol *kcontrol, static int rx51_set_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (rx51_dmic_func == ucontrol->value.integer.value[0]) return 0; rx51_dmic_func = ucontrol->value.integer.value[0]; - rx51_ext_control(codec); + rx51_ext_control(&card->dapm); return 1; } @@ -206,13 +205,13 @@ static int rx51_get_jack(struct snd_kcontrol *kcontrol, static int rx51_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (rx51_jack_func == ucontrol->value.integer.value[0]) return 0; rx51_jack_func = ucontrol->value.integer.value[0]; - rx51_ext_control(codec); + rx51_ext_control(&card->dapm); return 1; } @@ -297,7 +296,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "LINE1R"); /* Add RX-51 specific controls */ - err = snd_soc_add_controls(codec, aic34_rx51_controls, + err = snd_soc_add_card_controls(rtd->card, aic34_rx51_controls, ARRAY_SIZE(aic34_rx51_controls)); if (err < 0) return err; @@ -314,7 +313,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); - err = omap_mcbsp_st_add_controls(codec, 1); + err = omap_mcbsp_st_add_controls(rtd->cpu_dai); if (err < 0) return err; @@ -335,7 +334,7 @@ static int rx51_aic34b_init(struct snd_soc_dapm_context *dapm) { int err; - err = snd_soc_add_controls(dapm->codec, aic34_rx51_controlsb, + err = snd_soc_add_card_controls(dapm->card, aic34_rx51_controlsb, ARRAY_SIZE(aic34_rx51_controlsb)); if (err < 0) return err; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index bc21944851c..863367ad89c 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -45,10 +45,8 @@ static int corgi_jack_func; static int corgi_spk_func; -static void corgi_ext_control(struct snd_soc_codec *codec) +static void corgi_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; - /* set up jack connection */ switch (corgi_jack_func) { case CORGI_HP: @@ -104,7 +102,7 @@ static int corgi_startup(struct snd_pcm_substream *substream) mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - corgi_ext_control(codec); + corgi_ext_control(&codec->dapm); mutex_unlock(&codec->mutex); @@ -173,13 +171,13 @@ static int corgi_get_jack(struct snd_kcontrol *kcontrol, static int corgi_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (corgi_jack_func == ucontrol->value.integer.value[0]) return 0; corgi_jack_func = ucontrol->value.integer.value[0]; - corgi_ext_control(codec); + corgi_ext_control(&card->dapm); return 1; } @@ -193,13 +191,13 @@ static int corgi_get_spk(struct snd_kcontrol *kcontrol, static int corgi_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (corgi_spk_func == ucontrol->value.integer.value[0]) return 0; corgi_spk_func = ucontrol->value.integer.value[0]; - corgi_ext_control(codec); + corgi_ext_control(&card->dapm); return 1; } diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 3f7a8ecb972..aace19e0fe2 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -411,7 +411,7 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "VINR"); /* Add magician specific controls */ - err = snd_soc_add_controls(codec, uda1380_magician_controls, + err = snd_soc_add_codec_controls(codec, uda1380_magician_controls, ARRAY_SIZE(uda1380_magician_controls)); if (err < 0) return err; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index fd0ed10c6fe..d2cc8173503 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -43,10 +43,8 @@ static int poodle_jack_func; static int poodle_spk_func; -static void poodle_ext_control(struct snd_soc_codec *codec) +static void poodle_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; - /* set up jack connection */ if (poodle_jack_func == POODLE_HP) { /* set = unmute headphone */ @@ -81,7 +79,7 @@ static int poodle_startup(struct snd_pcm_substream *substream) mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - poodle_ext_control(codec); + poodle_ext_control(&codec->dapm); mutex_unlock(&codec->mutex); @@ -152,13 +150,13 @@ static int poodle_get_jack(struct snd_kcontrol *kcontrol, static int poodle_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (poodle_jack_func == ucontrol->value.integer.value[0]) return 0; poodle_jack_func = ucontrol->value.integer.value[0]; - poodle_ext_control(codec); + poodle_ext_control(&card->dapm); return 1; } @@ -172,13 +170,13 @@ static int poodle_get_spk(struct snd_kcontrol *kcontrol, static int poodle_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (poodle_spk_func == ucontrol->value.integer.value[0]) return 0; poodle_spk_func = ucontrol->value.integer.value[0]; - poodle_ext_control(codec); + poodle_ext_control(&card->dapm); return 1; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 564ef08a89f..2aec63f3706 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -197,7 +197,7 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "MONOOUT"); /* add tosa specific controls */ - err = snd_soc_add_controls(codec, tosa_controls, + err = snd_soc_add_codec_controls(codec, tosa_controls, ARRAY_SIZE(tosa_controls)); if (err < 0) return err; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 7ac0ba2025c..e34f4e80342 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -298,7 +298,7 @@ static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) if (ret) return ret; - ret = snd_soc_add_controls(codec, neo1973_gta02_wm8753_controls, + ret = snd_soc_add_card_controls(codec->card, neo1973_gta02_wm8753_controls, ARRAY_SIZE(neo1973_gta02_wm8753_controls)); if (ret) return ret; @@ -338,7 +338,7 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) return ret; /* add neo1973 specific controls */ - ret = snd_soc_add_controls(codec, neo1973_wm8753_controls, + ret = snd_soc_add_card_controls(rtd->card, neo1973_wm8753_controls, ARRAY_SIZE(neo1973_wm8753_controls)); if (ret) return ret; diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index a253bcc1646..656d5afe4ca 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -134,18 +134,18 @@ static const struct snd_kcontrol_new amp_unmute_controls[] = { void simtec_audio_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; if (pdata->amp_gpio > 0) { pr_debug("%s: adding amp routes\n", __func__); - snd_soc_add_controls(codec, amp_unmute_controls, + snd_soc_add_card_controls(card, amp_unmute_controls, ARRAY_SIZE(amp_unmute_controls)); } if (pdata->amp_gain[0] > 0) { pr_debug("%s: adding amp controls\n", __func__); - snd_soc_add_controls(codec, amp_gain_controls, + snd_soc_add_card_controls(card, amp_gain_controls, ARRAY_SIZE(amp_gain_controls)); } } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 091d5f37ae6..a3a47cdaac8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -996,7 +996,7 @@ static int soc_probe_codec(struct snd_soc_card *card, } if (driver->controls) - snd_soc_add_controls(codec, driver->controls, + snd_soc_add_codec_controls(codec, driver->controls, driver->num_controls); if (driver->dapm_routes) snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, @@ -1457,13 +1457,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } - /* We should have a non-codec control add function but we don't */ if (card->controls) - snd_soc_add_controls(list_first_entry(&card->codec_dev_list, - struct snd_soc_codec, - card_list), - card->controls, - card->num_controls); + snd_soc_add_card_controls(card, card->controls, card->num_controls); if (card->dapm_routes) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, @@ -2015,9 +2010,28 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, } EXPORT_SYMBOL_GPL(snd_soc_cnew); +static int snd_soc_add_controls(struct snd_card *card, struct device *dev, + const struct snd_kcontrol_new *controls, int num_controls, + const char *prefix, void *data) +{ + int err, i; + + for (i = 0; i < num_controls; i++) { + const struct snd_kcontrol_new *control = &controls[i]; + err = snd_ctl_add(card, snd_soc_cnew(control, data, + control->name, prefix)); + if (err < 0) { + dev_err(dev, "Failed to add %s: %d\n", control->name, err); + return err; + } + } + + return 0; +} + /** - * snd_soc_add_controls - add an array of controls to a codec. - * Convienience function to add a list of controls. Many codecs were + * snd_soc_add_codec_controls - add an array of controls to a codec. + * Convenience function to add a list of controls. Many codecs were * duplicating this code. * * @codec: codec to add controls to @@ -2026,31 +2040,19 @@ EXPORT_SYMBOL_GPL(snd_soc_cnew); * * Return 0 for success, else error. */ -int snd_soc_add_controls(struct snd_soc_codec *codec, +int snd_soc_add_codec_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = codec->card->snd_card; - int err, i; - for (i = 0; i < num_controls; i++) { - const struct snd_kcontrol_new *control = &controls[i]; - err = snd_ctl_add(card, snd_soc_cnew(control, codec, - control->name, - codec->name_prefix)); - if (err < 0) { - dev_err(codec->dev, "%s: Failed to add %s: %d\n", - codec->name, control->name, err); - return err; - } - } - - return 0; + return snd_soc_add_controls(card, codec->dev, controls, num_controls, + codec->name_prefix, codec); } -EXPORT_SYMBOL_GPL(snd_soc_add_controls); +EXPORT_SYMBOL_GPL(snd_soc_add_codec_controls); /** * snd_soc_add_platform_controls - add an array of controls to a platform. - * Convienience function to add a list of controls. + * Convenience function to add a list of controls. * * @platform: platform to add controls to * @controls: array of controls to add @@ -2062,22 +2064,52 @@ int snd_soc_add_platform_controls(struct snd_soc_platform *platform, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = platform->card->snd_card; - int err, i; - for (i = 0; i < num_controls; i++) { - const struct snd_kcontrol_new *control = &controls[i]; - err = snd_ctl_add(card, snd_soc_cnew(control, platform, - control->name, NULL)); - if (err < 0) { - dev_err(platform->dev, "Failed to add %s %d\n",control->name, err); - return err; - } - } - - return 0; + return snd_soc_add_controls(card, platform->dev, controls, num_controls, + NULL, platform); } EXPORT_SYMBOL_GPL(snd_soc_add_platform_controls); +/** + * snd_soc_add_card_controls - add an array of controls to a SoC card. + * Convenience function to add a list of controls. + * + * @soc_card: SoC card to add controls to + * @controls: array of controls to add + * @num_controls: number of elements in the array + * + * Return 0 for success, else error. + */ +int snd_soc_add_card_controls(struct snd_soc_card *soc_card, + const struct snd_kcontrol_new *controls, int num_controls) +{ + struct snd_card *card = soc_card->snd_card; + + return snd_soc_add_controls(card, soc_card->dev, controls, num_controls, + NULL, soc_card); +} +EXPORT_SYMBOL_GPL(snd_soc_add_card_controls); + +/** + * snd_soc_add_dai_controls - add an array of controls to a DAI. + * Convienience function to add a list of controls. + * + * @dai: DAI to add controls to + * @controls: array of controls to add + * @num_controls: number of elements in the array + * + * Return 0 for success, else error. + */ +int snd_soc_add_dai_controls(struct snd_soc_dai *dai, + const struct snd_kcontrol_new *controls, int num_controls) +{ + struct snd_card *card = dai->card->snd_card; + + return snd_soc_add_controls(card, dai->dev, controls, num_controls, + NULL, dai); +} +EXPORT_SYMBOL_GPL(snd_soc_add_dai_controls); + /** * snd_soc_info_enum_double - enumerated double mixer info callback * @kcontrol: mixer control -- cgit v1.2.3-18-g5258 From fc9e5c6f42f4706dfb9f06f369ddd81f38b0a3fa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Feb 2012 14:46:01 +0100 Subject: ALSA: hda - Remove a debug print in vmaster code Wrongly slipped in from the commit 9322ca54. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8a2f9dddbf0..65c01798d84 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2320,7 +2320,6 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves, name, suffix); name = tmpname; } - printk("XXX comparing %s vs %s\n", sctl->id.name, name); if (!strcmp(sctl->id.name, name)) { err = func(data, sctl); if (err) -- cgit v1.2.3-18-g5258 From 654a28c9dcf74a771318dacf237dd027944621b3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Feb 2012 13:15:18 +0000 Subject: ASoC: samsung: Enable accessory detect for WM8994 on Littlemill The WM8994 has a different accessory detect architecture, call its setup function too. We ignore the errors and the driver will check for chip type so it's safe to call the setup functions for both architectures. Signed-off-by: Mark Brown --- sound/soc/samsung/littlemill.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 9dd818bde06..e7416851bf7 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -189,6 +189,9 @@ static int littlemill_late_probe(struct snd_soc_card *card) /* This will check device compatibility itself */ wm8958_mic_detect(codec, &littlemill_headset, NULL, NULL); + /* As will this */ + wm8994_mic_detect(codec, &littlemill_headset, 1); + return 0; } -- cgit v1.2.3-18-g5258 From 839e5fadc68f5095e4fc76e8e618cc41affdf3d5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Feb 2012 15:22:16 +0000 Subject: ASoC: wm8994: Don't bring up to STANDBY by default In cases where we should enter STANDBY DAPM will power us up, otherwise there is no need to power up and we can remain in OFF. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8ae6585edbe..9bb8192de7f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2759,13 +2759,6 @@ static int wm8994_resume(struct snd_soc_codec *codec) codec->cache_only = 0; } - /* Restore the registers */ - ret = snd_soc_cache_sync(codec); - if (ret != 0) - dev_err(codec->dev, "Failed to sync cache: %d\n", ret); - - wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { if (!wm8994->fll_suspend[i].out) continue; @@ -3574,8 +3567,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->lrclk_shared[1] = 0; } - wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Latch volume updates (right only; we always do left then right). */ snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); -- cgit v1.2.3-18-g5258 From 40f02cd9f21dc2bd2c65713eb986139bb1ea0363 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 6 Feb 2012 16:05:14 +0000 Subject: ASoC: dapm: Export mixer|mux_update_power() to public API. Allow for the operation of custom mixer and mux DAPM widgets that can call snd_soc_dapm_mixer_update_power() and snd_soc_dapm_mux_update_power() directly after updating their status. This is useful with complex DAPM Mixer operations where we need to do additional work in addition to setting a few mixer register bits. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 21 +++++++++------------ 1 file changed, 9 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ec58a314656..e9776a133d8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1699,9 +1699,8 @@ static inline void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) #endif /* test and update the power status of a mux widget */ -static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int change, - int mux, struct soc_enum *e) +int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, + struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -1711,9 +1710,6 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, widget->id != snd_soc_dapm_value_mux) return -ENODEV; - if (!change) - return 0; - /* find dapm widget path assoc with kcontrol */ list_for_each_entry(path, &widget->dapm->card->paths, list) { if (path->kcontrol != kcontrol) @@ -1742,9 +1738,10 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_mux_update_power); /* test and update the power status of a mixer or switch widget */ -static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, +int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kcontrol, int connect) { struct snd_soc_dapm_path *path; @@ -1773,6 +1770,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_mixer_update_power); /* show dapm widget status in sys fs */ static ssize_t dapm_widget_show(struct device *dev, @@ -2357,7 +2355,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - dapm_mixer_update_power(widget, kcontrol, connect); + snd_soc_dapm_mixer_update_power(widget, kcontrol, connect); widget->dapm->update = NULL; } @@ -2448,7 +2446,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - dapm_mux_update_power(widget, kcontrol, change, mux, e); + snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } @@ -2509,8 +2507,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, widget->value = ucontrol->value.enumerated.item[0]; - dapm_mux_update_power(widget, kcontrol, change, - widget->value, e); + snd_soc_dapm_mux_update_power(widget, kcontrol, widget->value, e); } } @@ -2613,7 +2610,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, update.val = val; widget->dapm->update = &update; - dapm_mux_update_power(widget, kcontrol, change, mux, e); + snd_soc_dapm_mux_update_power(widget, kcontrol, mux, e); widget->dapm->update = NULL; } -- cgit v1.2.3-18-g5258 From 612a3fec2188c5a85c5057adf60c470a254e800b Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 6 Feb 2012 16:05:29 +0000 Subject: ASoC: dapm: Clean up header information. Fix some spelling mistakes in the header and remove the todo items. Most todo items are now available as kcontrol options now anyway. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index e9776a133d8..0c94027c4e3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -14,19 +14,13 @@ * dynamic configuration of codec internal audio paths and active * DACs/ADCs. * o Platform power domain - can support external components i.e. amps and - * mic/meadphone insertion events. + * mic/headphone insertion events. * o Automatic Mic Bias support * o Jack insertion power event initiation - e.g. hp insertion will enable * sinks, dacs, etc - * o Delayed powerdown of audio susbsystem to reduce pops between a quick + * o Delayed power down of audio subsystem to reduce pops between a quick * device reopen. * - * Todo: - * o DAPM power change sequencing - allow for configurable per - * codec sequences. - * o Support for analogue bias optimisation. - * o Support for reduced codec oversampling rates. - * o Support for reduced codec bias currents. */ #include -- cgit v1.2.3-18-g5258 From 24cace30c527c1dcbaddb738d9ee39a8ef491818 Mon Sep 17 00:00:00 2001 From: Felipe Contreras Date: Wed, 1 Feb 2012 03:06:20 +0200 Subject: ASoC: tlv320aic3x: remove unused code Looks like nobody is or will be using this code. Signed-off-by: Felipe Contreras Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 36 ------------------------------------ sound/soc/codecs/tlv320aic3x.h | 9 --------- 2 files changed, 45 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0bb7cb8815c..07efbedda1e 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1185,25 +1185,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec, return 0; } -void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state) -{ - u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; - u8 bit = gpio ? 3: 0; - u8 val = snd_soc_read(codec, reg) & ~(1 << bit); - snd_soc_write(codec, reg, val | (!!state << bit)); -} -EXPORT_SYMBOL_GPL(aic3x_set_gpio); - -int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio) -{ - u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG; - u8 val = 0, bit = gpio ? 2 : 1; - - aic3x_read(codec, reg, &val); - return (val >> bit) & 1; -} -EXPORT_SYMBOL_GPL(aic3x_get_gpio); - void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, int headset_debounce, int button_debounce) { @@ -1221,23 +1202,6 @@ void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); } -EXPORT_SYMBOL_GPL(aic3x_set_headset_detection); - -int aic3x_headset_detected(struct snd_soc_codec *codec) -{ - u8 val = 0; - aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val); - return (val >> 4) & 1; -} -EXPORT_SYMBOL_GPL(aic3x_headset_detected); - -int aic3x_button_pressed(struct snd_soc_codec *codec) -{ - u8 val = 0; - aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val); - return (val >> 5) & 1; -} -EXPORT_SYMBOL_GPL(aic3x_button_pressed); #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 06a19784b16..6f097fb6068 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -212,9 +212,6 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 -void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state); -int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio); - /* headset detection / button API */ /* The AIC3x supports detection of stereo headsets (GND + left + right signal) @@ -252,10 +249,4 @@ enum { #define AIC3X_BUTTON_DEBOUNCE_SHIFT 0 #define AIC3X_BUTTON_DEBOUNCE_MASK 3 -/* see the enums above for valid parameters to this function */ -void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, - int headset_debounce, int button_debounce); -int aic3x_headset_detected(struct snd_soc_codec *codec); -int aic3x_button_pressed(struct snd_soc_codec *codec); - #endif /* _AIC3X_H */ -- cgit v1.2.3-18-g5258 From 14ac91126b02cfe39f2bbda40fcbd94923bfabbb Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 6 Feb 2012 16:50:58 +0000 Subject: ASoC: max9768: Fix build and use new add control API. Fix build breakage by using the correct API call to add kcontrols. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/max9768.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 79e99018591..17b3ec2d05c 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -135,7 +135,7 @@ static int max9768_probe(struct snd_soc_codec *codec) } if (gpio_is_valid(max9768->mute_gpio)) { - ret = snd_soc_add_controls(codec, max9768_mute, + ret = snd_soc_add_codec_controls(codec, max9768_mute, ARRAY_SIZE(max9768_mute)); if (ret) return ret; -- cgit v1.2.3-18-g5258 From 5fab517476ad1e4af0043a7b8dd0bd4cdc58df9d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Feb 2012 18:37:08 +0000 Subject: ASoC: wm8994: We don't need to runtime resume by default This is the default state that the runtime PM infrastructure expects so instead just kick the runtime PM core to suspend us if we're not doing anything (as is default). Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9bb8192de7f..6b12f5da6b7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3387,7 +3387,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) WM8994_IRQ_MIC1_DET; pm_runtime_enable(codec->dev); - pm_runtime_resume(codec->dev); + pm_runtime_idle(codec->dev); /* By default use idle_bias_off, will override for WM8994 */ codec->dapm.idle_bias_off = 1; -- cgit v1.2.3-18-g5258 From 27060b3c64a1b9bc0b60c27da6153cf78919fa72 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Feb 2012 18:42:14 +0000 Subject: ASoC: wm8994: Unsuspend the device while reading GPIO statuses Otherwise we might get an error if the GPIO is configured as an input since that makes the register volatile and a suspended device can't be read from. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6b12f5da6b7..6a47c75119b 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3539,6 +3539,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->fll_locked_irq = false; } + /* Make sure we can read from the GPIOs if they're inputs */ + pm_runtime_get_sync(codec->dev); + /* Remember if AIFnLRCLK is configured as a GPIO. This should be * configured on init - if a system wants to do this dynamically * at runtime we can deal with that then. @@ -3567,6 +3570,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->lrclk_shared[1] = 0; } + pm_runtime_put(codec->dev); + /* Latch volume updates (right only; we always do left then right). */ snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); -- cgit v1.2.3-18-g5258 From 87092e3ca4ff7b2c2d7723b3402fe2f74b249bc4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Feb 2012 18:50:39 +0000 Subject: ASoC: wm8994: Bring WM8994 accessory detection up to date Make the mechanism used for WM8994 more like that for WM1811 and WM8958: provide the logic to distinguish between headphone and headset and hard code the reporting of sensible SND_JACK values. Should integration with other detection mechanisms be required we can add appropriate callbacks (though some integrations should be able to use the subsystem ones). Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 82 +++++++++++++++++++++++++++++++++++------------ sound/soc/codecs/wm8994.h | 5 ++- 2 files changed, 63 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6a47c75119b..8aed0e3c613 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2946,8 +2946,6 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) * @codec: WM8994 codec * @jack: jack to report detection events on * @micbias: microphone bias to detect on - * @det: value to report for presence detection - * @shrt: value to report for short detection * * Enable microphone detection via IRQ on the WM8994. If GPIOs are * being used to bring out signals to the processor then only platform @@ -2958,43 +2956,63 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) * and micbias2_lvl platform data members. */ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, - int micbias, int det, int shrt) + int micbias) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994_micdet *micdet; struct wm8994 *control = wm8994->wm8994; - int reg; + int reg, ret; - if (control->type != WM8994) + if (control->type != WM8994) { + dev_warn(codec->dev, "Not a WM8994\n"); return -EINVAL; + } switch (micbias) { case 1: micdet = &wm8994->micdet[0]; + if (jack) + ret = snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS1"); + else + ret = snd_soc_dapm_disable_pin(&codec->dapm, + "MICBIAS1"); break; case 2: micdet = &wm8994->micdet[1]; + if (jack) + ret = snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS1"); + else + ret = snd_soc_dapm_disable_pin(&codec->dapm, + "MICBIAS1"); break; default: + dev_warn(codec->dev, "Invalid MICBIAS %d\n", micbias); return -EINVAL; - } + } - dev_dbg(codec->dev, "Configuring microphone detection on %d: %x %x\n", - micbias, det, shrt); + if (ret != 0) + dev_warn(codec->dev, "Failed to configure MICBIAS%d: %d\n", + micbias, ret); + + dev_dbg(codec->dev, "Configuring microphone detection on %d %p\n", + micbias, jack); /* Store the configuration */ micdet->jack = jack; - micdet->det = det; - micdet->shrt = shrt; + micdet->detecting = true; /* If either of the jacks is set up then enable detection */ if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) reg = WM8994_MICD_ENA; - else + else reg = 0; snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, reg); + snd_soc_dapm_sync(&codec->dapm); + return 0; } EXPORT_SYMBOL_GPL(wm8994_mic_detect); @@ -3020,20 +3038,42 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) dev_dbg(codec->dev, "Microphone status: %x\n", reg); report = 0; - if (reg & WM8994_MIC1_DET_STS) - report |= priv->micdet[0].det; - if (reg & WM8994_MIC1_SHRT_STS) - report |= priv->micdet[0].shrt; + if (reg & WM8994_MIC1_DET_STS) { + if (priv->micdet[0].detecting) + report = SND_JACK_HEADSET; + } + if (reg & WM8994_MIC1_SHRT_STS) { + if (priv->micdet[0].detecting) + report = SND_JACK_HEADPHONE; + else + report |= SND_JACK_BTN_0; + } + if (report) + priv->micdet[0].detecting = false; + else + priv->micdet[0].detecting = true; + snd_soc_jack_report(priv->micdet[0].jack, report, - priv->micdet[0].det | priv->micdet[0].shrt); + SND_JACK_HEADSET | SND_JACK_BTN_0); report = 0; - if (reg & WM8994_MIC2_DET_STS) - report |= priv->micdet[1].det; - if (reg & WM8994_MIC2_SHRT_STS) - report |= priv->micdet[1].shrt; + if (reg & WM8994_MIC2_DET_STS) { + if (priv->micdet[1].detecting) + report = SND_JACK_HEADSET; + } + if (reg & WM8994_MIC2_SHRT_STS) { + if (priv->micdet[1].detecting) + report = SND_JACK_HEADPHONE; + else + report |= SND_JACK_BTN_0; + } + if (report) + priv->micdet[1].detecting = false; + else + priv->micdet[1].detecting = true; + snd_soc_jack_report(priv->micdet[1].jack, report, - priv->micdet[1].det | priv->micdet[1].shrt); + SND_JACK_HEADSET | SND_JACK_BTN_0); return IRQ_HANDLED; } diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index c3a42474ab1..f996d14766d 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -35,7 +35,7 @@ typedef void (*wm8958_micdet_cb)(u16 status, void *data); int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, - int micbias, int det, int shrt); + int micbias); int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8958_micdet_cb cb, void *cb_data); @@ -46,8 +46,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec); struct wm8994_micdet { struct snd_soc_jack *jack; - int det; - int shrt; + bool detecting; }; /* codec private data */ -- cgit v1.2.3-18-g5258 From 6035bbcfc0849450da9fb2389cf946303b41fa08 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Feb 2012 19:19:13 +0000 Subject: ASoC: tlv320aic3x: Remove even more unused code Now that we're not implementing GPIO or microphone detection support there's no users for the read() function any more. Signed-off-by: Mark Brown Acked-by: Jarkko Nikula --- sound/soc/codecs/tlv320aic3x.c | 24 ------------------------ 1 file changed, 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 07efbedda1e..8d20f6ec20f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -121,30 +121,6 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x02, /* 100 */ }; -/* - * read from the aic3x register space. Only use for this function is if - * wanting to read volatile bits from those registers that has both read-only - * and read/write bits. All other cases should use snd_soc_read. - */ -static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg, - u8 *value) -{ - u8 *cache = codec->reg_cache; - - if (codec->cache_only) - return -EINVAL; - if (reg >= AIC3X_CACHEREGNUM) - return -1; - - codec->cache_bypass = 1; - *value = snd_soc_read(codec, reg); - codec->cache_bypass = 0; - - cache[reg] = *value; - - return 0; -} - #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ -- cgit v1.2.3-18-g5258 From dc9c745437fc0d3ce7b2bd71594ea5ac48187f26 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Feb 2012 14:24:57 +0000 Subject: ASoC: wm_hubs: Convert most output drivers to OUT_DRV widgets No practical impact but now we have the control type we may as well use it for the slightly nicer sequencing. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c509911a59f..524828ba8ce 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -654,10 +654,10 @@ SND_SOC_DAPM_MIXER("SPKR Boost", SND_SOC_NOPM, 0, 0, right_speaker_boost, ARRAY_SIZE(right_speaker_boost)), SND_SOC_DAPM_SUPPLY("TSHUT", WM8993_POWER_MANAGEMENT_2, 14, 0, NULL, 0), -SND_SOC_DAPM_PGA("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, - NULL, 0), -SND_SOC_DAPM_PGA("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, - NULL, 0), +SND_SOC_DAPM_OUT_DRV("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, + NULL, 0), SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0, line1_mix, ARRAY_SIZE(line1_mix)), @@ -673,14 +673,14 @@ SND_SOC_DAPM_MIXER("LINEOUT2N Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("LINEOUT2P Mixer", SND_SOC_NOPM, 0, 0, line2p_mix, ARRAY_SIZE(line2p_mix)), -SND_SOC_DAPM_PGA("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, - NULL, 0), -SND_SOC_DAPM_PGA("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, - NULL, 0), -SND_SOC_DAPM_PGA("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, - NULL, 0), -SND_SOC_DAPM_PGA("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, - NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, + NULL, 0), +SND_SOC_DAPM_OUT_DRV("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, + NULL, 0), SND_SOC_DAPM_OUTPUT("SPKOUTLP"), SND_SOC_DAPM_OUTPUT("SPKOUTLN"), -- cgit v1.2.3-18-g5258 From 49915a54f0dceeff15a5d008dc9ce7a1d25f2a98 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Dec 2011 20:30:15 +0800 Subject: ASoC: Convert kirkwood-t5325 to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-t5325.c | 47 +++++++++++++++++++------------------ 1 file changed, 24 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index b47cc4e9b74..f8983635f7e 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -80,7 +80,6 @@ static struct snd_soc_dai_link t5325_dai[] = { }, }; - static struct snd_soc_card t5325 = { .name = "t5325", .owner = THIS_MODULE, @@ -93,38 +92,40 @@ static struct snd_soc_card t5325 = { .num_dapm_routes = ARRAY_SIZE(t5325_route), }; -static struct platform_device *t5325_snd_device; - -static int __init t5325_init(void) +static int __devinit t5325_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &t5325; int ret; - if (!machine_is_t5325()) - return 0; - - t5325_snd_device = platform_device_alloc("soc-audio", -1); - if (!t5325_snd_device) - return -ENOMEM; - - platform_set_drvdata(t5325_snd_device, - &t5325); - - ret = platform_device_add(t5325_snd_device); - if (ret) { - printk(KERN_ERR "%s: platform_device_add failed\n", __func__); - platform_device_put(t5325_snd_device); - } + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } -module_init(t5325_init); -static void __exit t5325_exit(void) +static int __devexit t5325_remove(struct platform_device *pdev) { - platform_device_unregister(t5325_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_exit(t5325_exit); + +static struct platform_driver t5325_driver = { + .driver = { + .name = "t5325-audio", + .owner = THIS_MODULE, + }, + .probe = t5325_probe, + .remove = __devexit_p(t5325_remove), +}; + +module_platform_driver(t5325_driver); MODULE_AUTHOR("Arnaud Patard "); MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:t5325-audio"); -- cgit v1.2.3-18-g5258 From 6d6761aa9b36e2da18669b9948ee6849064a84e9 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 28 Dec 2011 20:32:42 +0800 Subject: ASoC: Convert kirkwood-openrd to use snd_soc_register_card() Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-openrd.c | 46 ++++++++++++++++++------------------ 1 file changed, 23 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 55d2ed3df30..80bd59c33be 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -71,41 +71,41 @@ static struct snd_soc_card openrd_client = { .num_links = ARRAY_SIZE(openrd_client_dai), }; -static struct platform_device *openrd_client_snd_device; - -static int __init openrd_client_init(void) +static int __devinit openrd_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &openrd_client; int ret; - if (!machine_is_openrd_client() && !machine_is_openrd_ultimate()) - return 0; - - openrd_client_snd_device = platform_device_alloc("soc-audio", -1); - if (!openrd_client_snd_device) - return -ENOMEM; - - platform_set_drvdata(openrd_client_snd_device, - &openrd_client); - - ret = platform_device_add(openrd_client_snd_device); - if (ret) { - printk(KERN_ERR "%s: platform_device_add failed\n", __func__); - platform_device_put(openrd_client_snd_device); - } + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } -static void __exit openrd_client_exit(void) +static int __devexit openrd_remove(struct platform_device *pdev) { - platform_device_unregister(openrd_client_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_init(openrd_client_init); -module_exit(openrd_client_exit); +static struct platform_driver openrd_driver = { + .driver = { + .name = "openrd-client-audio", + .owner = THIS_MODULE, + }, + .probe = openrd_probe, + .remove = __devexit_p(openrd_remove), +}; + +module_platform_driver(openrd_driver); /* Module information */ MODULE_AUTHOR("Arnaud Patard "); MODULE_DESCRIPTION("ALSA SoC OpenRD Client"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:soc-audio"); +MODULE_ALIAS("platform:openrd-client-audio"); -- cgit v1.2.3-18-g5258 From 8bc039a1e15a72da8426b84293723fb7181f0b5e Mon Sep 17 00:00:00 2001 From: Seth Heasley Date: Mon, 23 Jan 2012 16:24:31 -0800 Subject: ALSA: hda - Add Lynx Point HD Audio Controller DeviceIDs This patch adds the HD Audio DeviceIDs for the Intel Lynx Point PCH. Signed-off-by: Seth Heasley Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d3bd3e74806..e354c161654 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -148,6 +148,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, PCH}," "{Intel, CPT}," "{Intel, PPT}," + "{Intel, LPT}," "{Intel, PBG}," "{Intel, SCH}," "{ATI, SB450}," @@ -3001,6 +3002,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE}, + /* Lynx Point */ + { PCI_DEVICE(0x8086, 0x8c20), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | -- cgit v1.2.3-18-g5258 From 4c6c0b5eee572a24345fdd1fac6aa670cc937a3a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Feb 2012 19:02:24 +0000 Subject: ASoC: wm8962: Support mono playback and record Make sure we generate enough BCLKs for I2S style modes by always having a minimum of two channels worth of clocks for the BCLK. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 25b6baed3a2..5e3795d7cdb 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2634,6 +2634,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, int adctl3 = 0; wm8962->bclk = snd_soc_params_to_bclk(params); + if (params_channels(params) == 1) + wm8962->bclk *= 2; + wm8962->lrclk = params_rate(params); for (i = 0; i < ARRAY_SIZE(sr_vals); i++) { @@ -3008,14 +3011,14 @@ static struct snd_soc_dai_driver wm8962_dai = { .name = "wm8962", .playback = { .stream_name = "Playback", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8962_RATES, .formats = WM8962_FORMATS, }, .capture = { .stream_name = "Capture", - .channels_min = 2, + .channels_min = 1, .channels_max = 2, .rates = WM8962_RATES, .formats = WM8962_FORMATS, -- cgit v1.2.3-18-g5258 From 945e5038455fef18e73914c149717878d78cb4c0 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 8 Feb 2012 20:33:31 +0000 Subject: ALSA: PCM - Add PCM creation API for internal PCMs. The new ASoC dynamic PCM core needs to create PCMs and substreams that are for use by internal ASoC drivers only and not visible to userspace for direct IO. These new PCMs are similar to regular PCMs expect they have no device nodes or procfs entries. The ASoC component drivers use them in exactly the same way as regular PCMs for PCM and DAI operations. The intention is that a dynamic PCM based driver will register both regular PCMs and internal PCMs. The regular PCMs will be used for all IO with userspace however the internal PCMs will be used by the driver to route digital audio through numerous back end DAI links (with potentially a DSP providing different hw_params, DAI formats based on the regular front end PCM params) to devices like CODECs, MODEMs, Bluetooth, FM, DMICs, etc This patch adds a new snd_pcm_new_internal() API call to create the internal PCM without device nodes or procfs. It also adds adds a new internal flag to snd_pcm. [fixed minor coding-style issues by tiwai] Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/core/pcm.c | 99 +++++++++++++++++++++++++++++++++++++++----------------- 1 file changed, 69 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8928ca871c2..6e4bfcc1425 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -650,7 +650,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) pstr->stream = stream; pstr->pcm = pcm; pstr->substream_count = substream_count; - if (substream_count > 0) { + if (substream_count > 0 && !pcm->internal) { err = snd_pcm_stream_proc_init(pstr); if (err < 0) { snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n"); @@ -674,15 +674,18 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) pstr->substream = substream; else prev->next = substream; - err = snd_pcm_substream_proc_init(substream); - if (err < 0) { - snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n"); - if (prev == NULL) - pstr->substream = NULL; - else - prev->next = NULL; - kfree(substream); - return err; + + if (!pcm->internal) { + err = snd_pcm_substream_proc_init(substream); + if (err < 0) { + snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n"); + if (prev == NULL) + pstr->substream = NULL; + else + prev->next = NULL; + kfree(substream); + return err; + } } substream->group = &substream->self_group; spin_lock_init(&substream->self_group.lock); @@ -696,25 +699,9 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) EXPORT_SYMBOL(snd_pcm_new_stream); -/** - * snd_pcm_new - create a new PCM instance - * @card: the card instance - * @id: the id string - * @device: the device index (zero based) - * @playback_count: the number of substreams for playback - * @capture_count: the number of substreams for capture - * @rpcm: the pointer to store the new pcm instance - * - * Creates a new PCM instance. - * - * The pcm operators have to be set afterwards to the new instance - * via snd_pcm_set_ops(). - * - * Returns zero if successful, or a negative error code on failure. - */ -int snd_pcm_new(struct snd_card *card, const char *id, int device, - int playback_count, int capture_count, - struct snd_pcm ** rpcm) +static int _snd_pcm_new(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, bool internal, + struct snd_pcm **rpcm) { struct snd_pcm *pcm; int err; @@ -735,6 +722,7 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device, } pcm->card = card; pcm->device = device; + pcm->internal = internal; if (id) strlcpy(pcm->id, id, sizeof(pcm->id)); if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) { @@ -756,8 +744,59 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device, return 0; } +/** + * snd_pcm_new - create a new PCM instance + * @card: the card instance + * @id: the id string + * @device: the device index (zero based) + * @playback_count: the number of substreams for playback + * @capture_count: the number of substreams for capture + * @rpcm: the pointer to store the new pcm instance + * + * Creates a new PCM instance. + * + * The pcm operators have to be set afterwards to the new instance + * via snd_pcm_set_ops(). + * + * Returns zero if successful, or a negative error code on failure. + */ +int snd_pcm_new(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, struct snd_pcm **rpcm) +{ + return _snd_pcm_new(card, id, device, playback_count, capture_count, + false, rpcm); +} EXPORT_SYMBOL(snd_pcm_new); +/** + * snd_pcm_new_internal - create a new internal PCM instance + * @card: the card instance + * @id: the id string + * @device: the device index (zero based - shared with normal PCMs) + * @playback_count: the number of substreams for playback + * @capture_count: the number of substreams for capture + * @rpcm: the pointer to store the new pcm instance + * + * Creates a new internal PCM instance with no userspace device or procfs + * entries. This is used by ASoC Back End PCMs in order to create a PCM that + * will only be used internally by kernel drivers. i.e. it cannot be opened + * by userspace. It provides existing ASoC components drivers with a substream + * and access to any private data. + * + * The pcm operators have to be set afterwards to the new instance + * via snd_pcm_set_ops(). + * + * Returns zero if successful, or a negative error code on failure. + */ +int snd_pcm_new_internal(struct snd_card *card, const char *id, int device, + int playback_count, int capture_count, + struct snd_pcm **rpcm) +{ + return _snd_pcm_new(card, id, device, playback_count, capture_count, + true, rpcm); +} +EXPORT_SYMBOL(snd_pcm_new_internal); + static void snd_pcm_free_stream(struct snd_pcm_str * pstr) { struct snd_pcm_substream *substream, *substream_next; @@ -994,7 +1033,7 @@ static int snd_pcm_dev_register(struct snd_device *device) } for (cidx = 0; cidx < 2; cidx++) { int devtype = -1; - if (pcm->streams[cidx].substream == NULL) + if (pcm->streams[cidx].substream == NULL || pcm->internal) continue; switch (cidx) { case SNDRV_PCM_STREAM_PLAYBACK: -- cgit v1.2.3-18-g5258 From b5d1d036eadb30996184cc335c798219dd5922a9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Feb 2012 20:10:56 +0000 Subject: ASoC: pcm: If pmdown_time is zero then shut down DAPM immediately Since we've already got logic to special case immediate teardown of the stream we may as well use it if the pmdown_time has been set to zero by the application layer instead of scheduling a work item with zero delay. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 121318defea..15816eccad3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -369,7 +369,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (codec->ignore_pmdown_time || + if (!rtd->pmdown_time || codec->ignore_pmdown_time || rtd->dai_link->ignore_pmdown_time) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, -- cgit v1.2.3-18-g5258 From 5124e69e2b31f4ded7ed9ac47b18804b7847f677 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Feb 2012 13:20:50 +0000 Subject: ASoC: core: Allow CODECs to set ignore_pmdown_time in the driver struct This is usually not a use case dependant flag anyway. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a3a47cdaac8..32dbcda5cb2 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3312,6 +3312,7 @@ int snd_soc_register_codec(struct device *dev, codec->volatile_register = codec_drv->volatile_register; codec->readable_register = codec_drv->readable_register; codec->writable_register = codec_drv->writable_register; + codec->ignore_pmdown_time = codec_drv->ignore_pmdown_time; codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.dev = dev; codec->dapm.codec = codec; -- cgit v1.2.3-18-g5258 From a5d3a21a9a8c5a2b8332768a3ce9636b630ef664 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Feb 2012 18:34:19 +0000 Subject: ASoC: twl6040: Move ignore_pmdown_time to driver struct It's set unconditionally. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Acked-by: Peter Ujfalusi --- sound/soc/codecs/twl6040.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 284dd2e9997..836eb14f7b8 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1540,7 +1540,6 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->codec = codec; codec->control_data = dev_get_drvdata(codec->dev->parent); - codec->ignore_pmdown_time = 1; if (pdata && pdata->hs_left_step && pdata->hs_right_step) { priv->hs_left_step = pdata->hs_left_step; @@ -1626,6 +1625,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { .reg_cache_size = ARRAY_SIZE(twl6040_reg), .reg_word_size = sizeof(u8), .reg_cache_default = twl6040_reg, + .ignore_pmdown_time = true, .controls = twl6040_snd_controls, .num_controls = ARRAY_SIZE(twl6040_snd_controls), -- cgit v1.2.3-18-g5258 From 17c0cee96f9dd76604e617b74ccca701eebd9727 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Feb 2012 18:35:43 +0000 Subject: ASoC: wm2200: Ignore pmdown_time The device is generally not succeptible to the issues that cause this to be an issue. Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 0db24758c46..a9388dfdbe0 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1916,6 +1916,7 @@ static struct snd_soc_codec_driver soc_codec_wm2200 = { .probe = wm2200_probe, .idle_bias_off = true, + .ignore_pmdown_time = true, .set_sysclk = wm2200_set_sysclk, .set_pll = wm2200_set_fll, -- cgit v1.2.3-18-g5258 From 1f5ff88327b5bc7f5476b4e03425074d07ad25d5 Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Thu, 9 Feb 2012 23:46:08 +0900 Subject: ASoC: Fix typo in twl4030.c Correct spelling "memroy" to "memory" in sound/soc/codecs/twl4030.c Signed-off-by: Masanari Iida Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 3039ba209d1..d7eee0d502e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2221,7 +2221,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); if (twl4030 == NULL) { - printk("Can not allocate memroy\n"); + printk("Can not allocate memory\n"); return -ENOMEM; } snd_soc_codec_set_drvdata(codec, twl4030); -- cgit v1.2.3-18-g5258 From 625136eb4a98e82f27c5d9dcb1b8cf8e0bb5bcef Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 10 Feb 2012 14:54:37 +0800 Subject: ASoC: Fix build error in sound/soc/omap/n810.c Fix below build error which is introduced by commit 022658 "ASoC: core: Add support for DAI and machine kcontrols". CC [M] sound/soc/omap/n810.o sound/soc/omap/n810.c: In function 'n810_set_input': sound/soc/omap/n810.c:194: error: 'codec' undeclared (first use in this function) sound/soc/omap/n810.c:194: error: (Each undeclared identifier is reported only once sound/soc/omap/n810.c:194: error: for each function it appears in.) sound/soc/omap/n810.c:188: warning: unused variable 'card' make[3]: *** [sound/soc/omap/n810.o] Error 1 make[2]: *** [sound/soc/omap] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/n810.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 1490227cd37..c292bf0fd19 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -191,7 +191,7 @@ static int n810_set_input(struct snd_kcontrol *kcontrol, return 0; n810_dmic_func = ucontrol->value.integer.value[0]; - n810_ext_control(codec); + n810_ext_control(&card->dapm); return 1; } -- cgit v1.2.3-18-g5258 From e2da26778c50761b16cc6e37828b37db8248f7ac Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Feb 2012 18:02:58 +0000 Subject: ASoC: wm8993: Correct typos in regmap conversion Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index f814d2711b5..43fa006e52c 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -61,8 +61,8 @@ static struct reg_default wm8993_reg_defaults[] = { { 18, 0x0000 }, /* R18 - GPIO CTRL 1 */ { 19, 0x0010 }, /* R19 - GPIO1 */ { 20, 0x0000 }, /* R20 - IRQ_DEBOUNCE */ - { 21, 0x8000 }, /* R22 - GPIOCTRL 2 */ - { 22, 0x0800 }, /* R23 - GPIO_POL */ + { 22, 0x8000 }, /* R22 - GPIOCTRL 2 */ + { 23, 0x0800 }, /* R23 - GPIO_POL */ { 24, 0x008B }, /* R24 - Left Line Input 1&2 Volume */ { 25, 0x008B }, /* R25 - Left Line Input 3&4 Volume */ { 26, 0x008B }, /* R26 - Right Line Input 1&2 Volume */ -- cgit v1.2.3-18-g5258 From 830eb8767d76b058118be74deac13e5f30b67892 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Fri, 10 Feb 2012 09:17:01 +0100 Subject: ASoC: mxs-saif: use devm_ functions The various devm_ functions allocate memory that is released when a driver detaches. This patch uses these functions for data that is allocated in the probe function of a platform device and is only freed in the remove function. Signed-off-by: Julia Lawall Acked-by: Dong Aisheng Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 51 +++++++++++------------------------------------- 1 file changed, 11 insertions(+), 40 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index f204dbac11d..12be05b1688 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -630,7 +630,7 @@ static int mxs_saif_probe(struct platform_device *pdev) if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; - saif = kzalloc(sizeof(*saif), GFP_KERNEL); + saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; @@ -655,29 +655,16 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", ret); - goto failed_clk; + return ret; } iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) { - ret = -ENODEV; - dev_err(&pdev->dev, "failed to get io resource: %d\n", - ret); - goto failed_get_resource; - } - if (!request_mem_region(iores->start, resource_size(iores), - "mxs-saif")) { - dev_err(&pdev->dev, "request_mem_region failed\n"); - ret = -EBUSY; - goto failed_get_resource; - } - - saif->base = ioremap(iores->start, resource_size(iores)); + saif->base = devm_request_and_ioremap(&pdev->dev, iores); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); ret = -ENODEV; - goto failed_ioremap; + goto failed_get_resource; } dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -685,7 +672,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = -ENODEV; dev_err(&pdev->dev, "failed to get dma resource: %d\n", ret); - goto failed_ioremap; + goto failed_get_resource; } saif->dma_param.chan_num = dmares->start; @@ -694,14 +681,15 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = saif->irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); - goto failed_get_irq1; + goto failed_get_resource; } saif->dev = &pdev->dev; - ret = request_irq(saif->irq, mxs_saif_irq, 0, "mxs-saif", saif); + ret = devm_request_irq(&pdev->dev, saif->irq, mxs_saif_irq, 0, + "mxs-saif", saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); - goto failed_get_irq1; + goto failed_get_resource; } saif->dma_param.chan_irq = platform_get_irq(pdev, 1); @@ -709,7 +697,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = saif->dma_param.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); - goto failed_get_irq2; + goto failed_get_resource; } platform_set_drvdata(pdev, saif); @@ -717,7 +705,7 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_register; + goto failed_get_resource; } saif->soc_platform_pdev = platform_device_alloc( @@ -740,36 +728,19 @@ failed_pdev_add: platform_device_put(saif->soc_platform_pdev); failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); -failed_register: -failed_get_irq2: - free_irq(saif->irq, saif); -failed_get_irq1: - iounmap(saif->base); -failed_ioremap: - release_mem_region(iores->start, resource_size(iores)); failed_get_resource: clk_put(saif->clk); -failed_clk: - kfree(saif); return ret; } static int __devexit mxs_saif_remove(struct platform_device *pdev) { - struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct mxs_saif *saif = platform_get_drvdata(pdev); platform_device_unregister(saif->soc_platform_pdev); - snd_soc_unregister_dai(&pdev->dev); - - iounmap(saif->base); - release_mem_region(res->start, resource_size(res)); - free_irq(saif->irq, saif); - clk_put(saif->clk); - kfree(saif); return 0; } -- cgit v1.2.3-18-g5258 From b803422979b1ab220ee51adc1c59b07eeea13fd7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Feb 2012 18:09:59 +0000 Subject: ASoC: wm8993: Add register default for INPUTS_CLAMP Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 43fa006e52c..ab4685a8a57 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -61,6 +61,7 @@ static struct reg_default wm8993_reg_defaults[] = { { 18, 0x0000 }, /* R18 - GPIO CTRL 1 */ { 19, 0x0010 }, /* R19 - GPIO1 */ { 20, 0x0000 }, /* R20 - IRQ_DEBOUNCE */ + { 21, 0x0000 }, /* R21 - Inputs Clamp */ { 22, 0x8000 }, /* R22 - GPIOCTRL 2 */ { 23, 0x0800 }, /* R23 - GPIO_POL */ { 24, 0x008B }, /* R24 - Left Line Input 1&2 Volume */ -- cgit v1.2.3-18-g5258 From 7fb7528acb219d27bc5a23d22b3ced816656eed2 Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sat, 11 Feb 2012 23:26:27 +0200 Subject: ASoC: alc5632: Coding style. Remove two extra empty lines. Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index f69fb426ad0..0d574aab1b0 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -437,8 +437,6 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"}, - - /* mono mixer */ {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, -- cgit v1.2.3-18-g5258 From 39be1aff5077b76477bfaf3eeebc4c69022784da Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sat, 11 Feb 2012 23:26:28 +0200 Subject: ASoC: alc5632: Fixed voice DAC volume step. Remove extra zero from volume step in DECLARE_TLV_DB_SCALE macro. Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 0d574aab1b0..fce5e76ec28 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -153,7 +153,7 @@ static const unsigned int boost_tlv[] = { /* 0db min scale, 6 db steps, no mute */ static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); /* 0db min scalem 0.75db steps, no mute */ -static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0); +static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 75, 0); static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = { /* left starts at bit 8, right at bit 0 */ -- cgit v1.2.3-18-g5258 From 0a2c056e55029f83dc48b08ec54e6713df5eadf7 Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sat, 11 Feb 2012 23:26:29 +0200 Subject: ASoC: alc5632: Add voice DAC playback switch Add voice DAC playback switch. Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index fce5e76ec28..6bb6075dd24 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -176,6 +176,8 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_AUX_OUT_VOL, 15, 7, 1, 1), SOC_SINGLE_TLV("Voice DAC Playback Volume", ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), + SOC_SINGLE("Voice DAC Playback Switch", + ALC5632_VOICE_DAC_VOL, 12, 1, 1), SOC_SINGLE_TLV("Phone Capture Volume", ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), SOC_DOUBLE_TLV("LineIn Capture Volume", -- cgit v1.2.3-18-g5258 From a15a9af28817034bd059c63033e72fd80e6ee6f5 Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sat, 11 Feb 2012 23:26:30 +0200 Subject: ASoC: alc5632: Rename capture switches to common scheme XXX2REC Rename capture switches to common scheme XXX2REC. Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 56 +++++++++++++++++++++++----------------------- 1 file changed, 28 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 6bb6075dd24..d56218fc7a3 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -246,24 +246,24 @@ SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1), /* Left Record Mixer */ static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = { -SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), -SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), -SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), -SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), -SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), -SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), -SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), +SOC_DAPM_SINGLE("MIC12REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LIL2REC Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("PH2REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPL2REC Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPK2REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MONO2REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1), }; /* Right Record Mixer */ static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = { -SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), -SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), -SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), -SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), -SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), -SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), -SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), +SOC_DAPM_SINGLE("MIC12REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("MIC22REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LIR2REC Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("PH2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPR2REC Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPK2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MONO2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), }; static const char *alc5632_spk_n_sour_sel[] = { @@ -449,22 +449,22 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"}, /* Left record mixer */ - {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, - {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"}, - {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, - {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, - {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, - {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, - {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + {"Left Capture Mix", "LIL2REC Capture Switch", "LINEINL"}, + {"Left Capture Mix", "PH2REC_L Capture Switch", "PHONEN"}, + {"Left Capture Mix", "MIC12REC_L Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "MIC22REC_L Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPL2REC Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPK2REC_L Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MONO2REC_L Capture Switch", "Mono Mix"}, /*Right record mixer */ - {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, - {"Right Capture Mix", "Right Phone Capture Switch", "PHONEP"}, - {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, - {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, - {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, - {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, - {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + {"Right Capture Mix", "LIR2REC Capture Switch", "LINEINR"}, + {"Right Capture Mix", "PH2REC_R Capture Switch", "PHONEP"}, + {"Right Capture Mix", "MIC12REC_R Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "MIC22REC_R Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPR2REC Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPK2REC_R Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MONO2REC_R Capture Switch", "Mono Mix"}, /* headphone left mux */ {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, -- cgit v1.2.3-18-g5258 From 6002c223193608186cdac9d5b7a64738aa4cc35c Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sat, 11 Feb 2012 23:26:31 +0200 Subject: ASoC: alc5632: Remove unexisting route from Phone Mix to Mono Mix There is no Phone Mix<->Mono Mix route in datasheet. Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index d56218fc7a3..11da8064bd1 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -443,7 +443,6 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, - {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"}, {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"}, -- cgit v1.2.3-18-g5258 From 75b3566f796f80918a43be53f369cfc97e949316 Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sat, 11 Feb 2012 23:26:32 +0200 Subject: ASoC: alc5632: Refactored DAPM routes to add voice support Refactored DAPM routes to add voice support. - Added undocumented register - Used AIF in/out - Added missed voice items and routes - Added DMIC input - Romoved unrelevant items Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 85 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/alc5632.h | 1 + 2 files changed, 70 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 11da8064bd1..b1e9af4f341 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -266,16 +266,20 @@ SOC_DAPM_SINGLE("SPK2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), SOC_DAPM_SINGLE("MONO2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), }; -static const char *alc5632_spk_n_sour_sel[] = { +static const char * const alc5632_spk_n_sour_sel[] = { "RN/-R", "RP/+R", "LN/-R", "Mute"}; -static const char *alc5632_hpl_out_input_sel[] = { +static const char * const alc5632_hpl_out_input_sel[] = { "Vmid", "HP Left Mix"}; -static const char *alc5632_hpr_out_input_sel[] = { +static const char * const alc5632_hpr_out_input_sel[] = { "Vmid", "HP Right Mix"}; -static const char *alc5632_spkout_input_sel[] = { +static const char * const alc5632_spkout_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; -static const char *alc5632_aux_out_input_sel[] = { +static const char * const alc5632_aux_out_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char * const alc5632_adcr_func_sel[] = { + "Stereo ADC", "Voice ADC"}; +static const char * const alc5632_i2s_out_sel[] = { + "ADC LR", "Voice Stereo Digital"}; /* auxout output mux */ static const struct soc_enum alc5632_aux_out_input_enum = @@ -314,6 +318,17 @@ static const struct soc_enum alc5632_amp_enum = static const struct snd_kcontrol_new alc5632_amp_mux_controls = SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum); +/* ADC output select */ +static const struct soc_enum alc5632_adcr_func_enum = + SOC_ENUM_SINGLE(ALC5632_DAC_FUNC_SELECT, 5, 2, alc5632_adcr_func_sel); +static const struct snd_kcontrol_new alc5632_adcr_func_controls = + SOC_DAPM_ENUM("ADCR Mux", alc5632_adcr_func_enum); + +/* I2S out select */ +static const struct soc_enum alc5632_i2s_out_enum = + SOC_ENUM_SINGLE(ALC5632_I2S_OUT_CTL, 5, 2, alc5632_i2s_out_sel); +static const struct snd_kcontrol_new alc5632_i2s_out_controls = + SOC_DAPM_ENUM("I2SOut Mux", alc5632_i2s_out_enum); static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = { /* Muxes */ @@ -327,6 +342,10 @@ SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &alc5632_hpr_out_mux_controls), SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, &alc5632_spkoutn_mux_controls), +SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, + &alc5632_adcr_func_controls), +SND_SOC_DAPM_MUX("I2SOut Mux", SND_SOC_NOPM, 0, 0, + &alc5632_i2s_out_controls), /* output mixers */ SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, @@ -354,20 +373,28 @@ SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0, &alc5632_captureR_mixer_controls[0], ARRAY_SIZE(alc5632_captureR_mixer_controls)), -SND_SOC_DAPM_DAC("Left DAC", "HiFi Playback", - ALC5632_PWR_MANAG_ADD2, 9, 0), -SND_SOC_DAPM_DAC("Right DAC", "HiFi Playback", - ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_AIF_IN("AIFRXL", "Left HiFi Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFRXR", "Right HiFi Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFTXL", "Left HiFi Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFTXR", "Right HiFi Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("VAIFRX", "Voice Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("VAIFTX", "Voice Capture", 0, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_DAC("Voice DAC", NULL, ALC5632_PWR_MANAG_ADD2, 10, 0), +SND_SOC_DAPM_DAC("Left DAC", NULL, ALC5632_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", NULL, ALC5632_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_ADC("Left ADC", NULL, ALC5632_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", NULL, ALC5632_PWR_MANAG_ADD2, 6, 0), + SND_SOC_DAPM_MIXER("DAC Left Channel", ALC5632_PWR_MANAG_ADD1, 15, 0, NULL, 0), SND_SOC_DAPM_MIXER("DAC Right Channel", ALC5632_PWR_MANAG_ADD1, 14, 0, NULL, 0), SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0), SND_SOC_DAPM_MIXER("Phone Mix", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), -SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", - ALC5632_PWR_MANAG_ADD2, 7, 0), -SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", - ALC5632_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_MIXER("Voice Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("ADCLR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0), @@ -395,10 +422,12 @@ SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), SND_SOC_DAPM_OUTPUT("SPKOUT"), SND_SOC_DAPM_OUTPUT("SPKOUTN"), + SND_SOC_DAPM_INPUT("LINEINL"), SND_SOC_DAPM_INPUT("LINEINR"), SND_SOC_DAPM_INPUT("PHONEP"), SND_SOC_DAPM_INPUT("PHONEN"), +SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_VMID("Vmid"), @@ -406,6 +435,10 @@ SND_SOC_DAPM_VMID("Vmid"), static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { + /* Playback streams */ + {"Left DAC", NULL, "AIFRXL"}, + {"Right DAC", NULL, "AIFRXR"}, + /* virtual mixer - mixes left & right channels */ {"I2S Mix", NULL, "Left DAC"}, {"I2S Mix", NULL, "Right DAC"}, @@ -428,7 +461,7 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"HP Mix", "PHONE2HP Playback Switch", "Phone Mix"}, {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, - + {"HP Mix", "VOICE2HP Playback Switch", "Voice Mix"}, {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"}, {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"}, @@ -438,6 +471,7 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, {"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"}, + {"Speaker Mix", "VOICE2SPK Playback Switch", "Voice Mix"}, /* mono mixer */ {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, @@ -446,6 +480,7 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, {"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"}, + {"Mono Mix", "VOICE2MONO Playback Switch", "Voice Mix"}, /* Left record mixer */ {"Left Capture Mix", "LIL2REC Capture Switch", "LINEINL"}, @@ -503,10 +538,28 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { /* left ADC */ {"Left ADC", NULL, "Left Capture Mix"}, + {"Left ADC", NULL, "DMICDAT"}, + {"ADCLR", NULL, "Left ADC"}, /* right ADC */ - {"Right ADC", NULL, "Right Capture Mix"}, - + {"Right ADC", NULL, "Right Capture Mix"}, + {"Right ADC", NULL, "DMICDAT"}, + {"ADCR Mux", "Stereo ADC", "Right ADC"}, + {"ADCR Mux", "Voice ADC", "Right ADC"}, + {"ADCLR", NULL, "ADCR Mux"}, + {"VAIFTX", NULL, "ADCR Mux"}, + + /* Digital I2S out */ + {"I2SOut Mux", "ADC LR", "ADCLR"}, + {"I2SOut Mux", "Voice Stereo Digital", "VAIFRX"}, + {"AIFTXL", NULL, "I2SOut Mux"}, + {"AIFTXR", NULL, "I2SOut Mux"}, + + /* Voice Mix */ + {"Voice DAC", NULL, "VAIFRX"}, + {"Voice Mix", NULL, "Voice DAC"}, + + /* Speaker Output */ {"SpeakerOut N Mux", "RN/-R", "Left Speaker"}, {"SpeakerOut N Mux", "RP/+R", "Left Speaker"}, {"SpeakerOut N Mux", "LN/-R", "Left Speaker"}, diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h index 357651ec074..1b5bda594ea 100644 --- a/sound/soc/codecs/alc5632.h +++ b/sound/soc/codecs/alc5632.h @@ -51,6 +51,7 @@ #define ALC5632_ADC_REC_MONOMIX (1 << 0) #define ALC5632_VOICE_DAC_VOL 0x18 /* voice dac vol */ +#define ALC5632_I2S_OUT_CTL 0x1A /* undocumented reg. found in path scheme */ /* ALC5632_OUTPUT_MIXER_CTRL : */ /* same remark as for reg 2 line vs speaker */ #define ALC5632_OUTPUT_MIXER_CTRL 0x1C /* out mix ctrl */ -- cgit v1.2.3-18-g5258 From 5e4e94a958636f002fd5ebcdfb9eb1fa08d9b242 Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sat, 11 Feb 2012 23:26:33 +0200 Subject: ASoC: alc5632: Add DMIC switches and controls Add DMIC switches and controls to ALC5632 codec. Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index b1e9af4f341..b6b95c8c7f6 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -196,8 +196,12 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), SOC_SINGLE_TLV("Mic 2 Boost Volume", ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), - SOC_SINGLE_TLV("Digital Boost Volume", + SOC_SINGLE_TLV("DMIC Boost Volume", ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), + SOC_SINGLE("DMIC En Capture Switch", + ALC5632_DIGI_BOOST_CTRL, 15, 1, 0), + SOC_SINGLE("DMIC PreFilter Capture Switch", + ALC5632_DIGI_BOOST_CTRL, 12, 1, 0), }; /* @@ -266,6 +270,14 @@ SOC_DAPM_SINGLE("SPK2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1), SOC_DAPM_SINGLE("MONO2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1), }; +/* Dmic Mixer */ +static const struct snd_kcontrol_new alc5632_dmicl_mixer_controls[] = { +SOC_DAPM_SINGLE("DMICL2ADC Capture Switch", ALC5632_DIGI_BOOST_CTRL, 7, 1, 1), +}; +static const struct snd_kcontrol_new alc5632_dmicr_mixer_controls[] = { +SOC_DAPM_SINGLE("DMICR2ADC Capture Switch", ALC5632_DIGI_BOOST_CTRL, 6, 1, 1), +}; + static const char * const alc5632_spk_n_sour_sel[] = { "RN/-R", "RP/+R", "LN/-R", "Mute"}; static const char * const alc5632_hpl_out_input_sel[] = { @@ -364,6 +376,12 @@ SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0, SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0, &alc5632_speaker_mixer_controls[0], ARRAY_SIZE(alc5632_speaker_mixer_controls)), +SND_SOC_DAPM_MIXER("DMICL Mix", SND_SOC_NOPM, 0, 0, + &alc5632_dmicl_mixer_controls[0], + ARRAY_SIZE(alc5632_dmicl_mixer_controls)), +SND_SOC_DAPM_MIXER("DMICR Mix", SND_SOC_NOPM, 0, 0, + &alc5632_dmicr_mixer_controls[0], + ARRAY_SIZE(alc5632_dmicr_mixer_controls)), /* input mixers */ SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0, @@ -538,12 +556,14 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { /* left ADC */ {"Left ADC", NULL, "Left Capture Mix"}, - {"Left ADC", NULL, "DMICDAT"}, + {"DMICL Mix", "DMICL2ADC Capture Switch", "DMICDAT"}, + {"Left ADC", NULL, "DMICL Mix"}, {"ADCLR", NULL, "Left ADC"}, /* right ADC */ {"Right ADC", NULL, "Right Capture Mix"}, - {"Right ADC", NULL, "DMICDAT"}, + {"DMICR Mix", "DMICR2ADC Capture Switch", "DMICDAT"}, + {"Right ADC", NULL, "DMICR Mix"}, {"ADCR Mux", "Stereo ADC", "Right ADC"}, {"ADCR Mux", "Voice ADC", "Right ADC"}, {"ADCLR", NULL, "ADCR Mux"}, -- cgit v1.2.3-18-g5258 From 9e75177901b03c3aea40e872f2dce361f2ffd10b Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sat, 11 Feb 2012 23:26:34 +0200 Subject: ASoC: alc5632: Fix I2S digital interface power for recording Fix I2S digital interface power for recording. Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index b6b95c8c7f6..f965560cd15 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -356,7 +356,7 @@ SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, &alc5632_spkoutn_mux_controls), SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0, &alc5632_adcr_func_controls), -SND_SOC_DAPM_MUX("I2SOut Mux", SND_SOC_NOPM, 0, 0, +SND_SOC_DAPM_MUX("I2SOut Mux", ALC5632_PWR_MANAG_ADD1, 11, 0, &alc5632_i2s_out_controls), /* output mixers */ -- cgit v1.2.3-18-g5258 From 2435d46fe61ecc1f00d1b9471808f399f465366c Mon Sep 17 00:00:00 2001 From: Marc Dietrich Date: Sat, 11 Feb 2012 23:26:35 +0200 Subject: ASoC: alc5632: Connect HP/HPL/HPR mix'es to HPOut Mix This patch should fix output through speakers using HP mixer. Signed-off-by: Andrey Danin Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index f965560cd15..69c652cfe85 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -482,6 +482,9 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = { {"HP Mix", "VOICE2HP Playback Switch", "Voice Mix"}, {"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"}, {"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"}, + {"HPOut Mix", NULL, "HP Mix"}, + {"HPOut Mix", NULL, "HPR Mix"}, + {"HPOut Mix", NULL, "HPL Mix"}, /* speaker mixer */ {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, -- cgit v1.2.3-18-g5258 From abb015be83ba88e2ed216d4cb78fe4d86bbd1701 Mon Sep 17 00:00:00 2001 From: Paul Fertser Date: Sat, 11 Feb 2012 23:26:36 +0200 Subject: ASoC: alc5632: Fix Boost Volume TLVs used for the external microphones This brings the TLVs in sync with the documentation and allows to properly manipulate mic boost controls with alsamixer. Signed-off-by: Paul Fertser Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 69c652cfe85..b77f4e7048c 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -145,10 +145,9 @@ static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); /* -16.5db min scale, 1.5db steps, no mute */ static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); static const unsigned int boost_tlv[] = { - TLV_DB_RANGE_HEAD(3), - 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), - 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), - 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), + TLV_DB_RANGE_HEAD(2), + 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), + 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), }; /* 0db min scale, 6 db steps, no mute */ static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); @@ -193,9 +192,9 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { SOC_DOUBLE_TLV("Rec Capture Volume", ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), SOC_SINGLE_TLV("Mic 1 Boost Volume", - ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv), + ALC5632_MIC_CTRL, 10, 3, 0, boost_tlv), SOC_SINGLE_TLV("Mic 2 Boost Volume", - ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv), + ALC5632_MIC_CTRL, 8, 3, 0, boost_tlv), SOC_SINGLE_TLV("DMIC Boost Volume", ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), SOC_SINGLE("DMIC En Capture Switch", -- cgit v1.2.3-18-g5258 From 5148aed2291dc4570b43812f80df0c4d67ce10c8 Mon Sep 17 00:00:00 2001 From: Paul Fertser Date: Sat, 11 Feb 2012 23:26:37 +0200 Subject: ASoC: alc5632: Fix Capture/Playback attributes for microphone inputs According to the mixer path diagram input sources' attenuators logically belong to the playback path and DMIC boost only affects capture. Signed-off-by: Paul Fertser Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index b77f4e7048c..b34761125fd 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -177,17 +177,17 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv), SOC_SINGLE("Voice DAC Playback Switch", ALC5632_VOICE_DAC_VOL, 12, 1, 1), - SOC_SINGLE_TLV("Phone Capture Volume", + SOC_SINGLE_TLV("Phone Playback Volume", ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv), - SOC_DOUBLE_TLV("LineIn Capture Volume", + SOC_DOUBLE_TLV("LineIn Playback Volume", ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), SOC_DOUBLE_TLV("Master Playback Volume", ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv), SOC_DOUBLE("Master Playback Switch", ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1), - SOC_SINGLE_TLV("Mic1 Capture Volume", + SOC_SINGLE_TLV("Mic1 Playback Volume", ALC5632_MIC_VOL, 8, 31, 1, vol_tlv), - SOC_SINGLE_TLV("Mic2 Capture Volume", + SOC_SINGLE_TLV("Mic2 Playback Volume", ALC5632_MIC_VOL, 0, 31, 1, vol_tlv), SOC_DOUBLE_TLV("Rec Capture Volume", ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv), @@ -195,7 +195,7 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = { ALC5632_MIC_CTRL, 10, 3, 0, boost_tlv), SOC_SINGLE_TLV("Mic 2 Boost Volume", ALC5632_MIC_CTRL, 8, 3, 0, boost_tlv), - SOC_SINGLE_TLV("DMIC Boost Volume", + SOC_SINGLE_TLV("DMIC Boost Capture Volume", ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv), SOC_SINGLE("DMIC En Capture Switch", ALC5632_DIGI_BOOST_CTRL, 15, 1, 0), -- cgit v1.2.3-18-g5258 From 0f613c21d652079d39a4906cbe311587705632b5 Mon Sep 17 00:00:00 2001 From: Andrey Danin Date: Sat, 11 Feb 2012 23:26:38 +0200 Subject: ASoC: alc5632: Allow 8kHz stream support. Signed-off-by: Andrey Danin Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index b34761125fd..e2111e0ccad 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -788,6 +788,7 @@ static int alc5632_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); switch (freq) { + case 4096000: case 8192000: case 11289600: case 12288000: -- cgit v1.2.3-18-g5258 From 5f2f38904019bbffb107767c55d9e781c94941ef Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Feb 2012 18:51:42 +0000 Subject: ASoC: wm_hubs: Improve single ended line output enable performance The enable of the single ended line outputs on wm_hubs devices performs better if the output is enabled prior to starting VMID. Since inactive outputs are held at VMID anyway there is little cost to doing this for unused outputs. Add callbacks into wm_hubs and keep track of which outputs are really active so we can disable them once we're active. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 4 ++ sound/soc/codecs/wm8994.c | 5 ++ sound/soc/codecs/wm_hubs.c | 113 +++++++++++++++++++++++++++++++++++++++++---- sound/soc/codecs/wm_hubs.h | 11 +++++ 4 files changed, 125 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index ab4685a8a57..1e69f63ede2 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1058,6 +1058,8 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); int ret; + wm_hubs_set_bias_level(codec, level); + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: @@ -1078,6 +1080,8 @@ static int wm8993_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(wm8993->regmap, false); regcache_sync(wm8993->regmap); + wm_hubs_vmid_ena(codec); + /* Bring up VMID with fast soft start */ snd_soc_update_bits(codec, WM8993_ANTIPOP2, WM8993_STARTUP_BIAS_ENA | diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e6eebf74792..21931a0c7ce 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -787,6 +787,8 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | (0x3 << WM8994_VMID_RAMP_SHIFT)); + wm_hubs_vmid_ena(codec); + /* Remove discharge for line out */ snd_soc_update_bits(codec, WM8994_ANTIPOP_1, WM8994_LINEOUT1_DISCH | @@ -2074,6 +2076,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + wm_hubs_set_bias_level(codec, level); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -2168,6 +2172,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, wm8994->cur_fw = NULL; break; } + codec->dapm.bias_level = level; return 0; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index a53daf64aa4..f7650c5cc5c 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -500,6 +500,36 @@ static int earpiece_event(struct snd_soc_dapm_widget *w, return 0; } +static int lineout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *control, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + bool *flag; + + switch (w->shift) { + case WM8993_LINEOUT1N_ENA_SHIFT: + flag = &hubs->lineout1n_ena; + break; + case WM8993_LINEOUT1P_ENA_SHIFT: + flag = &hubs->lineout1p_ena; + break; + case WM8993_LINEOUT2N_ENA_SHIFT: + flag = &hubs->lineout2n_ena; + break; + case WM8993_LINEOUT2P_ENA_SHIFT: + flag = &hubs->lineout2p_ena; + break; + default: + WARN(1, "Unknown line output"); + return -EINVAL; + } + + *flag = SND_SOC_DAPM_EVENT_ON(event); + + return 0; +} + static const struct snd_kcontrol_new in1l_pga[] = { SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0), SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0), @@ -675,14 +705,18 @@ SND_SOC_DAPM_MIXER("LINEOUT2N Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("LINEOUT2P Mixer", SND_SOC_NOPM, 0, 0, line2p_mix, ARRAY_SIZE(line2p_mix)), -SND_SOC_DAPM_OUT_DRV("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, - NULL, 0), -SND_SOC_DAPM_OUT_DRV("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, - NULL, 0), -SND_SOC_DAPM_OUT_DRV("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, - NULL, 0), -SND_SOC_DAPM_OUT_DRV("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, - NULL, 0), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0, + NULL, 0, lineout_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_OUTPUT("SPKOUTLP"), SND_SOC_DAPM_OUTPUT("SPKOUTLN"), @@ -949,6 +983,11 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, int jd_scthr, int jd_thr, int micbias1_lvl, int micbias2_lvl) { + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + + hubs->lineout1_se = !lineout1_diff; + hubs->lineout2_se = !lineout2_diff; + if (!lineout1_diff) snd_soc_update_bits(codec, WM8993_LINE_MIXER1, WM8993_LINEOUT1_MODE, @@ -978,6 +1017,64 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata); +void wm_hubs_vmid_ena(struct snd_soc_codec *codec) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + int val = 0; + + if (hubs->lineout1_se) + val |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se) + val |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA; + + /* Enable the line outputs while we power up */ + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, val, val); +} +EXPORT_SYMBOL_GPL(wm_hubs_vmid_ena); + +void wm_hubs_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + int val; + + switch (level) { + case SND_SOC_BIAS_ON: + /* Turn off any unneded single ended outputs */ + val = 0; + + if (hubs->lineout1_se && hubs->lineout1n_ena) + val |= WM8993_LINEOUT1N_ENA; + + if (hubs->lineout1_se && hubs->lineout1p_ena) + val |= WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se && hubs->lineout2n_ena) + val |= WM8993_LINEOUT2N_ENA; + + if (hubs->lineout2_se && hubs->lineout2p_ena) + val |= WM8993_LINEOUT2P_ENA; + + snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, + WM8993_LINEOUT1N_ENA | + WM8993_LINEOUT1P_ENA | + WM8993_LINEOUT2N_ENA | + WM8993_LINEOUT2P_ENA, + val); + + if (!hubs->lineout1n_ena && !hubs->lineout1p_ena && + !hubs->lineout2n_ena && !hubs->lineout2p_ena) + snd_soc_update_bits(codec, WM8993_ANTIPOP1, + WM8993_LINEOUT_VMID_BUF_ENA, 0); + break; + + default: + break; + } +} +EXPORT_SYMBOL_GPL(wm_hubs_set_bias_level); + MODULE_DESCRIPTION("Shared support for Wolfson hubs products"); MODULE_AUTHOR("Mark Brown "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index c674c7a502a..4140905c738 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -33,6 +33,14 @@ struct wm_hubs_data { bool class_w; u16 class_w_dcs; + bool lineout1_se; + bool lineout1n_ena; + bool lineout1p_ena; + + bool lineout2_se; + bool lineout2n_ena; + bool lineout2p_ena; + bool dcs_done_irq; struct completion dcs_done; }; @@ -46,5 +54,8 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int micbias1_lvl, int micbias2_lvl); extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); +extern void wm_hubs_vmid_ena(struct snd_soc_codec *codec); +extern void wm_hubs_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level); #endif -- cgit v1.2.3-18-g5258 From d60d6c3b65fa2156ec95d96f73e34cdb0c586458 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 10 Feb 2012 18:09:42 +0000 Subject: ASoC: wm_hubs: Clamp inputs to VMID while we ramp Reduces the amount of time taken to stabilise them. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.h | 9 +++++++++ sound/soc/codecs/wm_hubs.c | 10 ++++++++++ 2 files changed, 19 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.h b/sound/soc/codecs/wm8993.h index 2184617b961..4478b40c86e 100644 --- a/sound/soc/codecs/wm8993.h +++ b/sound/soc/codecs/wm8993.h @@ -31,6 +31,7 @@ #define WM8993_GPIO_CTRL_1 0x12 #define WM8993_GPIO1 0x13 #define WM8993_IRQ_DEBOUNCE 0x14 +#define WM8993_INPUTS_CLAMP_REG 0x15 #define WM8993_GPIOCTRL_2 0x16 #define WM8993_GPIO_POL 0x17 #define WM8993_LEFT_LINE_INPUT_1_2_VOLUME 0x18 @@ -655,6 +656,14 @@ #define WM8993_GPIO1_DB_SHIFT 0 /* GPIO1_DB */ #define WM8993_GPIO1_DB_WIDTH 1 /* GPIO1_DB */ +/* + * R21 (0x15) - Inputs Clamp + */ +#define WM8993_INPUTS_CLAMP 0x0040 /* INPUTS_CLAMP */ +#define WM8993_INPUTS_CLAMP_MASK 0x0040 /* INPUTS_CLAMP */ +#define WM8993_INPUTS_CLAMP_SHIFT 7 /* INPUTS_CLAMP */ +#define WM8993_INPUTS_CLAMP_WIDTH 1 /* INPUTS_CLAMP */ + /* * R22 (0x16) - GPIOCTRL 2 */ diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f7650c5cc5c..9742c666cd0 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1040,6 +1040,12 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, int val; switch (level) { + case SND_SOC_BIAS_STANDBY: + /* Clamp the inputs to VMID while we ramp to charge caps */ + snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, + WM8993_INPUTS_CLAMP, WM8993_INPUTS_CLAMP); + break; + case SND_SOC_BIAS_ON: /* Turn off any unneded single ended outputs */ val = 0; @@ -1067,6 +1073,10 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, !hubs->lineout2n_ena && !hubs->lineout2p_ena) snd_soc_update_bits(codec, WM8993_ANTIPOP1, WM8993_LINEOUT_VMID_BUF_ENA, 0); + + /* Remove the input clamps */ + snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, + WM8993_INPUTS_CLAMP, 0); break; default: -- cgit v1.2.3-18-g5258 From cc6d5a8c2b8c1a04c91bac0a5be02e64307518eb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 11 Feb 2012 23:09:53 +0000 Subject: ASoC: wm8994: VMID management improvements Raise the ramp time to 50ms to cover corner cases, use the startup bias generator, explicitly reset the ramp circuit when complete and reorder things all of which should improve performance somewhat for systems that are sensitive to noise from VMID. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 21931a0c7ce..aa94ca1f6a9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -778,29 +778,37 @@ static void vmid_reference(struct snd_soc_codec *codec) wm8994->vmid_refcount); if (wm8994->vmid_refcount == 1) { + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT_VMID_BUF_ENA | + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT_VMID_BUF_ENA); + /* Startup bias, VMID ramp & buffer */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_VMID_DISCH | WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | (0x3 << WM8994_VMID_RAMP_SHIFT)); wm_hubs_vmid_ena(codec); - /* Remove discharge for line out */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_1, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH, 0); - /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, WM8994_BIAS_ENA | 0x2); - msleep(20); + msleep(50); + + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_VMID_RAMP_MASK | WM8994_BIAS_SRC, + 0); } } -- cgit v1.2.3-18-g5258 From e85b26ce3a4cda67262ae6f7d918a63fd4dc153c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 11 Feb 2012 23:10:30 +0000 Subject: ASoC: wm8994: Actively discharge VMID when not in use Ensure we're in a known state when we restart. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index aa94ca1f6a9..3a69ec0d5af 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -838,6 +838,10 @@ static void vmid_dereference(struct snd_soc_codec *codec) WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0); + /* Discharge VMID */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_VMID_DISCH, WM8994_VMID_DISCH); + /* Discharge line */ snd_soc_update_bits(codec, WM8994_ANTIPOP_1, WM8994_LINEOUT1_DISCH | -- cgit v1.2.3-18-g5258 From a9c74173f4be2a536ef8d8c88642e41527f2d8b4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Feb 2012 11:41:42 +0100 Subject: ALSA: hda - Make is_jack_detectable() as non-inlined It's a bit too big and used too often as an inline function. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_jack.c | 14 ++++++++++++++ sound/pci/hda/hda_jack.h | 13 +------------ 2 files changed, 15 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 9d819c4b492..ac7e57c4039 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -19,6 +19,20 @@ #include "hda_local.h" #include "hda_jack.h" +bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) +{ + if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) + return false; + if (!codec->ignore_misc_bit && + (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & + AC_DEFCFG_MISC_NO_PRESENCE)) + return false; + if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) + return false; + return true; +} +EXPORT_SYMBOL_HDA(is_jack_detectable); + /* execute pin sense measurement */ static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid) { diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index f8f97c71c9c..c66655cf413 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -62,18 +62,7 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid, u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); -static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) -{ - if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) - return false; - if (!codec->ignore_misc_bit && - (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) & - AC_DEFCFG_MISC_NO_PRESENCE)) - return false; - if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP)) - return false; - return true; -} +bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, const char *name, int idx); -- cgit v1.2.3-18-g5258 From 71b1e9e43d5199f57c109e20c0f4dffc5c048130 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Feb 2012 11:55:02 +0100 Subject: ALSA: hda - Add codec->no_jack_detect flag Add a new flag to indicate that the codec has no jack-detection cap. This flag should be set for hardwares that have no jack-detect implementation although the codec chip itself supports it. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_jack.c | 2 ++ 2 files changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index e9f71dc0d46..654d2e41e25 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -852,6 +852,7 @@ struct hda_codec { unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ + unsigned int no_jack_detect:1; /* Machine has no jack-detection */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ unsigned int power_transition :1; /* power-state in transition */ diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index ac7e57c4039..d68948499fb 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -21,6 +21,8 @@ bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) { + if (codec->no_jack_detect) + return false; if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT)) return false; if (!codec->ignore_misc_bit && -- cgit v1.2.3-18-g5258 From e652f4c861fb7f1f59ff0828db0d85578471932d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Feb 2012 11:56:25 +0100 Subject: ALSA: hda - Suppress auto-mute feature on some machines with ALC861 A few machines with ALC861 & co are reported not to work properly with the auto-mute feature in software. The auto-mute feature is implemented in the hardware level, and the jack-detection never works with them. Also, rename the fixup index as ALC861_FIXUP_* to follow the standard. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 38 ++++++++++++++++++++++++++++++-------- 1 file changed, 30 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4a2a49fd92a..c6305984816 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5563,8 +5563,10 @@ static const struct hda_amp_list alc861_loopbacks[] = { /* Pin config fixes */ enum { - PINFIX_FSC_AMILO_PI1505, - PINFIX_ASUS_A6RP, + ALC861_FIXUP_FSC_AMILO_PI1505, + ALC861_FIXUP_AMP_VREF_0F, + ALC861_FIXUP_NO_JACK_DETECT, + ALC861_FIXUP_ASUS_A6RP, }; /* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */ @@ -5586,8 +5588,16 @@ static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec, spec->keep_vref_in_automute = 1; } +/* suppress the jack-detection */ +static void alc_fixup_no_jack_detect(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PRE_PROBE) + codec->no_jack_detect = 1; +} + static const struct alc_fixup alc861_fixups[] = { - [PINFIX_FSC_AMILO_PI1505] = { + [ALC861_FIXUP_FSC_AMILO_PI1505] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x0b, 0x0221101f }, /* HP */ @@ -5595,17 +5605,29 @@ static const struct alc_fixup alc861_fixups[] = { { } } }, - [PINFIX_ASUS_A6RP] = { + [ALC861_FIXUP_AMP_VREF_0F] = { .type = ALC_FIXUP_FUNC, .v.func = alc861_fixup_asus_amp_vref_0f, }, + [ALC861_FIXUP_NO_JACK_DETECT] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_no_jack_detect, + }, + [ALC861_FIXUP_ASUS_A6RP] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc861_fixup_asus_amp_vref_0f, + .chained = true, + .chain_id = ALC861_FIXUP_NO_JACK_DETECT, + } }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { - SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", PINFIX_ASUS_A6RP), - SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", PINFIX_ASUS_A6RP), - SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", PINFIX_ASUS_A6RP), - SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), + SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), + SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), + SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F), + SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", ALC861_FIXUP_AMP_VREF_0F), + SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", ALC861_FIXUP_FSC_AMILO_PI1505), {} }; -- cgit v1.2.3-18-g5258 From 1565cc358585be40608b46f18f7ac431a1aae2bc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Feb 2012 12:03:25 +0100 Subject: ALSA: hda - Add another jack-detection suppression for ASUS ALC892 Add the jack-detect suppression for an ASUS machine with ALC892 codec. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42655 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c6305984816..30ef877e628 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5875,6 +5875,7 @@ enum { ALC662_FIXUP_ASUS_MODE6, ALC662_FIXUP_ASUS_MODE7, ALC662_FIXUP_ASUS_MODE8, + ALC662_FIXUP_NO_JACK_DETECT, }; static const struct alc_fixup alc662_fixups[] = { @@ -6020,6 +6021,10 @@ static const struct alc_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_SKU_IGNORE }, + [ALC662_FIXUP_NO_JACK_DETECT] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc_fixup_no_jack_detect, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6028,6 +6033,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), -- cgit v1.2.3-18-g5258 From dd7d3a2417ba3358dfeb10c085c4d38035d352d4 Mon Sep 17 00:00:00 2001 From: Leon Romanovsky Date: Mon, 13 Feb 2012 21:27:36 +0200 Subject: ASoC: tegra+alc5632: Added digital microphone DAPM widget. ALC5632 codec supports digital microphone. This patch adds DAPM widget. Signed-off-by: Leon Romanovsky Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index c000f51c2ff..2a27725cc9b 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -102,6 +102,7 @@ static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { SND_SOC_DAPM_SPK("Int Spk", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Digital Mic", NULL), }; static const struct snd_kcontrol_new tegra_alc5632_controls[] = { -- cgit v1.2.3-18-g5258 From 2dbc34d827d17e2f4926d4424b5e41c25c70238a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 12 Feb 2012 18:39:49 +0000 Subject: ASoC: wm9081: Use module_i2c_driver Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 23 +---------------------- 1 file changed, 1 insertion(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 7b09b1f86db..ebd3a8a03a2 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1435,28 +1435,7 @@ static struct i2c_driver wm9081_i2c_driver = { }; #endif -static int __init wm9081_modinit(void) -{ - int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&wm9081_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM9081 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(wm9081_modinit); - -static void __exit wm9081_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&wm9081_i2c_driver); -#endif -} -module_exit(wm9081_exit); - +module_i2c_driver(wm9081_i2c_driver); MODULE_DESCRIPTION("ASoC WM9081 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.3-18-g5258 From d2dc0a7782d3f257789e8650b3fa2586b96c6436 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 12 Feb 2012 18:21:57 +0000 Subject: ASoC: wm8988: Convert to direct regmap API usage Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 140 +++++++++++++++++++++++++++++++++++++++------- 1 file changed, 120 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 4ef9d4cb7d7..2470321a0ee 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -33,24 +33,89 @@ * We can't read the WM8988 register space when we * are using 2 wire for device control, so we cache them instead. */ -static const u16 wm8988_reg[] = { - 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ - 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ - 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ - 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ - 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ - 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ - 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ - 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ - 0x0079, 0x0079, 0x0079, /* 40 */ +static const struct reg_default wm8988_reg_defaults[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x0079 }, + { 3, 0x0079 }, + { 5, 0x0008 }, + { 7, 0x000a }, + { 8, 0x0000 }, + { 10, 0x00ff }, + { 11, 0x00ff }, + { 12, 0x000f }, + { 13, 0x000f }, + { 16, 0x0000 }, + { 17, 0x007b }, + { 18, 0x0000 }, + { 19, 0x0032 }, + { 20, 0x0000 }, + { 21, 0x00c3 }, + { 22, 0x00c3 }, + { 23, 0x00c0 }, + { 24, 0x0000 }, + { 25, 0x0000 }, + { 26, 0x0000 }, + { 27, 0x0000 }, + { 31, 0x0000 }, + { 32, 0x0000 }, + { 33, 0x0000 }, + { 34, 0x0050 }, + { 35, 0x0050 }, + { 36, 0x0050 }, + { 37, 0x0050 }, + { 40, 0x0079 }, + { 41, 0x0079 }, + { 42, 0x0079 }, }; +static bool wm8988_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM8988_LINVOL: + case WM8988_RINVOL: + case WM8988_LOUT1V: + case WM8988_ROUT1V: + case WM8988_ADCDAC: + case WM8988_IFACE: + case WM8988_SRATE: + case WM8988_LDAC: + case WM8988_RDAC: + case WM8988_BASS: + case WM8988_TREBLE: + case WM8988_RESET: + case WM8988_3D: + case WM8988_ALC1: + case WM8988_ALC2: + case WM8988_ALC3: + case WM8988_NGATE: + case WM8988_LADC: + case WM8988_RADC: + case WM8988_ADCTL1: + case WM8988_ADCTL2: + case WM8988_PWR1: + case WM8988_PWR2: + case WM8988_ADCTL3: + case WM8988_ADCIN: + case WM8988_LADCIN: + case WM8988_RADCIN: + case WM8988_LOUTM1: + case WM8988_LOUTM2: + case WM8988_ROUTM1: + case WM8988_ROUTM2: + case WM8988_LOUT2V: + case WM8988_ROUT2V: + case WM8988_LPPB: + return true; + default: + return false; + } +} + /* codec private data */ struct wm8988_priv { + struct regmap *regmap; unsigned int sysclk; - enum snd_soc_control_type control_type; struct snd_pcm_hw_constraint_list *sysclk_constraints; }; @@ -661,6 +726,7 @@ static int wm8988_mute(struct snd_soc_dai *dai, int mute) static int wm8988_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); u16 pwr_reg = snd_soc_read(codec, WM8988_PWR1) & ~0x1c1; switch (level) { @@ -674,7 +740,7 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - snd_soc_cache_sync(codec); + regcache_sync(wm8988->regmap); /* VREF, VMID=2x5k */ snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); @@ -730,7 +796,10 @@ static struct snd_soc_dai_driver wm8988_dai = { static int wm8988_suspend(struct snd_soc_codec *codec) { + struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); + wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_mark_dirty(wm8988->regmap); return 0; } @@ -745,7 +814,8 @@ static int wm8988_probe(struct snd_soc_codec *codec) struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8988->control_type); + codec->control_data = wm8988->regmap; + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -781,9 +851,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .suspend = wm8988_suspend, .resume = wm8988_resume, .set_bias_level = wm8988_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm8988_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm8988_reg, .controls = wm8988_snd_controls, .num_controls = ARRAY_SIZE(wm8988_snd_controls), @@ -793,6 +860,18 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .num_dapm_routes = ARRAY_SIZE(wm8988_dapm_routes), }; +static struct regmap_config wm8988_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = WM8988_LPPB, + .writeable_reg = wm8988_writeable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = wm8988_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(wm8988_reg_defaults), +}; + #if defined(CONFIG_SPI_MASTER) static int __devinit wm8988_spi_probe(struct spi_device *spi) { @@ -804,18 +883,28 @@ static int __devinit wm8988_spi_probe(struct spi_device *spi) if (wm8988 == NULL) return -ENOMEM; - wm8988->control_type = SND_SOC_SPI; + wm8988->regmap = regmap_init_spi(spi, &wm8988_regmap); + if (IS_ERR(wm8988->regmap)) { + ret = PTR_ERR(wm8988->regmap); + dev_err(&spi->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + spi_set_drvdata(spi, wm8988); ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8988, &wm8988_dai, 1); + if (ret != 0) + regmap_exit(wm8988->regmap); return ret; } static int __devexit wm8988_spi_remove(struct spi_device *spi) { + struct wm8988_priv *wm8988 = spi_get_drvdata(spi); snd_soc_unregister_codec(&spi->dev); + regmap_exit(wm8988->regmap); return 0; } @@ -842,16 +931,27 @@ static __devinit int wm8988_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm8988); - wm8988->control_type = SND_SOC_I2C; + + wm8988->regmap = regmap_init_i2c(i2c, &wm8988_regmap); + if (IS_ERR(wm8988->regmap)) { + ret = PTR_ERR(wm8988->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8988, &wm8988_dai, 1); + if (ret != 0) + regmap_exit(wm8988->regmap); + return ret; } static __devexit int wm8988_i2c_remove(struct i2c_client *client) { + struct wm8988_priv *wm8988 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); + regmap_exit(wm8988->regmap); return 0; } -- cgit v1.2.3-18-g5258 From 1f5cafb2c88efc8d8ab611917282c04e85ceaf27 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 12 Feb 2012 18:22:15 +0000 Subject: ASoC: wm8988: Remove unneded -codec from driver name Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 2470321a0ee..6cdf6a2bc28 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -963,7 +963,7 @@ MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id); static struct i2c_driver wm8988_i2c_driver = { .driver = { - .name = "wm8988-codec", + .name = "wm8988", .owner = THIS_MODULE, }, .probe = wm8988_i2c_probe, -- cgit v1.2.3-18-g5258 From 68fcde97e42af0ddd4a3665aeec0f5286577f0b5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 12 Feb 2012 18:31:26 +0000 Subject: ASoC: wm9081: Move WM9081 IRQ platform data handling into I2C probe Better style and better supports idle_bias_off which we're going to implement. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index ebd3a8a03a2..c4b3fd923f2 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1258,7 +1258,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); int ret; - u16 reg; codec->control_data = wm9081->regmap; @@ -1268,16 +1267,6 @@ static int wm9081_probe(struct snd_soc_codec *codec) return ret; } - reg = 0; - if (wm9081->pdata.irq_high) - reg |= WM9081_IRQ_POL; - if (!wm9081->pdata.irq_cmos) - reg |= WM9081_IRQ_OP_CTRL; - snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL, - WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); - - wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Enable zero cross by default */ snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, WM9081_LINEOUTZC, WM9081_LINEOUTZC); @@ -1395,6 +1384,15 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), sizeof(wm9081->pdata)); + reg = 0; + if (wm9081->pdata.irq_high) + reg |= WM9081_IRQ_POL; + if (!wm9081->pdata.irq_cmos) + reg |= WM9081_IRQ_OP_CTRL; + regmap_update_bits(wm9081->regmap, WM9081_INTERRUPT_CONTROL, + WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); if (ret < 0) -- cgit v1.2.3-18-g5258 From da157875cd6f3f282b9404e35d1507d8bbbd4c34 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 12 Feb 2012 18:37:03 +0000 Subject: ASoC: wm9081: Use idle_bias_off The main role of the WM9081 is as a class D speaker amplifier so there is no concern about pops. There are also very few registers and a fast power up time so we can happily mark the driver as idle_bias_off. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 35 ++++++++++------------------------- 1 file changed, 10 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index c4b3fd923f2..076c126ed9b 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -824,6 +824,8 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { static int wm9081_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; @@ -841,6 +843,9 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + regcache_cache_only(wm9081->regmap, false); + regcache_sync(wm9081->regmap); + /* Disable LINEOUT discharge */ snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, WM9081_LINEOUT_DISCH, 0); @@ -892,6 +897,8 @@ static int wm9081_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, WM9081_LINEOUT_DISCH, WM9081_LINEOUT_DISCH); + + regcache_cache_only(wm9081->regmap, true); break; } @@ -1289,38 +1296,15 @@ static int wm9081_remove(struct snd_soc_codec *codec) return 0; } -#ifdef CONFIG_PM -static int wm9081_suspend(struct snd_soc_codec *codec) -{ - wm9081_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int wm9081_resume(struct snd_soc_codec *codec) -{ - struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); - - regcache_sync(wm9081->regmap); - - wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define wm9081_suspend NULL -#define wm9081_resume NULL -#endif - static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .probe = wm9081_probe, .remove = wm9081_remove, - .suspend = wm9081_suspend, - .resume = wm9081_resume, .set_sysclk = wm9081_set_sysclk, .set_bias_level = wm9081_set_bias_level, + .idle_bias_off = true, + .controls = wm9081_snd_controls, .num_controls = ARRAY_SIZE(wm9081_snd_controls), .dapm_widgets = wm9081_dapm_widgets, @@ -1392,6 +1376,7 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, regmap_update_bits(wm9081->regmap, WM9081_INTERRUPT_CONTROL, WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); + regcache_cache_only(wm9081->regmap, true); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); -- cgit v1.2.3-18-g5258 From 65f01ef09ee601aa04dfbe6c4f08193668461a6b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 14 Feb 2012 17:53:55 -0800 Subject: ASoC: wm8994: Use slow start for VMID Improves performance on power up. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3a69ec0d5af..74794818e1f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -794,7 +794,7 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | - (0x3 << WM8994_VMID_RAMP_SHIFT)); + (0x2 << WM8994_VMID_RAMP_SHIFT)); wm_hubs_vmid_ena(codec); -- cgit v1.2.3-18-g5258 From 6e8d5d2f17e707ecfabd33fd5fa167ac7739326e Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Wed, 15 Feb 2012 00:38:55 +0900 Subject: ALSA: usx2y: Fix typo in usbusx2yaudio.c and usx2yhwdeppcm.c Correct spelling "propably" to "probably" and "activ" to "active" in sound/usb/usx2y/usbusx2yaudio.c and usx2yhwdeppcm.c Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usbusx2yaudio.c | 4 ++-- sound/usb/usx2y/usx2yhwdeppcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 6ffb3713b60..520ef96d7c7 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -80,7 +80,7 @@ static int usX2Y_urb_capt_retire(struct snd_usX2Y_substream *subs) cp = (unsigned char*)urb->transfer_buffer + urb->iso_frame_desc[i].offset; if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */ snd_printk(KERN_ERR "active frame status %i. " - "Most propably some hardware problem.\n", + "Most probably some hardware problem.\n", urb->iso_frame_desc[i].status); return urb->iso_frame_desc[i].status; } @@ -300,7 +300,7 @@ static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, { snd_printk(KERN_ERR "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most propably some urb of usb-frame %i is still missing.\n" +"Most probably some urb of usb-frame %i is still missing.\n" "Cause could be too long delays in usb-hcd interrupt handling.\n", usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index a51340f6f2d..8e40b6e67e9 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -74,7 +74,7 @@ static int usX2Y_usbpcm_urb_capt_retire(struct snd_usX2Y_substream *subs) } for (i = 0; i < nr_of_packs(); i++) { if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */ - snd_printk(KERN_ERR "activ frame status %i. Most propably some hardware problem.\n", urb->iso_frame_desc[i].status); + snd_printk(KERN_ERR "active frame status %i. Most probably some hardware problem.\n", urb->iso_frame_desc[i].status); return urb->iso_frame_desc[i].status; } lens += urb->iso_frame_desc[i].actual_length / usX2Y->stride; -- cgit v1.2.3-18-g5258 From 6c120e19fa587710d80757a6e364961a017fb6c3 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 15 Feb 2012 15:15:34 +0000 Subject: ASoC: dapm - Make DAPM reset code a separate function. It's useful to export the DAPM reset as a static function for future use by other DAPM functions. e.g. The dynamic PCM query widgets resets the DAPM graph before working out active paths. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0c94027c4e3..227887e05b7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -169,6 +169,19 @@ static inline struct snd_soc_card *dapm_get_soc_card( return NULL; } +static void dapm_reset(struct snd_soc_card *card) +{ + struct snd_soc_dapm_widget *w; + + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); + + list_for_each_entry(w, &card->widgets, list) { + w->power_checked = false; + w->inputs = -1; + w->outputs = -1; + } +} + static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) { if (w->codec) @@ -1402,13 +1415,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } - memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); - - list_for_each_entry(w, &card->widgets, list) { - w->power_checked = false; - w->inputs = -1; - w->outputs = -1; - } + dapm_reset(card); /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. We -- cgit v1.2.3-18-g5258 From 8078d87f9d1383331289f78ea9b96b190d2a528f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 15 Feb 2012 15:15:35 +0000 Subject: ASoC: dapm: Notify stream event to all card components. Currently when DAPM widgets are power sequenced the stream_event() completion callback is only called for the stream_event originator DAPM context. Other components in the card may also be interested so make sure they are also notified of any widget power events. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 227887e05b7..63a5614d33c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1516,6 +1516,12 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) &async_domain); async_synchronize_full_domain(&async_domain); + /* do we need to notify any clients that DAPM event is complete */ + list_for_each_entry(d, &card->dapm_list, list) { + if (d->stream_event) + d->stream_event(d, event); + } + pop_dbg(dapm->dev, card->pop_time, "DAPM sequencing finished, waiting %dms\n", card->pop_time); pop_wait(card->pop_time); @@ -2854,10 +2860,6 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, } dapm_power_widgets(dapm, event); - - /* do we need to notify any clients that DAPM stream is complete */ - if (dapm->stream_event) - dapm->stream_event(dapm, event); } /** -- cgit v1.2.3-18-g5258 From 64e60f9f9e59f861fe9b79802b3cb03ca06b422f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 15 Feb 2012 15:15:37 +0000 Subject: ASoC: core: Convert CODEC debugfs init to use dev_warn() Update the codec debugfs initialisation to use dev_warn() instead of printk(KERN_WARNING). Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8bd999530a7..50b8b80acdf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -277,8 +277,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) codec->debugfs_codec_root = debugfs_create_dir(codec->name, debugfs_card_root); if (!codec->debugfs_codec_root) { - printk(KERN_WARNING - "ASoC: Failed to create codec debugfs directory\n"); + dev_warn(codec->dev, "Failed to create codec debugfs directory\n"); return; } @@ -291,8 +290,7 @@ static void soc_init_codec_debugfs(struct snd_soc_codec *codec) codec->debugfs_codec_root, codec, &codec_reg_fops); if (!codec->debugfs_reg) - printk(KERN_WARNING - "ASoC: Failed to create codec register debugfs file\n"); + dev_warn(codec->dev, "Failed to create codec register debugfs file\n"); snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root); } -- cgit v1.2.3-18-g5258 From 48a8c3943d1010c81d8144cc773f81c30bf59246 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 14 Feb 2012 17:11:15 -0800 Subject: ASoC: dapm: Convert pin switches to use snd_soc_card Since the addition of the non-CODEC control adds card controls like the DAPM pin switch have been broken as they are expecting the private data for the control to be the CODEC but it's now the card. Fix that for the pin switches, an audit of other drivers is required. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 63a5614d33c..0740cfc3d99 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2657,15 +2657,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_info_pin_switch); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&codec->mutex); + mutex_lock(&card->mutex); ucontrol->value.integer.value[0] = - snd_soc_dapm_get_pin_status(&codec->dapm, pin); + snd_soc_dapm_get_pin_status(&card->dapm, pin); - mutex_unlock(&codec->mutex); + mutex_unlock(&card->mutex); return 0; } @@ -2680,19 +2680,19 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_switch); int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock(&codec->mutex); + mutex_lock(&card->mutex); if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(&codec->dapm, pin); + snd_soc_dapm_enable_pin(&card->dapm, pin); else - snd_soc_dapm_disable_pin(&codec->dapm, pin); + snd_soc_dapm_disable_pin(&card->dapm, pin); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync(&card->dapm); - mutex_unlock(&codec->mutex); + mutex_unlock(&card->mutex); return 0; } -- cgit v1.2.3-18-g5258 From 731f1ab290ca1e59430ab222290d379222eb38a5 Mon Sep 17 00:00:00 2001 From: Sebastien Guiriec Date: Wed, 15 Feb 2012 15:25:31 +0000 Subject: ASoC: core: add platform DAPM debugfs support Allow platform widgets to be visible in debugfs like codec widgets. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 50b8b80acdf..6bad7cd4131 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -300,6 +300,27 @@ static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) debugfs_remove_recursive(codec->debugfs_codec_root); } +static void soc_init_platform_debugfs(struct snd_soc_platform *platform) +{ + struct dentry *debugfs_card_root = platform->card->debugfs_card_root; + + platform->debugfs_platform_root = debugfs_create_dir(platform->name, + debugfs_card_root); + if (!platform->debugfs_platform_root) { + dev_warn(platform->dev, + "Failed to create platform debugfs directory\n"); + return; + } + + snd_soc_dapm_debugfs_init(&platform->dapm, + platform->debugfs_platform_root); +} + +static void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) +{ + debugfs_remove_recursive(platform->debugfs_platform_root); +} + static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { @@ -433,6 +454,14 @@ static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) { } +static inline void soc_init_platform_debugfs(struct snd_soc_platform *platform) +{ +} + +static inline void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) +{ +} + static inline void soc_init_card_debugfs(struct snd_soc_card *card) { } @@ -920,6 +949,7 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) /* Make sure all DAPM widgets are freed */ snd_soc_dapm_free(&platform->dapm); + soc_cleanup_platform_debugfs(platform); platform->probed = 0; list_del(&platform->card_list); module_put(platform->dev->driver->owner); @@ -1037,6 +1067,8 @@ static int soc_probe_platform(struct snd_soc_card *card, if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; + soc_init_platform_debugfs(platform); + if (driver->dapm_widgets) snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); -- cgit v1.2.3-18-g5258 From 916be22c150f89639976700c2023ca9b2a5ccebd Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 16 Feb 2012 10:05:59 +0800 Subject: ASoC: Get correct revision id for wm2200 Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index a9388dfdbe0..acbdc5fde92 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2084,7 +2084,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, goto err_reset; } - wm2200->rev = ret & WM2200_DEVICE_REVISION_MASK; + wm2200->rev = reg & WM2200_DEVICE_REVISION_MASK; dev_info(&i2c->dev, "revision %c\n", wm2200->rev + 'A'); -- cgit v1.2.3-18-g5258 From 905b41956c81e5914470a13a166d4b4c4ee71f92 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Thu, 16 Feb 2012 10:33:45 +0800 Subject: ASoC: Show device id in the debug message Show the id we read when the id mismatch is detected. This is useful for debugging. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 172dcacb828..5a6516ee06a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3718,7 +3718,7 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, } if (reg != 0x6243) { dev_err(&i2c->dev, - "Device is not a WM8962, ID %x != 0x6243\n", ret); + "Device is not a WM8962, ID %x != 0x6243\n", reg); ret = -EINVAL; goto err_regmap; } diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 759ea69f0b4..c3bde4a6b7f 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3185,7 +3185,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, goto err_regmap; } if (reg != 0x8915) { - dev_err(&i2c->dev, "Device is not a WM8996, ID %x\n", ret); + dev_err(&i2c->dev, "Device is not a WM8996, ID %x\n", reg); ret = -EINVAL; goto err_regmap; } diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index e8280eecd4c..4b263b6edf1 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -639,7 +639,7 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, if (ret < 0) goto err; if (reg != 0x9093) { - dev_err(&i2c->dev, "Device is not a WM9090, ID=%x\n", ret); + dev_err(&i2c->dev, "Device is not a WM9090, ID=%x\n", reg); ret = -ENODEV; goto err; } -- cgit v1.2.3-18-g5258 From ca8f04247eaaec554528279686a514c6ce087bb9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 11:51:19 +0100 Subject: ALSA: hda/realtek - Add the fixup codes for ALC260 model=will The model=will for ALC260 requires the pin 0x0f to be a headphone and some special verbs for the COEF to turn on the amp. Now added these as fixup entries and removed the static model quirk. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc260_quirks.c | 43 ------------------------------------------- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++++++--- 2 files changed, 24 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 3b5170b9700..79aaae8e0d9 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -9,7 +9,6 @@ enum { ALC260_BASIC, ALC260_FUJITSU_S702X, ALC260_ACER, - ALC260_WILL, ALC260_REPLACER_672V, ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG @@ -236,23 +235,6 @@ static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { { } /* end */ }; -/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12, - * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17. - */ -static const struct snd_kcontrol_new alc260_will_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - { } /* end */ -}; - /* Replacer 672V ALC260 pin usage: Mic jack = 0x12, * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f. */ @@ -590,16 +572,6 @@ static const struct hda_verb alc260_favorit100_init_verbs[] = { { } }; -static const struct hda_verb alc260_will_verbs[] = { - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - {0x1a, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x1a, AC_VERB_SET_PROC_COEF, 0x3040}, - {} -}; - static const struct hda_verb alc260_replacer_672v_verbs[] = { {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, {0x1a, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -851,7 +823,6 @@ static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_FUJITSU_S702X] = "fujitsu", [ALC260_ACER] = "acer", - [ALC260_WILL] = "will", [ALC260_REPLACER_672V] = "replacer", [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG @@ -862,7 +833,6 @@ static const char * const alc260_models[ALC260_MODEL_LAST] = { static const struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), - SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), @@ -871,7 +841,6 @@ static const struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X), SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC), SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V), - SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL), {} }; @@ -924,18 +893,6 @@ static const struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources), .input_mux = alc260_favorit100_capture_sources, }, - [ALC260_WILL] = { - .mixers = { alc260_will_mixer }, - .init_verbs = { alc260_init_verbs, alc260_will_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids), - .adc_nids = alc260_adc_nids, - .dig_out_nid = ALC260_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .input_mux = &alc260_capture_source, - }, [ALC260_REPLACER_672V] = { .mixers = { alc260_replacer_672v_mixer }, .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 30ef877e628..f5f37103623 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4207,21 +4207,42 @@ static const struct hda_amp_list alc260_loopbacks[] = { * Pin config fixes */ enum { - PINFIX_HP_DC5750, + ALC260_FIXUP_HP_DC5750, + ALC260_FIXUP_HP_PIN_0F, + ALC260_FIXUP_COEF, }; static const struct alc_fixup alc260_fixups[] = { - [PINFIX_HP_DC5750] = { + [ALC260_FIXUP_HP_DC5750] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x11, 0x90130110 }, /* speaker */ { } } }, + [ALC260_FIXUP_HP_PIN_0F] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x0f, 0x01214000 }, /* HP */ + { } + } + }, + [ALC260_FIXUP_COEF] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3040 }, + { } + }, + .chained = true, + .chain_id = ALC260_FIXUP_HP_PIN_0F, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750), + SND_PCI_QUIRK(0x1025, 0x007f, "Acer Aspire 9500", ALC260_FIXUP_COEF), + SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), + SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} }; -- cgit v1.2.3-18-g5258 From 15317ab21686044f1af96dd329ba809a08f04b89 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:02:53 +0100 Subject: ALSA: hda/realtek - Replace ALC260 model=acer with the auto-parser The ALC260 model=acer needs GPIO1 setup. It could be selected well if the codec SSID is set properly by BIOS, but to make sure, enable it forcibly. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc260_quirks.c | 146 ------------------------------------------ sound/pci/hda/patch_realtek.c | 7 ++ 2 files changed, 7 insertions(+), 146 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 79aaae8e0d9..2f1594b3d4b 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -8,7 +8,6 @@ enum { ALC260_AUTO, ALC260_BASIC, ALC260_FUJITSU_S702X, - ALC260_ACER, ALC260_REPLACER_672V, ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG @@ -181,48 +180,6 @@ static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = { { } /* end */ }; -/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current - * versions of the ALC260 don't act on requests to enable mic bias from NID - * 0x0f (used to drive the headphone jack in these laptops). The ALC260 - * datasheet doesn't mention this restriction. At this stage it's not clear - * whether this behaviour is intentional or is a hardware bug in chip - * revisions available in early 2006. Therefore for now allow the - * "Headphone Jack Mode" control to span all choices, but if it turns out - * that the lack of mic bias for this NID is intentional we could change the - * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS. - * - * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006 - * don't appear to make the mic bias available from the "line" jack, even - * though the NID used for this jack (0x14) can supply it. The theory is - * that perhaps Acer have included blocking capacitors between the ALC260 - * and the output jack. If this turns out to be the case for all such - * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT - * to ALC_PIN_DIR_INOUT_NOMICBIAS. - * - * The C20x Tablet series have a mono internal speaker which is controlled - * via the chip's Mono sum widget and pin complex, so include the necessary - * controls for such models. On models without a "mono speaker" the control - * won't do anything. - */ -static const struct snd_kcontrol_new alc260_acer_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, - HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, - HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - { } /* end */ -}; - /* Maxdata Favorit 100XS: one output and one input (0x12) jack */ static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { @@ -401,94 +358,6 @@ static const struct hda_verb alc260_fujitsu_init_verbs[] = { { } }; -/* Initialisation sequence for ALC260 as configured in Acer TravelMate and - * similar laptops (adapted from Fujitsu init verbs). - */ -static const struct hda_verb alc260_acer_init_verbs[] = { - /* On TravelMate laptops, GPIO 0 enables the internal speaker and - * the headphone jack. Turn this on and rely on the standard mute - * methods whenever the user wants to turn these outputs off. - */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, - /* Internal speaker/Headphone jack is connected to Line-out pin */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Internal microphone/Mic jack is connected to Mic1 pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, - /* Line In jack is connected to Line1 pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Ensure all other unused pins are disabled and muted. */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Disable digital (SPDIF) pins */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum - * bus when acting as outputs. - */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* Unmute Line-out pin widget amp left and right - * (no equiv mixer ctrl) - */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute mono pin widget amp output (no equiv mixer ctrl) */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Mic1 and Line1 pin widget input buffers since they start as - * inputs. If the pin mode is changed by the user the pin mode control - * will take care of enabling the pin's input/output buffers as needed. - * Therefore there's no need to enable the input buffer at this - * stage. - */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - mic - * (on mic1 pin) - */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Do similar with the second ADC: mute capture input amp and - * set ADC connection to mic to match ALSA's default state. - */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; - /* Initialisation sequence for Maxdata Favorit 100XS * (adapted from Acer init verbs). */ @@ -822,7 +691,6 @@ static const struct hda_verb alc260_test_init_verbs[] = { static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_FUJITSU_S702X] = "fujitsu", - [ALC260_ACER] = "acer", [ALC260_REPLACER_672V] = "replacer", [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG @@ -832,8 +700,6 @@ static const char * const alc260_models[ALC260_MODEL_LAST] = { }; static const struct snd_pci_quirk alc260_cfg_tbl[] = { - SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), - SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), @@ -869,18 +735,6 @@ static const struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources), .input_mux = alc260_fujitsu_capture_sources, }, - [ALC260_ACER] = { - .mixers = { alc260_acer_mixer }, - .init_verbs = { alc260_acer_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_dual_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources), - .input_mux = alc260_acer_capture_sources, - }, [ALC260_FAVORIT100] = { .mixers = { alc260_favorit100_mixer }, .init_verbs = { alc260_favorit100_init_verbs }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f5f37103623..95ef722e407 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4210,6 +4210,7 @@ enum { ALC260_FIXUP_HP_DC5750, ALC260_FIXUP_HP_PIN_0F, ALC260_FIXUP_COEF, + ALC260_FIXUP_GPIO1, }; static const struct alc_fixup alc260_fixups[] = { @@ -4237,10 +4238,16 @@ static const struct alc_fixup alc260_fixups[] = { .chained = true, .chain_id = ALC260_FIXUP_HP_PIN_0F, }, + [ALC260_FIXUP_GPIO1] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio1_init_verbs, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x1025, 0x007f, "Acer Aspire 9500", ALC260_FIXUP_COEF), + SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} -- cgit v1.2.3-18-g5258 From 20f7d928fa6e51ca81648946ead6244c58a0b4c0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:35:16 +0100 Subject: ALSA: hda/realtek - Replace ALC260 model=replacer with the auto-parser The support for Replacer 627V in the auto-parser needs the unique unsol event handling: although the machine has a single output pin 0x0f, it's used for both the headphone and the speaker, and the driver needs to toggle the output route via GPIO 1. In addition, it needs a special COEF setup with 0x3050. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc260_quirks.c | 76 ------------------------------------------- sound/pci/hda/patch_realtek.c | 44 +++++++++++++++++++++++++ 2 files changed, 44 insertions(+), 76 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 2f1594b3d4b..55da43dddf3 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -8,7 +8,6 @@ enum { ALC260_AUTO, ALC260_BASIC, ALC260_FUJITSU_S702X, - ALC260_REPLACER_672V, ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG ALC260_TEST, @@ -192,23 +191,6 @@ static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { { } /* end */ }; -/* Replacer 672V ALC260 pin usage: Mic jack = 0x12, - * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f. - */ -static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - { } /* end */ -}; - /* * initialization verbs */ @@ -441,48 +423,6 @@ static const struct hda_verb alc260_favorit100_init_verbs[] = { { } }; -static const struct hda_verb alc260_replacer_672v_verbs[] = { - {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - {0x1a, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x1a, AC_VERB_SET_PROC_COEF, 0x3050}, - - {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - - {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc260_replacer_672v_automute(struct hda_codec *codec) -{ - unsigned int present; - - /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */ - present = snd_hda_jack_detect(codec, 0x0f); - if (present) { - snd_hda_codec_write_cache(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, 1); - snd_hda_codec_write_cache(codec, 0x0f, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_HP); - } else { - snd_hda_codec_write_cache(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, 0); - snd_hda_codec_write_cache(codec, 0x0f, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_OUT); - } -} - -static void alc260_replacer_672v_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) == ALC_HP_EVENT) - alc260_replacer_672v_automute(codec); -} - static const struct hda_verb alc260_hp_dc7600_verbs[] = { {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -691,7 +631,6 @@ static const struct hda_verb alc260_test_init_verbs[] = { static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_FUJITSU_S702X] = "fujitsu", - [ALC260_REPLACER_672V] = "replacer", [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", @@ -706,7 +645,6 @@ static const struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X), SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC), - SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V), {} }; @@ -747,20 +685,6 @@ static const struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources), .input_mux = alc260_favorit100_capture_sources, }, - [ALC260_REPLACER_672V] = { - .mixers = { alc260_replacer_672v_mixer }, - .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids), - .adc_nids = alc260_adc_nids, - .dig_out_nid = ALC260_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .input_mux = &alc260_capture_source, - .unsol_event = alc260_replacer_672v_unsol_event, - .init_hook = alc260_replacer_672v_automute, - }, #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = { .mixers = { alc260_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 95ef722e407..cfa6ad75834 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4211,8 +4211,35 @@ enum { ALC260_FIXUP_HP_PIN_0F, ALC260_FIXUP_COEF, ALC260_FIXUP_GPIO1, + ALC260_FIXUP_GPIO1_TOGGLE, + ALC260_FIXUP_REPLACER, }; +static void alc260_gpio1_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->hp_jack_present); +} + +static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action == ALC_FIXUP_ACT_PROBE) { + /* although the machine has only one output pin, we need to + * toggle GPIO1 according to the jack state + */ + spec->automute_hook = alc260_gpio1_automute; + spec->detect_hp = 1; + spec->automute_speaker = 1; + spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */ + snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT); + spec->unsol_event = alc_sku_unsol_event; + add_verb(codec->spec, alc_gpio1_init_verbs); + } +} + static const struct alc_fixup alc260_fixups[] = { [ALC260_FIXUP_HP_DC5750] = { .type = ALC_FIXUP_PINS, @@ -4242,6 +4269,22 @@ static const struct alc_fixup alc260_fixups[] = { .type = ALC_FIXUP_VERBS, .v.verbs = alc_gpio1_init_verbs, }, + [ALC260_FIXUP_GPIO1_TOGGLE] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc260_fixup_gpio1_toggle, + .chained = true, + .chain_id = ALC260_FIXUP_HP_PIN_0F, + }, + [ALC260_FIXUP_REPLACER] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, + { } + }, + .chained = true, + .chain_id = ALC260_FIXUP_GPIO1_TOGGLE, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -4249,6 +4292,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007f, "Acer Aspire 9500", ALC260_FIXUP_COEF), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), + SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} }; -- cgit v1.2.3-18-g5258 From 0a1c4fa2085de68a543f28827fb6614d28924540 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:42:30 +0100 Subject: ALSA: hda/realtek - Add the support for HP Presario B1900 HP Presario B1900 needs a similar hack like Replacer, toggling GPIO1 per the jack state, in addition to the COEF setup used for other Acer laptops. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cfa6ad75834..db1d8c888da 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4213,6 +4213,7 @@ enum { ALC260_FIXUP_GPIO1, ALC260_FIXUP_GPIO1_TOGGLE, ALC260_FIXUP_REPLACER, + ALC260_FIXUP_HP_B1900, }; static void alc260_gpio1_automute(struct hda_codec *codec) @@ -4285,6 +4286,12 @@ static const struct alc_fixup alc260_fixups[] = { .chained = true, .chain_id = ALC260_FIXUP_GPIO1_TOGGLE, }, + [ALC260_FIXUP_HP_B1900] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc260_fixup_gpio1_toggle, + .chained = true, + .chain_id = ALC260_FIXUP_COEF, + } }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -4292,6 +4299,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007f, "Acer Aspire 9500", ALC260_FIXUP_COEF), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), + SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} -- cgit v1.2.3-18-g5258 From b1f58085a9c01e8ffab954fd77a45f1143edf34d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:45:03 +0100 Subject: ALSA: hda/realtek - Drop model=favorit100 for ALC260 It's working with the auto-parser just with the standard GPIO 1 setup. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc260_quirks.c | 129 ------------------------------------------ sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 1 insertion(+), 129 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 55da43dddf3..94e7a270c5a 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -8,7 +8,6 @@ enum { ALC260_AUTO, ALC260_BASIC, ALC260_FUJITSU_S702X, - ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG ALC260_TEST, #endif @@ -103,25 +102,6 @@ static const struct hda_input_mux alc260_acer_capture_sources[2] = { }, }; -/* Maxdata Favorit 100XS */ -static const struct hda_input_mux alc260_favorit100_capture_sources[2] = { - { - .num_items = 2, - .items = { - { "Line/Mic", 0x0 }, - { "CD", 0x4 }, - }, - }, - { - .num_items = 3, - .items = { - { "Line/Mic", 0x0 }, - { "CD", 0x4 }, - { "Mixer", 0x5 }, - }, - }, -}; - /* * This is just place-holder, so there's something for alc_build_pcms to look * at when it calculates the maximum number of channels. ALC260 has no mixer @@ -179,18 +159,6 @@ static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = { { } /* end */ }; -/* Maxdata Favorit 100XS: one output and one input (0x12) jack - */ -static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - { } /* end */ -}; - /* * initialization verbs */ @@ -340,89 +308,6 @@ static const struct hda_verb alc260_fujitsu_init_verbs[] = { { } }; -/* Initialisation sequence for Maxdata Favorit 100XS - * (adapted from Acer init verbs). - */ -static const struct hda_verb alc260_favorit100_init_verbs[] = { - /* GPIO 0 enables the output jack. - * Turn this on and rely on the standard mute - * methods whenever the user wants to turn these outputs off. - */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, - /* Line/Mic input jack is connected to Mic1 pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, - /* Ensure all other unused pins are disabled and muted. */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Disable digital (SPDIF) pins */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum - * bus when acting as outputs. - */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* Unmute Line-out pin widget amp left and right - * (no equiv mixer ctrl) - */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Mic1 and Line1 pin widget input buffers since they start as - * inputs. If the pin mode is changed by the user the pin mode control - * will take care of enabling the pin's input/output buffers as needed. - * Therefore there's no need to enable the input buffer at this - * stage. - */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - mic - * (on mic1 pin) - */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Do similar with the second ADC: mute capture input amp and - * set ADC connection to mic to match ALSA's default state. - */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; - static const struct hda_verb alc260_hp_dc7600_verbs[] = { {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -631,7 +516,6 @@ static const struct hda_verb alc260_test_init_verbs[] = { static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_FUJITSU_S702X] = "fujitsu", - [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", #endif @@ -639,7 +523,6 @@ static const char * const alc260_models[ALC260_MODEL_LAST] = { }; static const struct snd_pci_quirk alc260_cfg_tbl[] = { - SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), @@ -673,18 +556,6 @@ static const struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources), .input_mux = alc260_fujitsu_capture_sources, }, - [ALC260_FAVORIT100] = { - .mixers = { alc260_favorit100_mixer }, - .init_verbs = { alc260_favorit100_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_dual_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources), - .input_mux = alc260_favorit100_capture_sources, - }, #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = { .mixers = { alc260_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index db1d8c888da..0d81eeb563c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4300,6 +4300,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), + SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} -- cgit v1.2.3-18-g5258 From c29b3f6dd7798964d77199af4925be72a3a48349 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:47:36 +0100 Subject: ALSA: hda/realtek - Drop model=fujitsu from ALC260 static quirks The model works with the auto-parser as is, thus now good to drop. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc260_quirks.c | 142 ------------------------------------------ 1 file changed, 142 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c index 94e7a270c5a..305341f892c 100644 --- a/sound/pci/hda/alc260_quirks.c +++ b/sound/pci/hda/alc260_quirks.c @@ -7,7 +7,6 @@ enum { ALC260_AUTO, ALC260_BASIC, - ALC260_FUJITSU_S702X, #ifdef CONFIG_SND_DEBUG ALC260_TEST, #endif @@ -50,33 +49,6 @@ static const struct hda_input_mux alc260_capture_source = { }, }; -/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack, - * headphone jack and the internal CD lines since these are the only pins at - * which audio can appear. For flexibility, also allow the option of - * recording the mixer output on the second ADC (ADC0 doesn't have a - * connection to the mixer output). - */ -static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = { - { - .num_items = 3, - .items = { - { "Mic/Line", 0x0 }, - { "CD", 0x4 }, - { "Headphone", 0x2 }, - }, - }, - { - .num_items = 4, - .items = { - { "Mic/Line", 0x0 }, - { "CD", 0x4 }, - { "Headphone", 0x2 }, - { "Mixer", 0x5 }, - }, - }, - -}; - /* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to * the Fujitsu S702x, but jacks are marked differently. */ @@ -142,23 +114,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = { { } /* end */ }; -/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, - * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10. - */ -static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT), - { } /* end */ -}; - /* * initialization verbs */ @@ -225,89 +180,6 @@ static const struct hda_verb alc260_init_verbs[] = { { } }; -/* Initialisation sequence for ALC260 as configured in Fujitsu S702x - * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD - * audio = 0x16, internal speaker = 0x10. - */ -static const struct hda_verb alc260_fujitsu_init_verbs[] = { - /* Disable all GPIOs */ - {0x01, AC_VERB_SET_GPIO_MASK, 0}, - /* Internal speaker is connected to headphone pin */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Headphone/Line-out jack connects to Line1 pin; make it an output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mic/Line-in jack is connected to mic1 pin, so make it an input */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Ensure all other unused pins are disabled and muted. */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - - /* Disable digital (SPDIF) pins */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure Line1 pin widget takes its input from the OUT1 sum bus - * when acting as an output. - */ - {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Line1 pin widget output buffer since it starts as an output. - * If the pin mode is changed by the user the pin mode control will - * take care of enabling the pin's input/output buffers as needed. - * Therefore there's no need to enable the input buffer at this - * stage. - */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute input buffer of pin widget used for Line-in (no equiv - * mixer ctrl) - */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - line - * in (on mic1 pin) - */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Do the same for the second ADC: mute capture input amp and - * set ADC connection to line in (on mic1 pin) - */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; - static const struct hda_verb alc260_hp_dc7600_verbs[] = { {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -515,7 +387,6 @@ static const struct hda_verb alc260_test_init_verbs[] = { */ static const char * const alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", - [ALC260_FUJITSU_S702X] = "fujitsu", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", #endif @@ -526,7 +397,6 @@ static const struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), - SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X), SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC), {} }; @@ -544,18 +414,6 @@ static const struct alc_config_preset alc260_presets[] = { .channel_mode = alc260_modes, .input_mux = &alc260_capture_source, }, - [ALC260_FUJITSU_S702X] = { - .mixers = { alc260_fujitsu_mixer }, - .init_verbs = { alc260_fujitsu_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_dual_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources), - .input_mux = alc260_fujitsu_capture_sources, - }, #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = { .mixers = { alc260_test_mixer }, -- cgit v1.2.3-18-g5258 From c3c2c9e7ff3e38bd9ff5b721b6ae8634fce42802 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 12:59:55 +0100 Subject: ALSA: hda/realtek - Remove leftover static quirks for ALC260 Now we can clean up all static quirks for ALC260. Also many codes in alc_quirks.c can be ripped off since they have been used only by ALC260 static quirks. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc260_quirks.c | 432 ------------------------------------------ sound/pci/hda/alc_quirks.c | 301 ----------------------------- sound/pci/hda/patch_realtek.c | 48 +---- 3 files changed, 8 insertions(+), 773 deletions(-) delete mode 100644 sound/pci/hda/alc260_quirks.c (limited to 'sound') diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c deleted file mode 100644 index 305341f892c..00000000000 --- a/sound/pci/hda/alc260_quirks.c +++ /dev/null @@ -1,432 +0,0 @@ -/* - * ALC260 quirk models - * included by patch_realtek.c - */ - -/* ALC260 models */ -enum { - ALC260_AUTO, - ALC260_BASIC, -#ifdef CONFIG_SND_DEBUG - ALC260_TEST, -#endif - ALC260_MODEL_LAST /* last tag */ -}; - -static const hda_nid_t alc260_dac_nids[1] = { - /* front */ - 0x02, -}; - -static const hda_nid_t alc260_adc_nids[1] = { - /* ADC0 */ - 0x04, -}; - -static const hda_nid_t alc260_adc_nids_alt[1] = { - /* ADC1 */ - 0x05, -}; - -/* NIDs used when simultaneous access to both ADCs makes sense. Note that - * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC. - */ -static const hda_nid_t alc260_dual_adc_nids[2] = { - /* ADC0, ADC1 */ - 0x04, 0x05 -}; - -#define ALC260_DIGOUT_NID 0x03 -#define ALC260_DIGIN_NID 0x06 - -static const struct hda_input_mux alc260_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to - * the Fujitsu S702x, but jacks are marked differently. - */ -static const struct hda_input_mux alc260_acer_capture_sources[2] = { - { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Headphone", 0x5 }, - }, - }, - { - .num_items = 5, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Headphone", 0x6 }, - { "Mixer", 0x5 }, - }, - }, -}; - -/* - * This is just place-holder, so there's something for alc_build_pcms to look - * at when it calculates the maximum number of channels. ALC260 has no mixer - * element which allows changing the channel mode, so the verb list is - * never used. - */ -static const struct hda_channel_mode alc260_modes[1] = { - { 2, NULL }, -}; - - -/* Mixer combinations - * - * basic: base_output + input + pc_beep + capture - * fujitsu: fujitsu + capture - * acer: acer + capture - */ - -static const struct snd_kcontrol_new alc260_base_output_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc260_input_mixer[] = { - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT), - { } /* end */ -}; - -/* - * initialization verbs - */ -static const struct hda_verb alc260_init_verbs[] = { - /* Line In pin widget for input */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* CD pin widget for input */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - /* Mic2 (front panel) pin widget for input and vref at 80% */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - /* LINE-2 is used for line-out in rear */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* select line-out */ - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* LINE-OUT pin */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* enable HP */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* enable Mono */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* set connection select to line in (default select for this ADC) */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* mute capture amp left and right */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* set connection select to line in (default select for this ADC) */ - {0x05, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* set vol=0 Line-Out mixer amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* unmute pin widget amp left and right (no gain on this amp) */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* set vol=0 HP mixer amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* unmute pin widget amp left and right (no gain on this amp) */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* set vol=0 Mono mixer amp left and right */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* unmute pin widget amp left and right (no gain on this amp) */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* unmute LINE-2 out pin */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & - * Line In 2 = 0x03 - */ - /* mute analog inputs */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */ - /* mute Front out path */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* mute Headphone out path */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* mute Mono out path */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - { } -}; - -static const struct hda_verb alc260_hp_dc7600_verbs[] = { - {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {} -}; - -/* Test configuration for debugging, modelled after the ALC880 test - * configuration. - */ -#ifdef CONFIG_SND_DEBUG -static const hda_nid_t alc260_test_dac_nids[1] = { - 0x02, -}; -static const hda_nid_t alc260_test_adc_nids[2] = { - 0x04, 0x05, -}; -/* For testing the ALC260, each input MUX needs its own definition since - * the signal assignments are different. This assumes that the first ADC - * is NID 0x04. - */ -static const struct hda_input_mux alc260_test_capture_sources[2] = { - { - .num_items = 7, - .items = { - { "MIC1 pin", 0x0 }, - { "MIC2 pin", 0x1 }, - { "LINE1 pin", 0x2 }, - { "LINE2 pin", 0x3 }, - { "CD pin", 0x4 }, - { "LINE-OUT pin", 0x5 }, - { "HP-OUT pin", 0x6 }, - }, - }, - { - .num_items = 8, - .items = { - { "MIC1 pin", 0x0 }, - { "MIC2 pin", 0x1 }, - { "LINE1 pin", 0x2 }, - { "LINE2 pin", 0x3 }, - { "CD pin", 0x4 }, - { "Mixer", 0x5 }, - { "LINE-OUT pin", 0x6 }, - { "HP-OUT pin", 0x7 }, - }, - }, -}; -static const struct snd_kcontrol_new alc260_test_mixer[] = { - /* Output driver widgets */ - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), - HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT), - HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT), - - /* Modes for retasking pin widgets - * Note: the ALC260 doesn't seem to act on requests to enable mic - * bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't - * mention this restriction. At this stage it's not clear whether - * this behaviour is intentional or is a hardware bug in chip - * revisions available at least up until early 2006. Therefore for - * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all - * choices, but if it turns out that the lack of mic bias for these - * NIDs is intentional we could change their modes from - * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS. - */ - ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT), - ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT), - - /* Loopback mixer controls */ - HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), - HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT), - - /* Controls for GPIO pins, assuming they are configured as outputs */ - ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01), - ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02), - ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04), - ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08), - - /* Switches to allow the digital IO pins to be enabled. The datasheet - * is ambigious as to which NID is which; testing on laptops which - * make this output available should provide clarification. - */ - ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01), - ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01), - - /* A switch allowing EAPD to be enabled. Some laptops seem to use - * this output to turn on an external amplifier. - */ - ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02), - ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02), - - { } /* end */ -}; -static const struct hda_verb alc260_test_init_verbs[] = { - /* Enable all GPIOs as outputs with an initial value of 0 */ - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, - {0x01, AC_VERB_SET_GPIO_MASK, 0x0f}, - - /* Enable retasking pins as output, initially without power amp */ - {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - - /* Disable digital (SPDIF) pins initially, but users can enable - * them via a mixer switch. In the case of SPDIF-out, this initverb - * payload also sets the generation to 0, output to be in "consumer" - * PCM format, copyright asserted, no pre-emphasis and no validity - * control. - */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - - /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the - * OUT1 sum bus when acting as an output. - */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0c, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - {0x0e, AC_VERB_SET_CONNECT_SEL, 0}, - - /* Start with output sum widgets muted and their output gains at min */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* Unmute retasking pin widget output buffers since the default - * state appears to be output. As the pin mode is changed by the - * user the pin mode control will take care of enabling the pin's - * input/output buffers as needed. - */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Also unmute the mono-out pin widget */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting (mic1 - * pin) - */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Do the same for the second ADC: mute capture input amp and - * set ADC connection to mic1 pin - */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ - - { } -}; -#endif - -/* - * ALC260 configurations - */ -static const char * const alc260_models[ALC260_MODEL_LAST] = { - [ALC260_BASIC] = "basic", -#ifdef CONFIG_SND_DEBUG - [ALC260_TEST] = "test", -#endif - [ALC260_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc260_cfg_tbl[] = { - SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC), - SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC), - SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC), - SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC), - {} -}; - -static const struct alc_config_preset alc260_presets[] = { - [ALC260_BASIC] = { - .mixers = { alc260_base_output_mixer, - alc260_input_mixer }, - .init_verbs = { alc260_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_dac_nids), - .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), - .adc_nids = alc260_dual_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .input_mux = &alc260_capture_source, - }, -#ifdef CONFIG_SND_DEBUG - [ALC260_TEST] = { - .mixers = { alc260_test_mixer }, - .init_verbs = { alc260_test_init_verbs }, - .num_dacs = ARRAY_SIZE(alc260_test_dac_nids), - .dac_nids = alc260_test_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids), - .adc_nids = alc260_test_adc_nids, - .num_channel_mode = ARRAY_SIZE(alc260_modes), - .channel_mode = alc260_modes, - .num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources), - .input_mux = alc260_test_capture_sources, - }, -#endif -}; - diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c index a18952ed431..b344603ac06 100644 --- a/sound/pci/hda/alc_quirks.c +++ b/sound/pci/hda/alc_quirks.c @@ -74,307 +74,6 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, return err; } -/* - * Control the mode of pin widget settings via the mixer. "pc" is used - * instead of "%" to avoid consequences of accidentally treating the % as - * being part of a format specifier. Maximum allowed length of a value is - * 63 characters plus NULL terminator. - * - * Note: some retasking pin complexes seem to ignore requests for input - * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these - * are requested. Therefore order this list so that this behaviour will not - * cause problems when mixer clients move through the enum sequentially. - * NIDs 0x0f and 0x10 have been observed to have this behaviour as of - * March 2006. - */ -static const char * const alc_pin_mode_names[] = { - "Mic 50pc bias", "Mic 80pc bias", - "Line in", "Line out", "Headphone out", -}; -static const unsigned char alc_pin_mode_values[] = { - PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP, -}; -/* The control can present all 5 options, or it can limit the options based - * in the pin being assumed to be exclusively an input or an output pin. In - * addition, "input" pins may or may not process the mic bias option - * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to - * accept requests for bias as of chip versions up to March 2006) and/or - * wiring in the computer. - */ -#define ALC_PIN_DIR_IN 0x00 -#define ALC_PIN_DIR_OUT 0x01 -#define ALC_PIN_DIR_INOUT 0x02 -#define ALC_PIN_DIR_IN_NOMICBIAS 0x03 -#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04 - -/* Info about the pin modes supported by the different pin direction modes. - * For each direction the minimum and maximum values are given. - */ -static const signed char alc_pin_mode_dir_info[5][2] = { - { 0, 2 }, /* ALC_PIN_DIR_IN */ - { 3, 4 }, /* ALC_PIN_DIR_OUT */ - { 0, 4 }, /* ALC_PIN_DIR_INOUT */ - { 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */ - { 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */ -}; -#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0]) -#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1]) -#define alc_pin_mode_n_items(_dir) \ - (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1) - -static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - unsigned int item_num = uinfo->value.enumerated.item; - unsigned char dir = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = alc_pin_mode_n_items(dir); - - if (item_numalc_pin_mode_max(dir)) - item_num = alc_pin_mode_min(dir); - strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]); - return 0; -} - -static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int i; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char dir = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; - unsigned int pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, - 0x00); - - /* Find enumerated value for current pinctl setting */ - i = alc_pin_mode_min(dir); - while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl) - i++; - *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir); - return 0; -} - -static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - signed int change; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char dir = (kcontrol->private_value >> 16) & 0xff; - long val = *ucontrol->value.integer.value; - unsigned int pinctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, - 0x00); - - if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir)) - val = alc_pin_mode_min(dir); - - change = pinctl != alc_pin_mode_values[val]; - if (change) { - /* Set pin mode to that requested */ - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - alc_pin_mode_values[val]); - - /* Also enable the retasking pin's input/output as required - * for the requested pin mode. Enum values of 2 or less are - * input modes. - * - * Dynamically switching the input/output buffers probably - * reduces noise slightly (particularly on input) so we'll - * do it. However, having both input and output buffers - * enabled simultaneously doesn't seem to be problematic if - * this turns out to be necessary in the future. - */ - if (val <= 2) { - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0, - HDA_AMP_MUTE, 0); - } else { - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, 0); - } - } - return change; -} - -#define ALC_PIN_MODE(xname, nid, dir) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_pin_mode_info, \ - .get = alc_pin_mode_get, \ - .put = alc_pin_mode_put, \ - .private_value = nid | (dir<<16) } - -/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged - * together using a mask with more than one bit set. This control is - * currently used only by the ALC260 test model. At this stage they are not - * needed for any "production" models. - */ -#ifdef CONFIG_SND_DEBUG -#define alc_gpio_data_info snd_ctl_boolean_mono_info - -static int alc_gpio_data_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; - unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_DATA, 0x00); - - *valp = (val & mask) != 0; - return 0; -} -static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - signed int change; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long val = *ucontrol->value.integer.value; - unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_GPIO_DATA, - 0x00); - - /* Set/unset the masked GPIO bit(s) as needed */ - change = (val == 0 ? 0 : mask) != (gpio_data & mask); - if (val == 0) - gpio_data &= ~mask; - else - gpio_data |= mask; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_GPIO_DATA, gpio_data); - - return change; -} -#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_gpio_data_info, \ - .get = alc_gpio_data_get, \ - .put = alc_gpio_data_put, \ - .private_value = nid | (mask<<16) } -#endif /* CONFIG_SND_DEBUG */ - -/* A switch control to allow the enabling of the digital IO pins on the - * ALC260. This is incredibly simplistic; the intention of this control is - * to provide something in the test model allowing digital outputs to be - * identified if present. If models are found which can utilise these - * outputs a more complete mixer control can be devised for those models if - * necessary. - */ -#ifdef CONFIG_SND_DEBUG -#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info - -static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; - unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT_1, 0x00); - - *valp = (val & mask) != 0; - return 0; -} -static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - signed int change; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long val = *ucontrol->value.integer.value; - unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT_1, - 0x00); - - /* Set/unset the masked control bit(s) as needed */ - change = (val == 0 ? 0 : mask) != (ctrl_data & mask); - if (val==0) - ctrl_data &= ~mask; - else - ctrl_data |= mask; - snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - ctrl_data); - - return change; -} -#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_spdif_ctrl_info, \ - .get = alc_spdif_ctrl_get, \ - .put = alc_spdif_ctrl_put, \ - .private_value = nid | (mask<<16) } -#endif /* CONFIG_SND_DEBUG */ - -/* A switch control to allow the enabling EAPD digital outputs on the ALC26x. - * Again, this is only used in the ALC26x test models to help identify when - * the EAPD line must be asserted for features to work. - */ -#ifdef CONFIG_SND_DEBUG -#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info - -static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; - unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_EAPD_BTLENABLE, 0x00); - - *valp = (val & mask) != 0; - return 0; -} - -static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - int change; - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = kcontrol->private_value & 0xffff; - unsigned char mask = (kcontrol->private_value >> 16) & 0xff; - long val = *ucontrol->value.integer.value; - unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_EAPD_BTLENABLE, - 0x00); - - /* Set/unset the masked control bit(s) as needed */ - change = (!val ? 0 : mask) != (ctrl_data & mask); - if (!val) - ctrl_data &= ~mask; - else - ctrl_data |= mask; - snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, - ctrl_data); - - return change; -} - -#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \ - { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_eapd_ctrl_info, \ - .get = alc_eapd_ctrl_get, \ - .put = alc_eapd_ctrl_put, \ - .private_value = nid | (mask<<16) } -#endif /* CONFIG_SND_DEBUG */ - static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0d81eeb563c..3ea42069b8e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4308,14 +4308,10 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc260_quirks.c" -#endif - static int patch_alc260(struct hda_codec *codec) { struct alc_spec *spec; - int err, board_config; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4325,38 +4321,13 @@ static int patch_alc260(struct hda_codec *codec) spec->mixer_nid = 0x07; - board_config = alc_board_config(codec, ALC260_MODEL_LAST, - alc260_models, alc260_cfg_tbl); - if (board_config < 0) { - snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } - - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc260_parse_auto_config(codec); - if (err < 0) - goto error; -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using base mode...\n"); - board_config = ALC260_BASIC; - } -#endif - } + alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - if (board_config != ALC_MODEL_AUTO) { - setup_preset(codec, &alc260_presets[board_config]); - spec->vmaster_nid = 0x08; - } + /* automatic parse from the BIOS config */ + err = alc260_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4377,10 +4348,7 @@ static int patch_alc260(struct hda_codec *codec) alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; - else - codec->patch_ops.build_controls = __alc_build_controls; + spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) -- cgit v1.2.3-18-g5258 From a7f3eedc88b547e0ec35ba4cc4ae61cd9bc760ac Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 13:03:18 +0100 Subject: ALSA: hda/realtek - Disable static fixups for ASUS with ALC269 We've enabled the static fixups for ASUS machines with ALC269 codec, just for making things compatible during the transition to the auto- parser. However, it seems that the static configurations do more harmful than good, as some of entries don't match with the actual hardware setups. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3ea42069b8e..b8e06eb96e1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5396,7 +5396,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -#if 1 +#if 0 /* Below is a quirk table taken from the old code. * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding -- cgit v1.2.3-18-g5258 From 0e66821f63b3bf1468f5793759beb09b3e1ffa4f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Thu, 16 Feb 2012 12:00:23 +0000 Subject: ASoC: spitz: Fix kcontrols to use card instead of codec Machine kcontrols now use card instead of codec for thier "chip". Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/spitz.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 90c5245c474..4d108dc52e2 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -44,9 +44,9 @@ static int spitz_jack_func; static int spitz_spk_func; static int spitz_mic_gpio; -static void spitz_ext_control(struct snd_soc_codec *codec) +static void spitz_ext_control(struct snd_soc_card *card) { - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &card->dapm; if (spitz_spk_func == SPITZ_SPK_ON) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); @@ -173,13 +173,13 @@ static int spitz_get_jack(struct snd_kcontrol *kcontrol, static int spitz_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (spitz_jack_func == ucontrol->value.integer.value[0]) return 0; spitz_jack_func = ucontrol->value.integer.value[0]; - spitz_ext_control(codec); + spitz_ext_control(card); return 1; } @@ -193,13 +193,13 @@ static int spitz_get_spk(struct snd_kcontrol *kcontrol, static int spitz_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (spitz_spk_func == ucontrol->value.integer.value[0]) return 0; spitz_spk_func = ucontrol->value.integer.value[0]; - spitz_ext_control(codec); + spitz_ext_control(card); return 1; } -- cgit v1.2.3-18-g5258 From 140547ef4ee9ad5f9ee9e6546f6027e8737c4149 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 17:23:46 +0100 Subject: ALSA: hda/realtek - Improve the signel-connection check When the connections from the pin selector contain only two widgets, a route to DAC and the aa-mixer, it's certainly a single connection. In such a case, get_dac_if_single() should return the connected DAC, too. This will improve the detection of the individual DAC assignment for each pin. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ffccc17895..a5697c3b30b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2972,8 +2972,12 @@ static bool alc_auto_is_dac_reachable(struct hda_codec *codec, static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) { + struct alc_spec *spec = codec->spec; hda_nid_t sel = alc_go_down_to_selector(codec, pin); - if (snd_hda_get_conn_list(codec, sel, NULL) == 1) + hda_nid_t srcs[5]; + int num = snd_hda_get_connections(codec, sel, srcs, + ARRAY_SIZE(srcs)); + if (num == 1 || (num == 2 && srcs[1] == spec->mixer_nid)) return alc_auto_look_for_dac(codec, pin); return 0; } -- cgit v1.2.3-18-g5258 From 1c4a54b4513c175ba1a56d0aba8d9cf8f231d407 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 16:45:59 +0100 Subject: ALSA: hda/realtek - Finer tuning of auto-parser with badness evaluation This patch improves the Realtek auto-parser for assigning the DACs and mixers in more suitable ways by evaluating the assignment with "badness" calculations. When assigning a DAC hinders the assignment of individual DACs for other pins, some badness point is given. Similarly, when it blocks the assignment of unique mixer controls, another badness point is added. Also, if no DAC, even shared DAC, can be assigned, more badness is pointed. Finally, comparing the accumulated badness, the best route is chosen among several trials. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 377 ++++++++++++++++++++++++++++++++---------- 1 file changed, 293 insertions(+), 84 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a5697c3b30b..4746afa25db 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2982,76 +2982,191 @@ static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) return 0; } -/* return 0 if no possible DAC is found, 1 if one or more found */ +/* mark up volume and mute control NIDs: used during badness parsing and + * at creating actual controls + */ +static inline unsigned int get_ctl_pos(unsigned int data) +{ + hda_nid_t nid = get_amp_nid_(data); + unsigned int dir; + if (snd_BUG_ON(nid >= MAX_VOL_NIDS)) + return 0; + dir = get_amp_direction_(data); + return (nid << 1) | dir; +} + +#define is_ctl_used(bits, data) \ + test_bit(get_ctl_pos(data), bits) +#define mark_ctl_usage(bits, data) \ + set_bit(get_ctl_pos(data), bits) + +static void clear_vol_marks(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + memset(spec->vol_ctls, 0, sizeof(spec->vol_ctls)); + memset(spec->sw_ctls, 0, sizeof(spec->sw_ctls)); +} + +/* badness definition */ +enum { + /* No primary DAC is found for the main output */ + BAD_NO_PRIMARY_DAC = 0x10000, + /* No DAC is found for the extra output */ + BAD_NO_DAC = 0x4000, + /* No individual DAC for extra output */ + BAD_NO_EXTRA_DAC = 0x1000, + /* No individual DAC for extra surrounds */ + BAD_NO_EXTRA_SURR_DAC = 0x200, + /* Primary DAC shared with main surrounds */ + BAD_SHARED_SURROUND = 0x100, + /* Volume widget is shared */ + BAD_SHARED_VOL = 0x10, + /* Primary DAC shared with main CLFE */ + BAD_SHARED_CLFE = 0x10, + /* Primary DAC shared with extra surrounds */ + BAD_SHARED_EXTRA_SURROUND = 0x10, + /* No possible multi-ios */ + BAD_MULTI_IO = 0x1, +}; + +static hda_nid_t alc_look_for_out_mute_nid(struct hda_codec *codec, + hda_nid_t pin, hda_nid_t dac); +static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec, + hda_nid_t pin, hda_nid_t dac); + +static int eval_shared_vol_badness(struct hda_codec *codec, hda_nid_t pin, + hda_nid_t dac) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t nid; + unsigned int val; + int badness = 0; + + nid = alc_look_for_out_vol_nid(codec, pin, dac); + if (nid) { + val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + if (is_ctl_used(spec->vol_ctls, nid)) + badness += BAD_SHARED_VOL; + else + mark_ctl_usage(spec->vol_ctls, val); + } else + badness += BAD_SHARED_VOL; + nid = alc_look_for_out_mute_nid(codec, pin, dac); + if (nid) { + unsigned int wid_type = get_wcaps_type(get_wcaps(codec, nid)); + if (wid_type == AC_WID_PIN || wid_type == AC_WID_AUD_OUT) + val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + else + val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT); + if (is_ctl_used(spec->sw_ctls, val)) + badness += BAD_SHARED_VOL; + else + mark_ctl_usage(spec->sw_ctls, val); + } else + badness += BAD_SHARED_VOL; + return badness; +} + +/* try to assign DACs to extra pins and return the resultant badness */ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, const hda_nid_t *pins, hda_nid_t *dacs) { + struct alc_spec *spec = codec->spec; int i; + int badness = 0; + hda_nid_t dac; if (num_outs && !dacs[0]) { - dacs[0] = alc_auto_look_for_dac(codec, pins[0]); - if (!dacs[0]) - return 0; + dac = dacs[0] = alc_auto_look_for_dac(codec, pins[0]); + if (!dacs[0]) { + dac = spec->private_dac_nids[0]; + if (!alc_auto_is_dac_reachable(codec, pins[0], dac)) + return BAD_NO_DAC; + badness += BAD_NO_EXTRA_DAC; + } + badness += eval_shared_vol_badness(codec, pins[0], dac); } for (i = 1; i < num_outs; i++) dacs[i] = get_dac_if_single(codec, pins[i]); for (i = 1; i < num_outs; i++) { - if (!dacs[i]) - dacs[i] = alc_auto_look_for_dac(codec, pins[i]); + dac = dacs[i]; + if (!dac) + dac = dacs[i] = alc_auto_look_for_dac(codec, pins[i]); + if (!dac) { + if (alc_auto_is_dac_reachable(codec, pins[i], dacs[0])) { + dac = dacs[0]; + badness += BAD_SHARED_EXTRA_SURROUND; + } else if (alc_auto_is_dac_reachable(codec, pins[i], + spec->private_dac_nids[0])) { + dac = spec->private_dac_nids[0]; + badness += BAD_NO_EXTRA_SURR_DAC; + } else + badness += BAD_NO_DAC; + } + if (dac) + badness += eval_shared_vol_badness(codec, pins[i], dac); } - return 1; + return badness; } static int alc_auto_fill_multi_ios(struct hda_codec *codec, unsigned int location, int offset); -static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec, - hda_nid_t pin, hda_nid_t dac); /* fill in the dac_nids table from the parsed pin configuration */ -static int alc_auto_fill_dac_nids(struct hda_codec *codec) +static int fill_and_eval_dacs(struct hda_codec *codec, + bool fill_hardwired) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int location, defcfg; - int num_pins; - bool redone = false; - int i; + int i, err, badness; - again: /* set num_dacs once to full for alc_auto_look_for_dac() */ spec->multiout.num_dacs = cfg->line_outs; - spec->multiout.hp_out_nid[0] = 0; - spec->multiout.extra_out_nid[0] = 0; - memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids)); spec->multiout.dac_nids = spec->private_dac_nids; + memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids)); + memset(spec->multiout.hp_out_nid, 0, sizeof(spec->multiout.hp_out_nid)); + memset(spec->multiout.extra_out_nid, 0, sizeof(spec->multiout.extra_out_nid)); spec->multi_ios = 0; + clear_vol_marks(codec); + badness = 0; /* fill hard-wired DACs first */ - if (!redone) { + if (fill_hardwired) { for (i = 0; i < cfg->line_outs; i++) spec->private_dac_nids[i] = get_dac_if_single(codec, cfg->line_out_pins[i]); - if (cfg->hp_outs) - spec->multiout.hp_out_nid[0] = - get_dac_if_single(codec, cfg->hp_pins[0]); - if (cfg->speaker_outs) - spec->multiout.extra_out_nid[0] = - get_dac_if_single(codec, cfg->speaker_pins[0]); + for (i = 0; i < cfg->hp_outs; i++) + spec->multiout.hp_out_nid[i] = + get_dac_if_single(codec, cfg->hp_pins[i]); + for (i = 0; i < cfg->speaker_outs; i++) + spec->multiout.extra_out_nid[i] = + get_dac_if_single(codec, cfg->speaker_pins[i]); } for (i = 0; i < cfg->line_outs; i++) { hda_nid_t pin = cfg->line_out_pins[i]; - if (spec->private_dac_nids[i]) - continue; - spec->private_dac_nids[i] = alc_auto_look_for_dac(codec, pin); - if (!spec->private_dac_nids[i] && !redone) { - /* if we can't find primary DACs, re-probe without - * checking the hard-wired DACs - */ - redone = true; - goto again; + hda_nid_t dac; + if (!spec->private_dac_nids[i]) + spec->private_dac_nids[i] = + alc_auto_look_for_dac(codec, pin); + dac = spec->private_dac_nids[i]; + if (!dac) { + if (!i) + badness += BAD_NO_PRIMARY_DAC; + else if (alc_auto_is_dac_reachable(codec, pin, + spec->private_dac_nids[0])) { + if (i == 1) + badness += BAD_SHARED_SURROUND; + else + badness += BAD_SHARED_CLFE; + dac = spec->private_dac_nids[0]; + } else + badness += BAD_NO_DAC; } + if (dac) + badness += eval_shared_vol_badness(codec, pin, dac); } /* re-count num_dacs and squash invalid entries */ @@ -3071,26 +3186,114 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) /* try to fill multi-io first */ defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]); location = get_defcfg_location(defcfg); - - num_pins = alc_auto_fill_multi_ios(codec, location, 0); - if (num_pins > 0) { - spec->multi_ios = num_pins; - spec->ext_channel_count = 2; - spec->multiout.num_dacs = num_pins + 1; - } + err = alc_auto_fill_multi_ios(codec, location, 0); + if (err < 0) + return err; + badness += err; } - if (cfg->line_out_type != AUTO_PIN_HP_OUT) - alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins, - spec->multiout.hp_out_nid); + if (cfg->line_out_type != AUTO_PIN_HP_OUT) { + err = alc_auto_fill_extra_dacs(codec, cfg->hp_outs, + cfg->hp_pins, + spec->multiout.hp_out_nid); + if (err < 0) + return err; + badness += err; + } if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { - int err = alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, - cfg->speaker_pins, - spec->multiout.extra_out_nid); - /* if no speaker volume is assigned, try again as the primary - * output - */ - if (!err && cfg->speaker_outs > 0 && + err = alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, + cfg->speaker_pins, + spec->multiout.extra_out_nid); + if (err < 0) + return err; + badness += err; + } + if (!spec->multi_ios && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && + cfg->hp_outs) { + /* try multi-ios with HP + inputs */ + defcfg = snd_hda_codec_get_pincfg(codec, cfg->hp_pins[0]); + location = get_defcfg_location(defcfg); + err = alc_auto_fill_multi_ios(codec, location, 1); + if (err < 0) + return err; + badness += err; + } + + return badness; +} + +#define DEBUG_BADNESS + +#ifdef DEBUG_BADNESS +#define debug_badness snd_printdd +#else +#define debug_badness(...) +#endif + +static void debug_show_configs(struct alc_spec *spec, struct auto_pin_cfg *cfg) +{ + debug_badness("multi_outs = %x/%x/%x/%x : %x/%x/%x/%x\n", + cfg->line_out_pins[0], cfg->line_out_pins[1], + cfg->line_out_pins[2], cfg->line_out_pins[2], + spec->multiout.dac_nids[0], + spec->multiout.dac_nids[1], + spec->multiout.dac_nids[2], + spec->multiout.dac_nids[3]); + debug_badness("hp_outs = %x/%x/%x/%x : %x/%x/%x/%x\n", + cfg->hp_pins[0], cfg->hp_pins[1], + cfg->hp_pins[2], cfg->hp_pins[2], + spec->multiout.hp_out_nid[0], + spec->multiout.hp_out_nid[1], + spec->multiout.hp_out_nid[2], + spec->multiout.hp_out_nid[3]); + debug_badness("spk_outs = %x/%x/%x/%x : %x/%x/%x/%x\n", + cfg->speaker_pins[0], cfg->speaker_pins[1], + cfg->speaker_pins[2], cfg->speaker_pins[3], + spec->multiout.extra_out_nid[0], + spec->multiout.extra_out_nid[1], + spec->multiout.extra_out_nid[2], + spec->multiout.extra_out_nid[3]); +} + +static int alc_auto_fill_dac_nids(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct auto_pin_cfg *best_cfg; + int best_badness = INT_MAX; + int badness; + bool fill_hardwired = true; + bool best_wired = true; + bool hp_spk_swapped = false; + + best_cfg = kmalloc(sizeof(*best_cfg), GFP_KERNEL); + if (!best_cfg) + return -ENOMEM; + *best_cfg = *cfg; + + for (;;) { + badness = fill_and_eval_dacs(codec, fill_hardwired); + if (badness < 0) + return badness; + debug_badness("==> lo_type=%d, wired=%d, badness=0x%x\n", + cfg->line_out_type, fill_hardwired, badness); + debug_show_configs(spec, cfg); + if (badness < best_badness) { + best_badness = badness; + *best_cfg = *cfg; + best_wired = fill_hardwired; + } + if (!badness) + break; + if (fill_hardwired) { + fill_hardwired = false; + continue; + } + if (hp_spk_swapped) + break; + hp_spk_swapped = true; + if (cfg->speaker_outs > 0 && cfg->line_out_type == AUTO_PIN_HP_OUT) { cfg->hp_outs = cfg->line_outs; memcpy(cfg->hp_pins, cfg->line_out_pins, @@ -3101,48 +3304,45 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->speaker_outs = 0; memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; - redone = false; - goto again; - } + fill_hardwired = true; + continue; + } + if (cfg->hp_outs > 0 && + cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { + cfg->speaker_outs = cfg->line_outs; + memcpy(cfg->speaker_pins, cfg->line_out_pins, + sizeof(cfg->speaker_pins)); + cfg->line_outs = cfg->hp_outs; + memcpy(cfg->line_out_pins, cfg->hp_pins, + sizeof(cfg->hp_pins)); + cfg->hp_outs = 0; + memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins)); + cfg->line_out_type = AUTO_PIN_HP_OUT; + fill_hardwired = true; + continue; + } + break; } - if (!spec->multi_ios && - cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && - cfg->hp_outs) { - /* try multi-ios with HP + inputs */ - defcfg = snd_hda_codec_get_pincfg(codec, cfg->hp_pins[0]); - location = get_defcfg_location(defcfg); - - num_pins = alc_auto_fill_multi_ios(codec, location, 1); - if (num_pins > 0) { - spec->multi_ios = num_pins; - spec->ext_channel_count = 2; - spec->multiout.num_dacs = num_pins + 1; - } + if (badness) { + *cfg = *best_cfg; + fill_and_eval_dacs(codec, best_wired); } + debug_badness("==> Best config: lo_type=%d, wired=%d\n", + cfg->line_out_type, best_wired); + debug_show_configs(spec, cfg); if (cfg->line_out_pins[0]) spec->vmaster_nid = alc_look_for_out_vol_nid(codec, cfg->line_out_pins[0], spec->multiout.dac_nids[0]); - return 0; -} -static inline unsigned int get_ctl_pos(unsigned int data) -{ - hda_nid_t nid = get_amp_nid_(data); - unsigned int dir; - if (snd_BUG_ON(nid >= MAX_VOL_NIDS)) - return 0; - dir = get_amp_direction_(data); - return (nid << 1) | dir; + /* clear the bitmap flags for creating controls */ + clear_vol_marks(codec); + kfree(best_cfg); + return 0; } -#define is_ctl_used(bits, data) \ - test_bit(get_ctl_pos(data), bits) -#define mark_ctl_usage(bits, data) \ - set_bit(get_ctl_pos(data), bits) - static int alc_auto_add_vol_ctl(struct hda_codec *codec, const char *pfx, int cidx, hda_nid_t nid, unsigned int chs) @@ -3539,6 +3739,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t prime_dac = spec->private_dac_nids[0]; int type, i, dacs, num_pins = 0; + int badness = 0; dacs = spec->multiout.num_dacs; for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) { @@ -3563,12 +3764,16 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, } if (!dac) dac = alc_auto_look_for_dac(codec, nid); - if (!dac) + if (!dac) { + badness += BAD_MULTI_IO; continue; + } spec->multi_io[num_pins].pin = nid; spec->multi_io[num_pins].dac = dac; num_pins++; spec->private_dac_nids[spec->multiout.num_dacs++] = dac; + if (num_pins >= 2) + break; } } spec->multiout.num_dacs = dacs; @@ -3577,9 +3782,13 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, memset(spec->private_dac_nids + dacs, 0, sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - dacs)); spec->private_dac_nids[0] = prime_dac; - return 0; + return badness; } - return num_pins; + + spec->multi_ios = num_pins; + spec->ext_channel_count = 2; + spec->multiout.num_dacs = num_pins + 1; + return 0; } static int alc_auto_ch_mode_info(struct snd_kcontrol *kcontrol, -- cgit v1.2.3-18-g5258 From 185d99f1924ee0047bcd524c58a01c9f8d58d673 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2012 18:39:45 +0100 Subject: ALSA: hda/realtek - Try harder to fit the single-connections So far, the Realtek driver tires to assign the single-connected routes for all pins only once at the beginning. However, since some DACs have been already mapped, the rest pins might have also single conections. In this patch, the driver does the single-connection assignment in a loop until all possbile single-connections are checked. This will improve the DAC assignment, e.g. for ASUS G72. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 124 +++++++++++++++++++++++++++++++----------- 1 file changed, 91 insertions(+), 33 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4746afa25db..29c1925e918 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2926,10 +2926,22 @@ static int alc_auto_select_dac(struct hda_codec *codec, hda_nid_t pin, return 0; } +static bool alc_is_dac_already_used(struct hda_codec *codec, hda_nid_t nid) +{ + struct alc_spec *spec = codec->spec; + if (found_in_nid_list(nid, spec->multiout.dac_nids, + ARRAY_SIZE(spec->private_dac_nids)) || + found_in_nid_list(nid, spec->multiout.hp_out_nid, + ARRAY_SIZE(spec->multiout.hp_out_nid)) || + found_in_nid_list(nid, spec->multiout.extra_out_nid, + ARRAY_SIZE(spec->multiout.extra_out_nid))) + return true; + return false; +} + /* look for an empty DAC slot */ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) { - struct alc_spec *spec = codec->spec; hda_nid_t srcs[5]; int i, num; @@ -2939,16 +2951,8 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) hda_nid_t nid = alc_auto_mix_to_dac(codec, srcs[i]); if (!nid) continue; - if (found_in_nid_list(nid, spec->multiout.dac_nids, - ARRAY_SIZE(spec->private_dac_nids))) - continue; - if (found_in_nid_list(nid, spec->multiout.hp_out_nid, - ARRAY_SIZE(spec->multiout.hp_out_nid))) - continue; - if (found_in_nid_list(nid, spec->multiout.extra_out_nid, - ARRAY_SIZE(spec->multiout.extra_out_nid))) - continue; - return nid; + if (!alc_is_dac_already_used(codec, nid)) + return nid; } return 0; } @@ -2974,12 +2978,23 @@ static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) { struct alc_spec *spec = codec->spec; hda_nid_t sel = alc_go_down_to_selector(codec, pin); - hda_nid_t srcs[5]; - int num = snd_hda_get_connections(codec, sel, srcs, + hda_nid_t nid, nid_found, srcs[5]; + int i, num = snd_hda_get_connections(codec, sel, srcs, ARRAY_SIZE(srcs)); - if (num == 1 || (num == 2 && srcs[1] == spec->mixer_nid)) + if (num == 1) return alc_auto_look_for_dac(codec, pin); - return 0; + nid_found = 0; + for (i = 0; i < num; i++) { + if (srcs[i] == spec->mixer_nid) + continue; + nid = alc_auto_mix_to_dac(codec, srcs[i]); + if (nid && !alc_is_dac_already_used(codec, nid)) { + if (nid_found) + return 0; + nid_found = nid; + } + } + return nid_found; } /* mark up volume and mute control NIDs: used during badness parsing and @@ -3076,16 +3091,30 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, int badness = 0; hda_nid_t dac; - if (num_outs && !dacs[0]) { - dac = dacs[0] = alc_auto_look_for_dac(codec, pins[0]); - if (!dacs[0]) { - dac = spec->private_dac_nids[0]; - if (!alc_auto_is_dac_reachable(codec, pins[0], dac)) - return BAD_NO_DAC; - badness += BAD_NO_EXTRA_DAC; + if (!num_outs) + return 0; + + if (!dacs[0]) + dacs[0] = alc_auto_look_for_dac(codec, pins[0]); + if (!dacs[0]) { + for (i = 1; i < num_outs; i++) { + dac = dacs[i]; + if (dac && alc_auto_is_dac_reachable(codec, pins[0], dac)) { + dacs[0] = dac; + dacs[i] = 0; + break; + } } - badness += eval_shared_vol_badness(codec, pins[0], dac); } + dac = dacs[0]; + if (!dac) { + dac = spec->private_dac_nids[0]; + if (!alc_auto_is_dac_reachable(codec, pins[0], dac)) + return BAD_NO_DAC; + badness += BAD_NO_EXTRA_DAC; + } + if (dac) + badness += eval_shared_vol_badness(codec, pins[0], dac); for (i = 1; i < num_outs; i++) dacs[i] = get_dac_if_single(codec, pins[i]); @@ -3113,6 +3142,21 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, static int alc_auto_fill_multi_ios(struct hda_codec *codec, unsigned int location, int offset); +static bool alc_map_singles(struct hda_codec *codec, int outs, + const hda_nid_t *pins, hda_nid_t *dacs) +{ + int i; + bool found = false; + for (i = 0; i < outs; i++) { + if (dacs[i]) + continue; + dacs[i] = get_dac_if_single(codec, pins[i]); + if (dacs[i]) + found = true; + } + return found; +} + /* fill in the dac_nids table from the parsed pin configuration */ static int fill_and_eval_dacs(struct hda_codec *codec, bool fill_hardwired) @@ -3120,7 +3164,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec, struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int location, defcfg; - int i, err, badness; + int i, j, err, badness; /* set num_dacs once to full for alc_auto_look_for_dac() */ spec->multiout.num_dacs = cfg->line_outs; @@ -3134,15 +3178,18 @@ static int fill_and_eval_dacs(struct hda_codec *codec, /* fill hard-wired DACs first */ if (fill_hardwired) { - for (i = 0; i < cfg->line_outs; i++) - spec->private_dac_nids[i] = - get_dac_if_single(codec, cfg->line_out_pins[i]); - for (i = 0; i < cfg->hp_outs; i++) - spec->multiout.hp_out_nid[i] = - get_dac_if_single(codec, cfg->hp_pins[i]); - for (i = 0; i < cfg->speaker_outs; i++) - spec->multiout.extra_out_nid[i] = - get_dac_if_single(codec, cfg->speaker_pins[i]); + bool mapped; + do { + mapped = alc_map_singles(codec, cfg->line_outs, + cfg->line_out_pins, + spec->private_dac_nids); + mapped |= alc_map_singles(codec, cfg->hp_outs, + cfg->hp_pins, + spec->multiout.hp_out_nid); + mapped |= alc_map_singles(codec, cfg->speaker_outs, + cfg->speaker_pins, + spec->multiout.extra_out_nid); + } while (mapped); } for (i = 0; i < cfg->line_outs; i++) { @@ -3152,6 +3199,17 @@ static int fill_and_eval_dacs(struct hda_codec *codec, spec->private_dac_nids[i] = alc_auto_look_for_dac(codec, pin); dac = spec->private_dac_nids[i]; + if (!dac && !i) { + for (j = 1; j < cfg->line_outs; j++) { + hda_nid_t dac2 = spec->private_dac_nids[j]; + if (dac2 && + alc_auto_is_dac_reachable(codec, pin, dac2)) { + dac = spec->private_dac_nids[0] = dac2; + spec->private_dac_nids[j] = 0; + break; + } + } + } if (!dac) { if (!i) badness += BAD_NO_PRIMARY_DAC; -- cgit v1.2.3-18-g5258 From 71de4d27c8e7e1d27dcab8bb983eff80ec8a3299 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 22:26:23 -0800 Subject: ASoC: wm8962: Update the clocking when setting system clock Make sure we update for any changes in cases where we reconfigure while live (eg, for analogue bypass). Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5a6516ee06a..a5b8a096358 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2705,6 +2705,8 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; + wm8962_configure_bclk(codec); + return 0; } -- cgit v1.2.3-18-g5258 From 1993502d24694c604414d42cccd3aa7a80c0b7ec Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 00:46:44 -0800 Subject: ASoC: wm8962: Only configure BCLK in hw_params when audio is active Otherwise we might not have a sensible clocking setup ready. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a5b8a096358..d285bc1556d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2675,7 +2675,8 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, WM8962_SAMPLE_RATE_INT_MODE | WM8962_SAMPLE_RATE_MASK, adctl3); - wm8962_configure_bclk(codec); + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) + wm8962_configure_bclk(codec); return 0; } -- cgit v1.2.3-18-g5258 From eeba1f8b6ab103c42c25751620fb8625bcff4c10 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 00:19:30 -0800 Subject: ASoC: wm8962: Log the selected SYSCLK ratio Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d285bc1556d..6b80e491ff7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2479,6 +2479,8 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) return; } + dev_dbg(codec->dev, "Selected sysclk ratio %d\n", sysclk_rates[i]); + snd_soc_update_bits(codec, WM8962_CLOCKING_4, WM8962_SYSCLK_RATE_MASK, clocking4); -- cgit v1.2.3-18-g5258 From 07fabd1bfb0036fa6a7c4b5d4db1540623ace742 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 00:19:47 -0800 Subject: ASoC: wm8962: Add new SYSCLK ratios for new device revisions Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6b80e491ff7..95a8b6aae22 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2445,7 +2445,7 @@ static const int bclk_divs[] = { }; static const int sysclk_rates[] = { - 64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536, + 64, 128, 192, 256, 384, 512, 768, 1024, 1408, 1536, 3072, 6144 }; static void wm8962_configure_bclk(struct snd_soc_codec *codec) -- cgit v1.2.3-18-g5258 From 5aa9b858ef4ad9dc5551abe62a92d16f9c7a9680 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 00:46:05 -0800 Subject: ASoC: wm8962: Clean up register dump cruft No longer needed with regmap. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 95a8b6aae22..c36178b3d44 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3630,20 +3630,11 @@ static int wm8962_remove(struct snd_soc_codec *codec) return 0; } -static int wm8962_soc_volatile(struct snd_soc_codec *codec, - unsigned int reg) -{ - return true; -} - - static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .probe = wm8962_probe, .remove = wm8962_remove, .set_bias_level = wm8962_set_bias_level, .set_pll = wm8962_set_fll, - .reg_cache_size = WM8962_MAX_REGISTER, - .volatile_register = wm8962_soc_volatile, .idle_bias_off = true, }; -- cgit v1.2.3-18-g5258 From a968d9db3b3a9329587b09bd15f4981473c63a9d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 27 Jan 2012 19:54:03 +0000 Subject: ASoC: wm8962: Don't automatically enable and disable FLL Only enable and disable the FLL when explicitly told to, supporting some additional use cases and making the driver behaviour more standard. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 72 ++++------------------------------------------- 1 file changed, 6 insertions(+), 66 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c36178b3d44..3dba53ace6b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1832,65 +1832,6 @@ SOC_SINGLE_TLV("SPKOUTR Mixer DACR Volume", WM8962_SPEAKER_MIXER_5, 4, 1, 0, inmix_tlv), }; -static int sysclk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - unsigned long timeout; - int src; - int fll; - - /* Ignore attempts to run the event during startup */ - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - return 0; - - src = snd_soc_read(codec, WM8962_CLOCKING2) & WM8962_SYSCLK_SRC_MASK; - - switch (src) { - case 0: /* MCLK */ - fll = 0; - break; - case 0x200: /* FLL */ - fll = 1; - break; - default: - dev_err(codec->dev, "Unknown SYSCLK source %x\n", src); - return -EINVAL; - } - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - if (fll) { - try_wait_for_completion(&wm8962->fll_lock); - - snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, - WM8962_FLL_ENA, WM8962_FLL_ENA); - - timeout = msecs_to_jiffies(5); - timeout = wait_for_completion_timeout(&wm8962->fll_lock, - timeout); - - if (wm8962->irq && timeout == 0) - dev_err(codec->dev, - "Timed out starting FLL\n"); - } - break; - - case SND_SOC_DAPM_POST_PMD: - if (fll) - snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, - WM8962_FLL_ENA, 0); - break; - - default: - BUG(); - return -EINVAL; - } - - return 0; -} - static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -2176,8 +2117,7 @@ SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_SUPPLY("MICBIAS", WM8962_PWR_MGMT_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Class G", WM8962_CHARGE_PUMP_B, 0, 1, NULL, 0), -SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Charge Pump", WM8962_CHARGE_PUMP_1, 0, 0, cp_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), @@ -2888,8 +2828,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, struct _fll_div fll_div; unsigned long timeout; int ret; - int fll1 = snd_soc_read(codec, WM8962_FLL_CONTROL_1) & WM8962_FLL_ENA; - int sysclk = snd_soc_read(codec, WM8962_CLOCKING2) & WM8962_SYSCLK_ENA; + int fll1 = 0; /* Any change? */ if (source == wm8962->fll_src && Fref == wm8962->fll_fref && @@ -2912,6 +2851,9 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (ret != 0) return ret; + /* Parameters good, disable so we can reprogram */ + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, 0); + switch (fll_id) { case WM8962_FLL_MCLK: case WM8962_FLL_BCLK: @@ -2950,12 +2892,10 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, try_wait_for_completion(&wm8962->fll_lock); - if (sysclk) - fll1 |= WM8962_FLL_ENA; snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | - WM8962_FLL_ENA, fll1); + WM8962_FLL_ENA, fll1 | WM8962_FLL_ENA); dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); -- cgit v1.2.3-18-g5258 From d23031a4d3fc13128b2f18867e8a19a1f8aa0eb2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Feb 2012 12:48:59 +0000 Subject: ASoC: wm8962: Convert to runtime PM for bias off management This allows userspace control of final power off, allowing policy decisions for register configuration retention. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 113 +++++++++++++++++++++++++++++----------------- 1 file changed, 72 insertions(+), 41 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3dba53ace6b..28d2e74ed01 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include @@ -2479,9 +2480,6 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) static int wm8962_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - int ret; - if (level == codec->dapm.bias_level) return 0; @@ -2498,51 +2496,15 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), - wm8962->supplies); - if (ret != 0) { - dev_err(codec->dev, - "Failed to enable supplies: %d\n", - ret); - return ret; - } - - regcache_cache_only(wm8962->regmap, false); - regcache_sync(wm8962->regmap); - - snd_soc_update_bits(codec, WM8962_ANTI_POP, - WM8962_STARTUP_BIAS_ENA | - WM8962_VMID_BUF_ENA, - WM8962_STARTUP_BIAS_ENA | - WM8962_VMID_BUF_ENA); - - /* Bias enable at 2*50k for ramp */ - snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK | - WM8962_BIAS_ENA, - WM8962_BIAS_ENA | 0x180); - - msleep(5); - } - /* VMID 2*250k */ snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x100); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, WM8962_PWR_MGMT_1, - WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, 0); - - snd_soc_update_bits(codec, WM8962_ANTI_POP, - WM8962_STARTUP_BIAS_ENA | - WM8962_VMID_BUF_ENA, 0); - - regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), - wm8962->supplies); break; } + codec->dapm.bias_level = level; return 0; } @@ -2844,6 +2806,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, 0); + pm_runtime_put(codec->dev); + return 0; } @@ -2892,6 +2856,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, try_wait_for_completion(&wm8962->fll_lock); + pm_runtime_get_sync(codec->dev); snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | @@ -3689,7 +3654,9 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, ret); } - regcache_cache_only(wm8962->regmap, true); + pm_runtime_set_active(&i2c->dev); + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8962, &wm8962_dai, 1); @@ -3721,6 +3688,69 @@ static __devexit int wm8962_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_PM_RUNTIME +static int wm8962_runtime_resume(struct device *dev) +{ + struct wm8962_priv *wm8962 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8962->supplies), + wm8962->supplies); + if (ret != 0) { + dev_err(dev, + "Failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(wm8962->regmap, false); + regcache_sync(wm8962->regmap); + + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA, + WM8962_STARTUP_BIAS_ENA | WM8962_VMID_BUF_ENA); + + /* Bias enable at 2*50k for ramp */ + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, + WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, + WM8962_BIAS_ENA | 0x180); + + msleep(5); + + /* VMID back to 2x250k for standby */ + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, + WM8962_VMID_SEL_MASK, 0x100); + + dev_crit(dev, "RESUME\n"); + + return 0; +} + +static int wm8962_runtime_suspend(struct device *dev) +{ + struct wm8962_priv *wm8962 = dev_get_drvdata(dev); + + dev_crit(dev, "SUSPEND\n"); + + regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, + WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, 0); + + regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, + WM8962_STARTUP_BIAS_ENA | + WM8962_VMID_BUF_ENA, 0); + + regcache_cache_only(wm8962->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), + wm8962->supplies); + + return 0; +} +#endif + +static struct dev_pm_ops wm8962_pm = { + SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) +}; + static const struct i2c_device_id wm8962_i2c_id[] = { { "wm8962", 0 }, { } @@ -3731,6 +3761,7 @@ static struct i2c_driver wm8962_i2c_driver = { .driver = { .name = "wm8962", .owner = THIS_MODULE, + .pm = &wm8962_pm, }, .probe = wm8962_i2c_probe, .remove = __devexit_p(wm8962_i2c_remove), -- cgit v1.2.3-18-g5258 From d69280be4160963796f3a899722c126ab4889f02 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 22:52:19 -0800 Subject: ASoC: ak4535: Remove -codec from driver name Redundant, the device is only a CODEC. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index dd15516763e..29d4dec1c25 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -443,7 +443,7 @@ MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id); static struct i2c_driver ak4535_i2c_driver = { .driver = { - .name = "ak4535-codec", + .name = "ak4535", .owner = THIS_MODULE, }, .probe = ak4535_i2c_probe, -- cgit v1.2.3-18-g5258 From b08c576434211f2e908f37844dc61ec6e0e1ecec Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 22:52:44 -0800 Subject: ASoC: ak4535: Make I2C usage unconditional Convert to module_i2c_driver() too. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 24 +----------------------- 1 file changed, 1 insertion(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 29d4dec1c25..71fca53c9bb 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -409,7 +409,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .num_dapm_routes = ARRAY_SIZE(ak4535_audio_map), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int ak4535_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -450,29 +449,8 @@ static struct i2c_driver ak4535_i2c_driver = { .remove = __devexit_p(ak4535_i2c_remove), .id_table = ak4535_i2c_id, }; -#endif -static int __init ak4535_modinit(void) -{ - int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&ak4535_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register AK4535 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(ak4535_modinit); - -static void __exit ak4535_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&ak4535_i2c_driver); -#endif -} -module_exit(ak4535_exit); +module_i2c_driver(ak4535_i2c_driver); MODULE_DESCRIPTION("Soc AK4535 driver"); MODULE_AUTHOR("Richard Purdie"); -- cgit v1.2.3-18-g5258 From 7e11a535169f2689141c20a916a96b5ddae7a508 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 22:53:20 -0800 Subject: ASoC: ak4535: Remove bitrotted driver version Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 71fca53c9bb..e13303a6500 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -27,8 +27,6 @@ #include "ak4535.h" -#define AK4535_VERSION "0.3" - /* codec private data */ struct ak4535_priv { unsigned int sysclk; @@ -372,8 +370,6 @@ static int ak4535_probe(struct snd_soc_codec *codec) struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); int ret; - printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4535->control_type); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); -- cgit v1.2.3-18-g5258 From a1fa92130c12c6a5abcfc8b960e442f87ebea475 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 23:03:44 -0800 Subject: ASoC: ak4535: Convert to direct regmap API usage I suspect the timer register may also be volatile. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4535.c | 66 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/ak4535.h | 2 -- 2 files changed, 55 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index e13303a6500..838ae8b22b5 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include @@ -29,20 +30,41 @@ /* codec private data */ struct ak4535_priv { + struct regmap *regmap; unsigned int sysclk; - enum snd_soc_control_type control_type; }; /* * ak4535 register cache */ -static const u8 ak4535_reg[AK4535_CACHEREGNUM] = { - 0x00, 0x80, 0x00, 0x03, - 0x02, 0x00, 0x11, 0x01, - 0x00, 0x40, 0x36, 0x10, - 0x00, 0x00, 0x57, 0x00, +static const struct reg_default ak4535_reg_defaults[] = { + { 0, 0x00 }, + { 1, 0x80 }, + { 2, 0x00 }, + { 3, 0x03 }, + { 4, 0x02 }, + { 5, 0x00 }, + { 6, 0x11 }, + { 7, 0x01 }, + { 8, 0x00 }, + { 9, 0x40 }, + { 10, 0x36 }, + { 11, 0x10 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x57 }, }; +static bool ak4535_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AK4535_STATUS: + return true; + default: + return false; + } +} + static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; @@ -370,7 +392,8 @@ static int ak4535_probe(struct snd_soc_codec *codec) struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4535->control_type); + codec->control_data = ak4535->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -390,15 +413,24 @@ static int ak4535_remove(struct snd_soc_codec *codec) return 0; } +static const struct regmap_config ak4535_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK4535_STATUS, + .volatile_reg = ak4535_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = ak4535_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4535_reg_defaults), +}; + static struct snd_soc_codec_driver soc_codec_dev_ak4535 = { .probe = ak4535_probe, .remove = ak4535_remove, .suspend = ak4535_suspend, .resume = ak4535_resume, .set_bias_level = ak4535_set_bias_level, - .reg_cache_size = ARRAY_SIZE(ak4535_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = ak4535_reg, .dapm_widgets = ak4535_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets), .dapm_routes = ak4535_audio_map, @@ -416,17 +448,29 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c, if (ak4535 == NULL) return -ENOMEM; + ak4535->regmap = regmap_init_i2c(i2c, &ak4535_regmap); + if (IS_ERR(ak4535->regmap)) { + ret = PTR_ERR(ak4535->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + i2c_set_clientdata(i2c, ak4535); - ak4535->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4535, &ak4535_dai, 1); + if (ret != 0) + regmap_exit(ak4535->regmap); + return ret; } static __devexit int ak4535_i2c_remove(struct i2c_client *client) { + struct ak4535_priv *ak4535 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + regmap_exit(ak4535->regmap); return 0; } diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h index 0431e5f634a..402de1d274b 100644 --- a/sound/soc/codecs/ak4535.h +++ b/sound/soc/codecs/ak4535.h @@ -34,6 +34,4 @@ #define AK4535_VOL 0xe #define AK4535_STATUS 0xf -#define AK4535_CACHEREGNUM 0x10 - #endif -- cgit v1.2.3-18-g5258 From 6f4530409199e7c3f8a4bdbd1391b7b25951e397 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 14:09:20 +0100 Subject: ALSA: hda/realtek - Show multi-io pins in debug prints Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 29c1925e918..4b2ecbcbe66 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3298,6 +3298,11 @@ static void debug_show_configs(struct alc_spec *spec, struct auto_pin_cfg *cfg) spec->multiout.dac_nids[1], spec->multiout.dac_nids[2], spec->multiout.dac_nids[3]); + if (spec->multi_ios > 0) + debug_badness("multi_ios(%d) = %x/%x : %x/%x\n", + spec->multi_ios, + spec->multi_io[0].pin, spec->multi_io[1].pin, + spec->multi_io[0].dac, spec->multi_io[1].dac); debug_badness("hp_outs = %x/%x/%x/%x : %x/%x/%x/%x\n", cfg->hp_pins[0], cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[2], -- cgit v1.2.3-18-g5258 From 276dd70baebe6334e603227c064a9beb07cb4e9e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 16:17:03 +0100 Subject: ALSA: hda/realtek - Adjust badness calculation for multi-ios Try harder to fit the multi-io pins also by checking the hard-wired connections for multi-ios. Also, the badness values are adjusted to prioritize the multi-ios as more valuable. These changes will enable the multi-io on some machines without losing the current capability. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 196 +++++++++++++++++++++++++++++------------- 1 file changed, 134 insertions(+), 62 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4b2ecbcbe66..d0c71d5be83 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2929,6 +2929,7 @@ static int alc_auto_select_dac(struct hda_codec *codec, hda_nid_t pin, static bool alc_is_dac_already_used(struct hda_codec *codec, hda_nid_t nid) { struct alc_spec *spec = codec->spec; + int i; if (found_in_nid_list(nid, spec->multiout.dac_nids, ARRAY_SIZE(spec->private_dac_nids)) || found_in_nid_list(nid, spec->multiout.hp_out_nid, @@ -2936,6 +2937,10 @@ static bool alc_is_dac_already_used(struct hda_codec *codec, hda_nid_t nid) found_in_nid_list(nid, spec->multiout.extra_out_nid, ARRAY_SIZE(spec->multiout.extra_out_nid))) return true; + for (i = 0; i < spec->multi_ios; i++) { + if (spec->multi_io[i].dac == nid) + return true; + } return false; } @@ -3028,20 +3033,20 @@ enum { BAD_NO_PRIMARY_DAC = 0x10000, /* No DAC is found for the extra output */ BAD_NO_DAC = 0x4000, + /* No possible multi-ios */ + BAD_MULTI_IO = 0x103, /* No individual DAC for extra output */ - BAD_NO_EXTRA_DAC = 0x1000, + BAD_NO_EXTRA_DAC = 0x102, /* No individual DAC for extra surrounds */ - BAD_NO_EXTRA_SURR_DAC = 0x200, + BAD_NO_EXTRA_SURR_DAC = 0x101, /* Primary DAC shared with main surrounds */ BAD_SHARED_SURROUND = 0x100, - /* Volume widget is shared */ - BAD_SHARED_VOL = 0x10, /* Primary DAC shared with main CLFE */ BAD_SHARED_CLFE = 0x10, /* Primary DAC shared with extra surrounds */ BAD_SHARED_EXTRA_SURROUND = 0x10, - /* No possible multi-ios */ - BAD_MULTI_IO = 0x1, + /* Volume widget is shared */ + BAD_SHARED_VOL = 0x10, }; static hda_nid_t alc_look_for_out_mute_nid(struct hda_codec *codec, @@ -3140,7 +3145,8 @@ static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, } static int alc_auto_fill_multi_ios(struct hda_codec *codec, - unsigned int location, int offset); + hda_nid_t reference_pin, + bool hardwired, int offset); static bool alc_map_singles(struct hda_codec *codec, int outs, const hda_nid_t *pins, hda_nid_t *dacs) @@ -3159,11 +3165,11 @@ static bool alc_map_singles(struct hda_codec *codec, int outs, /* fill in the dac_nids table from the parsed pin configuration */ static int fill_and_eval_dacs(struct hda_codec *codec, - bool fill_hardwired) + bool fill_hardwired, + bool fill_mio_first) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int location, defcfg; int i, j, err, badness; /* set num_dacs once to full for alc_auto_look_for_dac() */ @@ -3181,14 +3187,20 @@ static int fill_and_eval_dacs(struct hda_codec *codec, bool mapped; do { mapped = alc_map_singles(codec, cfg->line_outs, - cfg->line_out_pins, - spec->private_dac_nids); + cfg->line_out_pins, + spec->private_dac_nids); mapped |= alc_map_singles(codec, cfg->hp_outs, cfg->hp_pins, spec->multiout.hp_out_nid); mapped |= alc_map_singles(codec, cfg->speaker_outs, cfg->speaker_pins, spec->multiout.extra_out_nid); + if (fill_mio_first && cfg->line_outs == 1 && + cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + err = alc_auto_fill_multi_ios(codec, cfg->line_out_pins[0], true, 0); + if (!err) + mapped = true; + } } while (mapped); } @@ -3240,14 +3252,13 @@ static int fill_and_eval_dacs(struct hda_codec *codec, } } - if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + if (fill_mio_first && + cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { /* try to fill multi-io first */ - defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]); - location = get_defcfg_location(defcfg); - err = alc_auto_fill_multi_ios(codec, location, 0); + err = alc_auto_fill_multi_ios(codec, cfg->line_out_pins[0], false, 0); if (err < 0) return err; - badness += err; + /* we don't count badness at this stage yet */ } if (cfg->line_out_type != AUTO_PIN_HP_OUT) { @@ -3266,18 +3277,30 @@ static int fill_and_eval_dacs(struct hda_codec *codec, return err; badness += err; } - if (!spec->multi_ios && - cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && - cfg->hp_outs) { + if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + err = alc_auto_fill_multi_ios(codec, cfg->line_out_pins[0], false, 0); + if (err < 0) + return err; + badness += err; + } + if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { /* try multi-ios with HP + inputs */ - defcfg = snd_hda_codec_get_pincfg(codec, cfg->hp_pins[0]); - location = get_defcfg_location(defcfg); - err = alc_auto_fill_multi_ios(codec, location, 1); + err = alc_auto_fill_multi_ios(codec, cfg->hp_pins[0], false, 1); if (err < 0) return err; badness += err; } + if (spec->multi_ios == 2) { + for (i = 0; i < 2; i++) + spec->private_dac_nids[spec->multiout.num_dacs++] = + spec->multi_io[i].dac; + spec->ext_channel_count = 2; + } else if (spec->multi_ios) { + spec->multi_ios = 0; + badness += BAD_MULTI_IO; + } + return badness; } @@ -3326,8 +3349,8 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) struct auto_pin_cfg *best_cfg; int best_badness = INT_MAX; int badness; - bool fill_hardwired = true; - bool best_wired = true; + bool fill_hardwired = true, fill_mio_first = true; + bool best_wired = true, best_mio = true; bool hp_spk_swapped = false; best_cfg = kmalloc(sizeof(*best_cfg), GFP_KERNEL); @@ -3336,23 +3359,28 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) *best_cfg = *cfg; for (;;) { - badness = fill_and_eval_dacs(codec, fill_hardwired); + badness = fill_and_eval_dacs(codec, fill_hardwired, + fill_mio_first); if (badness < 0) return badness; - debug_badness("==> lo_type=%d, wired=%d, badness=0x%x\n", - cfg->line_out_type, fill_hardwired, badness); + debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n", + cfg->line_out_type, fill_hardwired, fill_mio_first, + badness); debug_show_configs(spec, cfg); if (badness < best_badness) { best_badness = badness; *best_cfg = *cfg; best_wired = fill_hardwired; + best_mio = fill_mio_first; } if (!badness) break; - if (fill_hardwired) { - fill_hardwired = false; + fill_mio_first = !fill_mio_first; + if (!fill_mio_first) + continue; + fill_hardwired = !fill_hardwired; + if (!fill_hardwired) continue; - } if (hp_spk_swapped) break; hp_spk_swapped = true; @@ -3389,10 +3417,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) if (badness) { *cfg = *best_cfg; - fill_and_eval_dacs(codec, best_wired); + fill_and_eval_dacs(codec, best_wired, best_mio); } - debug_badness("==> Best config: lo_type=%d, wired=%d\n", - cfg->line_out_type, best_wired); + debug_badness("==> Best config: lo_type=%d, wired=%d, mio=%d\n", + cfg->line_out_type, best_wired, best_mio); debug_show_configs(spec, cfg); if (cfg->line_out_pins[0]) @@ -3791,66 +3819,110 @@ static void alc_auto_init_extra_out(struct hda_codec *codec) } } +/* check whether the given pin can be a multi-io pin */ +static bool can_be_multiio_pin(struct hda_codec *codec, + unsigned int location, hda_nid_t nid) +{ + unsigned int defcfg, caps; + + defcfg = snd_hda_codec_get_pincfg(codec, nid); + if (get_defcfg_connect(defcfg) != AC_JACK_PORT_COMPLEX) + return false; + if (location && get_defcfg_location(defcfg) != location) + return false; + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_OUT)) + return false; + return true; +} + /* * multi-io helper + * + * When hardwired is set, try to fill ony hardwired pins, and returns + * zero if any pins are filled, non-zero if nothing found. + * When hardwired is off, try to fill possible input pins, and returns + * the badness value. */ static int alc_auto_fill_multi_ios(struct hda_codec *codec, - unsigned int location, - int offset) + hda_nid_t reference_pin, + bool hardwired, int offset) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - hda_nid_t prime_dac = spec->private_dac_nids[0]; - int type, i, dacs, num_pins = 0; + int type, i, j, dacs, num_pins, old_pins; + unsigned int defcfg = snd_hda_codec_get_pincfg(codec, reference_pin); + unsigned int location = get_defcfg_location(defcfg); int badness = 0; + old_pins = spec->multi_ios; + if (old_pins >= 2) + goto end_fill; + + num_pins = 0; + for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) { + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].type != type) + continue; + if (can_be_multiio_pin(codec, location, + cfg->inputs[i].pin)) + num_pins++; + } + } + if (num_pins < 2) + goto end_fill; + dacs = spec->multiout.num_dacs; for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) { for (i = 0; i < cfg->num_inputs; i++) { hda_nid_t nid = cfg->inputs[i].pin; hda_nid_t dac = 0; - unsigned int defcfg, caps; + if (cfg->inputs[i].type != type) continue; - defcfg = snd_hda_codec_get_pincfg(codec, nid); - if (get_defcfg_connect(defcfg) != AC_JACK_PORT_COMPLEX) - continue; - if (location && get_defcfg_location(defcfg) != location) + if (!can_be_multiio_pin(codec, location, nid)) continue; - caps = snd_hda_query_pin_caps(codec, nid); - if (!(caps & AC_PINCAP_OUT)) + for (j = 0; j < spec->multi_ios; j++) { + if (nid == spec->multi_io[j].pin) + break; + } + if (j < spec->multi_ios) continue; - if (offset && offset + num_pins < dacs) { - dac = spec->private_dac_nids[offset + num_pins]; + + if (offset && offset + spec->multi_ios < dacs) { + dac = spec->private_dac_nids[offset + spec->multi_ios]; if (!alc_auto_is_dac_reachable(codec, nid, dac)) dac = 0; } - if (!dac) + if (hardwired) + dac = get_dac_if_single(codec, nid); + else if (!dac) dac = alc_auto_look_for_dac(codec, nid); if (!dac) { - badness += BAD_MULTI_IO; + badness++; continue; } - spec->multi_io[num_pins].pin = nid; - spec->multi_io[num_pins].dac = dac; - num_pins++; - spec->private_dac_nids[spec->multiout.num_dacs++] = dac; - if (num_pins >= 2) + spec->multi_io[spec->multi_ios].pin = nid; + spec->multi_io[spec->multi_ios].dac = dac; + spec->multi_ios++; + if (spec->multi_ios >= 2) break; } } - spec->multiout.num_dacs = dacs; - if (num_pins < 2) { - /* clear up again */ - memset(spec->private_dac_nids + dacs, 0, - sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - dacs)); - spec->private_dac_nids[0] = prime_dac; + end_fill: + if (badness) + badness = BAD_MULTI_IO; + if (old_pins == spec->multi_ios) { + if (hardwired) + return 1; /* nothing found */ + else + return badness; /* no badness if nothing found */ + } + if (!hardwired && spec->multi_ios < 2) { + spec->multi_ios = old_pins; return badness; } - spec->multi_ios = num_pins; - spec->ext_channel_count = 2; - spec->multiout.num_dacs = num_pins + 1; return 0; } -- cgit v1.2.3-18-g5258 From dc6af52dea5ada1269095cad5ed2c04e92114399 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 16:18:59 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=lg with the auto-parser ALC880 model=lg could work fine with the auto-parser due to the recent rewrite, but it still needs the manual adjustment; namely, the BIOS leaves unused pins as some real active jacks. This confuses the parser. Thus we just cover these pins and override the pin-configs as a fix-up. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 158 ------------------------------------------ sound/pci/hda/patch_realtek.c | 14 ++++ 2 files changed, 14 insertions(+), 158 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 501501ef36a..3b88bc561e1 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -25,7 +25,6 @@ enum { ALC880_UNIWILL_P53, ALC880_CLEVO, ALC880_TCL_S700, - ALC880_LG, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -773,11 +772,6 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, } } -static void alc880_unsol_event(struct hda_codec *codec, unsigned int res) -{ - alc_exec_unsol_event(codec, res >> 28); -} - static void alc880_uniwill_p53_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -936,136 +930,6 @@ static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { { } }; -/* - * LG m1 express dual - * - * Pin assignment: - * Rear Line-In/Out (blue): 0x14 - * Build-in Mic-In: 0x15 - * Speaker-out: 0x17 - * HP-Out (green): 0x1b - * Mic-In/Out (red): 0x19 - * SPDIF-Out: 0x1e - */ - -/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */ -static const hda_nid_t alc880_lg_dac_nids[3] = { - 0x05, 0x02, 0x03 -}; - -/* seems analog CD is not working */ -static const struct hda_input_mux alc880_lg_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x1 }, - { "Line", 0x5 }, - { "Internal Mic", 0x6 }, - }, -}; - -/* 2,4,6 channel modes */ -static const struct hda_verb alc880_lg_ch2_init[] = { - /* set line-in and mic-in to input */ - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { } -}; - -static const struct hda_verb alc880_lg_ch4_init[] = { - /* set line-in to out and mic-in to input */ - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { } -}; - -static const struct hda_verb alc880_lg_ch6_init[] = { - /* set line-in and mic-in to output */ - { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - { } -}; - -static const struct hda_channel_mode alc880_lg_ch_modes[3] = { - { 2, alc880_lg_ch2_init }, - { 4, alc880_lg_ch4_init }, - { 6, alc880_lg_ch6_init }, -}; - -static const struct snd_kcontrol_new alc880_lg_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static const struct hda_verb alc880_lg_init_verbs[] = { - /* set capture source to mic-in */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* mute all amp mixer inputs */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* line-in to input */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* built-in mic */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* speaker-out */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* mic-in to input */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* HP-out */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x03}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* jack sense */ - {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - { } -}; - -/* toggle speaker-output according to the hp-jack state */ -static void alc880_lg_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x1b; - spec->autocfg.speaker_pins[0] = 0x17; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -#ifdef CONFIG_SND_HDA_POWER_SAVE -static const struct hda_amp_list alc880_lg_loopbacks[] = { - { 0x0b, HDA_INPUT, 1 }, - { 0x0b, HDA_INPUT, 6 }, - { 0x0b, HDA_INPUT, 7 }, - { } /* end */ -}; -#endif - /* * Test configuration for debugging * @@ -1352,7 +1216,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_UNIWILL_P53] = "uniwill-p53", [ALC880_FUJITSU] = "fujitsu", [ALC880_F1734] = "F1734", - [ALC880_LG] = "lg", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -1409,9 +1272,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), - SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG), - SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG), - SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), @@ -1673,24 +1533,6 @@ static const struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, - [ALC880_LG] = { - .mixers = { alc880_lg_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_lg_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids), - .dac_nids = alc880_lg_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes), - .channel_mode = alc880_lg_ch_modes, - .need_dac_fix = 1, - .input_mux = &alc880_lg_capture_source, - .unsol_event = alc880_unsol_event, - .setup = alc880_lg_setup, - .init_hook = alc_hp_automute, -#ifdef CONFIG_SND_HDA_POWER_SAVE - .loopbacks = alc880_lg_loopbacks, -#endif - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d0c71d5be83..a39146528c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4404,6 +4404,7 @@ static const struct hda_amp_list alc880_loopbacks[] = { enum { ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, + ALC880_FIXUP_LG, }; static const struct alc_fixup alc880_fixups[] = { @@ -4421,10 +4422,23 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_GPIO2, }, + [ALC880_FIXUP_LG] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + /* disable bogus unused pins */ + { 0x16, 0x411111f0 }, + { 0x18, 0x411111f0 }, + { 0x1a, 0x411111f0 }, + { } + } + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), + SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), {} }; -- cgit v1.2.3-18-g5258 From 059ad7602889aa724adb84298dccae92534b7697 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Fri, 17 Feb 2012 16:15:56 +0800 Subject: ASoC: Change spitz_ext_control to take dapm as argument. This fixes below build warning: CC sound/soc/pxa/spitz.o sound/soc/pxa/spitz.c: In function 'spitz_startup': sound/soc/pxa/spitz.c:116: warning: passing argument 1 of 'spitz_ext_control' from incompatible pointer type sound/soc/pxa/spitz.c:47: note: expected 'struct snd_soc_card *' but argument is of type 'struct snd_soc_codec *' Signed-off-by: Axel Lin Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/pxa/spitz.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 4d108dc52e2..fc052d8247f 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -44,10 +44,8 @@ static int spitz_jack_func; static int spitz_spk_func; static int spitz_mic_gpio; -static void spitz_ext_control(struct snd_soc_card *card) +static void spitz_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &card->dapm; - if (spitz_spk_func == SPITZ_SPK_ON) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else @@ -113,7 +111,7 @@ static int spitz_startup(struct snd_pcm_substream *substream) mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - spitz_ext_control(codec); + spitz_ext_control(&codec->dapm); mutex_unlock(&codec->mutex); @@ -179,7 +177,7 @@ static int spitz_set_jack(struct snd_kcontrol *kcontrol, return 0; spitz_jack_func = ucontrol->value.integer.value[0]; - spitz_ext_control(card); + spitz_ext_control(&card->dapm); return 1; } @@ -193,13 +191,13 @@ static int spitz_get_spk(struct snd_kcontrol *kcontrol, static int spitz_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (spitz_spk_func == ucontrol->value.integer.value[0]) return 0; spitz_spk_func = ucontrol->value.integer.value[0]; - spitz_ext_control(card); + spitz_ext_control(&card->dapm); return 1; } -- cgit v1.2.3-18-g5258 From ce0e9f0ede349097c849db9c3aa7e947fc443552 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 11:02:11 -0800 Subject: ASoC: dapm: Unexport snd_soc_dapm_new_control() Everything now uses snd_soc_dapm_new_controls() instead so we don't need to make it part of the external API. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0740cfc3d99..1bcce75058f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2707,8 +2707,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); * * Returns 0 for success else error. */ -int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget) +static int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; size_t name_len; @@ -2798,7 +2798,6 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->connected = 1; return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); /** * snd_soc_dapm_new_controls - create new dapm controls -- cgit v1.2.3-18-g5258 From 5ba06fc969d068dee9a59f1fa3dbe58e235fa913 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 11:07:13 -0800 Subject: ASoC: dapm: Refactor snd_soc_dapm_new_widget() to return the widget Let the caller fiddle with the widget after we're done in order to facilitate further refactoring. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 35 ++++++++++++++--------------------- 1 file changed, 14 insertions(+), 21 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1bcce75058f..295fa91d9d0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2698,24 +2698,16 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); -/** - * snd_soc_dapm_new_control - create new dapm control - * @dapm: DAPM context - * @widget: widget template - * - * Creates a new dapm control based upon the template. - * - * Returns 0 for success else error. - */ -static int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, - const struct snd_soc_dapm_widget *widget) +static struct snd_soc_dapm_widget * +snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_widget *widget) { struct snd_soc_dapm_widget *w; size_t name_len; int ret; if ((w = dapm_cnew_widget(widget)) == NULL) - return -ENOMEM; + return NULL; switch (w->id) { case snd_soc_dapm_regulator_supply: @@ -2724,7 +2716,7 @@ static int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, ret = PTR_ERR(w->priv); dev_err(dapm->dev, "Failed to request %s: %d\n", w->name, ret); - return ret; + return NULL; } break; default: @@ -2737,7 +2729,7 @@ static int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->name = kmalloc(name_len, GFP_KERNEL); if (w->name == NULL) { kfree(w); - return -ENOMEM; + return NULL; } if (dapm->codec && dapm->codec->name_prefix) snprintf(w->name, name_len, "%s %s", @@ -2796,7 +2788,7 @@ static int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, /* machine layer set ups unconnected pins and insertions */ w->connected = 1; - return 0; + return w; } /** @@ -2813,15 +2805,16 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget, int num) { - int i, ret; + struct snd_soc_dapm_widget *w; + int i; for (i = 0; i < num; i++) { - ret = snd_soc_dapm_new_control(dapm, widget); - if (ret < 0) { + w = snd_soc_dapm_new_control(dapm, widget); + if (!w) { dev_err(dapm->dev, - "ASoC: Failed to create DAPM control %s: %d\n", - widget->name, ret); - return ret; + "ASoC: Failed to create DAPM control %s\n", + widget->name); + return -ENOMEM; } widget++; } -- cgit v1.2.3-18-g5258 From 7bd3a6f34cdd4b1776ca34d0b6fab216e9323759 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 15:03:27 -0800 Subject: ASoC: dapm: Supply the DAI and substream when calling stream events In order to allow us to do something smarter than iterate through widgets doing strcmp() to work out what to power up for stream events change the interface used to generate them to be based on the combination of a DAI and a stream direction rather than just a simple string identifying the stream. At some point we'll probably want a set of channels too. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 30 ++++++++++++++++-------------- sound/soc/soc-dapm.c | 25 ++++++++++++++++--------- sound/soc/soc-pcm.c | 25 +++++++++---------------- 3 files changed, 41 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6bad7cd4131..7645b8a545c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -573,18 +573,20 @@ int snd_soc_suspend(struct device *dev) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai_driver *driver = card->rtd[i].codec_dai->driver; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; - if (driver->playback.stream_name != NULL) - snd_soc_dapm_stream_event(&card->rtd[i], driver->playback.stream_name, - SND_SOC_DAPM_STREAM_SUSPEND); + snd_soc_dapm_stream_event(&card->rtd[i], + SNDRV_PCM_STREAM_PLAYBACK, + codec_dai, + SND_SOC_DAPM_STREAM_SUSPEND); - if (driver->capture.stream_name != NULL) - snd_soc_dapm_stream_event(&card->rtd[i], driver->capture.stream_name, - SND_SOC_DAPM_STREAM_SUSPEND); + snd_soc_dapm_stream_event(&card->rtd[i], + SNDRV_PCM_STREAM_CAPTURE, + codec_dai, + SND_SOC_DAPM_STREAM_SUSPEND); } /* suspend all CODECs */ @@ -687,18 +689,18 @@ static void soc_resume_deferred(struct work_struct *work) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai_driver *driver = card->rtd[i].codec_dai->driver; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; - if (driver->playback.stream_name != NULL) - snd_soc_dapm_stream_event(&card->rtd[i], driver->playback.stream_name, - SND_SOC_DAPM_STREAM_RESUME); + snd_soc_dapm_stream_event(&card->rtd[i], + SNDRV_PCM_STREAM_PLAYBACK, codec_dai, + SND_SOC_DAPM_STREAM_RESUME); - if (driver->capture.stream_name != NULL) - snd_soc_dapm_stream_event(&card->rtd[i], driver->capture.stream_name, - SND_SOC_DAPM_STREAM_RESUME); + snd_soc_dapm_stream_event(&card->rtd[i], + SNDRV_PCM_STREAM_CAPTURE, codec_dai, + SND_SOC_DAPM_STREAM_RESUME); } /* unmute any active DACs */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 295fa91d9d0..97915eb711c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2823,17 +2823,27 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, - const char *stream, int event) + int stream, struct snd_soc_dai *dai, + int event) { struct snd_soc_dapm_widget *w; + const char *stream_name; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + stream_name = dai->driver->playback.stream_name; + else + stream_name = dai->driver->capture.stream_name; + + if (!stream_name) + return; list_for_each_entry(w, &dapm->card->widgets, list) { if (!w->sname || w->dapm != dapm) continue; dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", - w->name, w->sname, stream, event); - if (strstr(w->sname, stream)) { + w->name, w->sname, stream_name, event); + if (strstr(w->sname, stream_name)) { dapm_mark_dirty(w, "stream event"); switch(event) { case SND_SOC_DAPM_STREAM_START: @@ -2865,16 +2875,13 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, * * Returns 0 for success else error. */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, - const char *stream, int event) +int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + struct snd_soc_dai *dai, int event) { struct snd_soc_codec *codec = rtd->codec; - if (stream == NULL) - return 0; - mutex_lock(&codec->mutex); - soc_dapm_stream_event(&codec->dapm, stream, event); + soc_dapm_stream_event(&codec->dapm, stream, dai, event); mutex_unlock(&codec->mutex); return 0; } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 15816eccad3..0ad8dcacd2f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -307,9 +307,8 @@ static void close_delayed_work(struct work_struct *work) /* are we waiting on this codec DAI stream */ if (codec_dai->pop_wait == 1) { codec_dai->pop_wait = 0; - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_STOP); + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + codec_dai, SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -373,8 +372,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) rtd->dai_link->ignore_pmdown_time) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_STOP); + SNDRV_PCM_STREAM_PLAYBACK, + codec_dai, + SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ codec_dai->pop_wait = 1; @@ -383,9 +383,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } } else { /* capture streams can be powered down now */ - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_STOP); + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + codec_dai, SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -454,14 +453,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) cancel_delayed_work(&rtd->delayed_work); } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->playback.stream_name, - SND_SOC_DAPM_STREAM_START); - else - snd_soc_dapm_stream_event(rtd, - codec_dai->driver->capture.stream_name, - SND_SOC_DAPM_STREAM_START); + snd_soc_dapm_stream_event(rtd, substream->stream, codec_dai, + SND_SOC_DAPM_STREAM_START); snd_soc_dai_digital_mute(codec_dai, 0); -- cgit v1.2.3-18-g5258 From 3056557f3b2387d4ac99ca8af14956cd2bf003c2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 17:07:42 -0800 Subject: ASoC: dapm: Constify lots of names that are never modified Neater and avoids warnings when used in other places where const strings are desired. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7645b8a545c..77d230cba61 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2018,7 +2018,7 @@ EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); * Returns 0 for success, else error. */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, - void *data, char *long_name, + void *data, const char *long_name, const char *prefix) { struct snd_kcontrol_new template; -- cgit v1.2.3-18-g5258 From 888df395ebc5c88cde45478660197ca46665efe2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 19:37:51 -0800 Subject: ASoC: dapm: Implement and instantiate DAI widgets In order to allow us to do smarter things with DAI links create DAPM widgets which directly represent the DAIs in the DAPM graph. These are automatically created from the DAIs as we probe the card with references held in both directions between the widget and the DAI. The widgets are not made available for direct instantiation by drivers, they are created automatically from the DAIs. Drivers should be updated to create stream routes using DAPM maps rather than by annotating AIF and DAC widgets with streams. In order to ease transition to this model from existing drivers we automatically create DAPM routes between the DAI widgets and the existing stream widgets which are started and stopped by the DAI widgets, though the old stream handling mechanism is still in place. This also has the nice effect of removing non-DAPM devices as any device with a DAI acquires a widget automatically which will allow future simplifications to the core DAPM logic. The intention is that in future the AIF and DAI widgets will gain the ability to interact such that we are able to manage activity on individual channels independantly rather than powering up and down the entire AIF as we do currently. Currently we only generate these for CODECs, mostly as I have no systems with non-CODEC DAPM to integrate with. It should be a simple matter of programming to add the additional hookup for these. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 11 +++++ sound/soc/soc-dapm.c | 135 ++++++++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 138 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 77d230cba61..32ca75e2002 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1010,6 +1010,7 @@ static int soc_probe_codec(struct snd_soc_card *card, { int ret = 0; const struct snd_soc_codec_driver *driver = codec->driver; + struct snd_soc_dai *dai; codec->card = card; codec->dapm.card = card; @@ -1024,6 +1025,14 @@ static int soc_probe_codec(struct snd_soc_card *card, snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + /* Create DAPM widgets for each DAI stream */ + list_for_each_entry(dai, &dai_list, list) { + if (dai->dev != codec->dev) + continue; + + snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + } + codec->dapm.idle_bias_off = driver->idle_bias_off; if (driver->probe) { @@ -1500,6 +1509,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + snd_soc_dapm_link_dai_widgets(card); + if (card->controls) snd_soc_add_card_controls(card, card->controls, card->num_controls); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 97915eb711c..a4707d0fdf3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -52,6 +52,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, [snd_soc_dapm_micbias] = 2, + [snd_soc_dapm_dai] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, [snd_soc_dapm_mic] = 4, @@ -86,6 +87,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, + [snd_soc_dapm_dai] = 10, [snd_soc_dapm_regulator_supply] = 11, [snd_soc_dapm_supply] = 11, [snd_soc_dapm_post] = 12, @@ -365,6 +367,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_regulator_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai: case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: @@ -522,17 +525,17 @@ static int dapm_new_mixer(struct snd_soc_dapm_widget *w) * for widgets so cut the prefix off * the front of the widget name. */ - snprintf(path->long_name, name_len, "%s %s", - w->name + prefix_len, + snprintf((char *)path->long_name, name_len, + "%s %s", w->name + prefix_len, w->kcontrol_news[i].name); break; case snd_soc_dapm_mixer_named_ctl: - snprintf(path->long_name, name_len, "%s", - w->kcontrol_news[i].name); + snprintf((char *)path->long_name, name_len, + "%s", w->kcontrol_news[i].name); break; } - path->long_name[name_len - 1] = '\0'; + ((char *)path->long_name)[name_len - 1] = '\0'; path->kcontrol = snd_soc_cnew(&w->kcontrol_news[i], wlist, path->long_name, @@ -566,7 +569,7 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) struct snd_soc_dapm_widget_list *wlist; int shared, wlistentries; size_t wlistsize; - char *name; + const char *name; if (w->num_kcontrols != 1) { dev_err(dapm->dev, @@ -702,6 +705,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai: if (widget->active) { widget->outputs = snd_soc_dapm_suspend_check(widget); return widget->outputs; @@ -773,6 +777,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: + case snd_soc_dapm_dai: if (widget->active) { widget->inputs = snd_soc_dapm_suspend_check(widget); return widget->inputs; @@ -892,6 +897,13 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) return out != 0 && in != 0; } +static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) +{ + DAPM_UPDATE_STAT(w, power_checks); + + return w->active; +} + /* Check to see if an ADC has power */ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { @@ -2049,6 +2061,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_regulator_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); list_add(&path->list_source, &wsource->sinks); @@ -2732,10 +2745,10 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, return NULL; } if (dapm->codec && dapm->codec->name_prefix) - snprintf(w->name, name_len, "%s %s", + snprintf((char *)w->name, name_len, "%s %s", dapm->codec->name_prefix, widget->name); else - snprintf(w->name, name_len, "%s", widget->name); + snprintf((char *)w->name, name_len, "%s", widget->name); switch (w->id) { case snd_soc_dapm_switch: @@ -2771,6 +2784,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_regulator_supply: w->power_check = dapm_supply_check_power; break; + case snd_soc_dapm_dai: + w->power_check = dapm_dai_check_power; + break; default: w->power_check = dapm_always_on_check_power; break; @@ -2822,6 +2838,109 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_new_controls); +int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, + struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + + WARN_ON(dapm->dev != dai->dev); + + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + + if (dai->driver->playback.stream_name) { + template.id = snd_soc_dapm_dai; + template.name = dai->driver->playback.stream_name; + template.sname = dai->driver->playback.stream_name; + + dev_dbg(dai->dev, "adding %s widget\n", + template.name); + + w = snd_soc_dapm_new_control(dapm, &template); + if (!w) { + dev_err(dapm->dev, "Failed to create %s widget\n", + dai->driver->playback.stream_name); + } + + w->priv = dai; + dai->playback_widget = w; + } + + if (dai->driver->capture.stream_name) { + template.id = snd_soc_dapm_dai; + template.name = dai->driver->capture.stream_name; + template.sname = dai->driver->capture.stream_name; + + dev_dbg(dai->dev, "adding %s widget\n", + template.name); + + w = snd_soc_dapm_new_control(dapm, &template); + if (!w) { + dev_err(dapm->dev, "Failed to create %s widget\n", + dai->driver->capture.stream_name); + } + + w->priv = dai; + dai->capture_widget = w; + } + + return 0; +} + +int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) +{ + struct snd_soc_dapm_widget *dai_w, *w; + struct snd_soc_dai *dai; + struct snd_soc_dapm_route r; + + memset(&r, 0, sizeof(r)); + + /* For each DAI widget... */ + list_for_each_entry(dai_w, &card->widgets, list) { + if (dai_w->id != snd_soc_dapm_dai) + continue; + + dai = dai_w->priv; + + /* ...find all widgets with the same stream and link them */ + list_for_each_entry(w, &card->widgets, list) { + if (w->dapm != dai_w->dapm) + continue; + + if (w->id == snd_soc_dapm_dai) + continue; + + if (!w->sname) + continue; + + if (dai->driver->playback.stream_name && + strstr(w->sname, + dai->driver->playback.stream_name)) { + r.source = dai->playback_widget->name; + r.sink = w->name; + dev_dbg(dai->dev, "%s -> %s\n", + r.source, r.sink); + + snd_soc_dapm_add_route(w->dapm, &r); + } + + if (dai->driver->capture.stream_name && + strstr(w->sname, + dai->driver->capture.stream_name)) { + r.source = w->name; + r.sink = dai->capture_widget->name; + dev_dbg(dai->dev, "%s -> %s\n", + r.source, r.sink); + + snd_soc_dapm_add_route(w->dapm, &r); + } + } + } + + return 0; +} + static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, int stream, struct snd_soc_dai *dai, int event) -- cgit v1.2.3-18-g5258 From fe360685f9cf41a897c50fea50b4b95f3f622d7c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 19:43:20 -0800 Subject: ASoC: dapm: Convert stream events to use DAI widgets This means we don't need to walk through every single widget in the system for each stream event which is a bit less silly. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 43 +++++++++++++++++-------------------------- 1 file changed, 17 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a4707d0fdf3..86569044f66 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2946,38 +2946,29 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, int event) { struct snd_soc_dapm_widget *w; - const char *stream_name; if (stream == SNDRV_PCM_STREAM_PLAYBACK) - stream_name = dai->driver->playback.stream_name; + w = dai->playback_widget; else - stream_name = dai->driver->capture.stream_name; + w = dai->capture_widget; - if (!stream_name) + if (!w) return; - list_for_each_entry(w, &dapm->card->widgets, list) - { - if (!w->sname || w->dapm != dapm) - continue; - dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", - w->name, w->sname, stream_name, event); - if (strstr(w->sname, stream_name)) { - dapm_mark_dirty(w, "stream event"); - switch(event) { - case SND_SOC_DAPM_STREAM_START: - w->active = 1; - break; - case SND_SOC_DAPM_STREAM_STOP: - w->active = 0; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - case SND_SOC_DAPM_STREAM_RESUME: - case SND_SOC_DAPM_STREAM_PAUSE_PUSH: - case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: - break; - } - } + dapm_mark_dirty(w, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; } dapm_power_widgets(dapm, event); -- cgit v1.2.3-18-g5258 From 5567d8c621f47d83bf73ff1d35409139a4712859 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 21:43:29 -0800 Subject: ASoC: wm8994: Convert to use DAI widget routing rather than streams Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 74794818e1f..18c99cda7cc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1473,17 +1473,17 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, WM8994_POWER_MANAGEMENT_5, 12, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIF1DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIF2DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", NULL, 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", NULL, 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), SND_SOC_DAPM_MUX("AIF2DAC Mux", SND_SOC_NOPM, 0, 0, &aif2dac_mux), SND_SOC_DAPM_MUX("AIF2ADC Mux", SND_SOC_NOPM, 0, 0, &aif2adc_mux), -SND_SOC_DAPM_AIF_IN("AIF3DACDAT", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_AIF_OUT("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIF3DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF3ADCDAT", NULL, 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("TOCLK", WM8994_CLOCKING_1, 4, 0, NULL, 0), @@ -1598,6 +1598,14 @@ static const struct snd_soc_dapm_route intercon[] = { { "TOCLK", NULL, "CLK_SYS" }, + { "AIF1DACDAT", NULL, "AIF1 Playback" }, + { "AIF2DACDAT", NULL, "AIF2 Playback" }, + { "AIF3DACDAT", NULL, "AIF3 Playback" }, + + { "AIF1 Capture", NULL, "AIF1ADCDAT" }, + { "AIF2 Capture", NULL, "AIF2ADCDAT" }, + { "AIF3 Capture", NULL, "AIF3ADCDAT" }, + /* AIF1 outputs */ { "AIF1ADC1L", NULL, "AIF1ADC1L Mixer" }, { "AIF1ADC1L Mixer", "ADC/DMIC Switch", "ADCL Mux" }, -- cgit v1.2.3-18-g5258 From 1a8b2d9d5b6a9909ef0633990fb40c765a791692 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 11:50:07 -0800 Subject: ASoC: dapm: Only mark pin widgets as dirty if we actually change state Small optimisation for noop state updates. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 86569044f66..c9b088dab1c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1927,10 +1927,12 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, return -EINVAL; } + if (w->connected != status) + dapm_mark_dirty(w, "pin configuration"); + w->connected = status; if (status == 0) w->force = 0; - dapm_mark_dirty(w, "pin configuration"); return 0; } -- cgit v1.2.3-18-g5258 From 9d50a764b5cb3d06a01a076131700a34351d98a6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 22:43:39 -0800 Subject: ASoC: wm8962: Convert to module_i2c_driver() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 28d2e74ed01..2dd710f58b8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3768,17 +3768,7 @@ static struct i2c_driver wm8962_i2c_driver = { .id_table = wm8962_i2c_id, }; -static int __init wm8962_modinit(void) -{ - return i2c_add_driver(&wm8962_i2c_driver); -} -module_init(wm8962_modinit); - -static void __exit wm8962_exit(void) -{ - i2c_del_driver(&wm8962_i2c_driver); -} -module_exit(wm8962_exit); +module_i2c_driver(wm8962_i2c_driver); MODULE_DESCRIPTION("ASoC WM8962 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.3-18-g5258 From 7813561a39cbb4dfe5871ce13e1b54b5061c4625 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 22:43:52 -0800 Subject: ASoC: wm8993: Convert to module_i2c_driver() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 19 +------------------ 1 file changed, 1 insertion(+), 18 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 1e69f63ede2..d256a934064 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1859,24 +1859,7 @@ static struct i2c_driver wm8993_i2c_driver = { .id_table = wm8993_i2c_id, }; -static int __init wm8993_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8993_i2c_driver); - if (ret != 0) { - pr_err("WM8993: Unable to register I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8993_modinit); - -static void __exit wm8993_exit(void) -{ - i2c_del_driver(&wm8993_i2c_driver); -} -module_exit(wm8993_exit); - +module_i2c_driver(wm8993_i2c_driver); MODULE_DESCRIPTION("ASoC WM8993 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.3-18-g5258 From 8005f394ab3ffe2051c68dc434dd230f415cdd96 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 22:44:04 -0800 Subject: ASoC: wm8996: Convert to module_i2c_driver() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 20 +------------------- 1 file changed, 1 insertion(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c3bde4a6b7f..a60d2ec249a 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3264,25 +3264,7 @@ static struct i2c_driver wm8996_i2c_driver = { .id_table = wm8996_i2c_id, }; -static int __init wm8996_modinit(void) -{ - int ret; - - ret = i2c_add_driver(&wm8996_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8996 I2C driver: %d\n", - ret); - } - - return ret; -} -module_init(wm8996_modinit); - -static void __exit wm8996_exit(void) -{ - i2c_del_driver(&wm8996_i2c_driver); -} -module_exit(wm8996_exit); +module_i2c_driver(wm8996_i2c_driver); MODULE_DESCRIPTION("ASoC WM8996 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v1.2.3-18-g5258 From f02aab5d7fd53da95a78bb27bfbacc972ed75c10 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 16:33:56 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=w810 with auto-parser The Medion W810 with ALC880 has a typical BIOS bug, copying the pin-defaults without disabling the unused pins. At least, the pin 0x17 must be disabled. Also, it requires GPIO-2 setup. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 84 +------------------------------------------ sound/pci/hda/patch_realtek.c | 13 +++++++ 2 files changed, 14 insertions(+), 83 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 3b88bc561e1..41aecda30a8 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -10,7 +10,6 @@ enum { ALC880_3ST_DIG, ALC880_5ST, ALC880_5ST_DIG, - ALC880_W810, ALC880_Z71V, ALC880_6ST, ALC880_6ST_DIG, @@ -226,56 +225,11 @@ static const struct snd_kcontrol_new alc880_six_stack_mixer[] = { }; -/* - * ALC880 W810 model - * - * W810 has rear IO for: - * Front (DAC 02) - * Surround (DAC 03) - * Center/LFE (DAC 04) - * Digital out (06) - * - * The system also has a pair of internal speakers, and a headphone jack. - * These are both connected to Line2 on the codec, hence to DAC 02. - * - * There is a variable resistor to control the speaker or headphone - * volume. This is a hardware-only device without a software API. - * - * Plugging headphones in will disable the internal speakers. This is - * implemented in hardware, not via the driver using jack sense. In - * a similar fashion, plugging into the rear socket marked "front" will - * disable both the speakers and headphones. - * - * For input, there's a microphone jack, and an "audio in" jack. - * These may not do anything useful with this driver yet, because I - * haven't setup any initialization verbs for these yet... - */ - static const hda_nid_t alc880_w810_dac_nids[3] = { /* front, rear/surround, clfe */ 0x02, 0x03, 0x04 }; -/* fixed 6 channels */ -static const struct hda_channel_mode alc880_w810_modes[1] = { - { 6, NULL } -}; - -/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */ -static const struct snd_kcontrol_new alc880_w810_base_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - { } /* end */ -}; - - /* * Z710V model * @@ -593,27 +547,6 @@ static const struct hda_verb alc880_pin_5stack_init_verbs[] = { { } }; -/* - * W810 pin configuration: - * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b - */ -static const struct hda_verb alc880_pin_w810_init_verbs[] = { - /* hphone/speaker input selector: front DAC */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - { } -}; - /* * Z71V pin configuration: * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?) @@ -1204,7 +1137,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_CLEVO] = "clevo", [ALC880_5ST] = "5stack", [ALC880_5ST_DIG] = "5stack-digout", - [ALC880_W810] = "w810", [ALC880_Z71V] = "z71v", [ALC880_6ST] = "6stack", [ALC880_6ST_DIG] = "6stack-digout", @@ -1223,7 +1155,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { }; static const struct snd_pci_quirk alc880_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810), SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST), SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG), @@ -1265,7 +1196,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734), SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53), - SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), @@ -1377,18 +1307,6 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_sixstack_modes, .input_mux = &alc880_6stack_capture_source, }, - [ALC880_W810] = { - .mixers = { alc880_w810_base_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_w810_init_verbs, - alc880_gpio2_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_w810_dac_nids), - .dac_nids = alc880_w810_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_w810_modes), - .channel_mode = alc880_w810_modes, - .input_mux = &alc880_capture_source, - }, [ALC880_Z71V] = { .mixers = { alc880_z71v_mixer }, .init_verbs = { alc880_volume_init_verbs, @@ -1499,7 +1417,7 @@ static const struct alc_config_preset alc880_presets[] = { alc880_uniwill_p53_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), .dac_nids = alc880_asus_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_w810_modes), + .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, .input_mux = &alc880_capture_source, .unsol_event = alc880_uniwill_p53_unsol_event, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a39146528c2..1cad6748e33 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4405,6 +4405,7 @@ enum { ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, + ALC880_FIXUP_W810, }; static const struct alc_fixup alc880_fixups[] = { @@ -4432,9 +4433,21 @@ static const struct alc_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_W810] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + /* disable bogus unused pins */ + { 0x17, 0x411111f0 }, + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_GPIO2, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { + SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), -- cgit v1.2.3-18-g5258 From dc31b58dbc63a37685f153568b21ed65e3e22f0e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 17:27:53 +0100 Subject: ALSA: hda/realtek - Refactor the DAC filler function Refactor the DAC filling function to be used for both the primary line outputs and extra outputs using the individual badness tables. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 154 +++++++++++++++++++++--------------------- 1 file changed, 77 insertions(+), 77 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1cad6748e33..a0df05d0386 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2969,6 +2969,8 @@ static bool alc_auto_is_dac_reachable(struct hda_codec *codec, hda_nid_t srcs[5]; int i, num; + if (!pin || !dac) + return false; pin = alc_go_down_to_selector(codec, pin); num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs)); for (i = 0; i < num; i++) { @@ -3087,60 +3089,88 @@ static int eval_shared_vol_badness(struct hda_codec *codec, hda_nid_t pin, return badness; } -/* try to assign DACs to extra pins and return the resultant badness */ -static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs, - const hda_nid_t *pins, hda_nid_t *dacs) +struct badness_table { + int no_primary_dac; /* no primary DAC */ + int no_dac; /* no secondary DACs */ + int shared_primary; /* primary DAC is shared with main output */ + int shared_surr; /* secondary DAC shared with main or primary */ + int shared_clfe; /* third DAC shared with main or primary */ + int shared_surr_main; /* secondary DAC sahred with main/DAC0 */ +}; + +static struct badness_table main_out_badness = { + .no_primary_dac = BAD_NO_PRIMARY_DAC, + .no_dac = BAD_NO_DAC, + .shared_primary = BAD_NO_PRIMARY_DAC, + .shared_surr = BAD_SHARED_SURROUND, + .shared_clfe = BAD_SHARED_CLFE, + .shared_surr_main = BAD_SHARED_SURROUND, +}; + +static struct badness_table extra_out_badness = { + .no_primary_dac = BAD_NO_DAC, + .no_dac = BAD_NO_DAC, + .shared_primary = BAD_NO_EXTRA_DAC, + .shared_surr = BAD_SHARED_EXTRA_SURROUND, + .shared_clfe = BAD_SHARED_EXTRA_SURROUND, + .shared_surr_main = BAD_NO_EXTRA_SURR_DAC, +}; + +/* try to assign DACs to pins and return the resultant badness */ +static int alc_auto_fill_dacs(struct hda_codec *codec, int num_outs, + const hda_nid_t *pins, hda_nid_t *dacs, + const struct badness_table *bad) { struct alc_spec *spec = codec->spec; - int i; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i, j; int badness = 0; hda_nid_t dac; if (!num_outs) return 0; - if (!dacs[0]) - dacs[0] = alc_auto_look_for_dac(codec, pins[0]); - if (!dacs[0]) { - for (i = 1; i < num_outs; i++) { - dac = dacs[i]; - if (dac && alc_auto_is_dac_reachable(codec, pins[0], dac)) { - dacs[0] = dac; - dacs[i] = 0; - break; + for (i = 0; i < num_outs; i++) { + hda_nid_t pin = pins[i]; + if (!dacs[i]) + dacs[i] = alc_auto_look_for_dac(codec, pin); + if (!dacs[i] && !i) { + for (j = 1; j < num_outs; j++) { + if (alc_auto_is_dac_reachable(codec, pin, dacs[j])) { + dacs[0] = dacs[j]; + dacs[j] = 0; + break; + } } } - } - dac = dacs[0]; - if (!dac) { - dac = spec->private_dac_nids[0]; - if (!alc_auto_is_dac_reachable(codec, pins[0], dac)) - return BAD_NO_DAC; - badness += BAD_NO_EXTRA_DAC; - } - if (dac) - badness += eval_shared_vol_badness(codec, pins[0], dac); - - for (i = 1; i < num_outs; i++) - dacs[i] = get_dac_if_single(codec, pins[i]); - for (i = 1; i < num_outs; i++) { dac = dacs[i]; - if (!dac) - dac = dacs[i] = alc_auto_look_for_dac(codec, pins[i]); if (!dac) { - if (alc_auto_is_dac_reachable(codec, pins[i], dacs[0])) { + if (alc_auto_is_dac_reachable(codec, pin, dacs[0])) dac = dacs[0]; - badness += BAD_SHARED_EXTRA_SURROUND; - } else if (alc_auto_is_dac_reachable(codec, pins[i], + else if (cfg->line_outs > i && + alc_auto_is_dac_reachable(codec, pin, + spec->private_dac_nids[i])) + dac = spec->private_dac_nids[i]; + if (dac) { + if (!i) + badness += bad->shared_primary; + else if (i == 1) + badness += bad->shared_surr; + else + badness += bad->shared_clfe; + } else if (alc_auto_is_dac_reachable(codec, pin, spec->private_dac_nids[0])) { dac = spec->private_dac_nids[0]; - badness += BAD_NO_EXTRA_SURR_DAC; - } else - badness += BAD_NO_DAC; + badness += bad->shared_surr_main; + } else if (!i) + badness += bad->no_primary_dac; + else + badness += bad->no_dac; } if (dac) - badness += eval_shared_vol_badness(codec, pins[i], dac); + badness += eval_shared_vol_badness(codec, pin, dac); } + return badness; } @@ -3170,7 +3200,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec, { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - int i, j, err, badness; + int i, err, badness; /* set num_dacs once to full for alc_auto_look_for_dac() */ spec->multiout.num_dacs = cfg->line_outs; @@ -3204,40 +3234,9 @@ static int fill_and_eval_dacs(struct hda_codec *codec, } while (mapped); } - for (i = 0; i < cfg->line_outs; i++) { - hda_nid_t pin = cfg->line_out_pins[i]; - hda_nid_t dac; - if (!spec->private_dac_nids[i]) - spec->private_dac_nids[i] = - alc_auto_look_for_dac(codec, pin); - dac = spec->private_dac_nids[i]; - if (!dac && !i) { - for (j = 1; j < cfg->line_outs; j++) { - hda_nid_t dac2 = spec->private_dac_nids[j]; - if (dac2 && - alc_auto_is_dac_reachable(codec, pin, dac2)) { - dac = spec->private_dac_nids[0] = dac2; - spec->private_dac_nids[j] = 0; - break; - } - } - } - if (!dac) { - if (!i) - badness += BAD_NO_PRIMARY_DAC; - else if (alc_auto_is_dac_reachable(codec, pin, - spec->private_dac_nids[0])) { - if (i == 1) - badness += BAD_SHARED_SURROUND; - else - badness += BAD_SHARED_CLFE; - dac = spec->private_dac_nids[0]; - } else - badness += BAD_NO_DAC; - } - if (dac) - badness += eval_shared_vol_badness(codec, pin, dac); - } + badness += alc_auto_fill_dacs(codec, cfg->line_outs, cfg->line_out_pins, + spec->private_dac_nids, + &main_out_badness); /* re-count num_dacs and squash invalid entries */ spec->multiout.num_dacs = 0; @@ -3262,17 +3261,18 @@ static int fill_and_eval_dacs(struct hda_codec *codec, } if (cfg->line_out_type != AUTO_PIN_HP_OUT) { - err = alc_auto_fill_extra_dacs(codec, cfg->hp_outs, - cfg->hp_pins, - spec->multiout.hp_out_nid); + err = alc_auto_fill_dacs(codec, cfg->hp_outs, cfg->hp_pins, + spec->multiout.hp_out_nid, + &extra_out_badness); if (err < 0) return err; badness += err; } if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { - err = alc_auto_fill_extra_dacs(codec, cfg->speaker_outs, - cfg->speaker_pins, - spec->multiout.extra_out_nid); + err = alc_auto_fill_dacs(codec, cfg->speaker_outs, + cfg->speaker_pins, + spec->multiout.extra_out_nid, + &extra_out_badness); if (err < 0) return err; badness += err; -- cgit v1.2.3-18-g5258 From 27e917f82bfcf8c51a2c025ddfb69e0b5947f50b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 17:49:54 +0100 Subject: ALSA: hda/realtek - Drop ALC880 model=clevo Clevo machines with ALC880 are all well with proper BIOS setup. It seems still requiring the additional COEF setup for the EAPD. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 45 ------------------------------------------- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 2 files changed, 11 insertions(+), 45 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 41aecda30a8..b64d2464a78 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -22,7 +22,6 @@ enum { ALC880_UNIWILL_DIG, ALC880_UNIWILL, ALC880_UNIWILL_P53, - ALC880_CLEVO, ALC880_TCL_S700, #ifdef CONFIG_SND_DEBUG ALC880_TEST, @@ -809,35 +808,6 @@ static const struct hda_verb alc880_pin_asus_init_verbs[] = { #define alc880_gpio2_init_verbs alc_gpio2_init_verbs #define alc880_gpio3_init_verbs alc_gpio3_init_verbs -/* Clevo m520g init */ -static const struct hda_verb alc880_pin_clevo_init_verbs[] = { - /* headphone output */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* line-out */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Line-in */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* CD */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Mic1 (rear panel) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Mic2 (front panel) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* headphone */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* change to EAPD mode */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, - - { } -}; - static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { /* change to EAPD mode */ {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -1134,7 +1104,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_3ST] = "3stack", [ALC880_TCL_S700] = "tcl", [ALC880_3ST_DIG] = "3stack-digout", - [ALC880_CLEVO] = "clevo", [ALC880_5ST] = "5stack", [ALC880_5ST_DIG] = "5stack-digout", [ALC880_Z71V] = "z71v", @@ -1188,8 +1157,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO), - SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO), SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), @@ -1439,18 +1406,6 @@ static const struct alc_config_preset alc880_presets[] = { .setup = alc880_uniwill_p53_setup, .init_hook = alc_hp_automute, }, - [ALC880_CLEVO] = { - .mixers = { alc880_three_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_clevo_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a0df05d0386..4f8c3620799 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4406,6 +4406,7 @@ enum { ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, ALC880_FIXUP_W810, + ALC880_FIXUP_EAPD_COEF, }; static const struct alc_fixup alc880_fixups[] = { @@ -4443,10 +4444,20 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_GPIO2, }, + [ALC880_FIXUP_EAPD_COEF] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* change to EAPD mode */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3060 }, + {} + }, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), -- cgit v1.2.3-18-g5258 From b9368f5c10b15f2b79a58666849827edc1f2f3d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Feb 2012 17:54:44 +0100 Subject: ALSA: hda/realtek - Replace ALC880 model=tcl with auto-parser It needs a few extra setups for EAPD, but others look fairly straightforward. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 56 ------------------------------------------- sound/pci/hda/patch_realtek.c | 13 ++++++++++ 2 files changed, 13 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index b64d2464a78..56f8fa1e346 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -22,7 +22,6 @@ enum { ALC880_UNIWILL_DIG, ALC880_UNIWILL, ALC880_UNIWILL_P53, - ALC880_TCL_S700, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -344,20 +343,6 @@ static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { { } /* end */ }; -/* TCL S700 */ -static const struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { } /* end */ -}; - /* Uniwill */ static const struct snd_kcontrol_new alc880_uniwill_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -808,31 +793,6 @@ static const struct hda_verb alc880_pin_asus_init_verbs[] = { #define alc880_gpio2_init_verbs alc_gpio2_init_verbs #define alc880_gpio3_init_verbs alc_gpio3_init_verbs -static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { - /* change to EAPD mode */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3060}, - - /* Headphone output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Front output*/ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Line In pin widget for input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - - /* change to EAPD mode */ - {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, - {0x20, AC_VERB_SET_PROC_COEF, 0x3070}, - - { } -}; - /* * Test configuration for debugging * @@ -1102,7 +1062,6 @@ static const struct hda_verb alc880_test_init_verbs[] = { static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_3ST] = "3stack", - [ALC880_TCL_S700] = "tcl", [ALC880_3ST_DIG] = "3stack-digout", [ALC880_5ST] = "5stack", [ALC880_5ST_DIG] = "5stack-digout", @@ -1169,7 +1128,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), - SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), @@ -1216,20 +1174,6 @@ static const struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, - [ALC880_TCL_S700] = { - .mixers = { alc880_tcl_s700_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_tcl_S700_init_verbs, - alc880_gpio2_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */ - .num_adc_nids = 1, /* single ADC */ - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_capture_source, - }, [ALC880_5ST] = { .mixers = { alc880_three_stack_mixer, alc880_five_stack_mixer}, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f8c3620799..e6eec9a9ab4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4407,6 +4407,7 @@ enum { ALC880_FIXUP_LG, ALC880_FIXUP_W810, ALC880_FIXUP_EAPD_COEF, + ALC880_FIXUP_TCL_S700, }; static const struct alc_fixup alc880_fixups[] = { @@ -4453,6 +4454,17 @@ static const struct alc_fixup alc880_fixups[] = { {} }, }, + [ALC880_FIXUP_TCL_S700] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* change to EAPD mode */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x3070 }, + {} + }, + .chained = true, + .chain_id = ALC880_FIXUP_GPIO2, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { @@ -4463,6 +4475,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700), {} }; -- cgit v1.2.3-18-g5258 From d5a7f23f9c8be29833ef4d805976b6906c25c658 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 Feb 2012 13:12:21 -0800 Subject: ASoC: wm8996: Make sure we bounce /RESET to reset While it matches the current code only bringing the device out of reset isn't actually doing what the function says so make sure we set the GPIO high before we pull it low. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index a60d2ec249a..aba144f6994 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1709,6 +1709,7 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) static int wm8996_reset(struct wm8996_priv *wm8996) { if (wm8996->pdata.ldo_ena > 0) { + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); return 0; } else { -- cgit v1.2.3-18-g5258 From 0837fc624391378ea25c9f811c90d49b8808ba1a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 17 Feb 2012 19:40:38 -0200 Subject: ASoC: soc-core: Show the returned values on error messages Showing the returned values on error messages is useful information. While at it, use pr_err/pr_warn whenever possible. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 39 ++++++++++++++++++++++----------------- 1 file changed, 22 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 32ca75e2002..3ca70594e24 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -933,7 +933,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (codec_dai->driver->remove) { err = codec_dai->driver->remove(codec_dai); if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", codec_dai->name); + pr_err("asoc: failed to remove %s: %d\n", + codec_dai->name, err); } codec_dai->probed = 0; list_del(&codec_dai->card_list); @@ -945,7 +946,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (platform->driver->remove) { err = platform->driver->remove(platform); if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", platform->name); + pr_err("asoc: failed to remove %s: %d\n", + platform->name, err); } /* Make sure all DAPM widgets are freed */ @@ -968,7 +970,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (cpu_dai->driver->remove) { err = cpu_dai->driver->remove(cpu_dai); if (err < 0) - printk(KERN_ERR "asoc: failed to remove %s\n", cpu_dai->name); + pr_err("asoc: failed to remove %s: %d\n", + cpu_dai->name, err); } cpu_dai->probed = 0; list_del(&cpu_dai->card_list); @@ -1224,8 +1227,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe CPU DAI %s\n", - cpu_dai->name); + pr_err("asoc: failed to probe CPU DAI %s: %d\n", + cpu_dai->name, ret); module_put(cpu_dai->dev->driver->owner); return ret; } @@ -1256,8 +1259,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (codec_dai->driver->probe) { ret = codec_dai->driver->probe(codec_dai); if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe CODEC DAI %s\n", - codec_dai->name); + pr_err("asoc: failed to probe CODEC DAI %s: %d\n", + codec_dai->name, ret); return ret; } } @@ -1277,12 +1280,13 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); if (ret < 0) - printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); + pr_warn("asoc: failed to add pmdown_time sysfs:%d\n", ret); /* create the pcm */ ret = soc_new_pcm(rtd, num); if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm %s\n", dai_link->stream_name); + pr_err("asoc: can't create pcm %s :%d\n", + dai_link->stream_name, ret); return ret; } @@ -1315,7 +1319,7 @@ static int soc_register_ac97_dai_link(struct snd_soc_pcm_runtime *rtd) ret = soc_ac97_dev_register(rtd->codec); if (ret < 0) { - printk(KERN_ERR "asoc: AC97 device register failed\n"); + pr_err("asoc: AC97 device register failed:%d\n", ret); return ret; } @@ -1455,8 +1459,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, card->owner, 0, &card->snd_card); if (ret < 0) { - printk(KERN_ERR "asoc: can't create sound card for card %s\n", - card->name); + pr_err("asoc: can't create sound card for card %s: %d\n", + card->name, ret); mutex_unlock(&card->mutex); return; } @@ -1576,7 +1580,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_register(card->snd_card); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); + pr_err("asoc: failed to register soundcard for %s: %d\n", + card->name, ret); goto probe_aux_dev_err; } @@ -1585,7 +1590,8 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) for (i = 0; i < card->num_rtd; i++) { ret = soc_register_ac97_dai_link(&card->rtd[i]); if (ret < 0) { - printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + pr_err("asoc: failed to register AC97 %s: %d\n", + card->name, ret); while (--i >= 0) soc_unregister_ac97_dai_link(card->rtd[i].codec); goto probe_aux_dev_err; @@ -3083,7 +3089,7 @@ static inline char *fmt_multiple_name(struct device *dev, struct snd_soc_dai_driver *dai_drv) { if (dai_drv->name == NULL) { - printk(KERN_ERR "asoc: error - multiple DAI %s registered with no name\n", + pr_err("asoc: error - multiple DAI %s registered with no name\n", dev_name(dev)); return NULL; } @@ -3555,8 +3561,7 @@ static int __init snd_soc_init(void) #ifdef CONFIG_DEBUG_FS snd_soc_debugfs_root = debugfs_create_dir("asoc", NULL); if (IS_ERR(snd_soc_debugfs_root) || !snd_soc_debugfs_root) { - printk(KERN_WARNING - "ASoC: Failed to create debugfs directory\n"); + pr_warn("ASoC: Failed to create debugfs directory\n"); snd_soc_debugfs_root = NULL; } -- cgit v1.2.3-18-g5258 From a387419612f9c246701a5080bccecf3c04f65277 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 20 Feb 2012 08:04:50 +0800 Subject: ASoC: Add __devinit annotation for pxa2xx_ac97_probe This fixes below build warning: WARNING: vmlinux.o(.text+0x1e632c): Section mismatch in reference from the function pxa2xx_ac97_probe() to the function .devinit.text:pxa2xx_ac97_hw_probe() The function pxa2xx_ac97_probe() references the function __devinit pxa2xx_ac97_hw_probe(). This is often because pxa2xx_ac97_probe lacks a __devinit annotation or the annotation of pxa2xx_ac97_hw_probe is wrong. Also rename pxa_ac97_dai to pxa_ac97_dai_driver to fix below build warning: LD sound/soc/pxa/built-in.o WARNING: sound/soc/pxa/built-in.o(.data+0x18c): Section mismatch in reference from the variable pxa_ac97_dai to the function .devinit.text:pxa2xx_ac97_probe() The variable pxa_ac97_dai references the function __devinit pxa2xx_ac97_probe() If the reference is valid then annotate the variable with __init* or __refdata (see linux/init.h) or name the variable: *driver, *_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-ac97.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 837ff341fd6..4800d5fe568 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -103,7 +103,7 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) #define pxa2xx_ac97_resume NULL #endif -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) +static int __devinit pxa2xx_ac97_probe(struct snd_soc_dai *dai) { return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); } @@ -179,7 +179,7 @@ static const struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = { * There is only 1 physical AC97 interface for pxa2xx, but it * has extra fifo's that can be used for aux DACs and ADCs. */ -static struct snd_soc_dai_driver pxa_ac97_dai[] = { +static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, @@ -244,13 +244,13 @@ static __devinit int pxa2xx_ac97_dev_probe(struct platform_device *pdev) * driver to do interesting things with the clocking to get us up * and running. */ - return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai, - ARRAY_SIZE(pxa_ac97_dai)); + return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver, + ARRAY_SIZE(pxa_ac97_dai_driver)); } static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev) { - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai)); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver)); return 0; } -- cgit v1.2.3-18-g5258 From 7da9ced6066c654a22836c24bae509ef323e10a8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Fri, 3 Feb 2012 00:59:33 -0800 Subject: ASoC: fsi: Add DMAEngine support This patch supports DMAEngine to FSI driver. It supports only Tx case at this point. If platform/cpu doesn't support DMAEngine, FSI driver will use PIO transfer. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 232 +++++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 232 insertions(+) (limited to 'sound') diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 13746809c27..378cc5b056d 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -13,8 +13,11 @@ */ #include +#include #include #include +#include +#include #include #include #include @@ -53,6 +56,7 @@ /* DO_FMT */ /* DI_FMT */ +#define CR_BWS_MASK (0x3 << 20) /* FSI2 */ #define CR_BWS_24 (0x0 << 20) /* FSI2 */ #define CR_BWS_16 (0x1 << 20) /* FSI2 */ #define CR_BWS_20 (0x2 << 20) /* FSI2 */ @@ -68,6 +72,15 @@ #define CR_TDM (0x4 << 4) #define CR_TDM_D (0x5 << 4) +/* OUT_DMAC */ +/* IN_DMAC */ +#define VDMD_MASK (0x3 << 4) +#define VDMD_FRONT (0x0 << 4) /* Package in front */ +#define VDMD_BACK (0x1 << 4) /* Package in back */ +#define VDMD_STREAM (0x2 << 4) /* Stream mode(16bit * 2) */ + +#define DMA_ON (0x1 << 0) + /* DOFF_CTL */ /* DIFF_CTL */ #define IRQ_HALF 0x00100000 @@ -180,6 +193,14 @@ struct fsi_stream { */ struct fsi_stream_handler *handler; struct fsi_priv *priv; + + /* + * these are for DMAEngine + */ + struct dma_chan *chan; + struct sh_dmae_slave slave; /* see fsi_handler_init() */ + struct tasklet_struct tasklet; + dma_addr_t dma; }; struct fsi_priv { @@ -888,6 +909,212 @@ static irqreturn_t fsi_interrupt(int irq, void *data) return IRQ_HANDLED; } +/* + * dma data transfer handler + */ +static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) +{ + struct snd_pcm_runtime *runtime = io->substream->runtime; + struct snd_soc_dai *dai = fsi_get_dai(io->substream); + enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? + DMA_TO_DEVICE : DMA_FROM_DEVICE; + + io->dma = dma_map_single(dai->dev, runtime->dma_area, + snd_pcm_lib_buffer_bytes(io->substream), dir); + return 0; +} + +static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io) +{ + struct snd_soc_dai *dai = fsi_get_dai(io->substream); + enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? + DMA_TO_DEVICE : DMA_FROM_DEVICE; + + dma_unmap_single(dai->dev, io->dma, + snd_pcm_lib_buffer_bytes(io->substream), dir); + return 0; +} + +static void fsi_dma_complete(void *data) +{ + struct fsi_stream *io = (struct fsi_stream *)data; + struct fsi_priv *fsi = fsi_stream_to_priv(io); + struct snd_pcm_runtime *runtime = io->substream->runtime; + struct snd_soc_dai *dai = fsi_get_dai(io->substream); + enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? + DMA_TO_DEVICE : DMA_FROM_DEVICE; + + dma_sync_single_for_cpu(dai->dev, io->dma, + samples_to_bytes(runtime, io->period_samples), dir); + + io->buff_sample_pos += io->period_samples; + io->period_pos++; + + if (io->period_pos >= runtime->periods) { + io->period_pos = 0; + io->buff_sample_pos = 0; + } + + fsi_count_fifo_err(fsi); + fsi_stream_transfer(io); + + snd_pcm_period_elapsed(io->substream); +} + +static dma_addr_t fsi_dma_get_area(struct fsi_stream *io) +{ + struct snd_pcm_runtime *runtime = io->substream->runtime; + + return io->dma + samples_to_bytes(runtime, io->buff_sample_pos); +} + +static void fsi_dma_do_tasklet(unsigned long data) +{ + struct fsi_stream *io = (struct fsi_stream *)data; + struct fsi_priv *fsi = fsi_stream_to_priv(io); + struct dma_chan *chan; + struct snd_soc_dai *dai; + struct dma_async_tx_descriptor *desc; + struct scatterlist sg; + struct snd_pcm_runtime *runtime; + enum dma_data_direction dir; + dma_cookie_t cookie; + int is_play = fsi_stream_is_play(fsi, io); + int len; + dma_addr_t buf; + + if (!fsi_stream_is_working(fsi, io)) + return; + + dai = fsi_get_dai(io->substream); + chan = io->chan; + runtime = io->substream->runtime; + dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; + len = samples_to_bytes(runtime, io->period_samples); + buf = fsi_dma_get_area(io); + + dma_sync_single_for_device(dai->dev, io->dma, len, dir); + + sg_init_table(&sg, 1); + sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf)), + len , offset_in_page(buf)); + sg_dma_address(&sg) = buf; + sg_dma_len(&sg) = len; + + desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dir, + DMA_PREP_INTERRUPT | + DMA_CTRL_ACK); + if (!desc) { + dev_err(dai->dev, "device_prep_slave_sg() fail\n"); + return; + } + + desc->callback = fsi_dma_complete; + desc->callback_param = io; + + cookie = desc->tx_submit(desc); + if (cookie < 0) { + dev_err(dai->dev, "tx_submit() fail\n"); + return; + } + + dma_async_issue_pending(chan); + + /* + * FIXME + * + * In DMAEngine case, codec and FSI cannot be started simultaneously + * since FSI is using tasklet. + * Therefore, in capture case, probably FSI FIFO will have got + * overflow error in this point. + * in that case, DMA cannot start transfer until error was cleared. + */ + if (!is_play) { + if (ERR_OVER & fsi_reg_read(fsi, DIFF_ST)) { + fsi_reg_mask_set(fsi, DIFF_CTL, FIFO_CLR, FIFO_CLR); + fsi_reg_write(fsi, DIFF_ST, 0); + } + } +} + +static bool fsi_dma_filter(struct dma_chan *chan, void *param) +{ + struct sh_dmae_slave *slave = param; + + chan->private = slave; + + return true; +} + +static int fsi_dma_transfer(struct fsi_priv *fsi, struct fsi_stream *io) +{ + tasklet_schedule(&io->tasklet); + + return 0; +} + +static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, + int start) +{ + u32 bws; + u32 dma; + + switch (io->sample_width * start) { + case 2: + bws = CR_BWS_16; + dma = VDMD_STREAM | DMA_ON; + break; + case 4: + bws = CR_BWS_24; + dma = VDMD_BACK | DMA_ON; + break; + default: + bws = 0; + dma = 0; + } + + fsi_reg_mask_set(fsi, DO_FMT, CR_BWS_MASK, bws); + fsi_reg_write(fsi, OUT_DMAC, dma); +} + +static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) +{ + dma_cap_mask_t mask; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave); + if (!io->chan) + return -EIO; + + tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); + + return 0; +} + +static int fsi_dma_remove(struct fsi_priv *fsi, struct fsi_stream *io) +{ + tasklet_kill(&io->tasklet); + + fsi_stream_stop(fsi, io); + + if (io->chan) + dma_release_channel(io->chan); + + io->chan = NULL; + return 0; +} + +static struct fsi_stream_handler fsi_dma_push_handler = { + .init = fsi_dma_init, + .quit = fsi_dma_quit, + .probe = fsi_dma_probe, + .transfer = fsi_dma_transfer, + .remove = fsi_dma_remove, + .start_stop = fsi_dma_push_start_stop, +}; + /* * dai ops */ @@ -1304,6 +1531,11 @@ static void fsi_handler_init(struct fsi_priv *fsi) fsi->playback.priv = fsi; fsi->capture.handler = &fsi_pio_pop_handler; /* default PIO */ fsi->capture.priv = fsi; + + if (fsi->info->tx_id) { + fsi->playback.slave.slave_id = fsi->info->tx_id; + fsi->playback.handler = &fsi_dma_push_handler; + } } static int fsi_probe(struct platform_device *pdev) -- cgit v1.2.3-18-g5258 From 7913a49963ffa8849c14c805c26d9e63bb27ccaa Mon Sep 17 00:00:00 2001 From: Jeffrin Jose Date: Thu, 16 Feb 2012 21:50:49 +0530 Subject: ALSA: Fixed a trailing white space error This is a patch to the sound/core/misc.c file that fixes up a trailing white space issue found by the checkpatch.pl tool. Signed-off-by: Jeffrin Jose Signed-off-by: Takashi Iwai --- sound/core/misc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/misc.c b/sound/core/misc.c index 465f0ce772c..76816792540 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -72,7 +72,7 @@ void __snd_printk(unsigned int level, const char *path, int line, char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV"; #endif -#ifdef CONFIG_SND_DEBUG +#ifdef CONFIG_SND_DEBUG if (debug < level) return; #endif -- cgit v1.2.3-18-g5258 From 589876e243bb14343d09d9fd7f9ddf79f1d80158 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 15:47:55 +0100 Subject: ALSA: hda/realtek - Apply probe-fixup really after probing Move the call of alc_apply_fixup() with ALC_FIXUP_ACT_PROBE after the whole setups of patch_ops & co, so that the fix-up function may override the default setup. This will be needed for installing the own unsol event handler (e.g. for volume-knob controls). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e6eec9a9ab4..895113ee385 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4566,8 +4566,6 @@ static int patch_alc880(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -4578,6 +4576,8 @@ static int patch_alc880(struct hda_codec *codec) spec->loopback.amplist = alc880_loopbacks; #endif + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: @@ -4749,8 +4749,6 @@ static int patch_alc260(struct hda_codec *codec) set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); } - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -4759,6 +4757,8 @@ static int patch_alc260(struct hda_codec *codec) spec->loopback.amplist = alc260_loopbacks; #endif + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: @@ -5165,8 +5165,6 @@ static int patch_alc882(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - codec->patch_ops = alc_patch_ops; if (board_config == ALC_MODEL_AUTO) spec->init_hook = alc_auto_init_std; @@ -5178,6 +5176,8 @@ static int patch_alc882(struct hda_codec *codec) spec->loopback.amplist = alc882_loopbacks; #endif + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: @@ -5331,8 +5331,6 @@ static int patch_alc262(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -5342,6 +5340,8 @@ static int patch_alc262(struct hda_codec *codec) spec->loopback.amplist = alc262_loopbacks; #endif + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: @@ -5992,8 +5992,6 @@ static int patch_alc269(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); } - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - codec->patch_ops = alc_patch_ops; #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; @@ -6008,6 +6006,8 @@ static int patch_alc269(struct hda_codec *codec) codec->patch_ops.check_power_status = alc269_mic2_mute_check_ps; #endif + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: @@ -6146,8 +6146,6 @@ static int patch_alc861(struct hda_codec *codec) set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); } - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -6156,6 +6154,8 @@ static int patch_alc861(struct hda_codec *codec) spec->loopback.amplist = alc861_loopbacks; #endif + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: @@ -6270,8 +6270,6 @@ static int patch_alc861vd(struct hda_codec *codec) set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); } - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; @@ -6281,6 +6279,8 @@ static int patch_alc861vd(struct hda_codec *codec) spec->loopback.amplist = alc861vd_loopbacks; #endif + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: @@ -6659,8 +6659,6 @@ static int patch_alc662(struct hda_codec *codec) } } - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); - codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; @@ -6670,6 +6668,8 @@ static int patch_alc662(struct hda_codec *codec) spec->loopback.amplist = alc662_loopbacks; #endif + alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); + return 0; error: -- cgit v1.2.3-18-g5258 From cf5a22793cfa54c056655d374722dc5dfd496eca Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 16:31:07 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=futjisu with auto-parser Now adding the support for the volume-knob widget, we can move the static quirk for ALC880 model=fujitsu to the auto-parser completely. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 33 ------------------- sound/pci/hda/patch_realtek.c | 76 +++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 74 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 56f8fa1e346..f062eaae6b1 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -18,7 +18,6 @@ enum { ALC880_ASUS_DIG, ALC880_ASUS_W1V, ALC880_ASUS_DIG2, - ALC880_FUJITSU, ALC880_UNIWILL_DIG, ALC880_UNIWILL, ALC880_UNIWILL_P53, @@ -371,20 +370,6 @@ static const struct snd_kcontrol_new alc880_uniwill_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc880_fujitsu_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - static const struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), @@ -1074,7 +1059,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_ASUS_DIG2] = "asus-dig2", [ALC880_UNIWILL_DIG] = "uniwill", [ALC880_UNIWILL_P53] = "uniwill-p53", - [ALC880_FUJITSU] = "fujitsu", [ALC880_F1734] = "F1734", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", @@ -1125,9 +1109,7 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), - SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU), SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), - SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), @@ -1335,21 +1317,6 @@ static const struct alc_config_preset alc880_presets[] = { .setup = alc880_uniwill_p53_setup, .init_hook = alc_hp_automute, }, - [ALC880_FUJITSU] = { - .mixers = { alc880_fujitsu_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_uniwill_p53_init_verbs, - alc880_beep_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_capture_source, - .unsol_event = alc880_uniwill_p53_unsol_event, - .setup = alc880_uniwill_p53_setup, - .init_hook = alc_hp_automute, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 895113ee385..6a6436a54f0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -651,15 +651,51 @@ static void alc_exec_unsol_event(struct hda_codec *codec, int action) snd_hda_jack_report_sync(codec); } +/* update the master volume per volume-knob's unsol event */ +static void alc_update_knob_master(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int val; + struct snd_kcontrol *kctl; + struct snd_ctl_elem_value *uctl; + + kctl = snd_hda_find_mixer_ctl(codec, "Master Playback Volume"); + if (!kctl) + return; + uctl = kzalloc(sizeof(*uctl), GFP_KERNEL); + if (!uctl) + return; + val = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); + val &= HDA_AMP_VOLMASK; + uctl->value.integer.value[0] = val; + uctl->value.integer.value[1] = val; + kctl->put(kctl, uctl); + kfree(uctl); +} + /* unsolicited event for HP jack sensing */ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) { + int action; + if (codec->vendor_id == 0x10ec0880) res >>= 28; else res >>= 26; - res = snd_hda_jack_get_action(codec, res); - alc_exec_unsol_event(codec, res); + action = snd_hda_jack_get_action(codec, res); + if (res == ALC_DCVOL_EVENT) { + /* Execute the dc-vol event here as it requires the NID + * but we don't pass NID to alc_exec_unsol_event(). + * Once when we convert all static quirks to the auto-parser, + * this can be integerated into there. + */ + struct hda_jack_tbl *jack; + jack = snd_hda_jack_tbl_get_from_tag(codec, res); + if (jack) + alc_update_knob_master(codec, jack->nid); + return; + } + alc_exec_unsol_event(codec, action); } /* call init functions of standard auto-mute helpers */ @@ -4408,8 +4444,18 @@ enum { ALC880_FIXUP_W810, ALC880_FIXUP_EAPD_COEF, ALC880_FIXUP_TCL_S700, + ALC880_FIXUP_VOL_KNOB, + ALC880_FIXUP_FUJITSU, }; +/* enable the volume-knob widget support on NID 0x21 */ +static void alc880_fixup_vol_knob(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + if (action == ALC_FIXUP_ACT_PROBE) + snd_hda_jack_detect_enable(codec, 0x21, ALC_DCVOL_EVENT); +} + static const struct alc_fixup alc880_fixups[] = { [ALC880_FIXUP_GPIO2] = { .type = ALC_FIXUP_VERBS, @@ -4465,6 +4511,30 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_GPIO2, }, + [ALC880_FIXUP_VOL_KNOB] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc880_fixup_vol_knob, + }, + [ALC880_FIXUP_FUJITSU] = { + /* override all pins as BIOS on old Amilo is broken */ + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x0121411f }, /* HP */ + { 0x15, 0x99030120 }, /* speaker */ + { 0x16, 0x99030130 }, /* bass speaker */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x411111f0 }, /* N/A */ + { 0x19, 0x01a19950 }, /* mic-in */ + { 0x1a, 0x411111f0 }, /* N/A */ + { 0x1b, 0x411111f0 }, /* N/A */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0x01454140 }, /* SPDIF out */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_VOL_KNOB, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { @@ -4472,6 +4542,8 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), + SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FIXUP_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10b0, "FSC Amilo Pi1556", ALC880_FIXUP_FUJITSU), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), -- cgit v1.2.3-18-g5258 From cbdfb661ff5aee83daa8c60fb523cc4f679e1e42 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 20 Feb 2012 15:05:24 +0800 Subject: ASoC: imx/mxs: remove redundant SND_PCM selection SND_PCM is already selected by SND_SOC, there is no need for SND_IMX_SOC and SND_MXS_SOC to select it again. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 1 - sound/soc/mxs/Kconfig | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 738391757f2..91b66eff531 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC - select SND_PCM select FIQ select SND_SOC_AC97_BUS help diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index e4ba8d5f25f..21d20f3e026 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" depends on ARCH_MXS - select SND_PCM help Say Y or M if you want to add support for codecs attached to the MXS SAIF interface. -- cgit v1.2.3-18-g5258 From ba5338185dd522696f1c0d0957a724a1fdd1f39d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 16:36:52 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=F1734 with auto-parser Similar as the previous patch for model=fujitsu, we can now move the static quirk for F1734 to the auto-parser. The only difference is the default pin configurations: F1734 has less pins than Amilo's. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 85 ------------------------------------------- sound/pci/hda/patch_realtek.c | 24 ++++++++++++ 2 files changed, 24 insertions(+), 85 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index f062eaae6b1..2ab7c3b9bb9 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -13,7 +13,6 @@ enum { ALC880_Z71V, ALC880_6ST, ALC880_6ST_DIG, - ALC880_F1734, ALC880_ASUS, ALC880_ASUS_DIG, ALC880_ASUS_W1V, @@ -257,40 +256,6 @@ static const struct snd_kcontrol_new alc880_z71v_mixer[] = { { } /* end */ }; - -/* - * ALC880 F1734 model - * - * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d) - * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18 - */ - -static const hda_nid_t alc880_f1734_dac_nids[1] = { - 0x03 -}; -#define ALC880_F1734_HP_DAC 0x02 - -static const struct snd_kcontrol_new alc880_f1734_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct hda_input_mux alc880_f1734_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x1 }, - { "CD", 0x4 }, - }, -}; - - /* * ALC880 ASUS model * @@ -709,38 +674,6 @@ static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, alc_exec_unsol_event(codec, res); } -/* - * F1734 pin configuration: - * HP = 0x14, speaker-out = 0x15, mic = 0x18 - */ -static const struct hda_verb alc880_pin_f1734_init_verbs[] = { - {0x07, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_DCVOL_EVENT}, - - { } -}; - /* * ASUS pin configuration: * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a @@ -1059,7 +992,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_ASUS_DIG2] = "asus-dig2", [ALC880_UNIWILL_DIG] = "uniwill", [ALC880_UNIWILL_P53] = "uniwill-p53", - [ALC880_F1734] = "F1734", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -1103,13 +1035,10 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), - SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734), SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734), - SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), @@ -1212,20 +1141,6 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_2_jack_modes, .input_mux = &alc880_capture_source, }, - [ALC880_F1734] = { - .mixers = { alc880_f1734_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_f1734_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids), - .dac_nids = alc880_f1734_dac_nids, - .hp_nid = 0x02, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_f1734_capture_source, - .unsol_event = alc880_uniwill_p53_unsol_event, - .setup = alc880_uniwill_p53_setup, - .init_hook = alc_hp_automute, - }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, .init_verbs = { alc880_volume_init_verbs, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6a6436a54f0..2d102f70f78 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4446,6 +4446,7 @@ enum { ALC880_FIXUP_TCL_S700, ALC880_FIXUP_VOL_KNOB, ALC880_FIXUP_FUJITSU, + ALC880_FIXUP_F1734, }; /* enable the volume-knob widget support on NID 0x21 */ @@ -4535,14 +4536,37 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_VOL_KNOB, }, + [ALC880_FIXUP_F1734] = { + /* almost compatible with FUJITSU, but no bass and SPDIF */ + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x0121411f }, /* HP */ + { 0x15, 0x99030120 }, /* speaker */ + { 0x16, 0x411111f0 }, /* N/A */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x411111f0 }, /* N/A */ + { 0x19, 0x01a19950 }, /* mic-in */ + { 0x1a, 0x411111f0 }, /* N/A */ + { 0x1b, 0x411111f0 }, /* N/A */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0x411111f0 }, /* N/A */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_VOL_KNOB, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), + SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), + SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_FIXUP_F1734), SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FIXUP_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_FIXUP_F1734), SND_PCI_QUIRK(0x1734, 0x10b0, "FSC Amilo Pi1556", ALC880_FIXUP_FUJITSU), SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), -- cgit v1.2.3-18-g5258 From 7833c7e8b41d4c778e18481d7115dafa4bfaee0a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:11:38 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=uniwill-p53 with auto-parser Uniwill p53 has a sane BIOS setup but just needs the volume-knob handling like Fujitsu laptops with ALC880. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 94 ------------------------------------------- sound/pci/hda/patch_realtek.c | 1 + 2 files changed, 1 insertion(+), 94 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 2ab7c3b9bb9..2a00271e065 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -19,7 +19,6 @@ enum { ALC880_ASUS_DIG2, ALC880_UNIWILL_DIG, ALC880_UNIWILL, - ALC880_UNIWILL_P53, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -335,16 +334,6 @@ static const struct snd_kcontrol_new alc880_uniwill_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - /* * initialize the codec volumes, etc */ @@ -568,39 +557,6 @@ static const struct hda_verb alc880_uniwill_init_verbs[] = { { } }; -/* -* Uniwill P53 -* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19, - */ -static const struct hda_verb alc880_uniwill_p53_init_verbs[] = { - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_DCVOL_EVENT}, - - { } -}; - static const struct hda_verb alc880_beep_init_verbs[] = { { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) }, { } @@ -639,41 +595,6 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec, } } -static void alc880_uniwill_p53_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); - present &= HDA_AMP_VOLMASK; - snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0, - HDA_AMP_VOLMASK, present); - snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0, - HDA_AMP_VOLMASK, present); -} - -static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Looks like the unsol event is incompatible with the standard - * definition. 4bit tag is placed at 28 bit! - */ - res >>= 28; - if (res == ALC_DCVOL_EVENT) - alc880_uniwill_p53_dcvol_automute(codec); - else - alc_exec_unsol_event(codec, res); -} - /* * ASUS pin configuration: * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a @@ -991,7 +912,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_ASUS_DIG] = "asus-dig", [ALC880_ASUS_DIG2] = "asus-dig2", [ALC880_UNIWILL_DIG] = "uniwill", - [ALC880_UNIWILL_P53] = "uniwill-p53", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -1036,7 +956,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), - SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ @@ -1219,19 +1138,6 @@ static const struct alc_config_preset alc880_presets[] = { .setup = alc880_uniwill_setup, .init_hook = alc880_uniwill_init_hook, }, - [ALC880_UNIWILL_P53] = { - .mixers = { alc880_uniwill_p53_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_uniwill_p53_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .input_mux = &alc880_capture_source, - .unsol_event = alc880_uniwill_p53_unsol_event, - .setup = alc880_uniwill_p53_setup, - .init_hook = alc_hp_automute, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2d102f70f78..3c0a46ed9ca 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4562,6 +4562,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), + SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_FIXUP_VOL_KNOB), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_FIXUP_F1734), -- cgit v1.2.3-18-g5258 From 817de92f1b52358f28534bb0b0c373f75e4b4baa Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:20:48 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=uniwill with auto-parser The model=uniwill would work almost as is, but a couple of adjustments are needed to make the mutli-io working correctly. The headphone and speaker pins have to be marked properly in pin configs. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 118 ------------------------------------------ sound/pci/hda/alc_quirks.c | 12 ----- sound/pci/hda/patch_realtek.c | 12 +++++ 3 files changed, 12 insertions(+), 130 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 2a00271e065..c40f2446fcc 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -18,7 +18,6 @@ enum { ALC880_ASUS_W1V, ALC880_ASUS_DIG2, ALC880_UNIWILL_DIG, - ALC880_UNIWILL, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -306,34 +305,6 @@ static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { { } /* end */ }; -/* Uniwill */ -static const struct snd_kcontrol_new alc880_uniwill_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - /* * initialize the codec volumes, etc */ @@ -518,83 +489,11 @@ static const struct hda_verb alc880_pin_6stack_init_verbs[] = { { } }; -/* - * Uniwill pin configuration: - * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19, - * line = 0x1a - */ -static const struct hda_verb alc880_uniwill_init_verbs[] = { - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */ - /* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT}, - - { } -}; - static const struct hda_verb alc880_beep_init_verbs[] = { { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) }, { } }; -static void alc880_uniwill_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x16; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc880_uniwill_init_hook(struct hda_codec *codec) -{ - alc_hp_automute(codec); - alc88x_simple_mic_automute(codec); -} - -static void alc880_uniwill_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - /* Looks like the unsol event is incompatible with the standard - * definition. 4bit tag is placed at 28 bit! - */ - res >>= 28; - switch (res) { - case ALC_MIC_EVENT: - alc88x_simple_mic_automute(codec); - break; - default: - alc_exec_unsol_event(codec, res); - break; - } -} - /* * ASUS pin configuration: * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a @@ -911,7 +810,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_ASUS_W1V] = "asus-w1v", [ALC880_ASUS_DIG] = "asus-dig", [ALC880_ASUS_DIG2] = "asus-dig2", - [ALC880_UNIWILL_DIG] = "uniwill", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -955,7 +853,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), - SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ @@ -1123,21 +1020,6 @@ static const struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, - [ALC880_UNIWILL] = { - .mixers = { alc880_uniwill_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_uniwill_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - .unsol_event = alc880_uniwill_unsol_event, - .setup = alc880_uniwill_setup, - .init_hook = alc880_uniwill_init_hook, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c index b344603ac06..a63a517780d 100644 --- a/sound/pci/hda/alc_quirks.c +++ b/sound/pci/hda/alc_quirks.c @@ -165,15 +165,3 @@ static void alc_simple_setup_automute(struct alc_spec *spec, int mode) spec->automute_lo = spec->automute_lo_possible = !!lo_pin; spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0]; } - -/* auto-toggle front mic */ -static void alc88x_simple_mic_automute(struct hda_codec *codec) -{ - unsigned int present; - unsigned char bits; - - present = snd_hda_jack_detect(codec, 0x18); - bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); -} - diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c0a46ed9ca..ff4410cf75a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4447,6 +4447,7 @@ enum { ALC880_FIXUP_VOL_KNOB, ALC880_FIXUP_FUJITSU, ALC880_FIXUP_F1734, + ALC880_FIXUP_UNIWILL, }; /* enable the volume-knob widget support on NID 0x21 */ @@ -4556,12 +4557,23 @@ static const struct alc_fixup alc880_fixups[] = { .chained = true, .chain_id = ALC880_FIXUP_VOL_KNOB, }, + [ALC880_FIXUP_UNIWILL] = { + /* need to fix HP and speaker pins to be parsed correctly */ + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x0121411f }, /* HP */ + { 0x15, 0x99030120 }, /* speaker */ + { 0x16, 0x99030130 }, /* bass speaker */ + { } + }, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), + SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_FIXUP_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_FIXUP_VOL_KNOB), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810), SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM), -- cgit v1.2.3-18-g5258 From 967b88c47744f7ec424c71630c1f551d34e08eef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:31:02 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=uniwill-dig with auto-parser ALC880 model=uniwill-dig requires the fix-up of bogus BIOS pin default configurations. Other than that, it's pretty normal. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 14 -------------- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 2 files changed, 13 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index c40f2446fcc..59899f8b056 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -17,7 +17,6 @@ enum { ALC880_ASUS_DIG, ALC880_ASUS_W1V, ALC880_ASUS_DIG2, - ALC880_UNIWILL_DIG, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -852,7 +851,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ @@ -1008,18 +1006,6 @@ static const struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, - [ALC880_UNIWILL_DIG] = { - .mixers = { alc880_asus_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_asus_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), - .channel_mode = alc880_asus_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ff4410cf75a..e88c753743d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4448,6 +4448,7 @@ enum { ALC880_FIXUP_FUJITSU, ALC880_FIXUP_F1734, ALC880_FIXUP_UNIWILL, + ALC880_FIXUP_UNIWILL_DIG, }; /* enable the volume-knob widget support on NID 0x21 */ @@ -4567,11 +4568,23 @@ static const struct alc_fixup alc880_fixups[] = { { } }, }, + [ALC880_FIXUP_UNIWILL_DIG] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + /* disable bogus unused pins */ + { 0x17, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { 0x1b, 0x411111f0 }, + { 0x1f, 0x411111f0 }, + { } + } + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), + SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_FIXUP_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_FIXUP_VOL_KNOB), -- cgit v1.2.3-18-g5258 From 96e225f6922ecf3afafb55fdb0e6e771b3f71e94 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:41:51 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=z71v with auto-parser ASUS Z71V has a totally broken BIOS setup (at least the info I got), thus we need to override the whole pin-config table to make the auto-parser working correctly. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 58 ------------------------------------------- sound/pci/hda/patch_realtek.c | 20 +++++++++++++++ 2 files changed, 20 insertions(+), 58 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 59899f8b056..6caa2010a85 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -10,7 +10,6 @@ enum { ALC880_3ST_DIG, ALC880_5ST, ALC880_5ST_DIG, - ALC880_Z71V, ALC880_6ST, ALC880_6ST_DIG, ALC880_ASUS, @@ -223,36 +222,11 @@ static const hda_nid_t alc880_w810_dac_nids[3] = { 0x02, 0x03, 0x04 }; -/* - * Z710V model - * - * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d) - * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?), - * Line = 0x1a - */ - -static const hda_nid_t alc880_z71v_dac_nids[1] = { - 0x02 -}; -#define ALC880_Z71V_HP_DAC 0x03 - /* fixed 2 channels */ static const struct hda_channel_mode alc880_2_jack_modes[1] = { { 2, NULL } }; -static const struct snd_kcontrol_new alc880_z71v_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { } /* end */ -}; - /* * ALC880 ASUS model * @@ -440,24 +414,6 @@ static const struct hda_verb alc880_pin_5stack_init_verbs[] = { { } }; -/* - * Z71V pin configuration: - * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?) - */ -static const struct hda_verb alc880_pin_z71v_init_verbs[] = { - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - /* * 6-stack pin configuration: * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, @@ -802,7 +758,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_3ST_DIG] = "3stack-digout", [ALC880_5ST] = "5stack", [ALC880_5ST_DIG] = "5stack-digout", - [ALC880_Z71V] = "z71v", [ALC880_6ST] = "6stack", [ALC880_6ST_DIG] = "6stack-digout", [ALC880_ASUS] = "asus", @@ -831,7 +786,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V), /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */ SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG), @@ -943,18 +897,6 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_sixstack_modes, .input_mux = &alc880_6stack_capture_source, }, - [ALC880_Z71V] = { - .mixers = { alc880_z71v_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_z71v_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids), - .dac_nids = alc880_z71v_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .hp_nid = 0x03, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, - .input_mux = &alc880_capture_source, - }, [ALC880_ASUS] = { .mixers = { alc880_asus_mixer }, .init_verbs = { alc880_volume_init_verbs, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e88c753743d..71acd9b9a88 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4449,6 +4449,7 @@ enum { ALC880_FIXUP_F1734, ALC880_FIXUP_UNIWILL, ALC880_FIXUP_UNIWILL_DIG, + ALC880_FIXUP_Z71V, }; /* enable the volume-knob widget support on NID 0x21 */ @@ -4579,10 +4580,29 @@ static const struct alc_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_Z71V] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + /* set up the whole pins as BIOS is utterly broken */ + { 0x14, 0x99030120 }, /* speaker */ + { 0x15, 0x0121411f }, /* HP */ + { 0x16, 0x411111f0 }, /* N/A */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x01a19950 }, /* mic-in */ + { 0x19, 0x411111f0 }, /* N/A */ + { 0x1a, 0x01813031 }, /* line-in */ + { 0x1b, 0x411111f0 }, /* N/A */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0x0144111e }, /* SPDIF */ + { } + } + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), -- cgit v1.2.3-18-g5258 From 411225a01e57189b4116d5c61c0d64bd4b76e602 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:48:19 +0100 Subject: ALSA: hda/realtek - Rewrite ALC880 model=asus-w1v with auto-parser ASUS W1V has a sane pin-config table set by BIOS. The only missing piece is the setup of GPIO1. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 31 ------------------------------- sound/pci/hda/patch_realtek.c | 6 ++++++ 2 files changed, 6 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index 6caa2010a85..c8af01b7f85 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -14,7 +14,6 @@ enum { ALC880_6ST_DIG, ALC880_ASUS, ALC880_ASUS_DIG, - ALC880_ASUS_W1V, ALC880_ASUS_DIG2, #ifdef CONFIG_SND_DEBUG ALC880_TEST, @@ -263,21 +262,6 @@ static const struct snd_kcontrol_new alc880_asus_mixer[] = { { } /* end */ }; -/* - * ALC880 ASUS W1V model - * - * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e) - * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16, - * Mic = 0x18, Line = 0x1a, Line2 = 0x1b - */ - -/* additional mixers to alc880_asus_mixer */ -static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { - HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT), - { } /* end */ -}; - /* * initialize the codec volumes, etc */ @@ -761,7 +745,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_6ST] = "6stack", [ALC880_6ST_DIG] = "6stack-digout", [ALC880_ASUS] = "asus", - [ALC880_ASUS_W1V] = "asus-w1v", [ALC880_ASUS_DIG] = "asus-dig", [ALC880_ASUS_DIG2] = "asus-dig2", #ifdef CONFIG_SND_DEBUG @@ -780,7 +763,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V), SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG), @@ -935,19 +917,6 @@ static const struct alc_config_preset alc880_presets[] = { .need_dac_fix = 1, .input_mux = &alc880_capture_source, }, - [ALC880_ASUS_W1V] = { - .mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_asus_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), - .channel_mode = alc880_asus_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 71acd9b9a88..510ca928b84 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4438,6 +4438,7 @@ static const struct hda_amp_list alc880_loopbacks[] = { * ALC880 fix-ups */ enum { + ALC880_FIXUP_GPIO1, ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, @@ -4461,6 +4462,10 @@ static void alc880_fixup_vol_knob(struct hda_codec *codec, } static const struct alc_fixup alc880_fixups[] = { + [ALC880_FIXUP_GPIO1] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio1_init_verbs, + }, [ALC880_FIXUP_GPIO2] = { .type = ALC_FIXUP_VERBS, .v.verbs = alc_gpio2_init_verbs, @@ -4602,6 +4607,7 @@ static const struct alc_fixup alc880_fixups[] = { static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), + SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_FIXUP_GPIO1), SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), -- cgit v1.2.3-18-g5258 From 29e3fdcc84e5da04cb7e6a36fee0a772c91d3b28 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 17:56:57 +0100 Subject: ALSA: hda/realtek - Drop model=asus* from ALC880 It turned out that BIOS on most of ASUS mobo's set the pin-config tables reasonably well for the auto-parser. We'd need GPIO setups, but should work as is other than that. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 127 +----------------------------------------- sound/pci/hda/patch_realtek.c | 3 +- 2 files changed, 3 insertions(+), 127 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index c8af01b7f85..6917d78d4dc 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -12,9 +12,6 @@ enum { ALC880_5ST_DIG, ALC880_6ST, ALC880_6ST_DIG, - ALC880_ASUS, - ALC880_ASUS_DIG, - ALC880_ASUS_DIG2, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -226,42 +223,6 @@ static const struct hda_channel_mode alc880_2_jack_modes[1] = { { 2, NULL } }; -/* - * ALC880 ASUS model - * - * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e) - * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16, - * Mic = 0x18, Line = 0x1a - */ - -#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */ -#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */ - -static const struct snd_kcontrol_new alc880_asus_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - /* * initialize the codec volumes, etc */ @@ -433,38 +394,6 @@ static const struct hda_verb alc880_beep_init_verbs[] = { { } }; -/* - * ASUS pin configuration: - * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a - */ -static const struct hda_verb alc880_pin_asus_init_verbs[] = { - {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, - {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - /* Enable GPIO mask and set output */ #define alc880_gpio1_init_verbs alc_gpio1_init_verbs #define alc880_gpio2_init_verbs alc_gpio2_init_verbs @@ -744,9 +673,6 @@ static const char * const alc880_models[ALC880_MODEL_LAST] = { [ALC880_5ST_DIG] = "5stack-digout", [ALC880_6ST] = "6stack", [ALC880_6ST_DIG] = "6stack-digout", - [ALC880_ASUS] = "asus", - [ALC880_ASUS_DIG] = "asus-dig", - [ALC880_ASUS_DIG2] = "asus-dig2", #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = "test", #endif @@ -763,19 +689,7 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG), - /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */ - SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), - SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), - SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), - SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */ + SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), @@ -785,7 +699,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG), SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), @@ -879,44 +792,6 @@ static const struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_sixstack_modes, .input_mux = &alc880_6stack_capture_source, }, - [ALC880_ASUS] = { - .mixers = { alc880_asus_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_asus_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), - .channel_mode = alc880_asus_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - }, - [ALC880_ASUS_DIG] = { - .mixers = { alc880_asus_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_asus_init_verbs, - alc880_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), - .channel_mode = alc880_asus_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - }, - [ALC880_ASUS_DIG2] = { - .mixers = { alc880_asus_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_asus_init_verbs, - alc880_gpio2_init_verbs }, /* use GPIO2 */ - .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), - .dac_nids = alc880_asus_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_asus_modes), - .channel_mode = alc880_asus_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 510ca928b84..fce31b050f4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4607,8 +4607,9 @@ static const struct alc_fixup alc880_fixups[] = { static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810), - SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_FIXUP_GPIO1), SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1558, 0x5401, "Clevo GPIO2", ALC880_FIXUP_GPIO2), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG), SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734), -- cgit v1.2.3-18-g5258 From 67b6ec3196da235317ff1b9474f17379b78f3294 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 18:20:42 +0100 Subject: ALSA: hda/realtek - Drop all ALC880 static quirks Finally the all static quirks for ALC880 are converted to the auto-parser. Since we are never sure whether the BIOS on so many old machines are really correct, the quirk table entries are copied as they are, but just providing the proper pin-config values accordingly. Since alc880_quirks.c is removed, alc882_quirks.c has to be adjusted slightly to be built again. There might be some compile warnings due to the remaining alc882 quirks, but these shall be killed sooner or later, I don't care it much at this point. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc880_quirks.c | 808 ------------------------------------------ sound/pci/hda/alc882_quirks.c | 24 +- sound/pci/hda/patch_realtek.c | 219 +++++++++--- 3 files changed, 195 insertions(+), 856 deletions(-) delete mode 100644 sound/pci/hda/alc880_quirks.c (limited to 'sound') diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c deleted file mode 100644 index 6917d78d4dc..00000000000 --- a/sound/pci/hda/alc880_quirks.c +++ /dev/null @@ -1,808 +0,0 @@ -/* - * ALC880 quirk models - * included by patch_realtek.c - */ - -/* ALC880 board config type */ -enum { - ALC880_AUTO, - ALC880_3ST, - ALC880_3ST_DIG, - ALC880_5ST, - ALC880_5ST_DIG, - ALC880_6ST, - ALC880_6ST_DIG, -#ifdef CONFIG_SND_DEBUG - ALC880_TEST, -#endif - ALC880_MODEL_LAST /* last tag */ -}; - -/* - * ALC880 3-stack model - * - * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e) - * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18, - * F-Mic = 0x1b, HP = 0x19 - */ - -static const hda_nid_t alc880_dac_nids[4] = { - /* front, rear, clfe, rear_surr */ - 0x02, 0x05, 0x04, 0x03 -}; - -static const hda_nid_t alc880_adc_nids[3] = { - /* ADC0-2 */ - 0x07, 0x08, 0x09, -}; - -/* The datasheet says the node 0x07 is connected from inputs, - * but it shows zero connection in the real implementation on some devices. - * Note: this is a 915GAV bug, fixed on 915GLV - */ -static const hda_nid_t alc880_adc_nids_alt[2] = { - /* ADC1-2 */ - 0x08, 0x09, -}; - -#define ALC880_DIGOUT_NID 0x06 -#define ALC880_DIGIN_NID 0x0a -#define ALC880_PIN_CD_NID 0x1c - -static const struct hda_input_mux alc880_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x3 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -/* channel source setting (2/6 channel selection for 3-stack) */ -/* 2ch mode */ -static const struct hda_verb alc880_threestack_ch2_init[] = { - /* set line-in to input, mute it */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - /* set mic-in to input vref 80%, mute it */ - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* 6ch mode */ -static const struct hda_verb alc880_threestack_ch6_init[] = { - /* set line-in to output, unmute it */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - /* set mic-in to output, unmute it */ - { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } /* end */ -}; - -static const struct hda_channel_mode alc880_threestack_modes[2] = { - { 2, alc880_threestack_ch2_init }, - { 6, alc880_threestack_ch6_init }, -}; - -static const struct snd_kcontrol_new alc880_three_stack_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -/* - * ALC880 5-stack model - * - * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d), - * Side = 0x02 (0xd) - * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16 - * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19 - */ - -/* additional mixers to alc880_three_stack_mixer */ -static const struct snd_kcontrol_new alc880_five_stack_mixer[] = { - HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT), - { } /* end */ -}; - -/* channel source setting (6/8 channel selection for 5-stack) */ -/* 6ch mode */ -static const struct hda_verb alc880_fivestack_ch6_init[] = { - /* set line-in to input, mute it */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; - -/* 8ch mode */ -static const struct hda_verb alc880_fivestack_ch8_init[] = { - /* set line-in to output, unmute it */ - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } /* end */ -}; - -static const struct hda_channel_mode alc880_fivestack_modes[2] = { - { 6, alc880_fivestack_ch6_init }, - { 8, alc880_fivestack_ch8_init }, -}; - - -/* - * ALC880 6-stack model - * - * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e), - * Side = 0x05 (0x0f) - * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17, - * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b - */ - -static const hda_nid_t alc880_6st_dac_nids[4] = { - /* front, rear, clfe, rear_surr */ - 0x02, 0x03, 0x04, 0x05 -}; - -static const struct hda_input_mux alc880_6stack_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -/* fixed 8-channels */ -static const struct hda_channel_mode alc880_sixstack_modes[1] = { - { 8, NULL }, -}; - -static const struct snd_kcontrol_new alc880_six_stack_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - - -static const hda_nid_t alc880_w810_dac_nids[3] = { - /* front, rear/surround, clfe */ - 0x02, 0x03, 0x04 -}; - -/* fixed 2 channels */ -static const struct hda_channel_mode alc880_2_jack_modes[1] = { - { 2, NULL } -}; - -/* - * initialize the codec volumes, etc - */ - -/* - * generic initialization of ADC, input mixers and output mixers - */ -static const struct hda_verb alc880_volume_init_verbs[] = { - /* - * Unmute ADC0-2 and set the default input to mic-in - */ - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback - * mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for front - * panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - - /* - * Set up output mixers (0x0c - 0x0f) - */ - /* set vol=0 to output mixers */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - { } -}; - -/* - * 3-stack pin configuration: - * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b - */ -static const struct hda_verb alc880_pin_3stack_init_verbs[] = { - /* - * preset connection lists of input pins - * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround - */ - {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */ - - /* - * Set pin mode and muting - */ - /* set front pin widgets 0x14 for output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mic2 (as headphone out) for HP output */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Line In pin widget for input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line2 (as front mic) pin widget for input and vref at 80% */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - -/* - * 5-stack pin configuration: - * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19, - * line-in/side = 0x1a, f-mic = 0x1b - */ -static const struct hda_verb alc880_pin_5stack_init_verbs[] = { - /* - * preset connection lists of input pins - * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround - */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */ - - /* - * Set pin mode and muting - */ - /* set pin widgets 0x14-0x17 for output */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* unmute pins for output (no gain on this amp) */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - /* Mic1 (rear panel) pin widget for input and vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mic2 (as headphone out) for HP output */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Line In pin widget for input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line2 (as front mic) pin widget for input and vref at 80% */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - -/* - * 6-stack pin configuration: - * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, - * f-mic = 0x19, line = 0x1a, HP = 0x1b - */ -static const struct hda_verb alc880_pin_6stack_init_verbs[] = { - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ - - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - { } -}; - -static const struct hda_verb alc880_beep_init_verbs[] = { - { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) }, - { } -}; - -/* Enable GPIO mask and set output */ -#define alc880_gpio1_init_verbs alc_gpio1_init_verbs -#define alc880_gpio2_init_verbs alc_gpio2_init_verbs -#define alc880_gpio3_init_verbs alc_gpio3_init_verbs - -/* - * Test configuration for debugging - * - * Almost all inputs/outputs are enabled. I/O pins can be configured via - * enum controls. - */ -#ifdef CONFIG_SND_DEBUG -static const hda_nid_t alc880_test_dac_nids[4] = { - 0x02, 0x03, 0x04, 0x05 -}; - -static const struct hda_input_mux alc880_test_capture_source = { - .num_items = 7, - .items = { - { "In-1", 0x0 }, - { "In-2", 0x1 }, - { "In-3", 0x2 }, - { "In-4", 0x3 }, - { "CD", 0x4 }, - { "Front", 0x5 }, - { "Surround", 0x6 }, - }, -}; - -static const struct hda_channel_mode alc880_test_modes[4] = { - { 2, NULL }, - { 4, NULL }, - { 6, NULL }, - { 8, NULL }, -}; - -static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { - "N/A", "Line Out", "HP Out", - "In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%" - }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 8; - if (uinfo->value.enumerated.item >= 8) - uinfo->value.enumerated.item = 7; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = (hda_nid_t)kcontrol->private_value; - unsigned int pin_ctl, item = 0; - - pin_ctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - if (pin_ctl & AC_PINCTL_OUT_EN) { - if (pin_ctl & AC_PINCTL_HP_EN) - item = 2; - else - item = 1; - } else if (pin_ctl & AC_PINCTL_IN_EN) { - switch (pin_ctl & AC_PINCTL_VREFEN) { - case AC_PINCTL_VREF_HIZ: item = 3; break; - case AC_PINCTL_VREF_50: item = 4; break; - case AC_PINCTL_VREF_GRD: item = 5; break; - case AC_PINCTL_VREF_80: item = 6; break; - case AC_PINCTL_VREF_100: item = 7; break; - } - } - ucontrol->value.enumerated.item[0] = item; - return 0; -} - -static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = (hda_nid_t)kcontrol->private_value; - static const unsigned int ctls[] = { - 0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_50, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_80, - AC_PINCTL_IN_EN | AC_PINCTL_VREF_100, - }; - unsigned int old_ctl, new_ctl; - - old_ctl = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - new_ctl = ctls[ucontrol->value.enumerated.item[0]]; - if (old_ctl != new_ctl) { - int val; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - new_ctl); - val = ucontrol->value.enumerated.item[0] >= 3 ? - HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0, - HDA_AMP_MUTE, val); - return 1; - } - return 0; -} - -static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { - "Front", "Surround", "CLFE", "Side" - }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item >= 4) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = (hda_nid_t)kcontrol->private_value; - unsigned int sel; - - sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); - ucontrol->value.enumerated.item[0] = sel & 3; - return 0; -} - -static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = (hda_nid_t)kcontrol->private_value; - unsigned int sel; - - sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3; - if (ucontrol->value.enumerated.item[0] != sel) { - sel = ucontrol->value.enumerated.item[0] & 3; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, sel); - return 1; - } - return 0; -} - -#define PIN_CTL_TEST(xname,nid) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_test_pin_ctl_info, \ - .get = alc_test_pin_ctl_get, \ - .put = alc_test_pin_ctl_put, \ - .private_value = nid \ - } - -#define PIN_SRC_TEST(xname,nid) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ - .info = alc_test_pin_src_info, \ - .get = alc_test_pin_src_get, \ - .put = alc_test_pin_src_put, \ - .private_value = nid \ - } - -static const struct snd_kcontrol_new alc880_test_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), - PIN_CTL_TEST("Front Pin Mode", 0x14), - PIN_CTL_TEST("Surround Pin Mode", 0x15), - PIN_CTL_TEST("CLFE Pin Mode", 0x16), - PIN_CTL_TEST("Side Pin Mode", 0x17), - PIN_CTL_TEST("In-1 Pin Mode", 0x18), - PIN_CTL_TEST("In-2 Pin Mode", 0x19), - PIN_CTL_TEST("In-3 Pin Mode", 0x1a), - PIN_CTL_TEST("In-4 Pin Mode", 0x1b), - PIN_SRC_TEST("In-1 Pin Source", 0x18), - PIN_SRC_TEST("In-2 Pin Source", 0x19), - PIN_SRC_TEST("In-3 Pin Source", 0x1a), - PIN_SRC_TEST("In-4 Pin Source", 0x1b), - HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static const struct hda_verb alc880_test_init_verbs[] = { - /* Unmute inputs of 0x0c - 0x0f */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Vol output for 0x0c-0x0f */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Set output pins 0x14-0x17 */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Unmute output pins 0x14-0x17 */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Set input pins 0x18-0x1c */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Mute input pins 0x18-0x1b */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* ADC set up */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Analog input/passthru */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - { } -}; -#endif - -/* - */ - -static const char * const alc880_models[ALC880_MODEL_LAST] = { - [ALC880_3ST] = "3stack", - [ALC880_3ST_DIG] = "3stack-digout", - [ALC880_5ST] = "5stack", - [ALC880_5ST_DIG] = "5stack-digout", - [ALC880_6ST] = "6stack", - [ALC880_6ST_DIG] = "6stack-digout", -#ifdef CONFIG_SND_DEBUG - [ALC880_TEST] = "test", -#endif - [ALC880_AUTO] = "auto", -}; - -static const struct snd_pci_quirk alc880_cfg_tbl[] = { - SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST), - SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), - SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - - SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), - SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), - SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), - SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST), - SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST), - SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG), - SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */ - SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG), - /* default Intel */ - SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST), - SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG), - SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG), - {} -}; - -/* - * ALC880 codec presets - */ -static const struct alc_config_preset alc880_presets[] = { - [ALC880_3ST] = { - .mixers = { alc880_three_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_3stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - }, - [ALC880_3ST_DIG] = { - .mixers = { alc880_three_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_3stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), - .channel_mode = alc880_threestack_modes, - .need_dac_fix = 1, - .input_mux = &alc880_capture_source, - }, - [ALC880_5ST] = { - .mixers = { alc880_three_stack_mixer, - alc880_five_stack_mixer}, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_5stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes), - .channel_mode = alc880_fivestack_modes, - .input_mux = &alc880_capture_source, - }, - [ALC880_5ST_DIG] = { - .mixers = { alc880_three_stack_mixer, - alc880_five_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_5stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_dac_nids), - .dac_nids = alc880_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes), - .channel_mode = alc880_fivestack_modes, - .input_mux = &alc880_capture_source, - }, - [ALC880_6ST] = { - .mixers = { alc880_six_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_6stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids), - .dac_nids = alc880_6st_dac_nids, - .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes), - .channel_mode = alc880_sixstack_modes, - .input_mux = &alc880_6stack_capture_source, - }, - [ALC880_6ST_DIG] = { - .mixers = { alc880_six_stack_mixer }, - .init_verbs = { alc880_volume_init_verbs, - alc880_pin_6stack_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids), - .dac_nids = alc880_6st_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes), - .channel_mode = alc880_sixstack_modes, - .input_mux = &alc880_6stack_capture_source, - }, -#ifdef CONFIG_SND_DEBUG - [ALC880_TEST] = { - .mixers = { alc880_test_mixer }, - .init_verbs = { alc880_test_init_verbs }, - .num_dacs = ARRAY_SIZE(alc880_test_dac_nids), - .dac_nids = alc880_test_dac_nids, - .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_test_modes), - .channel_mode = alc880_test_modes, - .input_mux = &alc880_test_capture_source, - }, -#endif -}; - diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index bb364a53f54..0f4292688e1 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -34,8 +34,16 @@ static const hda_nid_t alc882_dac_nids[4] = { #define alc883_dac_nids alc882_dac_nids /* ADCs */ -#define alc882_adc_nids alc880_adc_nids -#define alc882_adc_nids_alt alc880_adc_nids_alt +static const hda_nid_t alc882_adc_nids[3] = { + /* ADC0-2 */ + 0x07, 0x08, 0x09, +}; + +static const hda_nid_t alc882_adc_nids_alt[2] = { + /* ADC1-2 */ + 0x08, 0x09, +}; + #define alc883_adc_nids alc882_adc_nids_alt static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; @@ -774,7 +782,7 @@ static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { static const struct alc_config_preset alc882_presets[] = { [ALC885_MBA21] = { .mixers = { alc885_mba21_mixer }, - .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs }, + .init_verbs = { alc885_mba21_init_verbs, alc_gpio1_init_verbs }, .num_dacs = 2, .dac_nids = alc882_dac_nids, .channel_mode = alc885_mba21_ch_modes, @@ -787,7 +795,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC885_MBP3] = { .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mbp3_init_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .num_dacs = 2, .dac_nids = alc882_dac_nids, .hp_nid = 0x04, @@ -803,7 +811,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC885_MB5] = { .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_mb5_init_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .channel_mode = alc885_mb5_6ch_modes, @@ -818,7 +826,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC885_MACMINI3] = { .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, .init_verbs = { alc885_macmini3_init_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .channel_mode = alc885_macmini3_6ch_modes, @@ -833,7 +841,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC885_IMAC91] = { .mixers = {alc885_imac91_mixer}, .init_verbs = { alc885_imac91_init_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .channel_mode = alc885_mba21_ch_modes, @@ -848,7 +856,7 @@ static const struct alc_config_preset alc882_presets[] = { [ALC889A_MB31] = { .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, - alc880_gpio1_init_verbs }, + alc_gpio1_init_verbs }, .adc_nids = alc883_adc_nids, .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), .capsrc_nids = alc883_capsrc_nids, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fce31b050f4..4ac1e3830af 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4451,6 +4451,15 @@ enum { ALC880_FIXUP_UNIWILL, ALC880_FIXUP_UNIWILL_DIG, ALC880_FIXUP_Z71V, + ALC880_FIXUP_3ST_BASE, + ALC880_FIXUP_3ST, + ALC880_FIXUP_3ST_DIG, + ALC880_FIXUP_5ST_BASE, + ALC880_FIXUP_5ST, + ALC880_FIXUP_5ST_DIG, + ALC880_FIXUP_6ST_BASE, + ALC880_FIXUP_6ST, + ALC880_FIXUP_6ST_DIG, }; /* enable the volume-knob widget support on NID 0x21 */ @@ -4603,6 +4612,114 @@ static const struct alc_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_3ST_BASE] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x01014010 }, /* line-out */ + { 0x15, 0x411111f0 }, /* N/A */ + { 0x16, 0x411111f0 }, /* N/A */ + { 0x17, 0x411111f0 }, /* N/A */ + { 0x18, 0x01a19c30 }, /* mic-in */ + { 0x19, 0x0121411f }, /* HP */ + { 0x1a, 0x01813031 }, /* line-in */ + { 0x1b, 0x02a19c40 }, /* front-mic */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + /* 0x1e is filled in below */ + { 0x1f, 0x411111f0 }, /* N/A */ + { } + } + }, + [ALC880_FIXUP_3ST] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x411111f0 }, /* N/A */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_3ST_BASE, + }, + [ALC880_FIXUP_3ST_DIG] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x0144111e }, /* SPDIF */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_3ST_BASE, + }, + [ALC880_FIXUP_5ST_BASE] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x01014010 }, /* front */ + { 0x15, 0x411111f0 }, /* N/A */ + { 0x16, 0x01011411 }, /* CLFE */ + { 0x17, 0x01016412 }, /* surr */ + { 0x18, 0x01a19c30 }, /* mic-in */ + { 0x19, 0x0121411f }, /* HP */ + { 0x1a, 0x01813031 }, /* line-in */ + { 0x1b, 0x02a19c40 }, /* front-mic */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + /* 0x1e is filled in below */ + { 0x1f, 0x411111f0 }, /* N/A */ + { } + } + }, + [ALC880_FIXUP_5ST] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x411111f0 }, /* N/A */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_5ST_BASE, + }, + [ALC880_FIXUP_5ST_DIG] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x0144111e }, /* SPDIF */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_5ST_BASE, + }, + [ALC880_FIXUP_6ST_BASE] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x14, 0x01014010 }, /* front */ + { 0x15, 0x01016412 }, /* surr */ + { 0x16, 0x01011411 }, /* CLFE */ + { 0x17, 0x01012414 }, /* side */ + { 0x18, 0x01a19c30 }, /* mic-in */ + { 0x19, 0x02a19c40 }, /* front-mic */ + { 0x1a, 0x01813031 }, /* line-in */ + { 0x1b, 0x0121411f }, /* HP */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + /* 0x1e is filled in below */ + { 0x1f, 0x411111f0 }, /* N/A */ + { } + } + }, + [ALC880_FIXUP_6ST] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x411111f0 }, /* N/A */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_6ST_BASE, + }, + [ALC880_FIXUP_6ST_DIG] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x1e, 0x0144111e }, /* SPDIF */ + { } + }, + .chained = true, + .chain_id = ALC880_FIXUP_6ST_BASE, + }, }; static const struct snd_pci_quirk alc880_fixup_tbl[] = { @@ -4625,6 +4742,60 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700), + + /* Below is the copied entries from alc880_quirks.c. + * It's not quite sure whether BIOS sets the correct pin-config table + * on these machines, thus they are kept to be compatible with + * the old static quirks. Once when it's confirmed to work without + * these overrides, it'd be better to remove. + */ + SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_FIXUP_6ST), + SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_FIXUP_3ST), + SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_FIXUP_3ST), + SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_FIXUP_3ST), + SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_FIXUP_5ST), + SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_FIXUP_5ST), + SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_FIXUP_5ST), + SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_FIXUP_6ST_DIG), /* broken BIOS */ + SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_FIXUP_6ST_DIG), + SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_FIXUP_3ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_FIXUP_5ST_DIG), + /* default Intel */ + SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_FIXUP_3ST), + SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_FIXUP_5ST_DIG), + SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_FIXUP_6ST_DIG), + {} +}; + +static const struct alc_model_fixup alc880_fixup_models[] = { + {.id = ALC880_FIXUP_3ST, .name = "3stack"}, + {.id = ALC880_FIXUP_3ST_DIG, .name = "3stack-digout"}, + {.id = ALC880_FIXUP_5ST, .name = "5stack"}, + {.id = ALC880_FIXUP_5ST_DIG, .name = "5stack-digout"}, + {.id = ALC880_FIXUP_6ST, .name = "6stack"}, + {.id = ALC880_FIXUP_6ST_DIG, .name = "6stack-digout"}, {} }; @@ -4647,14 +4818,9 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { /* * OK, here we have finally the patch for ALC880 */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc880_quirks.c" -#endif - static int patch_alc880(struct hda_codec *codec) { struct alc_spec *spec; - int board_config; int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -4666,38 +4832,14 @@ static int patch_alc880(struct hda_codec *codec) spec->mixer_nid = 0x0b; spec->need_dac_fix = 1; - board_config = alc_board_config(codec, ALC880_MODEL_LAST, - alc880_models, alc880_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, NULL, alc880_fixup_tbl, alc880_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } - - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc880_parse_auto_config(codec); - if (err < 0) - goto error; -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS - else if (!err) { - printk(KERN_INFO - "hda_codec: Cannot set up configuration " - "from BIOS. Using 3-stack mode...\n"); - board_config = ALC880_3ST; - } -#endif - } + alc_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl, + alc880_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - if (board_config != ALC_MODEL_AUTO) { - spec->vmaster_nid = 0x0c; - setup_preset(codec, &alc880_presets[board_config]); - } + /* automatic parse from the BIOS config */ + err = alc880_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -4716,10 +4858,7 @@ static int patch_alc880(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; - else - codec->patch_ops.build_controls = __alc_build_controls; + spec->init_hook = alc_auto_init_std; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc880_loopbacks; -- cgit v1.2.3-18-g5258 From 9155f82a6a26da4a5b8d2d29f1d31836906b4712 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 18:41:02 +0100 Subject: ALSA: hda/realtek - Add model=fixup not to apply fix-ups If anyone wants to debug the driver and avoid the existing fix-ups, pass model=nofixup option. Then the driver will skip to pick up the fixup list. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4ac1e3830af..3c6f5b5161f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1556,6 +1556,13 @@ static void alc_pick_fixup(struct hda_codec *codec, int id = -1; const char *name = NULL; + /* when model=nofixup is given, don't pick up any fixups */ + if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { + spec->fixup_list = NULL; + spec->fixup_id = -1; + return; + } + if (codec->modelname && models) { while (models->name) { if (!strcmp(codec->modelname, models->name)) { -- cgit v1.2.3-18-g5258 From 65b30bee5824a888dbd636c19bce64570d4a88e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 22:50:18 -0800 Subject: ASoC: ak4104: Remove uninformative print on probe() Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index d27b5e4cce9..6f938929154 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -224,7 +224,6 @@ static int ak4104_probe(struct snd_soc_codec *codec) if (ret < 0) return ret; - dev_info(codec->dev, "SPI device initialized\n"); return 0; } -- cgit v1.2.3-18-g5258 From 38d78bafb98adb7b256ba24014471c5562ba4ee1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 16 Feb 2012 22:50:35 -0800 Subject: ASoC: ak4104: Convert to module_spi_driver() Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 6f938929154..f12c1154849 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -289,17 +289,7 @@ static struct spi_driver ak4104_spi_driver = { .remove = __devexit_p(ak4104_spi_remove), }; -static int __init ak4104_init(void) -{ - return spi_register_driver(&ak4104_spi_driver); -} -module_init(ak4104_init); - -static void __exit ak4104_exit(void) -{ - spi_unregister_driver(&ak4104_spi_driver); -} -module_exit(ak4104_exit); +module_spi_driver(ak4104_spi_driver); MODULE_AUTHOR("Daniel Mack "); MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver"); -- cgit v1.2.3-18-g5258 From fdd48f9d1961d579ded4d820348af5f3f0f94463 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 09:12:25 +0000 Subject: ASoC: wm8776: Add WM8775 device ID to the WM8776 The WM8775 is register compatible with the WM8776 so can be supported with the same driver though it is an ADC only part. Add the device ID to the WM8776 driver, further updates will be added in the future. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8776.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 33e97d1d8f4..a19db5a0a17 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -30,6 +30,11 @@ #include "wm8776.h" +enum wm8776_chip_type { + WM8775 = 1, + WM8776, +}; + /* codec private data */ struct wm8776_priv { enum snd_soc_control_type control_type; @@ -512,7 +517,8 @@ static __devexit int wm8776_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id wm8776_i2c_id[] = { - { "wm8776", 0 }, + { "wm8775", WM8775 }, + { "wm8776", WM8776 }, { } }; MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); -- cgit v1.2.3-18-g5258 From 26422625d55d94f60e48fa62e05f3b5aa5ad98f4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 09:36:49 +0000 Subject: ASoC: wm_hubs: Convert headphone driver to output driver widget Mostly for neatness, though it may help with sequencing in some situations. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 9742c666cd0..82b7e9dece2 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -670,9 +670,8 @@ SND_SOC_DAPM_PGA("Right Output PGA", WM8993_POWER_MANAGEMENT_3, 6, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Headphone Supply", SND_SOC_NOPM, 0, 0, hp_supply_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Headphone PGA", SND_SOC_NOPM, 0, 0, - NULL, 0, - hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_OUT_DRV_E("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0, + hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, earpiece_mixer, ARRAY_SIZE(earpiece_mixer)), -- cgit v1.2.3-18-g5258 From 3b8a079516328c06a50396dbe36a0b9f4ea4e9d4 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 21 Feb 2012 09:34:20 +0200 Subject: ASoC: twl4030: Debug code cleanup Replace the printk(KERN_ERR* instances with dev_err in the driver. While we are here clean up some of the debug messages as well. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 33 +++++++++++++++++---------------- 1 file changed, 17 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index d7eee0d502e..170cf9a8fc7 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1002,8 +1002,8 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, unsigned short mask, bitmask; if (twl4030->configured) { - printk(KERN_ERR "twl4030 operation mode cannot be " - "changed on-the-fly\n"); + dev_err(codec->dev, + "operation mode cannot be changed on-the-fly\n"); return -EBUSY; } @@ -1800,7 +1800,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, mode |= TWL4030_APLL_RATE_96000; break; default: - printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", + dev_err(codec->dev, "%s: unknown rate %d\n", __func__, params_rate(params)); return -EINVAL; } @@ -1817,7 +1817,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, format |= TWL4030_DATA_WIDTH_32S_24W; break; default: - printk(KERN_ERR "TWL4030 hw params: unknown format %d\n", + dev_err(codec->dev, "%s: unknown format %d\n", __func__, params_format(params)); return -EINVAL; } @@ -1867,13 +1867,13 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, case 38400000: break; default: - dev_err(codec->dev, "Unsupported APLL mclk: %u\n", freq); + dev_err(codec->dev, "Unsupported HFCLKIN: %u\n", freq); return -EINVAL; } if ((freq / 1000) != twl4030->sysclk) { dev_err(codec->dev, - "Mismatch in APLL mclk: %u (configured: %u)\n", + "Mismatch in HFCLKIN: %u (configured: %u)\n", freq, twl4030->sysclk * 1000); return -EINVAL; } @@ -1983,9 +1983,9 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, * not available. */ if (twl4030->sysclk != 26000) { - dev_err(codec->dev, "The board is configured for %u Hz, while" - "the Voice interface needs 26MHz APLL mclk\n", - twl4030->sysclk * 1000); + dev_err(codec->dev, + "%s: HFCLKIN is %u KHz, voice interface needs 26MHz\n", + __func__, twl4030->sysclk); return -EINVAL; } @@ -1996,8 +1996,8 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, & TWL4030_OPT_MODE; if (mode != TWL4030_OPTION_2) { - printk(KERN_ERR "TWL4030 voice startup: " - "the codec mode is not option2\n"); + dev_err(codec->dev, "%s: the codec mode is not option2\n", + __func__); return -EINVAL; } @@ -2038,7 +2038,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, mode |= TWL4030_SEL_16K; break; default: - printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + dev_err(codec->dev, "%s: unknown rate %d\n", __func__, params_rate(params)); return -EINVAL; } @@ -2067,13 +2067,14 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); if (freq != 26000000) { - dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice" - "interface needs 26MHz APLL mclk\n", freq); + dev_err(codec->dev, + "%s: HFCLKIN is %u KHz, voice interface needs 26MHz\n", + __func__, freq / 1000); return -EINVAL; } if ((freq / 1000) != twl4030->sysclk) { dev_err(codec->dev, - "Mismatch in APLL mclk: %u (configured: %u)\n", + "Mismatch in HFCLKIN: %u (configured: %u)\n", freq, twl4030->sysclk * 1000); return -EINVAL; } @@ -2221,7 +2222,7 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); if (twl4030 == NULL) { - printk("Can not allocate memory\n"); + dev_err(codec->dev, "Can not allocate memory\n"); return -ENOMEM; } snd_soc_codec_set_drvdata(codec, twl4030); -- cgit v1.2.3-18-g5258 From 1a97b7f22774b454531f013638b181803fba470f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Feb 2012 11:11:48 +0100 Subject: ALSA: hda/realtek - Remove the last static quirks for ALC882 Resitance is futile. The remaining static model quirks for Apple machines with ALC882-compatible codecs are converted to the auto-parser now. We can remove all alc*_quirks.c finally. Signed-off-by: Takashi Iwai --- sound/pci/hda/alc882_quirks.c | 874 ------------------------------------------ sound/pci/hda/alc_quirks.c | 167 -------- sound/pci/hda/patch_realtek.c | 143 ++++--- 3 files changed, 92 insertions(+), 1092 deletions(-) delete mode 100644 sound/pci/hda/alc882_quirks.c delete mode 100644 sound/pci/hda/alc_quirks.c (limited to 'sound') diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c deleted file mode 100644 index 0f4292688e1..00000000000 --- a/sound/pci/hda/alc882_quirks.c +++ /dev/null @@ -1,874 +0,0 @@ -/* - * ALC882/ALC883/ALC888/ALC889 quirk models - * included by patch_realtek.c - */ - -/* ALC882 models */ -enum { - ALC882_AUTO, - ALC885_MBA21, - ALC885_MBP3, - ALC885_MB5, - ALC885_MACMINI3, - ALC885_IMAC91, - ALC889A_MB31, - ALC882_MODEL_LAST, -}; - -#define ALC882_DIGOUT_NID 0x06 -#define ALC882_DIGIN_NID 0x0a -#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID -#define ALC883_DIGIN_NID ALC882_DIGIN_NID -#define ALC1200_DIGOUT_NID 0x10 - - -static const struct hda_channel_mode alc882_ch_modes[1] = { - { 8, NULL } -}; - -/* DACs */ -static const hda_nid_t alc882_dac_nids[4] = { - /* front, rear, clfe, rear_surr */ - 0x02, 0x03, 0x04, 0x05 -}; -#define alc883_dac_nids alc882_dac_nids - -/* ADCs */ -static const hda_nid_t alc882_adc_nids[3] = { - /* ADC0-2 */ - 0x07, 0x08, 0x09, -}; - -static const hda_nid_t alc882_adc_nids_alt[2] = { - /* ADC1-2 */ - 0x08, 0x09, -}; - -#define alc883_adc_nids alc882_adc_nids_alt - -static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 }; -#define alc883_capsrc_nids alc882_capsrc_nids_alt - -/* input MUX */ -/* FIXME: should be a matrix-type input source selection */ - -static const struct hda_input_mux alc882_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -#define alc883_capture_source alc882_capture_source - -static const struct hda_input_mux mb5_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x1 }, - { "Line", 0x7 }, - { "CD", 0x4 }, - }, -}; - -static const struct hda_input_mux macmini3_capture_source = { - .num_items = 2, - .items = { - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static const struct hda_input_mux alc883_3stack_6ch_intel = { - .num_items = 4, - .items = { - { "Mic", 0x1 }, - { "Front Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -static const struct hda_input_mux alc889A_mb31_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - /* Front Mic (0x01) unused */ - { "Line", 0x2 }, - /* Line 2 (0x03) unused */ - /* CD (0x04) unused? */ - }, -}; - -static const struct hda_input_mux alc889A_imac91_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x01 }, - { "Line", 0x2 }, /* Not sure! */ - }, -}; - -/* Macbook Air 2,1 */ - -static const struct hda_channel_mode alc885_mba21_ch_modes[1] = { - { 2, NULL }, -}; - -/* - * macbook pro ALC885 can switch LineIn to LineOut without losing Mic - */ - -/* - * 2ch mode - */ -static const struct hda_verb alc885_mbp_ch2_init[] = { - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - { } /* end */ -}; - -/* - * 4ch mode - */ -static const struct hda_verb alc885_mbp_ch4_init[] = { - { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - { } /* end */ -}; - -static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = { - { 2, alc885_mbp_ch2_init }, - { 4, alc885_mbp_ch4_init }, -}; - -/* - * 2ch - * Speakers/Woofer/HP = Front - * LineIn = Input - */ -static const struct hda_verb alc885_mb5_ch2_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - { } /* end */ -}; - -/* - * 6ch mode - * Speakers/HP = Front - * Woofer = LFE - * LineIn = Surround - */ -static const struct hda_verb alc885_mb5_ch6_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - { } /* end */ -}; - -static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = { - { 2, alc885_mb5_ch2_init }, - { 6, alc885_mb5_ch6_init }, -}; - -#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes - -/* Macbook Air 2,1 same control for HP and internal Speaker */ - -static const struct snd_kcontrol_new alc885_mba21_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT), - { } -}; - - -static const struct snd_kcontrol_new alc885_mbp3_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc885_mb5_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc885_macmini3_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc885_imac91_mixer[] = { - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT), - { } /* end */ -}; - - -static const struct snd_kcontrol_new alc882_chmode_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static const struct hda_verb alc882_base_init_verbs[] = { - /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* CLFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Side mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - - /* Front Pin: output 0 (0x0c) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Rear Pin: output 1 (0x0d) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* CLFE Pin: output 2 (0x0e) */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* Side Pin: output 3 (0x0f) */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: input */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line-2 In: Headphone output (output 0 - 0x0c) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* CD pin widget for input */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* ADC2: mute amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC3: mute amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - - { } -}; - -#define alc883_init_verbs alc882_base_init_verbs - -/* Macbook 5,1 */ -static const struct hda_verb alc885_mb5_init_verbs[] = { - /* DACs */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Front mixer */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Surround mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* LFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* HP mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Front Pin (0x0c) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* LFE Pin (0x0e) */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* HP Pin (0x0f) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)}, - { } -}; - -/* Macmini 3,1 */ -static const struct hda_verb alc885_macmini3_init_verbs[] = { - /* DACs */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Front mixer */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Surround mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* LFE mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* HP mixer */ - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Front Pin (0x0c) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* LFE Pin (0x0e) */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, - /* HP Pin (0x0f) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - /* Line In pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - { } -}; - - -static const struct hda_verb alc885_mba21_init_verbs[] = { - /*Internal and HP Speaker Mixer*/ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /*Internal Speaker Pin (0x0c)*/ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP Pin: output 0 (0x0e) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)}, - /* Line in (is hp when jack connected)*/ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - - { } - }; - - -/* Macbook Pro rev3 */ -static const struct hda_verb alc885_mbp3_init_verbs[] = { - /* Front mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* HP mixer */ - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Front Pin: output 0 (0x0c) */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP Pin: output 0 (0x0e) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - /* Mic (rear) pin: input vref at 80% */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Line In pin: use output 1 when in LineOut mode */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, - - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Input mixer3 */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* ADC1: mute amp left and right */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC2: mute amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* ADC3: mute amp left and right */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - - { } -}; - -/* iMac 9,1 */ -static const struct hda_verb alc885_imac91_init_verbs[] = { - /* Internal Speaker Pin (0x0c) */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) }, - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* HP Pin: Rear */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)}, - /* Line in Rear */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Front Mic pin: input vref at 80% */ - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Rear mixer */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, - { } -}; - -/* Toggle speaker-output according to the hp-jack state */ -static void alc885_imac24_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x18; - spec->autocfg.speaker_pins[1] = 0x1a; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -#define alc885_mb5_setup alc885_imac24_setup -#define alc885_macmini3_setup alc885_imac24_setup - -/* Macbook Air 2,1 */ -static void alc885_mba21_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x18; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - - - -static void alc885_mbp3_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -static void alc885_imac91_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x18; - spec->autocfg.speaker_pins[1] = 0x1a; - alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); -} - -/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */ -static const struct hda_verb alc889A_mb31_ch2_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ - { } /* end */ -}; - -/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */ -static const struct hda_verb alc889A_mb31_ch4_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */ - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ - { } /* end */ -}; - -/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */ -static const struct hda_verb alc889A_mb31_ch5_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */ - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */ - { } /* end */ -}; - -/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */ -static const struct hda_verb alc889A_mb31_ch6_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */ - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */ - { } /* end */ -}; - -static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = { - { 2, alc889A_mb31_ch2_init }, - { 4, alc889A_mb31_ch4_init }, - { 5, alc889A_mb31_ch5_init }, - { 6, alc889A_mb31_ch6_init }, -}; - -static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc889A_mb31_mixer[] = { - /* Output mixers */ - HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00, - HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT), - /* Output switches */ - HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT), - /* Boost mixers */ - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT), - /* Input mixers */ - HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc883_chmode_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = alc_ch_mode_info, - .get = alc_ch_mode_get, - .put = alc_ch_mode_put, - }, - { } /* end */ -}; - -static const struct hda_verb alc889A_mb31_verbs[] = { - /* Init rear pin (used as headphone output) */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */ - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN}, - /* Init line pin (used as output in 4ch and 6ch mode) */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */ - /* Init line 2 pin (used as headphone out by default) */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */ - { } /* end */ -}; - -/* Mute speakers according to the headphone jack state */ -static void alc889A_mb31_automute(struct hda_codec *codec) -{ - unsigned int present; - - /* Mute only in 2ch or 4ch mode */ - if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0) - == 0x00) { - present = snd_hda_jack_detect(codec, 0x15); - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - } -} - -static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) == ALC_HP_EVENT) - alc889A_mb31_automute(codec); -} - -static void alc882_unsol_event(struct hda_codec *codec, unsigned int res) -{ - alc_exec_unsol_event(codec, res >> 26); -} - -/* - * configuration and preset - */ -static const char * const alc882_models[ALC882_MODEL_LAST] = { - [ALC885_MB5] = "mb5", - [ALC885_MACMINI3] = "macmini3", - [ALC885_MBA21] = "mba21", - [ALC885_MBP3] = "mbp3", - [ALC885_IMAC91] = "imac91", - [ALC889A_MB31] = "mb31", - [ALC882_AUTO] = "auto", -}; - -/* codec SSID table for Intel Mac */ -static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = { - SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31), - SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21), - SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31), - SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3), - SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91), - SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5), - SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5), - /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2, - * so apparently no perfect solution yet - */ - SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5), - SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5), - SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3), - {} /* terminator */ -}; - -static const struct alc_config_preset alc882_presets[] = { - [ALC885_MBA21] = { - .mixers = { alc885_mba21_mixer }, - .init_verbs = { alc885_mba21_init_verbs, alc_gpio1_init_verbs }, - .num_dacs = 2, - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mba21_ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), - .input_mux = &alc882_capture_source, - .unsol_event = alc882_unsol_event, - .setup = alc885_mba21_setup, - .init_hook = alc_hp_automute, - }, - [ALC885_MBP3] = { - .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_mbp3_init_verbs, - alc_gpio1_init_verbs }, - .num_dacs = 2, - .dac_nids = alc882_dac_nids, - .hp_nid = 0x04, - .channel_mode = alc885_mbp_4ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes), - .input_mux = &alc882_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc882_unsol_event, - .setup = alc885_mbp3_setup, - .init_hook = alc_hp_automute, - }, - [ALC885_MB5] = { - .mixers = { alc885_mb5_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_mb5_init_verbs, - alc_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mb5_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes), - .input_mux = &mb5_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc882_unsol_event, - .setup = alc885_mb5_setup, - .init_hook = alc_hp_automute, - }, - [ALC885_MACMINI3] = { - .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer }, - .init_verbs = { alc885_macmini3_init_verbs, - alc_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_macmini3_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes), - .input_mux = &macmini3_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc882_unsol_event, - .setup = alc885_macmini3_setup, - .init_hook = alc_hp_automute, - }, - [ALC885_IMAC91] = { - .mixers = {alc885_imac91_mixer}, - .init_verbs = { alc885_imac91_init_verbs, - alc_gpio1_init_verbs }, - .num_dacs = ARRAY_SIZE(alc882_dac_nids), - .dac_nids = alc882_dac_nids, - .channel_mode = alc885_mba21_ch_modes, - .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes), - .input_mux = &alc889A_imac91_capture_source, - .dig_out_nid = ALC882_DIGOUT_NID, - .dig_in_nid = ALC882_DIGIN_NID, - .unsol_event = alc882_unsol_event, - .setup = alc885_imac91_setup, - .init_hook = alc_hp_automute, - }, - [ALC889A_MB31] = { - .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer}, - .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs, - alc_gpio1_init_verbs }, - .adc_nids = alc883_adc_nids, - .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), - .capsrc_nids = alc883_capsrc_nids, - .dac_nids = alc883_dac_nids, - .num_dacs = ARRAY_SIZE(alc883_dac_nids), - .channel_mode = alc889A_mb31_6ch_modes, - .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes), - .input_mux = &alc889A_mb31_capture_source, - .dig_out_nid = ALC883_DIGOUT_NID, - .unsol_event = alc889A_mb31_unsol_event, - .init_hook = alc889A_mb31_automute, - }, -}; - - diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c deleted file mode 100644 index a63a517780d..00000000000 --- a/sound/pci/hda/alc_quirks.c +++ /dev/null @@ -1,167 +0,0 @@ -/* - * Common codes for Realtek codec quirks - * included by patch_realtek.c - */ - -/* - * configuration template - to be copied to the spec instance - */ -struct alc_config_preset { - const struct snd_kcontrol_new *mixers[5]; /* should be identical size - * with spec - */ - const struct snd_kcontrol_new *cap_mixer; /* capture mixer */ - const struct hda_verb *init_verbs[5]; - unsigned int num_dacs; - const hda_nid_t *dac_nids; - hda_nid_t dig_out_nid; /* optional */ - hda_nid_t hp_nid; /* optional */ - const hda_nid_t *slave_dig_outs; - unsigned int num_adc_nids; - const hda_nid_t *adc_nids; - const hda_nid_t *capsrc_nids; - hda_nid_t dig_in_nid; - unsigned int num_channel_mode; - const struct hda_channel_mode *channel_mode; - int need_dac_fix; - int const_channel_count; - unsigned int num_mux_defs; - const struct hda_input_mux *input_mux; - void (*unsol_event)(struct hda_codec *, unsigned int); - void (*setup)(struct hda_codec *); - void (*init_hook)(struct hda_codec *); -#ifdef CONFIG_SND_HDA_POWER_SAVE - const struct hda_amp_list *loopbacks; - void (*power_hook)(struct hda_codec *codec); -#endif -}; - -/* - * channel mode setting - */ -static int alc_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode, - spec->num_channel_mode); -} - -static int alc_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - spec->ext_channel_count); -} - -static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct alc_spec *spec = codec->spec; - int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, - spec->num_channel_mode, - &spec->ext_channel_count); - if (err >= 0 && !spec->const_channel_count) { - spec->multiout.max_channels = spec->ext_channel_count; - if (spec->need_dac_fix) - spec->multiout.num_dacs = spec->multiout.max_channels / 2; - } - return err; -} - -static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - struct auto_pin_cfg *cfg = &spec->autocfg; - - if (!cfg->line_outs) { - while (cfg->line_outs < AUTO_CFG_MAX_OUTS && - cfg->line_out_pins[cfg->line_outs]) - cfg->line_outs++; - } - if (!cfg->speaker_outs) { - while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS && - cfg->speaker_pins[cfg->speaker_outs]) - cfg->speaker_outs++; - } - if (!cfg->hp_outs) { - while (cfg->hp_outs < AUTO_CFG_MAX_OUTS && - cfg->hp_pins[cfg->hp_outs]) - cfg->hp_outs++; - } -} - -/* - * set up from the preset table - */ -static void setup_preset(struct hda_codec *codec, - const struct alc_config_preset *preset) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++) - add_mixer(spec, preset->mixers[i]); - spec->cap_mixer = preset->cap_mixer; - for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i]; - i++) - add_verb(spec, preset->init_verbs[i]); - - spec->channel_mode = preset->channel_mode; - spec->num_channel_mode = preset->num_channel_mode; - spec->need_dac_fix = preset->need_dac_fix; - spec->const_channel_count = preset->const_channel_count; - - if (preset->const_channel_count) - spec->multiout.max_channels = preset->const_channel_count; - else - spec->multiout.max_channels = spec->channel_mode[0].channels; - spec->ext_channel_count = spec->channel_mode[0].channels; - - spec->multiout.num_dacs = preset->num_dacs; - spec->multiout.dac_nids = preset->dac_nids; - spec->multiout.dig_out_nid = preset->dig_out_nid; - spec->multiout.slave_dig_outs = preset->slave_dig_outs; - spec->multiout.hp_nid = preset->hp_nid; - - spec->num_mux_defs = preset->num_mux_defs; - if (!spec->num_mux_defs) - spec->num_mux_defs = 1; - spec->input_mux = preset->input_mux; - - spec->num_adc_nids = preset->num_adc_nids; - spec->adc_nids = preset->adc_nids; - spec->capsrc_nids = preset->capsrc_nids; - spec->dig_in_nid = preset->dig_in_nid; - - spec->unsol_event = preset->unsol_event; - spec->init_hook = preset->init_hook; -#ifdef CONFIG_SND_HDA_POWER_SAVE - spec->power_hook = preset->power_hook; - spec->loopback.amplist = preset->loopbacks; -#endif - - if (preset->setup) - preset->setup(codec); - - alc_fixup_autocfg_pin_nums(codec); -} - -static void alc_simple_setup_automute(struct alc_spec *spec, int mode) -{ - int lo_pin = spec->autocfg.line_out_pins[0]; - - if (lo_pin == spec->autocfg.speaker_pins[0] || - lo_pin == spec->autocfg.hp_pins[0]) - lo_pin = 0; - spec->automute_mode = mode; - spec->detect_hp = !!spec->autocfg.hp_pins[0]; - spec->detect_lo = !!lo_pin; - spec->automute_lo = spec->automute_lo_possible = !!lo_pin; - spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0]; -} diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c6f5b5161f..c5216b58d21 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4807,21 +4807,6 @@ static const struct alc_model_fixup alc880_fixup_models[] = { }; -/* - * board setups - */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#define alc_board_config \ - snd_hda_check_board_config -#define alc_board_codec_sid_config \ - snd_hda_check_board_codec_sid_config -#include "alc_quirks.c" -#else -#define alc_board_config(codec, nums, models, tbl) -1 -#define alc_board_codec_sid_config(codec, nums, models, tbl) -1 -#define setup_preset(codec, x) /* NOP */ -#endif - /* * OK, here we have finally the patch for ALC880 */ @@ -5091,6 +5076,8 @@ enum { ALC882_FIXUP_EAPD, ALC883_FIXUP_EAPD, ALC883_FIXUP_ACER_EAPD, + ALC882_FIXUP_GPIO1, + ALC882_FIXUP_GPIO2, ALC882_FIXUP_GPIO3, ALC889_FIXUP_COEF, ALC882_FIXUP_ASUS_W2JC, @@ -5099,6 +5086,8 @@ enum { ALC882_FIXUP_ASPIRE_8930G_VERBS, ALC885_FIXUP_MACPRO_GPIO, ALC889_FIXUP_DAC_ROUTE, + ALC889_FIXUP_MBP_VREF, + ALC889_FIXUP_IMAC91_VREF, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -5169,6 +5158,51 @@ static void alc889_fixup_dac_route(struct hda_codec *codec, } } +/* Set VREF on HP pin */ +static void alc889_fixup_mbp_vref(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static hda_nid_t nids[2] = { 0x14, 0x15 }; + int i; + + if (action != ALC_FIXUP_ACT_INIT) + return; + for (i = 0; i < ARRAY_SIZE(nids); i++) { + unsigned int val = snd_hda_codec_get_pincfg(codec, nids[i]); + if (get_defcfg_device(val) != AC_JACK_HP_OUT) + continue; + val = snd_hda_codec_read(codec, nids[i], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + val |= AC_PINCTL_VREF_80; + snd_hda_codec_write(codec, nids[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + spec->keep_vref_in_automute = 1; + break; + } +} + +/* Set VREF on speaker pins on imac91 */ +static void alc889_fixup_imac91_vref(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static hda_nid_t nids[2] = { 0x18, 0x1a }; + int i; + + if (action != ALC_FIXUP_ACT_INIT) + return; + for (i = 0; i < ARRAY_SIZE(nids); i++) { + unsigned int val; + val = snd_hda_codec_read(codec, nids[i], 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + val |= AC_PINCTL_VREF_50; + snd_hda_codec_write(codec, nids[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + } + spec->keep_vref_in_automute = 1; +} + static const struct alc_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = ALC_FIXUP_PINS, @@ -5247,6 +5281,14 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [ALC882_FIXUP_GPIO1] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio1_init_verbs, + }, + [ALC882_FIXUP_GPIO2] = { + .type = ALC_FIXUP_VERBS, + .v.verbs = alc_gpio2_init_verbs, + }, [ALC882_FIXUP_GPIO3] = { .type = ALC_FIXUP_VERBS, .v.verbs = alc_gpio3_init_verbs, @@ -5320,6 +5362,18 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_FUNC, .v.func = alc889_fixup_dac_route, }, + [ALC889_FIXUP_MBP_VREF] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_mbp_vref, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, + }, + [ALC889_FIXUP_IMAC91_VREF] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc889_fixup_imac91_vref, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -5353,11 +5407,26 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), /* All Apple entries are in codec SSIDs */ + SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), + SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), @@ -5382,14 +5451,10 @@ static int alc882_parse_auto_config(struct hda_codec *codec) /* */ -#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS -#include "alc882_quirks.c" -#endif - static int patch_alc882(struct hda_codec *codec) { struct alc_spec *spec; - int err, board_config; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -5413,36 +5478,15 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - board_config = alc_board_config(codec, ALC882_MODEL_LAST, - alc882_models, NULL); - if (board_config < 0) - board_config = alc_board_codec_sid_config(codec, - ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl); - - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = ALC_MODEL_AUTO; - } - - if (board_config == ALC_MODEL_AUTO) { - alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); - alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); - } + alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); + alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); - if (board_config == ALC_MODEL_AUTO) { - /* automatic parse from the BIOS config */ - err = alc882_parse_auto_config(codec); - if (err < 0) - goto error; - } - - if (board_config != ALC_MODEL_AUTO) { - setup_preset(codec, &alc882_presets[board_config]); - spec->vmaster_nid = 0x0c; - } + /* automatic parse from the BIOS config */ + err = alc882_parse_auto_config(codec); + if (err < 0) + goto error; if (!spec->no_analog && !spec->adc_nids) { alc_auto_fill_adc_caps(codec); @@ -5461,10 +5505,7 @@ static int patch_alc882(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - if (board_config == ALC_MODEL_AUTO) - spec->init_hook = alc_auto_init_std; - else - codec->patch_ops.build_controls = __alc_build_controls; + spec->init_hook = alc_auto_init_std; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) -- cgit v1.2.3-18-g5258 From 164f73ee93131f67a61eaca6a6f6180580c39445 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Feb 2012 11:27:09 +0100 Subject: ALSA: hda/realtek - Parse aa-loopback items dynamically Similarly in patch_via.c, parse the active analog-loopback connections and create a list dynamically rather than static arrays. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 104 +++++++++--------------------------------- 1 file changed, 22 insertions(+), 82 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c5216b58d21..eba50dff613 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -200,6 +200,8 @@ struct alc_spec { hda_nid_t vmaster_nid; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; + int num_loopbacks; + struct hda_amp_list loopback_list[8]; #endif /* for PLL fix */ @@ -2690,6 +2692,25 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch, return channel_name[ch]; } +#ifdef CONFIG_SND_HDA_POWER_SAVE +/* add the powersave loopback-list entry */ +static void add_loopback_list(struct alc_spec *spec, hda_nid_t mix, int idx) +{ + struct hda_amp_list *list; + + if (spec->num_loopbacks >= ARRAY_SIZE(spec->loopback_list) - 1) + return; + list = spec->loopback_list + spec->num_loopbacks; + list->nid = mix; + list->dir = HDA_INPUT; + list->idx = idx; + spec->num_loopbacks++; + spec->loopback.amplist = spec->loopback_list; +} +#else +#define add_loopback_list(spec, mix, idx) /* NOP */ +#endif + /* create input playback/capture controls for the given pin */ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ctlname, int ctlidx, @@ -2705,6 +2726,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; + add_loopback_list(spec, mix_nid, idx); return 0; } @@ -4430,17 +4452,6 @@ static int alc880_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids); } -#ifdef CONFIG_SND_HDA_POWER_SAVE -static const struct hda_amp_list alc880_loopbacks[] = { - { 0x0b, HDA_INPUT, 0 }, - { 0x0b, HDA_INPUT, 1 }, - { 0x0b, HDA_INPUT, 2 }, - { 0x0b, HDA_INPUT, 3 }, - { 0x0b, HDA_INPUT, 4 }, - { } /* end */ -}; -#endif - /* * ALC880 fix-ups */ @@ -4851,10 +4862,6 @@ static int patch_alc880(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!spec->loopback.amplist) - spec->loopback.amplist = alc880_loopbacks; -#endif alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -4876,17 +4883,6 @@ static int alc260_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc260_ignore, alc260_ssids); } -#ifdef CONFIG_SND_HDA_POWER_SAVE -static const struct hda_amp_list alc260_loopbacks[] = { - { 0x07, HDA_INPUT, 0 }, - { 0x07, HDA_INPUT, 1 }, - { 0x07, HDA_INPUT, 2 }, - { 0x07, HDA_INPUT, 3 }, - { 0x07, HDA_INPUT, 4 }, - { } /* end */ -}; -#endif - /* * Pin config fixes */ @@ -5032,10 +5028,6 @@ static int patch_alc260(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!spec->loopback.amplist) - spec->loopback.amplist = alc260_loopbacks; -#endif alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -5058,9 +5050,6 @@ static int patch_alc260(struct hda_codec *codec) * In addition, an independent DAC for the multi-playback (not used in this * driver yet). */ -#ifdef CONFIG_SND_HDA_POWER_SAVE -#define alc882_loopbacks alc880_loopbacks -#endif /* * Pin config fixes @@ -5507,11 +5496,6 @@ static int patch_alc882(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; spec->init_hook = alc_auto_init_std; -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!spec->loopback.amplist) - spec->loopback.amplist = alc882_loopbacks; -#endif - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); return 0; @@ -5608,10 +5592,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = { }; -#ifdef CONFIG_SND_HDA_POWER_SAVE -#define alc262_loopbacks alc880_loopbacks -#endif - /* */ static int patch_alc262(struct hda_codec *codec) @@ -5671,11 +5651,6 @@ static int patch_alc262(struct hda_codec *codec) spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!spec->loopback.amplist) - spec->loopback.amplist = alc262_loopbacks; -#endif - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); return 0; @@ -5793,10 +5768,6 @@ static int patch_alc268(struct hda_codec *codec) /* * ALC269 */ -#ifdef CONFIG_SND_HDA_POWER_SAVE -#define alc269_loopbacks alc880_loopbacks -#endif - static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = { .substreams = 1, .channels_min = 2, @@ -6336,8 +6307,6 @@ static int patch_alc269(struct hda_codec *codec) spec->shutup = alc269_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE - if (!spec->loopback.amplist) - spec->loopback.amplist = alc269_loopbacks; if (alc269_mic2_for_mute_led(codec)) codec->patch_ops.check_power_status = alc269_mic2_mute_check_ps; #endif @@ -6362,17 +6331,6 @@ static int alc861_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc861_ignore, alc861_ssids); } -#ifdef CONFIG_SND_HDA_POWER_SAVE -static const struct hda_amp_list alc861_loopbacks[] = { - { 0x15, HDA_INPUT, 0 }, - { 0x15, HDA_INPUT, 1 }, - { 0x15, HDA_INPUT, 2 }, - { 0x15, HDA_INPUT, 3 }, - { } /* end */ -}; -#endif - - /* Pin config fixes */ enum { ALC861_FIXUP_FSC_AMILO_PI1505, @@ -6486,8 +6444,6 @@ static int patch_alc861(struct hda_codec *codec) spec->init_hook = alc_auto_init_std; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->power_hook = alc_power_eapd; - if (!spec->loopback.amplist) - spec->loopback.amplist = alc861_loopbacks; #endif alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6506,10 +6462,6 @@ static int patch_alc861(struct hda_codec *codec) * * In addition, an independent DAC */ -#ifdef CONFIG_SND_HDA_POWER_SAVE -#define alc861vd_loopbacks alc880_loopbacks -#endif - static int alc861vd_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc861vd_ignore[] = { 0x1d, 0 }; @@ -6610,10 +6562,6 @@ static int patch_alc861vd(struct hda_codec *codec) spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!spec->loopback.amplist) - spec->loopback.amplist = alc861vd_loopbacks; -#endif alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6635,9 +6583,6 @@ static int patch_alc861vd(struct hda_codec *codec) * In addition, an independent DAC for the multi-playback (not used in this * driver yet). */ -#ifdef CONFIG_SND_HDA_POWER_SAVE -#define alc662_loopbacks alc880_loopbacks -#endif /* * BIOS auto configuration @@ -6999,11 +6944,6 @@ static int patch_alc662(struct hda_codec *codec) spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (!spec->loopback.amplist) - spec->loopback.amplist = alc662_loopbacks; -#endif - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); return 0; -- cgit v1.2.3-18-g5258 From 29fdc3605c947d037f3333afe3c295f8708640b0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 10:50:50 +0000 Subject: ASoC: wm8994: Enable headphone startup mode 1 for WM1811 and WM8958 The latest recommendation for optimal performance. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 18c99cda7cc..e9a405a1fda 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3492,11 +3492,13 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: wm8994->hubs.dcs_readback_mode = 1; + wm8994->hubs.hp_startup_mode = 1; break; case WM1811: wm8994->hubs.dcs_readback_mode = 2; wm8994->hubs.no_series_update = 1; + wm8994->hubs.hp_startup_mode = 1; switch (wm8994->revision) { case 0: -- cgit v1.2.3-18-g5258 From 5803a326465e38ee3cab8badbd8947732a8277f5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Feb 2012 11:59:45 +0100 Subject: ALSA: hda/realtek - Fix possible Oops with NULL input_mux When BIOS is damn crazy and gives no pin-config at all, the driver might lead to a NULL dereference. Let's add a NULL check for such a case. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eba50dff613..997cc8127a0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -302,6 +302,9 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx, int i, type, num_conns; hda_nid_t nid; + if (!spec->input_mux) + return 0; + mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; if (!imux->num_items && mux_idx > 0) -- cgit v1.2.3-18-g5258 From c96f0bf4adc0663a69cdb0e2b73d33e6be312d1c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Feb 2012 12:12:57 +0100 Subject: ALSA: hda/realtek - Create individual mute switches for shared DAC Even if the outputs are using shared DACs, we can still create individual mute siwtches since they are assigned per pin. This allows to create, e.g. Speaker and Bass Speaker mute switches while the single volume is used for these outputs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 50 +++++++++++++++++-------------------------- 1 file changed, 20 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 997cc8127a0..3cedb26f9cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3717,41 +3717,31 @@ static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins, return alc_auto_create_extra_out(codec, *pins, dac, pfx, 0); } - if (dacs[num_pins - 1]) { - /* OK, we have a multi-output system with individual volumes */ - for (i = 0; i < num_pins; i++) { - if (num_pins >= 3) { - snprintf(name, sizeof(name), "%s %s", - pfx, channel_name[i]); - err = alc_auto_create_extra_out(codec, pins[i], dacs[i], - name, 0); - } else { - err = alc_auto_create_extra_out(codec, pins[i], dacs[i], - pfx, i); - } - if (err < 0) - return err; - } - return 0; - } - - /* Let's create a bind-controls */ - ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_sw); - if (!ctl) - return -ENOMEM; - n = 0; for (i = 0; i < num_pins; i++) { - if (get_wcaps(codec, pins[i]) & AC_WCAP_OUT_AMP) - ctl->values[n++] = - HDA_COMPOSE_AMP_VAL(pins[i], 3, 0, HDA_OUTPUT); - } - if (n) { - snprintf(name, sizeof(name), "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_SW, name, 0, (long)ctl); + hda_nid_t dac; + if (dacs[num_pins - 1]) + dac = dacs[i]; /* with individual volumes */ + else + dac = 0; + if (num_pins == 2 && i == 1 && !strcmp(pfx, "Speaker")) { + err = alc_auto_create_extra_out(codec, pins[i], dac, + "Bass Speaker", 0); + } else if (num_pins >= 3) { + snprintf(name, sizeof(name), "%s %s", + pfx, channel_name[i]); + err = alc_auto_create_extra_out(codec, pins[i], dac, + name, 0); + } else { + err = alc_auto_create_extra_out(codec, pins[i], dac, + pfx, i); + } if (err < 0) return err; } + if (dacs[num_pins - 1]) + return 0; + /* Let's create a bind-controls for volumes */ ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_vol); if (!ctl) return -ENOMEM; -- cgit v1.2.3-18-g5258 From 689cabf6d07c82003310c221f719130f3a4f29c7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Feb 2012 12:35:27 +0100 Subject: ALSA: hda/realtek - Fix the possible conflicts of Bass Speaker name When the multi-io is added to the two speaker output configuration, the parser would try to add yet another "Bass Speaker" control since it checks only cfg->line_outs. Add a workaround for it by simply passing the channel name in the case of multi-io outputs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3cedb26f9cf..e5c04593d36 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3613,14 +3613,17 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec, dac = spec->multiout.dac_nids[i]; if (!dac) continue; - if (i >= cfg->line_outs) + if (i >= cfg->line_outs) { pin = spec->multi_io[i - 1].pin; - else + index = 0; + name = channel_name[i]; + } else { pin = cfg->line_out_pins[i]; + name = alc_get_line_out_pfx(spec, i, true, &index); + } sw = alc_look_for_out_mute_nid(codec, pin, dac); vol = alc_look_for_out_vol_nid(codec, pin, dac); - name = alc_get_line_out_pfx(spec, i, true, &index); if (!name || !strcmp(name, "CLFE")) { /* Center/LFE */ err = alc_auto_add_vol_ctl(codec, "Center", 0, vol, 1); -- cgit v1.2.3-18-g5258 From f568291ef571522202e8b9f893ab33694bb2fc31 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Feb 2012 12:37:00 +0100 Subject: ALSA: hda/realtek - Fix the wrong offset for two-speaker systems When the machine has two speakers but wants to put more multi-io jacks, the parser shouldn't consider about the shared DAC but try to assign the individual DACs. Otherwise the channel mapping would be fairly confused and lead to the wrong DACs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e5c04593d36..e82911ab0f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3353,7 +3353,11 @@ static int fill_and_eval_dacs(struct hda_codec *codec, } if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { /* try multi-ios with HP + inputs */ - err = alc_auto_fill_multi_ios(codec, cfg->hp_pins[0], false, 1); + int offset = 0; + if (cfg->line_outs >= 3) + offset = 1; + err = alc_auto_fill_multi_ios(codec, cfg->hp_pins[0], false, + offset); if (err < 0) return err; badness += err; -- cgit v1.2.3-18-g5258 From 070cff4cfd267b9d266f4f8362ea99532234eb21 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Feb 2012 12:54:17 +0100 Subject: ALSA: hda/realtek - Small code cleanups A few clean-ups for post-static-quirk time: - Call alc_auto_init_std() statically in alc_init() instead of setting spec->init_hook in each caller. spec->init_hook field is left unused for any future use. - Move the call of set_capture_mixer() to to alc_parse_auto_config() instead of each caller. - Get rid of the ADC-filling and imux check in each parser function. This is no longer needed since the auto-parser always check ADCs and imux. It was only for the static quirks. - Kill unused defines Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 140 ++---------------------------------------- 1 file changed, 5 insertions(+), 135 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e82911ab0f8..e142f6f5c49 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -222,8 +222,6 @@ struct alc_spec { struct snd_array bind_ctls; }; -#define ALC_MODEL_AUTO 0 /* common for all chips */ - static bool check_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int bits) { @@ -1074,45 +1072,6 @@ static bool alc_check_dyn_adc_switch(struct hda_codec *codec) return true; } -/* rebuild imux for matching with the given auto-mic pins (if not yet) */ -static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - struct hda_input_mux *imux; - static char * const texts[3] = { - "Mic", "Internal Mic", "Dock Mic" - }; - int i; - - if (!spec->auto_mic) - return false; - imux = &spec->private_imux[0]; - if (spec->input_mux == imux) - return true; - spec->imux_pins[0] = spec->ext_mic_pin; - spec->imux_pins[1] = spec->int_mic_pin; - spec->imux_pins[2] = spec->dock_mic_pin; - for (i = 0; i < 3; i++) { - strcpy(imux->items[i].label, texts[i]); - if (spec->imux_pins[i]) { - hda_nid_t pin = spec->imux_pins[i]; - int c; - for (c = 0; c < spec->num_adc_nids; c++) { - hda_nid_t cap = get_capsrc(spec, c); - int idx = get_connection_index(codec, cap, pin); - if (idx >= 0) { - imux->items[i].index = idx; - break; - } - } - imux->num_items = i + 1; - } - } - spec->num_mux_defs = 1; - spec->input_mux = imux; - return true; -} - /* check whether all auto-mic pins are valid; setup indices if OK */ static bool alc_auto_mic_check_imux(struct hda_codec *codec) { @@ -2092,6 +2051,7 @@ static int alc_build_controls(struct hda_codec *codec) */ static void alc_init_special_input_src(struct hda_codec *codec); +static void alc_auto_init_std(struct hda_codec *codec); static int alc_init(struct hda_codec *codec) { @@ -2104,6 +2064,7 @@ static int alc_init(struct hda_codec *codec) for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); alc_init_special_input_src(codec); + alc_auto_init_std(codec); if (spec->init_hook) spec->init_hook(codec); @@ -4442,6 +4403,9 @@ static int alc_parse_auto_config(struct hda_codec *codec, if (spec->kctls.list) add_mixer(spec, spec->kctls.list); + if (!spec->no_analog && !spec->cap_mixer) + set_capture_mixer(codec); + return 1; } @@ -4844,15 +4808,6 @@ static int patch_alc880(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); - } - - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) @@ -4861,7 +4816,6 @@ static int patch_alc880(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - spec->init_hook = alc_auto_init_std; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -5009,15 +4963,6 @@ static int patch_alc260(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); - } - - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) @@ -5026,7 +4971,6 @@ static int patch_alc260(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -5477,15 +5421,6 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); - } - - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) @@ -5494,7 +5429,6 @@ static int patch_alc882(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - spec->init_hook = alc_auto_init_std; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -5631,15 +5565,6 @@ static int patch_alc262(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); - } - - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) @@ -5648,7 +5573,6 @@ static int patch_alc262(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -5745,17 +5669,7 @@ static int patch_alc268(struct hda_codec *codec) (0 << AC_AMPCAP_MUTE_SHIFT)); } - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); - } - - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - codec->patch_ops = alc_patch_ops; - spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; return 0; @@ -6283,15 +6197,6 @@ static int patch_alc269(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); - } - - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) @@ -6303,7 +6208,6 @@ static int patch_alc269(struct hda_codec *codec) #ifdef CONFIG_PM codec->patch_ops.resume = alc269_resume; #endif - spec->init_hook = alc_auto_init_std; spec->shutup = alc269_shutup; #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -6424,15 +6328,6 @@ static int patch_alc861(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); - } - - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x23); if (err < 0) @@ -6441,7 +6336,6 @@ static int patch_alc861(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - spec->init_hook = alc_auto_init_std; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->power_hook = alc_power_eapd; #endif @@ -6542,15 +6436,6 @@ static int patch_alc861vd(struct hda_codec *codec) add_verb(spec, alc660vd_eapd_verbs); } - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); - } - - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - if (!spec->no_analog) { err = snd_hda_attach_beep_device(codec, 0x23); if (err < 0) @@ -6560,7 +6445,6 @@ static int patch_alc861vd(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; - spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6912,15 +6796,6 @@ static int patch_alc662(struct hda_codec *codec) if (err < 0) goto error; - if (!spec->no_analog && !spec->adc_nids) { - alc_auto_fill_adc_caps(codec); - alc_rebuild_imux_for_auto_mic(codec); - alc_remove_invalid_adc_nids(codec); - } - - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - if (!spec->no_analog && has_cdefine_beep(codec)) { err = snd_hda_attach_beep_device(codec, 0x1); if (err < 0) @@ -6941,7 +6816,6 @@ static int patch_alc662(struct hda_codec *codec) } codec->patch_ops = alc_patch_ops; - spec->init_hook = alc_auto_init_std; spec->shutup = alc_eapd_shutup; alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); @@ -6984,11 +6858,7 @@ static int patch_alc680(struct hda_codec *codec) return err; } - if (!spec->no_analog && !spec->cap_mixer) - set_capture_mixer(codec); - codec->patch_ops = alc_patch_ops; - spec->init_hook = alc_auto_init_std; return 0; } -- cgit v1.2.3-18-g5258 From 55a27786856458a785e1ed7221aee22a06def877 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 13:45:53 +0000 Subject: ASoC: wm8994: Actively discharge idle MICBIAS with jack detect This minimises the chance of any external capacitors that are fitted being discharged into headphones as they insert. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e9a405a1fda..77085c1047d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3224,6 +3224,9 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_MECHANICAL, SND_JACK_MECHANICAL); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_DISCH, 0); + /* * Start off measument of microphone impedence to find * out what's actually there. @@ -3235,6 +3238,9 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) } else { dev_dbg(codec->dev, "Jack not detected\n"); + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_DISCH, WM8958_MICB2_DISCH); + snd_soc_jack_report(wm8994->micdet[0].jack, 0, SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); @@ -3320,6 +3326,9 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, * otherwise jump straight to microphone detection. */ if (wm8994->jackdet) { + snd_soc_update_bits(codec, WM8958_MICBIAS2, + WM8958_MICB2_DISCH, + WM8958_MICB2_DISCH); snd_soc_update_bits(codec, WM8994_LDO_1, WM8994_LDO1_DISCH, 0); wm1811_jackdet_set_mode(codec, -- cgit v1.2.3-18-g5258 From 07fb9d9e935a07aaed557c58d795c18fcd99aab4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 16:23:35 +0000 Subject: ASoC: wm8994: Support external capacitors on MICBIAS2 with jack detection When an external capacitor is connected to MICBIAS2 on devices with jack detection (which is not required but may be done in some systems) then the loading may mean that better performance is obtained when the microphone bias is enabled normally rather than using the low power mode. Provide platform data allowing systems to indicate if they require this. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 77085c1047d..0b1c271468a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3172,6 +3172,14 @@ static void wm8958_default_micdet(u16 status, void *data) wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK); + + if (wm8994->pdata->jd_ext_cap) { + mutex_lock(&codec->mutex); + snd_soc_dapm_disable_pin(&codec->dapm, + "MICBIAS2"); + snd_soc_dapm_sync(&codec->dapm); + mutex_unlock(&codec->mutex); + } } } @@ -3227,6 +3235,15 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); + /* If required for an external cap force MICBIAS on */ + if (wm8994->pdata->jd_ext_cap) { + mutex_lock(&codec->mutex); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS2"); + snd_soc_dapm_sync(&codec->dapm); + mutex_unlock(&codec->mutex); + } + /* * Start off measument of microphone impedence to find * out what's actually there. @@ -3241,6 +3258,13 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, WM8958_MICB2_DISCH); + if (wm8994->pdata->jd_ext_cap) { + mutex_lock(&codec->mutex); + snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); + snd_soc_dapm_sync(&codec->dapm); + mutex_unlock(&codec->mutex); + } + snd_soc_jack_report(wm8994->micdet[0].jack, 0, SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); -- cgit v1.2.3-18-g5258 From e54e2f81da0aaf9e39d6f9b25fa08ce0bec9ca06 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Feb 2012 17:47:55 +0100 Subject: ASoC: imx-pcm: Remove empty prepare callback Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 11 ----------- 1 file changed, 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 5780c9b9d56..bc1df166385 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -186,16 +186,6 @@ static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params; - - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - return 0; -} - static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -291,7 +281,6 @@ static struct snd_pcm_ops imx_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_imx_pcm_hw_params, .hw_free = snd_imx_pcm_hw_free, - .prepare = snd_imx_pcm_prepare, .trigger = snd_imx_pcm_trigger, .pointer = snd_imx_pcm_pointer, .mmap = snd_imx_pcm_mmap, -- cgit v1.2.3-18-g5258 From f6914024575fc3fd9773531ca74d1bcb0ddaf88f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Feb 2012 17:47:56 +0100 Subject: ASoC: imx-pcm: Remove unused fields from imx_pcm_runtime_data struct Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index bc1df166385..ec139441552 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -35,11 +35,7 @@ struct imx_pcm_runtime_data { int period_bytes; int periods; - int dma; unsigned long offset; - unsigned long size; - void *buf; - int period_time; struct dma_async_tx_descriptor *desc; struct dma_chan *dma_chan; struct imx_dma_data dma_data; @@ -144,19 +140,14 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, return ret; chan = iprtd->dma_chan; - iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); iprtd->period_bytes = params_period_bytes(params); iprtd->offset = 0; - iprtd->period_time = HZ / (params_rate(params) / - params_period_size(params)); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); dma_addr = runtime->dma_addr; - iprtd->buf = (unsigned int *)substream->dma_buffer.area; - iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr, iprtd->period_bytes * iprtd->periods, iprtd->period_bytes, -- cgit v1.2.3-18-g5258 From 8949490f70cf2cda615fb0fd0ddc299b531e6e48 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 21 Feb 2012 17:47:57 +0100 Subject: ASoC: mxs-pcm: Remove unused fields from struct mxs_pcm_runtime_data Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 5 ----- sound/soc/mxs/mxs-pcm.h | 4 ---- 2 files changed, 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 105f42a394d..06c18ecffbb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -119,19 +119,14 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, return ret; chan = iprtd->dma_chan; - iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); iprtd->period_bytes = params_period_bytes(params); iprtd->offset = 0; - iprtd->period_time = HZ / (params_rate(params) / - params_period_size(params)); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); dma_addr = runtime->dma_addr; - iprtd->buf = substream->dma_buffer.area; - iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr, iprtd->period_bytes * iprtd->periods, iprtd->period_bytes, diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index f55ac4f7a76..ba75e103bb3 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -29,11 +29,7 @@ struct mxs_pcm_dma_params { struct mxs_pcm_runtime_data { int period_bytes; int periods; - int dma; unsigned long offset; - unsigned long size; - void *buf; - int period_time; struct dma_async_tx_descriptor *desc; struct dma_chan *dma_chan; struct mxs_dma_data dma_data; -- cgit v1.2.3-18-g5258 From 2b4bdee2920fb3894f9116f76343f8b31f9e4da8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 Feb 2012 14:33:29 -0800 Subject: ASoC: io: Retrieve val_bytes from the regmap API Allow us to build infrastructure which needs to know the size of a value without requiring regmap based drivers to supply this information to both ASoC and regmap by asking regmap for the value. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-io.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 39ba5070ff9..4d8dc6a27d4 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -114,6 +114,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, enum snd_soc_control_type control) { struct regmap_config config; + int ret; memset(&config, 0, sizeof(config)); codec->write = hw_write; @@ -141,6 +142,11 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ codec->using_regmap = true; + + ret = regmap_get_val_bytes(codec->control_data); + /* Errors are legitimate for non-integer byte multiples */ + if (ret > 0) + codec->val_bytes = ret; break; default: -- cgit v1.2.3-18-g5258 From 71d08516b80638a69d5efea4e8cb832c053f9dd9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Oct 2011 18:31:26 +0100 Subject: ASoC: core: Add SND_SOC_BYTES control for coefficient blocks Allow devices to export blocks of registers to the application layer, intended for use for reading and writing coefficient data which can't usefully be worked with by the kernel at runtime (for example, due to requiring complex and expensive calculations or being the results of callibration procedures). Currently drivers are using platform data to provide configurations for coefficient blocks which isn't at all convenient for runtime management or configuration development. Currently only devices using regmap are supported, an error will be generated for any attempt to work with a byte control on a non-regmap device. There's no fundamental block to other devices so support could be added if required. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3ca70594e24..a9786ab7050 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2736,6 +2736,55 @@ int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r_sx); +int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_bytes *params = (void *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = params->num_regs * codec->val_bytes; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_info); + +int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int ret; + + if (codec->using_regmap) + ret = regmap_raw_read(codec->control_data, params->base, + ucontrol->value.bytes.data, + params->num_regs * codec->val_bytes); + else + ret = -EINVAL; + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_get); + +int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_bytes *params = (void *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int ret; + + if (codec->using_regmap) + ret = regmap_raw_write(codec->control_data, params->base, + ucontrol->value.bytes.data, + params->num_regs * codec->val_bytes); + else + ret = -EINVAL; + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_put); + /** * snd_soc_dai_set_sysclk - configure DAI system or master clock. * @dai: DAI -- cgit v1.2.3-18-g5258 From f831b055ececb3172f7fe498db5ca1fb43ff644d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 Feb 2012 16:20:33 -0800 Subject: ASoC: core: Add support for masking out parts of coefficient blocks Chip designers frequently include things like the enable and disable controls for algorithms in the register blocks which also hold the coefficients. Since it's desirable to split out the enable/disable control from userspace the plain SND_SOC_BYTES() isn't optimal for these devices. Add a SND_SOC_BYTES_MASK() which allows a bitmask from the first word of the block to be excluded from the control. This supports the needs of devices I've looked at and lets us have a reasonably simple API. Further controls can be added in future if that's needed. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 74 +++++++++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 67 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a9786ab7050..fc0fd3485e7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2763,6 +2763,25 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, else ret = -EINVAL; + /* Hide any masked bytes to ensure consistent data reporting */ + if (ret == 0 && params->mask) { + switch (codec->val_bytes) { + case 1: + ucontrol->value.bytes.data[0] &= ~params->mask; + break; + case 2: + ((u16 *)(&ucontrol->value.bytes.data))[0] + &= ~params->mask; + break; + case 4: + ((u32 *)(&ucontrol->value.bytes.data))[0] + &= ~params->mask; + break; + default: + return -EINVAL; + } + } + return ret; } EXPORT_SYMBOL_GPL(snd_soc_bytes_get); @@ -2772,14 +2791,55 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, { struct soc_bytes *params = (void *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - int ret; + int ret, len; + unsigned int val; + void *data; - if (codec->using_regmap) - ret = regmap_raw_write(codec->control_data, params->base, - ucontrol->value.bytes.data, - params->num_regs * codec->val_bytes); - else - ret = -EINVAL; + if (!codec->using_regmap) + return -EINVAL; + + data = ucontrol->value.bytes.data; + len = params->num_regs * codec->val_bytes; + + /* + * If we've got a mask then we need to preserve the register + * bits. We shouldn't modify the incoming data so take a + * copy. + */ + if (params->mask) { + ret = regmap_read(codec->control_data, params->base, &val); + if (ret != 0) + return ret; + + val &= params->mask; + + data = kmemdup(data, len, GFP_KERNEL); + if (!data) + return -ENOMEM; + + switch (codec->val_bytes) { + case 1: + ((u8 *)data)[0] &= ~params->mask; + ((u8 *)data)[0] |= val; + break; + case 2: + ((u16 *)data)[0] &= cpu_to_be16(~params->mask); + ((u16 *)data)[0] |= cpu_to_be16(val); + break; + case 4: + ((u32 *)data)[0] &= cpu_to_be32(~params->mask); + ((u32 *)data)[0] |= cpu_to_be32(val); + break; + default: + return -EINVAL; + } + } + + ret = regmap_raw_write(codec->control_data, params->base, + data, len); + + if (params->mask) + kfree(data); return ret; } -- cgit v1.2.3-18-g5258 From 29e3cc15972389dc8f101f2788cdf56114330ad1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 19:13:10 +0000 Subject: ASoC: wm8996: Implement DRC coefficient configuration Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index aba144f6994..adaaf80bd9e 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -716,10 +716,16 @@ SOC_SINGLE("DSP2 EQ Switch", WM8996_DSP2_RX_EQ_GAINS_1, 0, 1, 0), SOC_SINGLE("DSP1 DRC TXL Switch", WM8996_DSP1_DRC_1, 0, 1, 0), SOC_SINGLE("DSP1 DRC TXR Switch", WM8996_DSP1_DRC_1, 1, 1, 0), SOC_SINGLE("DSP1 DRC RX Switch", WM8996_DSP1_DRC_1, 2, 1, 0), +SND_SOC_BYTES_MASK("DSP1 DRC", WM8996_DSP1_DRC_1, 5, + WM8996_DSP1RX_DRC_ENA | WM8996_DSP1TXL_DRC_ENA | + WM8996_DSP1TXR_DRC_ENA), SOC_SINGLE("DSP2 DRC TXL Switch", WM8996_DSP2_DRC_1, 0, 1, 0), SOC_SINGLE("DSP2 DRC TXR Switch", WM8996_DSP2_DRC_1, 1, 1, 0), SOC_SINGLE("DSP2 DRC RX Switch", WM8996_DSP2_DRC_1, 2, 1, 0), +SND_SOC_BYTES_MASK("DSP2 DRC", WM8996_DSP2_DRC_1, 5, + WM8996_DSP2RX_DRC_ENA | WM8996_DSP2TXL_DRC_ENA | + WM8996_DSP2TXR_DRC_ENA), }; static const struct snd_kcontrol_new wm8996_eq_controls[] = { -- cgit v1.2.3-18-g5258 From cdaaf301dae0077e36c20f3fc5cdb8774ead3c9d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 10 Oct 2011 18:31:44 +0100 Subject: ASoC: wm5100: Implement DRC coefficient configuration Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 2339aa0e945..b9c185ce64e 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -709,6 +709,8 @@ WM5100_MIXER_CONTROLS("EQ4", WM5100_EQ4MIX_INPUT_1_SOURCE), WM5100_MIXER_CONTROLS("DRC1L", WM5100_DRC1LMIX_INPUT_1_SOURCE), WM5100_MIXER_CONTROLS("DRC1R", WM5100_DRC1RMIX_INPUT_1_SOURCE), +SND_SOC_BYTES_MASK("DRC", WM5100_DRC1_CTRL1, 5, + WM5100_DRCL_ENA | WM5100_DRCR_ENA), WM5100_MIXER_CONTROLS("LHPF1", WM5100_HPLP1MIX_INPUT_1_SOURCE), WM5100_MIXER_CONTROLS("LHPF2", WM5100_HPLP2MIX_INPUT_1_SOURCE), -- cgit v1.2.3-18-g5258 From 1ec1cdfbb37add5af839d50ae8729961a8a307c1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 Feb 2012 13:02:48 -0800 Subject: ASoC: wm8996: Convert to use DAPM routes for stream connections Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 74 +++++++++++++++++++++++++---------------------- 1 file changed, 39 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index adaaf80bd9e..fb5c07a9ec9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1175,41 +1175,25 @@ SND_SOC_DAPM_DAC("DAC2R", NULL, WM8996_POWER_MANAGEMENT_5, 2, 0), SND_SOC_DAPM_DAC("DAC1L", NULL, WM8996_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8996_POWER_MANAGEMENT_5, 0, 0), -SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 0, - WM8996_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 1, - WM8996_POWER_MANAGEMENT_4, 8, 0), - -SND_SOC_DAPM_AIF_OUT("AIF2TX1", "AIF2 Capture", 0, - WM8996_POWER_MANAGEMENT_6, 9, 0), -SND_SOC_DAPM_AIF_OUT("AIF2TX0", "AIF2 Capture", 1, - WM8996_POWER_MANAGEMENT_6, 8, 0), - -SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 5, - WM8996_POWER_MANAGEMENT_4, 5, 0), -SND_SOC_DAPM_AIF_IN("AIF1RX4", "AIF1 Playback", 4, - WM8996_POWER_MANAGEMENT_4, 4, 0), -SND_SOC_DAPM_AIF_IN("AIF1RX3", "AIF1 Playback", 3, - WM8996_POWER_MANAGEMENT_4, 3, 0), -SND_SOC_DAPM_AIF_IN("AIF1RX2", "AIF1 Playback", 2, - WM8996_POWER_MANAGEMENT_4, 2, 0), -SND_SOC_DAPM_AIF_IN("AIF1RX1", "AIF1 Playback", 1, - WM8996_POWER_MANAGEMENT_4, 1, 0), -SND_SOC_DAPM_AIF_IN("AIF1RX0", "AIF1 Playback", 0, - WM8996_POWER_MANAGEMENT_4, 0, 0), - -SND_SOC_DAPM_AIF_OUT("AIF1TX5", "AIF1 Capture", 5, - WM8996_POWER_MANAGEMENT_6, 5, 0), -SND_SOC_DAPM_AIF_OUT("AIF1TX4", "AIF1 Capture", 4, - WM8996_POWER_MANAGEMENT_6, 4, 0), -SND_SOC_DAPM_AIF_OUT("AIF1TX3", "AIF1 Capture", 3, - WM8996_POWER_MANAGEMENT_6, 3, 0), -SND_SOC_DAPM_AIF_OUT("AIF1TX2", "AIF1 Capture", 2, - WM8996_POWER_MANAGEMENT_6, 2, 0), -SND_SOC_DAPM_AIF_OUT("AIF1TX1", "AIF1 Capture", 1, - WM8996_POWER_MANAGEMENT_6, 1, 0), -SND_SOC_DAPM_AIF_OUT("AIF1TX0", "AIF1 Capture", 0, - WM8996_POWER_MANAGEMENT_6, 0, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, WM8996_POWER_MANAGEMENT_4, 9, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX0", NULL, 1, WM8996_POWER_MANAGEMENT_4, 8, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, WM8996_POWER_MANAGEMENT_6, 9, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX0", NULL, 1, WM8996_POWER_MANAGEMENT_6, 8, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 5, WM8996_POWER_MANAGEMENT_4, 5, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 4, WM8996_POWER_MANAGEMENT_4, 4, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 3, WM8996_POWER_MANAGEMENT_4, 3, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 2, WM8996_POWER_MANAGEMENT_4, 2, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 1, WM8996_POWER_MANAGEMENT_4, 1, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX0", NULL, 0, WM8996_POWER_MANAGEMENT_4, 0, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 5, WM8996_POWER_MANAGEMENT_6, 5, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 4, WM8996_POWER_MANAGEMENT_6, 4, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 3, WM8996_POWER_MANAGEMENT_6, 3, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 2, WM8996_POWER_MANAGEMENT_6, 2, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 1, WM8996_POWER_MANAGEMENT_6, 1, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX0", NULL, 0, WM8996_POWER_MANAGEMENT_6, 0, 0), /* We route as stereo pairs so define some dummy widgets to squash * things down for now. RXA = 0,1, RXB = 2,3 and so on */ @@ -1284,6 +1268,26 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "MICB2", NULL, "MICB2 Audio" }, { "MICB2", NULL, "Bandgap" }, + { "AIF1RX0", NULL, "AIF1 Playback" }, + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + + { "AIF2RX0", NULL, "AIF2 Playback" }, + { "AIF2RX1", NULL, "AIF2 Playback" }, + + { "AIF1 Capture", NULL, "AIF1TX0" }, + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + + { "AIF2 Capture", NULL, "AIF2TX0" }, + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "IN1L PGA", NULL, "IN2LN" }, { "IN1L PGA", NULL, "IN2LP" }, { "IN1L PGA", NULL, "IN1LN" }, -- cgit v1.2.3-18-g5258 From f7085641eeaeaab5216bab143287d2288940d9a2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 21 Feb 2012 16:24:00 +0000 Subject: ASoC: wm8994: Move wm_hubs callback before we start ramping VMID Allows the generic code to set up for that. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0b1c271468a..33bc718f164 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -784,6 +784,8 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_LINEOUT2_DISCH, WM8994_LINEOUT_VMID_BUF_ENA); + wm_hubs_vmid_ena(codec); + /* Startup bias, VMID ramp & buffer */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, WM8994_BIAS_SRC | @@ -796,8 +798,6 @@ static void vmid_reference(struct snd_soc_codec *codec) WM8994_VMID_BUF_ENA | (0x2 << WM8994_VMID_RAMP_SHIFT)); - wm_hubs_vmid_ena(codec); - /* Main bias enable, VMID=2x40k */ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, WM8994_BIAS_ENA | -- cgit v1.2.3-18-g5258 From 8d8bbc6f17b2a28c58de804064dbdab036d4318e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Feb 2012 12:26:38 +0100 Subject: ALSA: hda/via - Don't create duplicated boost controls The driver may create duplicated mic boost controls when there are multiple mics with the very same type / location, and this leads to the error at actual kcontrol creation. It needs to check the validity of the created control and add a proper index if it's duplicated. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c7eb4d7d05c..93d52fc605f 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -2474,6 +2474,8 @@ static int create_mic_boost_ctls(struct hda_codec *codec) { struct via_spec *spec = codec->spec; const struct auto_pin_cfg *cfg = &spec->autocfg; + const char *prev_label = NULL; + int type_idx = 0; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -2488,8 +2490,13 @@ static int create_mic_boost_ctls(struct hda_codec *codec) if (caps == -1 || !(caps & AC_AMPCAP_NUM_STEPS)) continue; label = hda_get_autocfg_input_label(codec, cfg, i); + if (prev_label && !strcmp(label, prev_label)) + type_idx++; + else + type_idx = 0; + prev_label = label; snprintf(name, sizeof(name), "%s Boost Volume", label); - err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, + err = __via_add_control(spec, VIA_CTL_WIDGET_VOL, name, type_idx, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT)); if (err < 0) return err; -- cgit v1.2.3-18-g5258 From 77e314f72241daeac575158f946e905191611f0e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Feb 2012 12:34:08 +0100 Subject: ALSA: hda/via - Add a few sanity checks Added sanity checks in a few places not to assume the pins having the certain amp caps or the input-source being always assigned to a mux. No actual bugs have been triggered by these, but surely better to be a bit more robust. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 93d52fc605f..06214fdc948 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -550,7 +550,10 @@ static void via_auto_init_output(struct hda_codec *codec, pin = path->path[path->depth - 1]; init_output_pin(codec, pin, pin_type); - caps = query_amp_caps(codec, pin, HDA_OUTPUT); + if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP) + caps = query_amp_caps(codec, pin, HDA_OUTPUT); + else + caps = 0; if (caps & AC_AMPCAP_MUTE) { unsigned int val; val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; @@ -645,6 +648,10 @@ static void via_auto_init_analog_input(struct hda_codec *codec) /* init ADCs */ for (i = 0; i < spec->num_adc_nids; i++) { + hda_nid_t nid = spec->adc_nids[i]; + if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP) || + !(query_amp_caps(codec, nid, HDA_INPUT) & AC_AMPCAP_MUTE)) + continue; snd_hda_codec_write(codec, spec->adc_nids[i], 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); @@ -1508,6 +1515,8 @@ static int via_build_controls(struct hda_codec *codec) /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { + if (!spec->mux_nids[i]) + continue; err = snd_hda_add_nid(codec, kctl, i, spec->mux_nids[i]); if (err < 0) return err; -- cgit v1.2.3-18-g5258 From 91a38540f504cdde7cb62668f7a6d52e3bd0178b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 22 Feb 2012 10:49:05 +0100 Subject: ASoC: imx-ssi: Set dma data early Move the call to snd_soc_dai_set_dma_data from the hw_params callback to the startup callback. This allows us to use the dma data in the pcm driver's open callback. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 28 ++++++++++++++++++++-------- 1 file changed, 20 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 01d1f749cf0..dbf43f5f0ae 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -233,6 +233,23 @@ static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, return 0; } +static int imx_ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); + struct imx_pcm_dma_params *dma_data; + + /* Tx/Rx config */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &ssi->dma_params_tx; + else + dma_data = &ssi->dma_params_rx; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + + return 0; +} + /* * Should only be called when port is inactive (i.e. SSIEN = 0), * although can be called multiple times by upper layers. @@ -242,23 +259,17 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct imx_ssi *ssi = snd_soc_dai_get_drvdata(cpu_dai); - struct imx_pcm_dma_params *dma_data; u32 reg, sccr; /* Tx/Rx config */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = SSI_STCCR; - dma_data = &ssi->dma_params_tx; - } else { + else reg = SSI_SRCCR; - dma_data = &ssi->dma_params_rx; - } if (ssi->flags & IMX_SSI_SYN) reg = SSI_STCCR; - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; /* DAI data (word) size */ @@ -343,6 +354,7 @@ static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, } static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { + .startup = imx_ssi_startup, .hw_params = imx_ssi_hw_params, .set_fmt = imx_ssi_set_dai_fmt, .set_clkdiv = imx_ssi_set_dai_clkdiv, -- cgit v1.2.3-18-g5258 From 4564d10f3066a1abf5053936684e2c8495163def Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 22 Feb 2012 10:49:06 +0100 Subject: ASoC: imx-pcm: Request DMA channel early Request the DMA channel in the pcm open callback. This allows us to let open fail if there is no dma channel available. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 78 +++++++++++++++++------------------------ 1 file changed, 32 insertions(+), 46 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index ec139441552..f974e61fa68 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -65,17 +65,13 @@ static bool filter(struct dma_chan *chan, void *param) return true; } -static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct imx_pcm_dma_params *dma_params; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - struct dma_slave_config slave_config; dma_cap_mask_t mask; - enum dma_slave_buswidth buswidth; - int ret; dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); @@ -84,13 +80,29 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, iprtd->dma_data.dma_request = dma_params->dma; /* Try to grab a DMA channel */ - if (!iprtd->dma_chan) { - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); - if (!iprtd->dma_chan) - return -EINVAL; - } + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); + if (!iprtd->dma_chan) + return -EINVAL; + + return 0; +} + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct dma_chan *chan = iprtd->dma_chan; + struct imx_pcm_dma_params *dma_params; + struct dma_slave_config slave_config; + enum dma_slave_buswidth buswidth; + unsigned long dma_addr; + int ret; + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -116,29 +128,10 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.src_maxburst = dma_params->burstsize; } - ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); + ret = dmaengine_slave_config(chan, &slave_config); if (ret) return ret; - return 0; -} - -static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - unsigned long dma_addr; - struct dma_chan *chan; - struct imx_pcm_dma_params *dma_params; - int ret; - - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - ret = imx_ssi_dma_alloc(substream, params); - if (ret) - return ret; - chan = iprtd->dma_chan; iprtd->periods = params_periods(params); iprtd->period_bytes = params_period_bytes(params); @@ -164,19 +157,6 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - - if (iprtd->dma_chan) { - dma_release_channel(iprtd->dma_chan); - iprtd->dma_chan = NULL; - } - - return 0; -} - static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -251,6 +231,12 @@ static int snd_imx_open(struct snd_pcm_substream *substream) return ret; } + ret = imx_ssi_dma_alloc(substream); + if (ret < 0) { + kfree(iprtd); + return ret; + } + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); return 0; @@ -261,6 +247,7 @@ static int snd_imx_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; + dma_release_channel(iprtd->dma_chan); kfree(iprtd); return 0; @@ -271,7 +258,6 @@ static struct snd_pcm_ops imx_pcm_ops = { .close = snd_imx_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_imx_pcm_hw_params, - .hw_free = snd_imx_pcm_hw_free, .trigger = snd_imx_pcm_trigger, .pointer = snd_imx_pcm_pointer, .mmap = snd_imx_pcm_mmap, -- cgit v1.2.3-18-g5258 From 95a771ca16ad63cfd665bebadbc543857db6fa4e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 22 Feb 2012 10:49:07 +0100 Subject: ASoC: mxs-pcm: Request DMA channel early Request the DMA channel in the PCM open callback instead of the hwparams callback, this allows us to let open fail if no dma channel is available. This also fixes a bug where the channel will be requested multiple times if hwparams is called multiple times. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 28 ++++++++-------------------- 1 file changed, 8 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 06c18ecffbb..5b8c8d31406 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -85,8 +85,7 @@ static bool filter(struct dma_chan *chan, void *param) return true; } -static int mxs_dma_alloc(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int mxs_dma_alloc(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_pcm_runtime *runtime = substream->runtime; @@ -112,11 +111,7 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, struct mxs_pcm_runtime_data *iprtd = runtime->private_data; unsigned long dma_addr; struct dma_chan *chan; - int ret; - ret = mxs_dma_alloc(substream, params); - if (ret) - return ret; chan = iprtd->dma_chan; iprtd->periods = params_periods(params); @@ -143,19 +138,6 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int snd_mxs_pcm_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mxs_pcm_runtime_data *iprtd = runtime->private_data; - - if (iprtd->dma_chan) { - dma_release_channel(iprtd->dma_chan); - iprtd->dma_chan = NULL; - } - - return 0; -} - static int snd_mxs_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -208,6 +190,12 @@ static int snd_mxs_open(struct snd_pcm_substream *substream) return ret; } + ret = mxs_dma_alloc(substream); + if (ret) { + kfree(iprtd); + return ret; + } + snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); return 0; @@ -218,6 +206,7 @@ static int snd_mxs_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + dma_release_channel(iprtd->dma_chan); kfree(iprtd); return 0; @@ -239,7 +228,6 @@ static struct snd_pcm_ops mxs_pcm_ops = { .close = snd_mxs_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_mxs_pcm_hw_params, - .hw_free = snd_mxs_pcm_hw_free, .trigger = snd_mxs_pcm_trigger, .pointer = snd_mxs_pcm_pointer, .mmap = snd_mxs_pcm_mmap, -- cgit v1.2.3-18-g5258 From c596758f57aa33e5e89c006867ae66fa2f9c357c Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Wed, 22 Feb 2012 15:20:45 +0100 Subject: ALSA: snd-usb-6fire: remove driver version information Remove unused driver version information from the individual files. Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/chip.c | 3 +-- sound/usb/6fire/chip.h | 1 - sound/usb/6fire/comm.c | 1 - sound/usb/6fire/comm.h | 1 - sound/usb/6fire/common.h | 1 - sound/usb/6fire/control.c | 1 - sound/usb/6fire/control.h | 1 - sound/usb/6fire/firmware.c | 1 - sound/usb/6fire/midi.c | 1 - sound/usb/6fire/midi.h | 1 - sound/usb/6fire/pcm.c | 1 - sound/usb/6fire/pcm.h | 1 - 12 files changed, 1 insertion(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c index 8af92e3e9c1..fc8cc823e43 100644 --- a/sound/usb/6fire/chip.c +++ b/sound/usb/6fire/chip.c @@ -5,7 +5,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify @@ -29,7 +28,7 @@ #include MODULE_AUTHOR("Torsten Schenk "); -MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.0"); +MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver"); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); diff --git a/sound/usb/6fire/chip.h b/sound/usb/6fire/chip.h index d11e5cb520f..bde02d105a5 100644 --- a/sound/usb/6fire/chip.h +++ b/sound/usb/6fire/chip.h @@ -3,7 +3,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c index c994daa57af..6c3d531a250 100644 --- a/sound/usb/6fire/comm.c +++ b/sound/usb/6fire/comm.c @@ -5,7 +5,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h index edc5dc84b88..d2af0a5ddcf 100644 --- a/sound/usb/6fire/comm.h +++ b/sound/usb/6fire/comm.h @@ -3,7 +3,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/common.h b/sound/usb/6fire/common.h index 7dbeb4a3783..b6eb03ed1c2 100644 --- a/sound/usb/6fire/common.h +++ b/sound/usb/6fire/common.h @@ -3,7 +3,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c index ac828eff1a6..8111844c4b1 100644 --- a/sound/usb/6fire/control.c +++ b/sound/usb/6fire/control.c @@ -5,7 +5,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h index 8f5aeead2e3..0dcb1d2f522 100644 --- a/sound/usb/6fire/control.h +++ b/sound/usb/6fire/control.h @@ -3,7 +3,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 3b5f517a397..6f9715ab32f 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -5,7 +5,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c index 13f4509dce2..f0e5179b242 100644 --- a/sound/usb/6fire/midi.c +++ b/sound/usb/6fire/midi.c @@ -5,7 +5,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h index 97a7bf66913..5114eccc1d8 100644 --- a/sound/usb/6fire/midi.h +++ b/sound/usb/6fire/midi.h @@ -3,7 +3,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index d144cdb2f15..c97d05f0e96 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -5,7 +5,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff --git a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h index 2bee8137400..3104301b257 100644 --- a/sound/usb/6fire/pcm.h +++ b/sound/usb/6fire/pcm.h @@ -3,7 +3,6 @@ * * Author: Torsten Schenk * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify -- cgit v1.2.3-18-g5258 From 8e247a9c90e65b25b5b064e2159d9c4c2c173a5e Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Wed, 22 Feb 2012 15:20:54 +0100 Subject: ALSA: snd-usb-6fire: add tlv to controls Remove the soft log-conversion and add a dB scale according to the DAC documentation instead. Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/control.c | 34 +++++++++------------------------- 1 file changed, 9 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c index 8111844c4b1..b00b8bb88c6 100644 --- a/sound/usb/6fire/control.c +++ b/sound/usb/6fire/control.c @@ -15,6 +15,7 @@ #include #include +#include #include "control.h" #include "comm.h" @@ -23,26 +24,6 @@ static char *opt_coax_texts[2] = { "Optical", "Coax" }; static char *line_phono_texts[2] = { "Line", "Phono" }; -/* - * calculated with $value\[i\] = 128 \cdot sqrt[3]{\frac{i}{128}}$ - * this is done because the linear values cause rapid degredation - * of volume in the uppermost region. - */ -static const u8 log_volume_table[128] = { - 0x00, 0x19, 0x20, 0x24, 0x28, 0x2b, 0x2e, 0x30, 0x32, 0x34, - 0x36, 0x38, 0x3a, 0x3b, 0x3d, 0x3e, 0x40, 0x41, 0x42, 0x43, - 0x44, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4c, 0x4d, 0x4e, - 0x4e, 0x4f, 0x50, 0x51, 0x52, 0x53, 0x53, 0x54, 0x55, 0x56, - 0x56, 0x57, 0x58, 0x58, 0x59, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c, - 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x62, 0x62, - 0x63, 0x63, 0x64, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x68, - 0x68, 0x69, 0x69, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c, - 0x6d, 0x6d, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71, - 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75, - 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79, - 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c, - 0x7d, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f }; - /* * data that needs to be sent to device. sets up card internal stuff. * values dumped from windows driver and filtered by trial'n'error. @@ -69,6 +50,8 @@ static const int rates_altsetting[] = { 1, 1, 2, 2, 3, 3 }; static const u16 rates_6fire_vl[] = {0x00, 0x01, 0x00, 0x01, 0x00, 0x01}; static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00}; +static DECLARE_TLV_DB_MINMAX(tlv_output, -9000, 0); + enum { DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; @@ -78,8 +61,7 @@ static void usb6fire_control_master_vol_update(struct control_runtime *rt) struct comm_runtime *comm_rt = rt->chip->comm; if (comm_rt) { /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 0x7f - - log_volume_table[rt->master_vol]); + comm_rt->write8(comm_rt, 0x12, 0x0f, 180 - rt->master_vol); /* unmute */ comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); } @@ -170,7 +152,7 @@ static int usb6fire_control_master_vol_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 127; + uinfo->value.integer.max = 180; return 0; } @@ -291,10 +273,12 @@ static struct __devinitdata snd_kcontrol_new elements[] = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Volume", .index = 0, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = usb6fire_control_master_vol_info, .get = usb6fire_control_master_vol_get, - .put = usb6fire_control_master_vol_put + .put = usb6fire_control_master_vol_put, + .tlv = { .p = tlv_output } }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, -- cgit v1.2.3-18-g5258 From f90ffbf3c68a69714b4273b203d4deb5ae81d8d6 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Wed, 22 Feb 2012 15:21:12 +0100 Subject: ALSA: snd-usb-6fire: add individual volume control for analog channels Add a stereo volume control for every analog output pair 1/2, 3/4, 5/6. Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/control.c | 146 +++++++++++++++++++++++++++++++++++++++------- sound/usb/6fire/control.h | 3 +- 2 files changed, 126 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c index b00b8bb88c6..c22cc29e33d 100644 --- a/sound/usb/6fire/control.c +++ b/sound/usb/6fire/control.c @@ -7,6 +7,10 @@ * Created: Jan 01, 2011 * Copyright: (C) Torsten Schenk * + * Thanks to: + * - Holger Ruckdeschel: he found out how to control individual channel + * volumes and introduced mute switch + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -39,7 +43,7 @@ init_data[] = { { 0x22, 0x03, 0x00 }, { 0x20, 0x03, 0x08 }, { 0x22, 0x04, 0x00 }, { 0x20, 0x04, 0x08 }, { 0x22, 0x05, 0x01 }, { 0x20, 0x05, 0x08 }, { 0x22, 0x04, 0x01 }, { 0x12, 0x04, 0x00 }, { 0x12, 0x05, 0x00 }, - { 0x12, 0x0d, 0x78 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, + { 0x12, 0x0d, 0x38 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, { 0x12, 0x23, 0x00 }, { 0x12, 0x06, 0x02 }, { 0x12, 0x03, 0x00 }, { 0x12, 0x02, 0x00 }, { 0x22, 0x03, 0x01 }, { 0 } /* TERMINATING ENTRY */ @@ -56,15 +60,18 @@ enum { DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; -static void usb6fire_control_master_vol_update(struct control_runtime *rt) +static void usb6fire_control_output_vol_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; - if (comm_rt) { - /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 180 - rt->master_vol); - /* unmute */ - comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); - } + int i; + + if (comm_rt) + for (i = 0; i < 6; i++) + if (!(rt->ovol_updated & (1 << i))) { + comm_rt->write8(comm_rt, 0x12, 0x0f + i, + 180 - rt->output_vol[i]); + rt->ovol_updated |= 1 << i; + } } static void usb6fire_control_line_phono_update(struct control_runtime *rt) @@ -146,34 +153,58 @@ static int usb6fire_control_streaming_update(struct control_runtime *rt) return -EINVAL; } -static int usb6fire_control_master_vol_info(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = 2; uinfo->value.integer.min = 0; uinfo->value.integer.max = 180; return 0; } -static int usb6fire_control_master_vol_put(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned int ch = kcontrol->private_value; int changed = 0; - if (rt->master_vol != ucontrol->value.integer.value[0]) { - rt->master_vol = ucontrol->value.integer.value[0]; - usb6fire_control_master_vol_update(rt); + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + if (rt->output_vol[ch] != ucontrol->value.integer.value[0]) { + rt->output_vol[ch] = ucontrol->value.integer.value[0]; + rt->ovol_updated &= ~(1 << ch); changed = 1; } + if (rt->output_vol[ch + 1] != ucontrol->value.integer.value[1]) { + rt->output_vol[ch + 1] = ucontrol->value.integer.value[1]; + rt->ovol_updated &= ~(2 << ch); + changed = 1; + } + + if (changed) + usb6fire_control_output_vol_update(rt); + return changed; } -static int usb6fire_control_master_vol_get(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = rt->master_vol; + unsigned int ch = kcontrol->private_value; + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + ucontrol->value.integer.value[0] = rt->output_vol[ch]; + ucontrol->value.integer.value[1] = rt->output_vol[ch + 1]; return 0; } @@ -268,18 +299,47 @@ static int usb6fire_control_digital_thru_get(struct snd_kcontrol *kcontrol, return 0; } -static struct __devinitdata snd_kcontrol_new elements[] = { +static struct __devinitdata snd_kcontrol_new vol_elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Volume", + .name = "Analog Playback Volume", .index = 0, + .private_value = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .tlv = { .p = tlv_output } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Volume", + .index = 1, + .private_value = 2, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .tlv = { .p = tlv_output } + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Volume", + .index = 2, + .private_value = 4, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = usb6fire_control_master_vol_info, - .get = usb6fire_control_master_vol_get, - .put = usb6fire_control_master_vol_put, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, .tlv = { .p = tlv_output } }, + {} +}; + +static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line/Phono Capture Route", @@ -310,6 +370,40 @@ static struct __devinitdata snd_kcontrol_new elements[] = { {} }; +static int usb6fire_control_add_virtual( + struct control_runtime *rt, + struct snd_card *card, + char *name, + struct snd_kcontrol_new *elems) +{ + int ret; + int i; + struct snd_kcontrol *vmaster = + snd_ctl_make_virtual_master(name, tlv_output); + struct snd_kcontrol *control; + + if (!vmaster) + return -ENOMEM; + ret = snd_ctl_add(card, vmaster); + if (ret < 0) + return ret; + + i = 0; + while (elems[i].name) { + control = snd_ctl_new1(&elems[i], rt); + if (!control) + return -ENOMEM; + ret = snd_ctl_add(card, control); + if (ret < 0) + return ret; + ret = snd_ctl_add_slave(vmaster, control); + if (ret < 0) + return ret; + i++; + } + return 0; +} + int __devinit usb6fire_control_init(struct sfire_chip *chip) { int i; @@ -335,9 +429,17 @@ int __devinit usb6fire_control_init(struct sfire_chip *chip) usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); - usb6fire_control_master_vol_update(rt); + usb6fire_control_output_vol_update(rt); usb6fire_control_streaming_update(rt); + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Volume", vol_elements); + if (ret) { + kfree(rt); + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + return ret; + } + i = 0; while (elements[i].name) { ret = snd_ctl_add(chip->card, snd_ctl_new1(&elements[i], rt)); diff --git a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h index 0dcb1d2f522..ce024113c98 100644 --- a/sound/usb/6fire/control.h +++ b/sound/usb/6fire/control.h @@ -43,7 +43,8 @@ struct control_runtime { bool line_phono_switch; bool digital_thru_switch; bool usb_streaming; - u8 master_vol; + u8 output_vol[6]; + u8 ovol_updated; }; int __devinit usb6fire_control_init(struct sfire_chip *chip); -- cgit v1.2.3-18-g5258 From d97c735a1047fa06165e55da32154cf0e6b9419c Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Wed, 22 Feb 2012 15:21:23 +0100 Subject: ALSA: snd-usb-6fire: add mute control for analog outputs Add a mute control for every analog output channel. Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/control.c | 95 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/6fire/control.h | 1 + 2 files changed, 96 insertions(+) (limited to 'sound') diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c index c22cc29e33d..a2bbf48c641 100644 --- a/sound/usb/6fire/control.c +++ b/sound/usb/6fire/control.c @@ -74,6 +74,14 @@ static void usb6fire_control_output_vol_update(struct control_runtime *rt) } } +static void usb6fire_control_output_mute_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + + if (comm_rt) + comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -208,6 +216,51 @@ static int usb6fire_control_output_vol_get(struct snd_kcontrol *kcontrol, return 0; } +static int usb6fire_control_output_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned int ch = kcontrol->private_value; + u8 old = rt->output_mute; + u8 value = 0; + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + rt->output_mute &= ~(3 << ch); + if (ucontrol->value.integer.value[0]) + value |= 1; + if (ucontrol->value.integer.value[1]) + value |= 2; + rt->output_mute |= value << ch; + + if (rt->output_mute != old) + usb6fire_control_output_mute_update(rt); + + return rt->output_mute != old; +} + +static int usb6fire_control_output_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned int ch = kcontrol->private_value; + u8 value = rt->output_mute >> ch; + + if (ch > 4) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return -EINVAL; + } + + ucontrol->value.integer.value[0] = 1 & value; + value >>= 1; + ucontrol->value.integer.value[1] = 1 & value; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -339,6 +392,40 @@ static struct __devinitdata snd_kcontrol_new vol_elements[] = { {} }; +static struct __devinitdata snd_kcontrol_new mute_elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Switch", + .index = 0, + .private_value = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Switch", + .index = 1, + .private_value = 2, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Playback Switch", + .index = 2, + .private_value = 4, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + }, + {} +}; + static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -430,13 +517,21 @@ int __devinit usb6fire_control_init(struct sfire_chip *chip) usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); + usb6fire_control_output_mute_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, "Master Playback Volume", vol_elements); if (ret) { + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); kfree(rt); + return ret; + } + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Switch", mute_elements); + if (ret) { snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + kfree(rt); return ret; } diff --git a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h index ce024113c98..9f9eb647bc6 100644 --- a/sound/usb/6fire/control.h +++ b/sound/usb/6fire/control.h @@ -45,6 +45,7 @@ struct control_runtime { bool usb_streaming; u8 output_vol[6]; u8 ovol_updated; + u8 output_mute; }; int __devinit usb6fire_control_init(struct sfire_chip *chip); -- cgit v1.2.3-18-g5258 From 06bb4e7435019ff9b6dbc9b1d02d8babb36d8177 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Wed, 22 Feb 2012 15:21:30 +0100 Subject: ALSA: snd-usb-6fire: add analog input volume control Add a stereo volume control for analog input channel pair 1/2. Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/control.c | 71 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/6fire/control.h | 2 ++ 2 files changed, 73 insertions(+) (limited to 'sound') diff --git a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c index a2bbf48c641..07ed914d5e7 100644 --- a/sound/usb/6fire/control.c +++ b/sound/usb/6fire/control.c @@ -55,6 +55,7 @@ static const u16 rates_6fire_vl[] = {0x00, 0x01, 0x00, 0x01, 0x00, 0x01}; static const u16 rates_6fire_vh[] = {0x11, 0x11, 0x10, 0x10, 0x00, 0x00}; static DECLARE_TLV_DB_MINMAX(tlv_output, -9000, 0); +static DECLARE_TLV_DB_MINMAX(tlv_input, -1500, 1500); enum { DIGITAL_THRU_ONLY_SAMPLERATE = 3 @@ -82,6 +83,20 @@ static void usb6fire_control_output_mute_update(struct control_runtime *rt) comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); } +static void usb6fire_control_input_vol_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + int i; + + if (comm_rt) + for (i = 0; i < 2; i++) + if (!(rt->ivol_updated & (1 << i))) { + comm_rt->write8(comm_rt, 0x12, 0x1c + i, + rt->input_vol[i] & 0x3f); + rt->ivol_updated |= 1 << i; + } +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -261,6 +276,50 @@ static int usb6fire_control_output_mute_get(struct snd_kcontrol *kcontrol, return 0; } +static int usb6fire_control_input_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 30; + return 0; +} + +static int usb6fire_control_input_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + int changed = 0; + + if (rt->input_vol[0] != ucontrol->value.integer.value[0]) { + rt->input_vol[0] = ucontrol->value.integer.value[0] - 15; + rt->ivol_updated &= ~(1 << 0); + changed = 1; + } + if (rt->input_vol[1] != ucontrol->value.integer.value[1]) { + rt->input_vol[1] = ucontrol->value.integer.value[1] - 15; + rt->ivol_updated &= ~(1 << 1); + changed = 1; + } + + if (changed) + usb6fire_control_input_vol_update(rt); + + return changed; +} + +static int usb6fire_control_input_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = rt->input_vol[0] + 15; + ucontrol->value.integer.value[1] = rt->input_vol[1] + 15; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -454,6 +513,17 @@ static struct __devinitdata snd_kcontrol_new elements[] = { .get = usb6fire_control_digital_thru_get, .put = usb6fire_control_digital_thru_put }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Capture Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .info = usb6fire_control_input_vol_info, + .get = usb6fire_control_input_vol_get, + .put = usb6fire_control_input_vol_put, + .tlv = { .p = tlv_input } + }, {} }; @@ -518,6 +588,7 @@ int __devinit usb6fire_control_init(struct sfire_chip *chip) usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); usb6fire_control_output_mute_update(rt); + usb6fire_control_input_vol_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, diff --git a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h index 9f9eb647bc6..9a596d95474 100644 --- a/sound/usb/6fire/control.h +++ b/sound/usb/6fire/control.h @@ -46,6 +46,8 @@ struct control_runtime { u8 output_vol[6]; u8 ovol_updated; u8 output_mute; + s8 input_vol[2]; + u8 ivol_updated; }; int __devinit usb6fire_control_init(struct sfire_chip *chip); -- cgit v1.2.3-18-g5258 From 6edc59e602b36cd3c95a426ef6e8cad0344af8c7 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Thu, 23 Feb 2012 15:07:44 +0800 Subject: ALSA: hda - add id for Atom Cedar Trail HDMI codec [the order sorted by tiwai] Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1168ebd3fb5..540cd13f7f1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1912,6 +1912,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi }, { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi }, +{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi }, { .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi }, {} /* terminator */ }; @@ -1958,6 +1959,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862803"); MODULE_ALIAS("snd-hda-codec-id:80862804"); MODULE_ALIAS("snd-hda-codec-id:80862805"); MODULE_ALIAS("snd-hda-codec-id:80862806"); +MODULE_ALIAS("snd-hda-codec-id:80862880"); MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_LICENSE("GPL"); -- cgit v1.2.3-18-g5258 From afad95f82582cdd685cc9ba919eb55150a0ec909 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 Feb 2012 12:04:41 -0800 Subject: ASoC: ak4104: Use snd_soc_update_bits() for read/modify/write Don't use the internal I/O functions directly. Signed-off-by: Mark Brown Acked-by: Daniel Mack --- sound/soc/codecs/ak4104.c | 38 +++++++++++++++----------------------- 1 file changed, 15 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index f12c1154849..d6d9e40cbde 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -110,12 +110,6 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; int val = 0; - val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); - if (val < 0) - return val; - - val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1); - /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: @@ -135,7 +129,13 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) return -EINVAL; - return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); + ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, + AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, + val); + if (ret < 0) + return ret; + + return 0; } static int ak4104_hw_params(struct snd_pcm_substream *substream, @@ -211,16 +211,15 @@ static int ak4104_probe(struct snd_soc_codec *codec) return -ENODEV; /* set power-up and non-reset bits */ - val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); - val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN; - ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); + ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); if (ret < 0) return ret; /* enable transmitter */ - val = ak4104_read_reg_cache(codec, AK4104_REG_TX); - val |= AK4104_TX_TXE; - ret = ak4104_spi_write(codec, AK4104_REG_TX, val); + ret = snd_soc_update_bits(codec, AK4104_REG_TX, + AK4104_TX_TXE, AK4104_TX_TXE); if (ret < 0) return ret; @@ -229,17 +228,10 @@ static int ak4104_probe(struct snd_soc_codec *codec) static int ak4104_remove(struct snd_soc_codec *codec) { - int val, ret; - - val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1); - if (val < 0) - return val; - - /* clear power-up and non-reset bits */ - val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); - ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val); + snd_soc_update_bits(codec, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); - return ret; + return 0; } static struct snd_soc_codec_driver soc_codec_device_ak4104 = { -- cgit v1.2.3-18-g5258 From 34baf2202059a7a7624aa02a104b64a9134aa885 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 Feb 2012 12:05:51 -0800 Subject: ASoC: ak4104: Use snd_soc_write() rather than internal write function Signed-off-by: Mark Brown Acked-by: Daniel Mack --- sound/soc/codecs/ak4104.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index d6d9e40cbde..34a840c5d2b 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -148,7 +148,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, /* set the IEC958 bits: consumer mode, no copyright bit */ val |= IEC958_AES0_CON_NOT_COPYRIGHT; - ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val); + snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val); val = 0; @@ -167,7 +167,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val); + return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val); } static const struct snd_soc_dai_ops ak4101_dai_ops = { -- cgit v1.2.3-18-g5258 From 2901d6ebe1f9da781a5bbb64d7c2a46c4b8e7364 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 17 Feb 2012 12:14:18 -0800 Subject: ASoC: ak4104: Convert to direct regmap API usage Since the cache is currently open coded this is more of a win than for most devices. Signed-off-by: Mark Brown Acked-by: Daniel Mack --- sound/soc/codecs/ak4104.c | 119 +++++++++++++++++----------------------------- 1 file changed, 44 insertions(+), 75 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 34a840c5d2b..ceb96ecf558 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -46,69 +46,15 @@ #define DRV_NAME "ak4104-codec" struct ak4104_private { - enum snd_soc_control_type control_type; - void *control_data; + struct regmap *regmap; }; -static int ak4104_fill_cache(struct snd_soc_codec *codec) -{ - int i; - u8 *reg_cache = codec->reg_cache; - struct spi_device *spi = codec->control_data; - - for (i = 0; i < codec->driver->reg_cache_size; i++) { - int ret = spi_w8r8(spi, i | AK4104_READ); - if (ret < 0) { - dev_err(&spi->dev, "SPI write failure\n"); - return ret; - } - - reg_cache[i] = ret; - } - - return 0; -} - -static unsigned int ak4104_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u8 *reg_cache = codec->reg_cache; - - if (reg >= codec->driver->reg_cache_size) - return -EINVAL; - - return reg_cache[reg]; -} - -static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 *cache = codec->reg_cache; - struct spi_device *spi = codec->control_data; - - if (reg >= codec->driver->reg_cache_size) - return -EINVAL; - - /* only write to the hardware if value has changed */ - if (cache[reg] != value) { - u8 tmp[2] = { (reg & AK4104_REG_MASK) | AK4104_WRITE, value }; - - if (spi_write(spi, tmp, sizeof(tmp))) { - dev_err(&spi->dev, "SPI write failed\n"); - return -EIO; - } - - cache[reg] = value; - } - - return 0; -} - static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; int val = 0; + int ret; /* set DAI format */ switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -192,23 +138,12 @@ static struct snd_soc_dai_driver ak4104_dai = { static int ak4104_probe(struct snd_soc_codec *codec) { struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); - int ret, val; - - codec->control_data = ak4104->control_data; + int ret; - /* read all regs and fill the cache */ - ret = ak4104_fill_cache(codec); - if (ret < 0) { - dev_err(codec->dev, "failed to fill register cache\n"); + codec->control_data = ak4104->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) return ret; - } - - /* read the 'reserved' register - according to the datasheet, it - * should contain 0x5b. Not a good way to verify the presence of - * the device, but there is no hardware ID register. */ - if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) != - AK4104_RESERVED_VAL) - return -ENODEV; /* set power-up and non-reset bits */ ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, @@ -237,13 +172,23 @@ static int ak4104_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, - .reg_cache_size = AK4104_NUM_REGS, - .reg_word_size = sizeof(u8), +}; + +static const struct regmap_config ak4104_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK4104_NUM_REGS - 1, + .read_flag_mask = AK4104_READ, + .write_flag_mask = AK4104_WRITE, + + .cache_type = REGCACHE_RBTREE, }; static int ak4104_spi_probe(struct spi_device *spi) { struct ak4104_private *ak4104; + unsigned int val; int ret; spi->bits_per_word = 8; @@ -257,17 +202,41 @@ static int ak4104_spi_probe(struct spi_device *spi) if (ak4104 == NULL) return -ENOMEM; - ak4104->control_data = spi; - ak4104->control_type = SND_SOC_SPI; + ak4104->regmap = regmap_init_spi(spi, &ak4104_regmap); + if (IS_ERR(ak4104->regmap)) { + ret = PTR_ERR(ak4104->regmap); + return ret; + } + + /* read the 'reserved' register - according to the datasheet, it + * should contain 0x5b. Not a good way to verify the presence of + * the device, but there is no hardware ID register. */ + ret = regmap_read(ak4104->regmap, AK4104_REG_RESERVED, &val); + if (ret != 0) + goto err; + if (val != AK4104_RESERVED_VAL) { + ret = -ENODEV; + goto err; + } + spi_set_drvdata(spi, ak4104); ret = snd_soc_register_codec(&spi->dev, &soc_codec_device_ak4104, &ak4104_dai, 1); + if (ret != 0) + goto err; + + return 0; + +err: + regmap_exit(ak4104->regmap); return ret; } static int __devexit ak4104_spi_remove(struct spi_device *spi) { + struct ak4104_private *ak4101 = spi_get_drvdata(spi); + regmap_exit(ak4101->regmap); snd_soc_unregister_codec(&spi->dev); return 0; } -- cgit v1.2.3-18-g5258 From d690516c6dcef11b87c35ce294ae1fcf9797276f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Feb 2012 20:23:01 +0000 Subject: ASoC: wm8962: Remove mistakenly committed debug logging Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 2dd710f58b8..dc5f19e1065 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3720,8 +3720,6 @@ static int wm8962_runtime_resume(struct device *dev) regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK, 0x100); - dev_crit(dev, "RESUME\n"); - return 0; } @@ -3729,8 +3727,6 @@ static int wm8962_runtime_suspend(struct device *dev) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); - dev_crit(dev, "SUSPEND\n"); - regmap_update_bits(wm8962->regmap, WM8962_PWR_MGMT_1, WM8962_VMID_SEL_MASK | WM8962_BIAS_ENA, 0); -- cgit v1.2.3-18-g5258 From 8a236f3f1a0e65de526c5e169eb8d7a758ffde9e Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Thu, 23 Feb 2012 23:23:06 +0900 Subject: ALSA: ctxfi: Fix typo in ctvmem.c Correct spelling "virtural" to "virtual" in sound/pci/ctxfi/ctvmem.c Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- sound/pci/ctxfi/ctvmem.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index b78f3fc3c33..6109490b83e 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -36,7 +36,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size) size = CT_PAGE_ALIGN(size); if (size > vm->size) { - printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural " + printk(KERN_ERR "ctxfi: Fail! No sufficient device virtual " "memory space available!\n"); return NULL; } -- cgit v1.2.3-18-g5258 From 0512615db6dbf5ab7d5b6f46ebbe657707bb9dad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Feb 2012 21:49:37 +0000 Subject: ASoC: wm8962: Convert interrupt handler to direct regmap usage Avoids potential locking issues with anything that needs the CODEC lock. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 59 +++++++++++++++++++++++++++++++---------------- 1 file changed, 39 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index dc5f19e1065..6af794510c6 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2969,54 +2969,73 @@ static void wm8962_mic_work(struct work_struct *work) static irqreturn_t wm8962_irq(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - int mask; - int active; - int reg; + struct device *dev = data; + struct wm8962_priv *wm8962 = dev_get_drvdata(dev); + unsigned int mask; + unsigned int active; + int reg, ret; + + ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2_MASK, + &mask); + if (ret != 0) { + dev_err(dev, "Failed to read interrupt mask: %d\n", + ret); + return IRQ_NONE; + } - mask = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2_MASK); + ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, &active); + if (ret != 0) { + dev_err(dev, "Failed to read interrupt: %d\n", ret); + return IRQ_NONE; + } - active = snd_soc_read(codec, WM8962_INTERRUPT_STATUS_2); active &= ~mask; if (!active) return IRQ_NONE; /* Acknowledge the interrupts */ - snd_soc_write(codec, WM8962_INTERRUPT_STATUS_2, active); + ret = regmap_write(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, active); + if (ret != 0) + dev_warn(dev, "Failed to ack interrupt: %d\n", ret); if (active & WM8962_FLL_LOCK_EINT) { - dev_dbg(codec->dev, "FLL locked\n"); + dev_dbg(dev, "FLL locked\n"); complete(&wm8962->fll_lock); } if (active & WM8962_FIFOS_ERR_EINT) - dev_err(codec->dev, "FIFO error\n"); + dev_err(dev, "FIFO error\n"); if (active & WM8962_TEMP_SHUT_EINT) { - dev_crit(codec->dev, "Thermal shutdown\n"); + dev_crit(dev, "Thermal shutdown\n"); - reg = snd_soc_read(codec, WM8962_THERMAL_SHUTDOWN_STATUS); + ret = regmap_read(wm8962->regmap, + WM8962_THERMAL_SHUTDOWN_STATUS, ®); + if (ret != 0) { + dev_warn(dev, "Failed to read thermal status: %d\n", + ret); + reg = 0; + } if (reg & WM8962_TEMP_ERR_HP) - dev_crit(codec->dev, "Headphone thermal error\n"); + dev_crit(dev, "Headphone thermal error\n"); if (reg & WM8962_TEMP_WARN_HP) - dev_crit(codec->dev, "Headphone thermal warning\n"); + dev_crit(dev, "Headphone thermal warning\n"); if (reg & WM8962_TEMP_ERR_SPK) - dev_crit(codec->dev, "Speaker thermal error\n"); + dev_crit(dev, "Speaker thermal error\n"); if (reg & WM8962_TEMP_WARN_SPK) - dev_crit(codec->dev, "Speaker thermal warning\n"); + dev_crit(dev, "Speaker thermal warning\n"); } if (active & (WM8962_MICSCD_EINT | WM8962_MICD_EINT)) { - dev_dbg(codec->dev, "Microphone event detected\n"); + dev_dbg(dev, "Microphone event detected\n"); #ifndef CONFIG_SND_SOC_WM8962_MODULE - trace_snd_soc_jack_irq(dev_name(codec->dev)); + trace_snd_soc_jack_irq(dev_name(dev)); #endif - pm_wakeup_event(codec->dev, 300); + pm_wakeup_event(dev, 300); schedule_delayed_work(&wm8962->mic_work, msecs_to_jiffies(250)); @@ -3497,7 +3516,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) ret = request_threaded_irq(wm8962->irq, NULL, wm8962_irq, trigger | IRQF_ONESHOT, - "wm8962", codec); + "wm8962", codec->dev); if (ret != 0) { dev_err(codec->dev, "Failed to request IRQ %d: %d\n", wm8962->irq, ret); -- cgit v1.2.3-18-g5258 From b1dd5897f53647390a7622795541bc89f8a84fe2 Mon Sep 17 00:00:00 2001 From: Viresh Kumar Date: Fri, 24 Feb 2012 16:25:49 +0530 Subject: ASoC: core: Don't overwrite .poweroff in snd_soc_pm_ops SET_SYSTEM_SLEEP_PM_OPS writes .poweroff = *_resume once. Then we overwrite it again explicitly as .poweroff = snd_soc_poweroff. Even though it works, as the second one overwrites the first one, this is not the correct way. Fix this by expanding SET_SYSTEM_SLEEP_PM_OPS in our structure. Signed-off-by: Viresh Kumar Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fc0fd3485e7..1bdc67e0bd1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1718,8 +1718,12 @@ int snd_soc_poweroff(struct device *dev) EXPORT_SYMBOL_GPL(snd_soc_poweroff); const struct dev_pm_ops snd_soc_pm_ops = { - SET_SYSTEM_SLEEP_PM_OPS(snd_soc_suspend, snd_soc_resume) + .suspend = snd_soc_suspend, + .resume = snd_soc_resume, + .freeze = snd_soc_suspend, + .thaw = snd_soc_resume, .poweroff = snd_soc_poweroff, + .restore = snd_soc_resume, }; EXPORT_SYMBOL_GPL(snd_soc_pm_ops); -- cgit v1.2.3-18-g5258 From adef39c0ea2e5deae5c4f2917b23694b68535e45 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Fri, 24 Feb 2012 21:34:22 +0100 Subject: ALSA: snd-usb-6fire: Select missing SND_VMASTER option in Kconfig Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 3efc21c3d67..ff77b28f3da 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -106,6 +106,7 @@ config SND_USB_6FIRE select BITREVERSE select SND_RAWMIDI select SND_PCM + select SND_VMASTER help Say Y here to include support for TerraTec 6fire DMX USB interface. -- cgit v1.2.3-18-g5258 From 1e1d7e593e46f884b359769b29b6d92eb3d0f1ee Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Wed, 22 Feb 2012 16:40:26 +0800 Subject: ASoC: imx: let SND_MXC_SOC_FIQ select FIQ CONFIG_FIQ is only needed when CONFIG_SND_MXC_SOC_FIQ is selected to build imx-pcm-fiq.c, so let SND_MXC_SOC_FIQ select FIQ. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 91b66eff531..e89139ffbce 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC - select FIQ select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to @@ -11,6 +10,7 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC config SND_MXC_SOC_FIQ + select FIQ tristate config SND_MXC_SOC_MX2 -- cgit v1.2.3-18-g5258 From 07a38b1b4ee505a70c31eb015f1f031dcdb854ab Mon Sep 17 00:00:00 2001 From: Paul Gortmaker Date: Sat, 25 Feb 2012 16:12:30 -0500 Subject: ASoC: fix trivial build error in mpc5200_dma.c Add the obvious header to fix this: sound/soc/fsl/mpc5200_dma.c:301: error: implicit declaration of function 'DMA_BIT_MASK' sound/soc/fsl/mpc5200_dma.c:301: error: initializer element is not constant Signed-off-by: Paul Gortmaker Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 33adbf1e40d..9a3f7c5ab68 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -8,6 +8,7 @@ #include #include +#include #include #include -- cgit v1.2.3-18-g5258 From 2c823d14bfad16e75d16674a312a779a1485a2bd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 27 Feb 2012 15:24:10 +0000 Subject: ASoC: wm8753: Convert to devm_kzalloc() Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 21ed75de41f..59c28dbc5fc 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1557,7 +1557,8 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) struct wm8753_priv *wm8753; int ret; - wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); + wm8753 = devm_kzalloc(&spi->dev, sizeof(struct wm8753_priv), + GFP_KERNEL); if (wm8753 == NULL) return -ENOMEM; @@ -1580,7 +1581,6 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) err_regmap: regmap_exit(wm8753->regmap); err: - kfree(wm8753); return ret; } @@ -1612,7 +1612,8 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, struct wm8753_priv *wm8753; int ret; - wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); + wm8753 = devm_kzalloc(&i2c->dev, sizeof(struct wm8753_priv), + GFP_KERNEL); if (wm8753 == NULL) return -ENOMEM; @@ -1632,10 +1633,10 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); goto err_regmap; } + err_regmap: regmap_exit(wm8753->regmap); err: - kfree(wm8753); return ret; } @@ -1645,7 +1646,6 @@ static __devexit int wm8753_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(wm8753->regmap); - kfree(wm8753); return 0; } -- cgit v1.2.3-18-g5258 From 53803aead010a314f76a8a6fa132fdcc5edf55ed Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Feb 2012 19:48:47 +0000 Subject: ALSA: Use a define for the number of jack switch types This is intended to facilitate the merge of the two jack detection mechanisms. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/core/jack.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/jack.c b/sound/core/jack.c index 26edf63b265..471e1e3b0a9 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -25,7 +25,7 @@ #include #include -static int jack_switch_types[] = { +static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { SW_HEADPHONE_INSERT, SW_MICROPHONE_INSERT, SW_LINEOUT_INSERT, @@ -128,7 +128,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, jack->type = type; - for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) + for (i = 0; i < SND_JACK_SWITCH_TYPES; i++) if (type & (1 << i)) input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -- cgit v1.2.3-18-g5258 From 5556e147083fb4d473d5c1a82f73205b8b145cd9 Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Mon, 27 Feb 2012 16:47:37 -0600 Subject: ALSA: hda - Fix audio playback support on HP Zephyr system Enables port E of IDT 92HD91 codec as output and sets correct output phase between ports E and D and high pass filter. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 4c769405d72..8c346ac59d4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -99,6 +99,7 @@ enum { STAC_DELL_VOSTRO_3500, STAC_92HD83XXX_HP_cNB11_INTQUAD, STAC_HP_DV7_4000, + STAC_HP_ZEPHYR, STAC_92HD83XXX_MODELS }; @@ -894,6 +895,13 @@ static const struct hda_verb stac92hd83xxx_core_init[] = { {} }; +static const struct hda_verb stac92hd83xxx_hp_zephyr_init[] = { + { 0x22, 0x785, 0x43 }, + { 0x22, 0x782, 0xe0 }, + { 0x22, 0x795, 0x00 }, + {} +}; + static const struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -1621,6 +1629,12 @@ static const unsigned int hp_dv7_4000_pin_configs[10] = { 0x40f000f0, 0x40f000f0, }; +static const unsigned int hp_zephyr_pin_configs[10] = { + 0x01813050, 0x0421201f, 0x04a1205e, 0x96130310, + 0x96130310, 0x0101401f, 0x1111611f, 0xd5a30130, + 0, 0, +}; + static const unsigned int hp_cNB11_intquad_pin_configs[10] = { 0x40f000f0, 0x0221101f, 0x02a11020, 0x92170110, 0x40f000f0, 0x92170110, 0x40f000f0, 0xd5a30130, @@ -1634,6 +1648,7 @@ static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_DELL_VOSTRO_3500] = dell_vostro_3500_pin_configs, [STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs, [STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs, + [STAC_HP_ZEPHYR] = hp_zephyr_pin_configs, }; static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { @@ -1644,6 +1659,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_DELL_VOSTRO_3500] = "dell-vostro-3500", [STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad", [STAC_HP_DV7_4000] = "hp-dv7-4000", + [STAC_HP_ZEPHYR] = "hp-zephyr", }; static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1696,6 +1712,14 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3593, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561, + "HP", STAC_HP_ZEPHYR), + {} /* terminator */ +}; + +static const struct snd_pci_quirk stac92hd83xxx_codec_id_cfg_tbl[] = { + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561, + "HP", STAC_HP_ZEPHYR), {} /* terminator */ }; @@ -5565,6 +5589,12 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) STAC_92HD83XXX_MODELS, stac92hd83xxx_models, stac92hd83xxx_cfg_tbl); + /* check codec subsystem id if not found */ + if (spec->board_config < 0) + spec->board_config = + snd_hda_check_board_codec_sid_config(codec, + STAC_92HD83XXX_MODELS, stac92hd83xxx_models, + stac92hd83xxx_codec_id_cfg_tbl); again: if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", @@ -5575,6 +5605,12 @@ again: codec->patch_ops = stac92xx_patch_ops; + switch (spec->board_config) { + case STAC_HP_ZEPHYR: + spec->init = stac92hd83xxx_hp_zephyr_init; + break; + } + if (find_mute_led_cfg(codec, -1/*no default cfg*/)) snd_printd("mute LED gpio %d polarity %d\n", spec->gpio_led, -- cgit v1.2.3-18-g5258 From a6f2fd557f993aecc93d51afd9e339524107937f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Feb 2012 11:58:40 +0100 Subject: ALSA: hda - Add position_fix=4 (COMBO) option This patch adds a new position_fix option value, 4, as a combo mode to use LPIB for playbacks and POSBUF for captures. It's the way recommended by Intel hardware guys. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e354c161654..6e958bf9419 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -84,7 +84,7 @@ module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(position_fix, int, NULL, 0444); MODULE_PARM_DESC(position_fix, "DMA pointer read method." - "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO)."); + "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO)."); module_param_array(bdl_pos_adj, int, NULL, 0644); MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset."); module_param_array(probe_mask, int, NULL, 0444); @@ -330,6 +330,7 @@ enum { POS_FIX_LPIB, POS_FIX_POSBUF, POS_FIX_VIACOMBO, + POS_FIX_COMBO, }; /* Defines for ATI HD Audio support in SB450 south bridge */ @@ -2520,6 +2521,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix) case POS_FIX_LPIB: case POS_FIX_POSBUF: case POS_FIX_VIACOMBO: + case POS_FIX_COMBO: return fix; } @@ -2699,6 +2701,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->position_fix[0] = chip->position_fix[1] = check_position_fix(chip, position_fix[dev]); + /* combo mode uses LPIB for playback */ + if (chip->position_fix[0] == POS_FIX_COMBO) { + chip->position_fix[0] = POS_FIX_LPIB; + chip->position_fix[1] = POS_FIX_AUTO; + } + check_probe_mask(chip, dev); chip->single_cmd = single_cmd; -- cgit v1.2.3-18-g5258 From fb1a6453bc413841876eac2ed1d7fd9e74b118fc Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 24 Feb 2012 22:09:36 +0800 Subject: ASoC: fsl: correct get_dma_channel parameter name The second parameter of function get_dma_channel is actually a property name rather than a compatible string, so rename it for less confusing. Signed-off-by: Shawn Guo Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 4 ++-- sound/soc/fsl/p1022_ds.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 0ea4a5a96e0..15e06e9d362 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -273,7 +273,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) } static int get_dma_channel(struct device_node *ssi_np, - const char *compatible, + const char *name, struct snd_soc_dai_link *dai, unsigned int *dma_channel_id, unsigned int *dma_id) @@ -283,7 +283,7 @@ static int get_dma_channel(struct device_node *ssi_np, const u32 *iprop; int ret; - dma_channel_np = get_node_by_phandle_name(ssi_np, compatible, + dma_channel_np = get_node_by_phandle_name(ssi_np, name, "fsl,ssi-dma-channel"); if (!dma_channel_np) return -EINVAL; diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index a5d4e80a9cf..d32ec4646d2 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -276,7 +276,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) } static int get_dma_channel(struct device_node *ssi_np, - const char *compatible, + const char *name, struct snd_soc_dai_link *dai, unsigned int *dma_channel_id, unsigned int *dma_id) @@ -286,7 +286,7 @@ static int get_dma_channel(struct device_node *ssi_np, const u32 *iprop; int ret; - dma_channel_np = get_node_by_phandle_name(ssi_np, compatible, + dma_channel_np = get_node_by_phandle_name(ssi_np, name, "fsl,ssi-dma-channel"); if (!dma_channel_np) return -EINVAL; -- cgit v1.2.3-18-g5258 From 7c59bc55e476947c1ea8a019e97aa648de53f5a0 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 24 Feb 2012 22:09:37 +0800 Subject: ASoC: fsl: align mpc8610_hpcd with p1022_ds on getting codec node Align mpc8610_hpcd with p1022_ds on getting codec node by just calling of_parse_phandle. The bonus point of doing that is we can save exporting get_node_by_phandle_name() when we consolidate the common bits between mpc8610_hpcd and p1022_ds into a module, which can be shared by more machine drivers added later. Signed-off-by: Shawn Guo Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/mpc8610_hpcd.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 15e06e9d362..93256b39ccb 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -336,12 +336,8 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) const char *sprop; const u32 *iprop; - /* We are only interested in SSIs with a codec phandle in them, - * so let's make sure this SSI has one. The MPC8610 HPCD only - * knows about the CS4270 codec, so reject anything else. - */ - codec_np = get_node_by_phandle_name(np, "codec-handle", - "cirrus,cs4270"); + /* Find the codec node for this SSI. */ + codec_np = of_parse_phandle(np, "codec-handle", 0); if (!codec_np) { dev_err(dev, "invalid codec node\n"); return -EINVAL; -- cgit v1.2.3-18-g5258 From 64902b29cbdd04e030621f556d44702272faa96d Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 24 Feb 2012 22:09:38 +0800 Subject: ASoC: Remove unnecessary -codec from cs4270 driver name Similar to what commit 1e3ad57 (ASoC: Remove redundant -codec from WM8776 driver name) does for wm8776 driver, this patch does the same thing for cs4270 driver. Signed-off-by: Shawn Guo Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 ++-- sound/soc/pxa/raumfeld.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 6baccd285df..1d672f52866 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -715,7 +715,7 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); */ static struct i2c_driver cs4270_i2c_driver = { .driver = { - .name = "cs4270-codec", + .name = "cs4270", .owner = THIS_MODULE, }, .id_table = cs4270_id, diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 93256b39ccb..fcf9302f59b 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -245,7 +245,7 @@ static int get_parent_cell_index(struct device_node *np) * 'struct device' It's ugly and hackish, but it works. * * The dev_name for such devices include the bus number and I2C address. For - * example, "cs4270-codec.0-004f". + * example, "cs4270.0-004f". */ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) { @@ -267,7 +267,7 @@ static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) if (!i2c) return -ENODEV; - snprintf(buf, len, "%s-codec.%u-%04x", temp, i2c->adapter->nr, addr); + snprintf(buf, len, "%s.%u-%04x", temp, i2c->adapter->nr, addr); return 0; } diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index ba1545188ec..08370659549 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -232,7 +232,7 @@ static struct snd_soc_ops raumfeld_ak4104_ops = { .cpu_dai_name = "pxa-ssp-dai.0", \ .platform_name = "pxa-pcm-audio", \ .codec_dai_name = "cs4270-hifi", \ - .codec_name = "cs4270-codec.0-0048", \ + .codec_name = "cs4270.0-0048", \ .ops = &raumfeld_cs4270_ops, \ } -- cgit v1.2.3-18-g5258 From 3e93f5efaf9cd48bae97ae6436cbc5f91be8003c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 28 Feb 2012 21:49:55 +0100 Subject: ALSA: hda - Enable docking-station SPDIF for Thinkpad The docking-station of Thinkpad X200 & co supports also an SPDIF output, and the corresponding pin 0x1c has to be enabled for using it. Reported-and-tested-by: Sebastian Glita Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 266e5a68baf..6bbdbb6dd4e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4355,6 +4355,7 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ { 0x19, 0x2121103f }, /* dock-HP */ + { 0x1c, 0x21440100 }, /* dock SPDIF out */ {} }; -- cgit v1.2.3-18-g5258 From b9e67e5ef3c14453e10f41468b6e601d37291a82 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 28 Feb 2012 19:03:37 +0000 Subject: ASoC: wm8994: Make sure we don't have MICBIAS on during jackdet mode Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 33bc718f164..691e89753a6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3170,9 +3170,6 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); - wm1811_jackdet_set_mode(codec, - WM1811_JACKDET_MODE_JACK); - if (wm8994->pdata->jd_ext_cap) { mutex_lock(&codec->mutex); snd_soc_dapm_disable_pin(&codec->dapm, @@ -3180,6 +3177,9 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_dapm_sync(&codec->dapm); mutex_unlock(&codec->mutex); } + + wm1811_jackdet_set_mode(codec, + WM1811_JACKDET_MODE_JACK); } } @@ -3235,23 +3235,24 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); - /* If required for an external cap force MICBIAS on */ - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } - /* * Start off measument of microphone impedence to find * out what's actually there. */ wm8994->mic_detecting = true; wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_MIC); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); + + /* If required for an external cap force MICBIAS on */ + if (wm8994->pdata->jd_ext_cap) { + mutex_lock(&codec->mutex); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS2"); + snd_soc_dapm_sync(&codec->dapm); + mutex_unlock(&codec->mutex); + } } else { dev_dbg(codec->dev, "Jack not detected\n"); -- cgit v1.2.3-18-g5258 From 07cafff288266c3aa082f4bda3d47989e73ee85d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2012 12:30:59 +0100 Subject: ALSA: hda/conexant - Clear unsol events on unused pins It seems that Lenovo machines (or codec chip itself?) leave the unsol event tags and the enablement-flag from other pins bogusly even on the unused pins. Although this shouldn't be too critical, it's better to clear them up sanely. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 45 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6bbdbb6dd4e..f3b79031fcc 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3931,6 +3931,50 @@ static void enable_unsol_pins(struct hda_codec *codec, int num_pins, snd_hda_jack_detect_enable(codec, pins[i], action); } +static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums) +{ + int i; + for (i = 0; i < nums; i++) + if (list[i] == nid) + return true; + return false; +} + +/* is the given NID found in any of autocfg items? */ +static bool found_in_autocfg(struct auto_pin_cfg *cfg, hda_nid_t nid) +{ + int i; + + if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) || + found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) || + found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs) || + found_in_nid_list(nid, cfg->dig_out_pins, cfg->dig_outs)) + return true; + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].pin == nid) + return true; + if (cfg->dig_in_pin == nid) + return true; + return false; +} + +/* clear unsol-event tags on unused pins; Conexant codecs seem to leave + * invalid unsol tags by some reason + */ +static void clear_unsol_on_unused_pins(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + if (!found_in_autocfg(cfg, pin->nid)) + snd_hda_codec_write(codec, pin->nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, 0); + } +} + static void cx_auto_init_output(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -3971,6 +4015,7 @@ static void cx_auto_init_output(struct hda_codec *codec) /* turn on all EAPDs if no individual EAPD control is available */ if (!spec->pin_eapd_ctrls) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); + clear_unsol_on_unused_pins(codec); } static void cx_auto_init_input(struct hda_codec *codec) -- cgit v1.2.3-18-g5258 From 1dd4c8e42aa08f21de3cdb4f7aa0841fbd7e3f58 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Feb 2012 17:45:12 +0000 Subject: ASoC: wm8996: Fix /RESET bounce ordering We want to leave the device out of rather than in reset. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index fb5c07a9ec9..1226f92bbb0 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1719,8 +1719,8 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) static int wm8996_reset(struct wm8996_priv *wm8996) { if (wm8996->pdata.ldo_ena > 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); return 0; } else { return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, -- cgit v1.2.3-18-g5258 From 6449c9f858cd68efc93b1d71455e5e475dab69e4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Feb 2012 17:28:39 +0000 Subject: ASoC: wm8996: Remove stub register cache Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1226f92bbb0..9376b19941b 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3054,12 +3054,6 @@ static int wm8996_remove(struct snd_soc_codec *codec) return 0; } -static int wm8996_soc_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - return true; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .probe = wm8996_probe, .remove = wm8996_remove, @@ -3073,8 +3067,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { .dapm_routes = wm8996_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm8996_dapm_routes), .set_pll = wm8996_set_fll, - .reg_cache_size = WM8996_MAX_REGISTER, - .volatile_register = wm8996_soc_volatile_register, }; #define WM8996_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ -- cgit v1.2.3-18-g5258 From e778ba07edd03bc5000e22bc72113e06a7ded694 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Feb 2012 15:39:56 +0000 Subject: ASoC: wm_hubs: Bomb out if we can't read back the DC servo result Should have no practical impact but it's safer than trying to soldier on. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 82b7e9dece2..7cffdd4b70f 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -172,7 +172,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) break; default: WARN(1, "Unknown DCS readback method\n"); - break; + return; } dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); -- cgit v1.2.3-18-g5258 From 62172f4a750e28c5573b3402c3fe3f029b5dd67b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Feb 2012 15:26:54 +0000 Subject: ASoC: wm8994: Remove stub of register access code Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 691e89753a6..bc1f1199a87 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3935,20 +3935,12 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) return 0; } -static int wm8994_soc_volatile(struct snd_soc_codec *codec, - unsigned int reg) -{ - return true; -} - static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .probe = wm8994_codec_probe, .remove = wm8994_codec_remove, .suspend = wm8994_suspend, .resume = wm8994_resume, .set_bias_level = wm8994_set_bias_level, - .reg_cache_size = WM8994_MAX_REGISTER, - .volatile_register = wm8994_soc_volatile, }; static int __devinit wm8994_probe(struct platform_device *pdev) -- cgit v1.2.3-18-g5258 From c90887fe982e46d1b23e151636616d4e4a0077a4 Mon Sep 17 00:00:00 2001 From: Jassi Brar Date: Sat, 25 Feb 2012 16:42:34 +0530 Subject: ASoC: Samsung: Merge two identical if-else clauses Saves two lines and a hell of a lot of embarrassment looking at the code. Signed-off-by: Jassi Brar Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 87a874dc7a3..ea5ccaec83f 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -761,15 +761,13 @@ static int i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: local_irq_save(flags); - if (capture) + if (capture) { i2s_rxctrl(i2s, 0); - else - i2s_txctrl(i2s, 0); - - if (capture) i2s_fifo(i2s, FIC_RXFLUSH); - else + } else { + i2s_txctrl(i2s, 0); i2s_fifo(i2s, FIC_TXFLUSH); + } local_irq_restore(flags); break; -- cgit v1.2.3-18-g5258 From df8ad33558eb74e1a5d37b0313d7230d4ccde631 Mon Sep 17 00:00:00 2001 From: Jaswinder Singh Date: Sat, 25 Feb 2012 16:24:36 +0530 Subject: ASoC: Samsung: Update email id of the author I moved on from a great employer and the email-id no longer exists. Update email-id to a personal one, assuming I don't move on from myself anytime soon. And when I do, people don't get the eulogies bounced. Signed-off-by: Jaswinder Singh Signed-off-by: Mark Brown --- sound/soc/samsung/ac97.c | 4 ++-- sound/soc/samsung/i2s.c | 4 ++-- sound/soc/samsung/i2s.h | 2 +- sound/soc/samsung/pcm.c | 4 ++-- sound/soc/samsung/smdk_wm8580.c | 4 ++-- sound/soc/samsung/smdk_wm9713.c | 4 ++-- 6 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 7b9bf93e370..3d04c1fa678 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -4,7 +4,7 @@ * Evolved from s3c2443-ac97.c * * Copyright (c) 2010 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh + * Author: Jaswinder Singh * Credits: Graeme Gregory, Sean Choi * * This program is free software; you can redistribute it and/or modify @@ -511,7 +511,7 @@ static struct platform_driver s3c_ac97_driver = { module_platform_driver(s3c_ac97_driver); -MODULE_AUTHOR("Jaswinder Singh, "); +MODULE_AUTHOR("Jaswinder Singh, "); MODULE_DESCRIPTION("AC97 driver for the Samsung SoC"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:samsung-ac97"); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ea5ccaec83f..6553b19c70c 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -3,7 +3,7 @@ * ALSA SoC Audio Layer - Samsung I2S Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd. - * Jaswinder Singh + * Jaswinder Singh * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -1141,7 +1141,7 @@ static struct platform_driver samsung_i2s_driver = { module_platform_driver(samsung_i2s_driver); /* Module information */ -MODULE_AUTHOR("Jaswinder Singh, "); +MODULE_AUTHOR("Jaswinder Singh, "); MODULE_DESCRIPTION("Samsung I2S Interface"); MODULE_ALIAS("platform:samsung-i2s"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/i2s.h b/sound/soc/samsung/i2s.h index 8e15f6a616d..d420a7ca56c 100644 --- a/sound/soc/samsung/i2s.h +++ b/sound/soc/samsung/i2s.h @@ -3,7 +3,7 @@ * ALSA SoC Audio Layer - Samsung I2S Controller driver * * Copyright (c) 2010 Samsung Electronics Co. Ltd. - * Jaswinder Singh + * Jaswinder Singh * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 56780206c00..b7b2a1f9142 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -3,7 +3,7 @@ * ALSA SoC Audio Layer - S3C PCM-Controller driver * * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh + * Author: Jaswinder Singh * based upon I2S drivers by Ben Dooks. * * This program is free software; you can redistribute it and/or modify @@ -639,7 +639,7 @@ static struct platform_driver s3c_pcm_driver = { module_platform_driver(s3c_pcm_driver); /* Module information */ -MODULE_AUTHOR("Jaswinder Singh, "); +MODULE_AUTHOR("Jaswinder Singh, "); MODULE_DESCRIPTION("S3C PCM Controller Driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:samsung-pcm"); diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index bff8758e7f2..ade2809cf39 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -2,7 +2,7 @@ * smdk_wm8580.c * * Copyright (c) 2009 Samsung Electronics Co. Ltd - * Author: Jaswinder Singh + * Author: Jaswinder Singh * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -253,6 +253,6 @@ static void __exit smdk_audio_exit(void) } module_exit(smdk_audio_exit); -MODULE_AUTHOR("Jaswinder Singh, jassi.brar@samsung.com"); +MODULE_AUTHOR("Jaswinder Singh, jassisinghbrar@gmail.com"); MODULE_DESCRIPTION("ALSA SoC SMDK WM8580"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/smdk_wm9713.c b/sound/soc/samsung/smdk_wm9713.c index 8e26a730fcd..55b2ca7f329 100644 --- a/sound/soc/samsung/smdk_wm9713.c +++ b/sound/soc/samsung/smdk_wm9713.c @@ -2,7 +2,7 @@ * smdk_wm9713.c -- SoC audio for SMDK * * Copyright 2010 Samsung Electronics Co. Ltd. - * Author: Jaswinder Singh Brar + * Author: Jaswinder Singh Brar * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License as @@ -103,6 +103,6 @@ module_init(smdk_init); module_exit(smdk_exit); /* Module information */ -MODULE_AUTHOR("Jaswinder Singh Brar, jassi.brar@samsung.com"); +MODULE_AUTHOR("Jaswinder Singh Brar, jassisinghbrar@gmail.com"); MODULE_DESCRIPTION("ALSA SoC SMDK+WM9713"); MODULE_LICENSE("GPL"); -- cgit v1.2.3-18-g5258 From df0c016d0bbeea99f93b5f7836d6418a19feefa7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 1 Mar 2012 12:07:15 +0000 Subject: ASoC: wm8962: Remove register default for PLL2 The initial value can be changed depending on system configuration. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 6af794510c6..245c5f8023d 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -208,8 +208,6 @@ static struct reg_default wm8962_reg[] = { { 126, 0x000D }, /* R126 - Analogue Clocking3 */ { 127, 0x0000 }, /* R127 - PLL Software Reset */ - { 129, 0x0000 }, /* R129 - PLL2 */ - { 131, 0x0000 }, /* R131 - PLL 4 */ { 136, 0x0067 }, /* R136 - PLL 9 */ -- cgit v1.2.3-18-g5258 From 378ec0ca38527d3ca3683f7f38f3243c8d0a90f3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 1 Mar 2012 19:01:43 +0000 Subject: ASoC: wm8994: Disable debounce of jack detection on inserted jack Don't debounce jack detection for inserted jacks, giving improved responsiveness. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bc1f1199a87..81d62a38475 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3235,6 +3235,10 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); + /* Disable debounce while inserted */ + snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, + WM1811_JACKDET_DB, 0); + /* * Start off measument of microphone impedence to find * out what's actually there. @@ -3270,6 +3274,10 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) SND_JACK_MECHANICAL | SND_JACK_HEADSET | wm8994->btn_mask); + /* Enable debounce while removed */ + snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, + WM1811_JACKDET_DB, WM1811_JACKDET_DB); + wm8994->mic_detecting = false; wm8994->jack_mic = false; snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, -- cgit v1.2.3-18-g5258 From 1355ab147fa38e4b3841469c51422e2343a877b2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 1 Mar 2012 16:40:51 +0000 Subject: ASoC: wm8962: Run the headphone in class G mode when sidetone is enabled Class W mode with sidetone is not fully supported. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 245c5f8023d..445d2090661 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2230,9 +2230,11 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "STL", "Left", "ADCL" }, { "STL", "Right", "ADCR" }, + { "STL", NULL, "Class G" }, { "STR", "Left", "ADCL" }, { "STR", "Right", "ADCR" }, + { "STR", NULL, "Class G" }, { "DACL", NULL, "SYSCLK" }, { "DACL", NULL, "TOCLK" }, -- cgit v1.2.3-18-g5258 From e7f73a1613567ac82314f33956c0f3810bf1efb2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 22 Feb 2012 10:49:08 +0100 Subject: ASoC: Add dmaengine PCM helper functions This patch adds a set of functions which are intended to be used when implementing a dmaengine based sound PCM driver. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/Kconfig | 3 + sound/soc/Makefile | 3 + sound/soc/soc-dmaengine-pcm.c | 287 ++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 293 insertions(+) create mode 100644 sound/soc/soc-dmaengine-pcm.c (limited to 'sound') diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 35e662d270e..91c985599d3 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -25,6 +25,9 @@ if SND_SOC config SND_SOC_AC97_BUS bool +config SND_SOC_DMAENGINE_PCM + bool + # All the supported SoCs source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 9ea8ac827ad..2feaf376e94 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,9 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-io.o +snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o +obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o + obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ obj-$(CONFIG_SND_SOC) += atmel/ diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c new file mode 100644 index 00000000000..0526cf82b54 --- /dev/null +++ b/sound/soc/soc-dmaengine-pcm.c @@ -0,0 +1,287 @@ +/* + * Copyright (C) 2012, Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Based on: + * imx-pcm-dma-mx2.c, Copyright 2009 Sascha Hauer + * mxs-pcm.c, Copyright (C) 2011 Freescale Semiconductor, Inc. + * ep93xx-pcm.c, Copyright (C) 2006 Lennert Buytenhek + * Copyright (C) 2006 Applied Data Systems + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 675 Mass Ave, Cambridge, MA 02139, USA. + * + */ +#include +#include +#include +#include +#include +#include +#include + +#include + +struct dmaengine_pcm_runtime_data { + struct dma_chan *dma_chan; + + unsigned int pos; + + void *data; +}; + +static inline struct dmaengine_pcm_runtime_data *substream_to_prtd( + const struct snd_pcm_substream *substream) +{ + return substream->runtime->private_data; +} + +/** + * snd_dmaengine_pcm_set_data - Set dmaengine substream private data + * @substream: PCM substream + * @data: Data to set + */ +void snd_dmaengine_pcm_set_data(struct snd_pcm_substream *substream, void *data) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + prtd->data = data; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_set_data); + +/** + * snd_dmaengine_pcm_get_data - Get dmaeinge substream private data + * @substream: PCM substream + * + * Returns the data previously set with snd_dmaengine_pcm_set_data + */ +void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + return prtd->data; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_get_data); + +struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + return prtd->dma_chan; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_get_chan); + +/** + * snd_hwparams_to_dma_slave_config - Convert hw_params to dma_slave_config + * @substream: PCM substream + * @params: hw_params + * @slave_config: DMA slave config + * + * This function can be used to initialize a dma_slave_config from a substream + * and hw_params in a dmaengine based PCM driver implementation. + */ +int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, + const struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config) +{ + enum dma_slave_buswidth buswidth; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; + break; + case SNDRV_PCM_FORMAT_S16_LE: + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config->direction = DMA_MEM_TO_DEV; + slave_config->dst_addr_width = buswidth; + } else { + slave_config->direction = DMA_DEV_TO_MEM; + slave_config->src_addr_width = buswidth; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); + +static void dmaengine_pcm_dma_complete(void *arg) +{ + struct snd_pcm_substream *substream = arg; + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + prtd->pos += snd_pcm_lib_period_bytes(substream); + if (prtd->pos >= snd_pcm_lib_buffer_bytes(substream)) + prtd->pos = 0; + + snd_pcm_period_elapsed(substream); +} + +static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dma_chan *chan = prtd->dma_chan; + struct dma_async_tx_descriptor *desc; + enum dma_transfer_direction direction; + + direction = snd_pcm_substream_to_dma_direction(substream); + + desc = chan->device->device_prep_dma_cyclic(chan, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), direction); + + if (!desc) + return -ENOMEM; + + desc->callback = dmaengine_pcm_dma_complete; + desc->callback_param = substream; + dmaengine_submit(desc); + + return 0; +} + +/** + * snd_dmaengine_pcm_trigger - dmaengine based PCM trigger implementation + * @substream: PCM substream + * @cmd: Trigger command + * + * Returns 0 on success, a negative error code otherwise. + * + * This function can be used as the PCM trigger callback for dmaengine based PCM + * driver implementations. + */ +int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = dmaengine_pcm_prepare_and_submit(substream); + if (ret) + return ret; + dma_async_issue_pending(prtd->dma_chan); + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dmaengine_resume(prtd->dma_chan); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dmaengine_pause(prtd->dma_chan); + break; + case SNDRV_PCM_TRIGGER_STOP: + dmaengine_terminate_all(prtd->dma_chan); + break; + default: + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); + +/** + * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation + * @substream: PCM substream + * + * This function can be used as the PCM pointer callback for dmaengine based PCM + * driver implementations. + */ +snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + return bytes_to_frames(substream->runtime, prtd->pos); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); + +static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd, + dma_filter_fn filter_fn, void *filter_data) +{ + dma_cap_mask_t mask; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + dma_cap_set(DMA_CYCLIC, mask); + prtd->dma_chan = dma_request_channel(mask, filter_fn, filter_data); + + if (!prtd->dma_chan) + return -ENXIO; + + return 0; +} + +/** + * snd_dmaengine_pcm_open - Open a dmaengine based PCM substream + * @substream: PCM substream + * @filter_fn: Filter function used to request the DMA channel + * @filter_data: Data passed to the DMA filter function + * + * Returns 0 on success, a negative error code otherwise. + * + * This function will request a DMA channel using the passed filter function and + * data. The function should usually be called from the pcm open callback. + * + * Note that this function will use private_data field of the substream's + * runtime. So it is not availabe to your pcm driver implementation. If you need + * to keep additional data attached to a substream use + * snd_dmaeinge_pcm_{set,get}_data. + */ +int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, + dma_filter_fn filter_fn, void *filter_data) +{ + struct dmaengine_pcm_runtime_data *prtd; + int ret; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + prtd = kzalloc(sizeof(*prtd), GFP_KERNEL); + if (!prtd) + return -ENOMEM; + + ret = dmaengine_pcm_request_channel(prtd, filter_fn, filter_data); + if (ret < 0) { + kfree(prtd); + return ret; + } + + substream->runtime->private_data = prtd; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open); + +/** + * snd_dmaengine_pcm_close - Close a dmaengine based PCM substream + * @substream: PCM substream + */ +int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + + dma_release_channel(prtd->dma_chan); + kfree(prtd); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close); -- cgit v1.2.3-18-g5258 From c307e8e32e11aae0f561db02898a090c9b7c9b70 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 22 Feb 2012 10:49:09 +0100 Subject: ASoC: imx-pcm-dma: Use dmaengine PCM helper functions Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 1 + sound/soc/imx/imx-pcm-dma-mx2.c | 176 ++++++---------------------------------- 2 files changed, 28 insertions(+), 149 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index e89139ffbce..192861c2b4a 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -14,6 +14,7 @@ config SND_MXC_SOC_FIQ tristate config SND_MXC_SOC_MX2 + select SND_SOC_DMAENGINE_PCM tristate config SND_MXC_SOC_WM1133_EV1 diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index f974e61fa68..4cd5462049b 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -27,104 +27,42 @@ #include #include #include +#include #include #include "imx-ssi.h" -struct imx_pcm_runtime_data { - int period_bytes; - int periods; - unsigned long offset; - struct dma_async_tx_descriptor *desc; - struct dma_chan *dma_chan; - struct imx_dma_data dma_data; -}; - -static void audio_dma_irq(void *data) -{ - struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - - iprtd->offset += iprtd->period_bytes; - iprtd->offset %= iprtd->period_bytes * iprtd->periods; - - snd_pcm_period_elapsed(substream); -} - static bool filter(struct dma_chan *chan, void *param) { - struct imx_pcm_runtime_data *iprtd = param; - if (!imx_dma_is_general_purpose(chan)) return false; - chan->private = &iprtd->dma_data; + chan->private = param; - return true; -} - -static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params; - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - dma_cap_mask_t mask; - - dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - iprtd->dma_data.peripheral_type = IMX_DMATYPE_SSI; - iprtd->dma_data.priority = DMA_PRIO_HIGH; - iprtd->dma_data.dma_request = dma_params->dma; - - /* Try to grab a DMA channel */ - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); - if (!iprtd->dma_chan) - return -EINVAL; - - return 0; + return true; } static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - struct dma_chan *chan = iprtd->dma_chan; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); struct imx_pcm_dma_params *dma_params; struct dma_slave_config slave_config; - enum dma_slave_buswidth buswidth; - unsigned long dma_addr; int ret; dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: - buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; - break; - case SNDRV_PCM_FORMAT_S20_3LE: - case SNDRV_PCM_FORMAT_S24_LE: - buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - break; - default: - return 0; - } + ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + if (ret) + return ret; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config.direction = DMA_MEM_TO_DEV; slave_config.dst_addr = dma_params->dma_addr; - slave_config.dst_addr_width = buswidth; slave_config.dst_maxburst = dma_params->burstsize; } else { - slave_config.direction = DMA_DEV_TO_MEM; slave_config.src_addr = dma_params->dma_addr; - slave_config.src_addr_width = buswidth; slave_config.src_maxburst = dma_params->burstsize; } @@ -132,68 +70,11 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, if (ret) return ret; - - iprtd->periods = params_periods(params); - iprtd->period_bytes = params_period_bytes(params); - iprtd->offset = 0; - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - dma_addr = runtime->dma_addr; - - iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr, - iprtd->period_bytes * iprtd->periods, - iprtd->period_bytes, - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - if (!iprtd->desc) { - dev_err(&chan->dev->device, "cannot prepare slave dma\n"); - return -EINVAL; - } - - iprtd->desc->callback = audio_dma_irq; - iprtd->desc->callback_param = substream; - - return 0; -} - -static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dmaengine_submit(iprtd->desc); - - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dmaengine_terminate_all(iprtd->dma_chan); - - break; - default: - return -EINVAL; - } - return 0; } -static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; - - pr_debug("%s: %ld %ld\n", __func__, iprtd->offset, - bytes_to_frames(substream->runtime, iprtd->offset)); - - return bytes_to_frames(substream->runtime, iprtd->offset); -} - static struct snd_pcm_hardware snd_imx_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -215,40 +96,37 @@ static struct snd_pcm_hardware snd_imx_hardware = { static int snd_imx_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params; + struct imx_dma_data *dma_data; int ret; - iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); - if (iprtd == NULL) - return -ENOMEM; - runtime->private_data = iprtd; + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); - ret = snd_pcm_hw_constraint_integer(substream->runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) { - kfree(iprtd); - return ret; - } + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - ret = imx_ssi_dma_alloc(substream); - if (ret < 0) { - kfree(iprtd); - return ret; + dma_data = kzalloc(sizeof(*dma_data), GFP_KERNEL); + dma_data->peripheral_type = IMX_DMATYPE_SSI; + dma_data->priority = DMA_PRIO_HIGH; + dma_data->dma_request = dma_params->dma; + + ret = snd_dmaengine_pcm_open(substream, filter, dma_data); + if (ret) { + kfree(dma_data); + return 0; } - snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + snd_dmaengine_pcm_set_data(substream, dma_data); return 0; } static int snd_imx_close(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct imx_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); - dma_release_channel(iprtd->dma_chan); - kfree(iprtd); + snd_dmaengine_pcm_close(substream); + kfree(dma_data); return 0; } @@ -258,8 +136,8 @@ static struct snd_pcm_ops imx_pcm_ops = { .close = snd_imx_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_imx_pcm_hw_params, - .trigger = snd_imx_pcm_trigger, - .pointer = snd_imx_pcm_pointer, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, .mmap = snd_imx_pcm_mmap, }; -- cgit v1.2.3-18-g5258 From 016ab467aa53639d68b03386885c481b8761018e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 22 Feb 2012 10:49:10 +0100 Subject: ASoC: mxs-pcm: Use dmaengine PCM helper functions Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Acked-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/mxs/Kconfig | 1 + sound/soc/mxs/mxs-pcm.c | 136 +++++++++--------------------------------------- sound/soc/mxs/mxs-pcm.h | 12 ----- 3 files changed, 25 insertions(+), 124 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 21d20f3e026..99a997f19bb 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,6 +1,7 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" depends on ARCH_MXS + select SND_SOC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the MXS SAIF interface. diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 5b8c8d31406..6ca1f46d84a 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -34,10 +34,16 @@ #include #include #include +#include #include #include "mxs-pcm.h" +struct mxs_pcm_dma_data { + struct mxs_dma_data dma_data; + struct mxs_pcm_dma_params *dma_params; +}; + static struct snd_pcm_hardware snd_mxs_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | @@ -58,21 +64,10 @@ static struct snd_pcm_hardware snd_mxs_hardware = { }; -static void audio_dma_irq(void *data) -{ - struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct mxs_pcm_runtime_data *iprtd = runtime->private_data; - - iprtd->offset += iprtd->period_bytes; - iprtd->offset %= iprtd->period_bytes * iprtd->periods; - snd_pcm_period_elapsed(substream); -} - static bool filter(struct dma_chan *chan, void *param) { - struct mxs_pcm_runtime_data *iprtd = param; - struct mxs_pcm_dma_params *dma_params = iprtd->dma_params; + struct mxs_pcm_dma_data *pcm_dma_data = param; + struct mxs_pcm_dma_params *dma_params = pcm_dma_data->dma_params; if (!mxs_dma_is_apbx(chan)) return false; @@ -80,134 +75,51 @@ static bool filter(struct dma_chan *chan, void *param) if (chan->chan_id != dma_params->chan_num) return false; - chan->private = &iprtd->dma_data; + chan->private = &pcm_dma_data->dma_data; return true; } -static int mxs_dma_alloc(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct mxs_pcm_runtime_data *iprtd = runtime->private_data; - dma_cap_mask_t mask; - - iprtd->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - iprtd->dma_data.chan_irq = iprtd->dma_params->chan_irq; - iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); - if (!iprtd->dma_chan) - return -EINVAL; - - return 0; -} - static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct mxs_pcm_runtime_data *iprtd = runtime->private_data; - unsigned long dma_addr; - struct dma_chan *chan; - - chan = iprtd->dma_chan; - - iprtd->periods = params_periods(params); - iprtd->period_bytes = params_period_bytes(params); - iprtd->offset = 0; - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - dma_addr = runtime->dma_addr; - - iprtd->desc = chan->device->device_prep_dma_cyclic(chan, dma_addr, - iprtd->period_bytes * iprtd->periods, - iprtd->period_bytes, - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); - if (!iprtd->desc) { - dev_err(&chan->dev->device, "cannot prepare slave dma\n"); - return -EINVAL; - } - - iprtd->desc->callback = audio_dma_irq; - iprtd->desc->callback_param = substream; - return 0; } -static int snd_mxs_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mxs_pcm_runtime_data *iprtd = runtime->private_data; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - dmaengine_submit(iprtd->desc); - - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - dmaengine_terminate_all(iprtd->dma_chan); - - break; - default: - return -EINVAL; - } - - return 0; -} - -static snd_pcm_uframes_t snd_mxs_pcm_pointer( - struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct mxs_pcm_runtime_data *iprtd = runtime->private_data; - - return bytes_to_frames(substream->runtime, iprtd->offset); -} - static int snd_mxs_open(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct mxs_pcm_runtime_data *iprtd; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mxs_pcm_dma_data *pcm_dma_data; int ret; - iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); - if (iprtd == NULL) + pcm_dma_data = kzalloc(sizeof(*pcm_dma_data), GFP_KERNEL); + if (pcm_dma_data == NULL) return -ENOMEM; - runtime->private_data = iprtd; - ret = snd_pcm_hw_constraint_integer(substream->runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) { - kfree(iprtd); - return ret; - } + pcm_dma_data->dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + pcm_dma_data->dma_data.chan_irq = pcm_dma_data->dma_params->chan_irq; - ret = mxs_dma_alloc(substream); + ret = snd_dmaengine_pcm_open(substream, filter, pcm_dma_data); if (ret) { - kfree(iprtd); + kfree(pcm_dma_data); return ret; } snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); + snd_dmaengine_pcm_set_data(substream, pcm_dma_data); + return 0; } static int snd_mxs_close(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct mxs_pcm_runtime_data *iprtd = runtime->private_data; + struct mxs_pcm_dma_data *pcm_dma_data = snd_dmaengine_pcm_get_data(substream); - dma_release_channel(iprtd->dma_chan); - kfree(iprtd); + snd_dmaengine_pcm_close(substream); + kfree(pcm_dma_data); return 0; } @@ -228,8 +140,8 @@ static struct snd_pcm_ops mxs_pcm_ops = { .close = snd_mxs_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_mxs_pcm_hw_params, - .trigger = snd_mxs_pcm_trigger, - .pointer = snd_mxs_pcm_pointer, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, .mmap = snd_mxs_pcm_mmap, }; diff --git a/sound/soc/mxs/mxs-pcm.h b/sound/soc/mxs/mxs-pcm.h index ba75e103bb3..5f01a9124b3 100644 --- a/sound/soc/mxs/mxs-pcm.h +++ b/sound/soc/mxs/mxs-pcm.h @@ -19,21 +19,9 @@ #ifndef _MXS_PCM_H #define _MXS_PCM_H -#include - struct mxs_pcm_dma_params { int chan_irq; int chan_num; }; -struct mxs_pcm_runtime_data { - int period_bytes; - int periods; - unsigned long offset; - struct dma_async_tx_descriptor *desc; - struct dma_chan *dma_chan; - struct mxs_dma_data dma_data; - struct mxs_pcm_dma_params *dma_params; -}; - #endif -- cgit v1.2.3-18-g5258 From 02db110351019eef2d6b10e08162dd370542e9fd Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 2 Mar 2012 16:13:44 +0000 Subject: ASoC: core: cleanup platform debugfs on probe failure. Make sure we cleanup the platform debugfs when probe fails. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1bdc67e0bd1..29dbbd793fc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1112,6 +1112,7 @@ static int soc_probe_platform(struct snd_soc_card *card, return 0; err_probe: + soc_cleanup_platform_debugfs(platform); module_put(platform->dev->driver->owner); return ret; -- cgit v1.2.3-18-g5258 From fe4085e84f17a57a533a210a626e0cc9ead381f9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 2 Mar 2012 13:07:41 +0000 Subject: ASoC: core: Log a warning when machines use soc-audio snd_soc_register_card() has been available and strongly preferred since 2.6.38 but we're still seeing new drivers using it and the conversion rate for older machines has been low. Help address both issues by logging a warning when the soc-audio device probes. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 29dbbd793fc..7978f6c01ef 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1645,6 +1645,10 @@ static int soc_probe(struct platform_device *pdev) if (!card) return -EINVAL; + dev_warn(&pdev->dev, + "ASoC machine %s should use snd_soc_register_card()\n", + card->name); + /* Bodge while we unpick instantiation */ card->dev = &pdev->dev; -- cgit v1.2.3-18-g5258 From 2bc16ed8e02ba39dc5010b4a2b2a606e4b87abbd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Mar 2012 23:24:39 +0000 Subject: ASoC: wm8994: Push wm8994 private data allocation out into device probe Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 81d62a38475..73cd8b922d1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3472,23 +3472,16 @@ static irqreturn_t wm8994_temp_shut(int irq, void *data) static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control = dev_get_drvdata(codec->dev->parent); - struct wm8994_priv *wm8994; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; unsigned int reg; int ret, i; + wm8994->codec = codec; codec->control_data = control->regmap; - wm8994 = devm_kzalloc(codec->dev, sizeof(struct wm8994_priv), - GFP_KERNEL); - if (wm8994 == NULL) - return -ENOMEM; - snd_soc_codec_set_drvdata(codec, wm8994); - snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - wm8994->wm8994 = dev_get_drvdata(codec->dev->parent); - wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; mutex_init(&wm8994->accdet_lock); @@ -3953,6 +3946,17 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { static int __devinit wm8994_probe(struct platform_device *pdev) { + struct wm8994_priv *wm8994; + + wm8994 = devm_kzalloc(&pdev->dev, sizeof(struct wm8994_priv), + GFP_KERNEL); + if (wm8994 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm8994); + + wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + wm8994->pdata = dev_get_platdata(pdev->dev.parent); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8994, wm8994_dai, ARRAY_SIZE(wm8994_dai)); } -- cgit v1.2.3-18-g5258 From b16db745b51a1ecd3fd526a2ff35d61f2962bd7f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Mar 2012 15:33:23 +0000 Subject: ASoC: wm8994: Suppress noop updates of FLL K Using snd_soc_write() means we always write to the register even if it already contains the newly calculated value. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 73cd8b922d1..a567a4d9b5d 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1918,7 +1918,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_OUTDIV_MASK | WM8994_FLL1_FRATIO_MASK, reg); - snd_soc_write(codec, WM8994_FLL1_CONTROL_3 + reg_offset, fll.k); + snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_3 + reg_offset, + WM8994_FLL1_K_MASK, fll.k); snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_4 + reg_offset, WM8994_FLL1_N_MASK, -- cgit v1.2.3-18-g5258 From 67109cbea1f92d369849dc88b6c9aca0f66c044e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 29 Feb 2012 16:40:08 +0000 Subject: ASoC: wm_hubs: Disable cache of the DC servo calibration for WM1811 The WM1811 DC servo is able to run much faster than previous devices so the benefit of skipping calibration is not useful. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/codecs/wm_hubs.h | 1 + 3 files changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a567a4d9b5d..1fef87d6a28 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3535,6 +3535,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_readback_mode = 2; wm8994->hubs.no_series_update = 1; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.no_cache_class_w = true; switch (wm8994->revision) { case 0: diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 7cffdd4b70f..c08d1c2f346 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -207,7 +207,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* Save the callibrated offset if we're in class W mode and * therefore don't have any analogue signal mixed in. */ - if (hubs->class_w) + if (hubs->class_w && !hubs->no_cache_class_w) hubs->class_w_dcs = dcs_cfg; } diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 4140905c738..5705276f494 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -30,6 +30,7 @@ struct wm_hubs_data { int series_startup; int no_series_update; + bool no_cache_class_w; bool class_w; u16 class_w_dcs; -- cgit v1.2.3-18-g5258 From 7d464b201fd2f82902028437314a10db85e48ed8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Mar 2012 18:46:06 +0000 Subject: ASoC: wm8994: Make sure we sync DAPM on WM8958 detection mode changes Normally this will have no effect as we set detection up at system startup before DAPM syncs take effect, this will only be useful if the system enables and disables detection at runtime. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1fef87d6a28..17baacbcf2f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3330,6 +3330,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } snd_soc_dapm_force_enable_pin(&codec->dapm, "CLK_SYS"); + snd_soc_dapm_sync(&codec->dapm); wm8994->micdet[0].jack = jack; wm8994->jack_cb = cb; @@ -3376,6 +3377,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); snd_soc_dapm_disable_pin(&codec->dapm, "CLK_SYS"); + snd_soc_dapm_sync(&codec->dapm); } return 0; -- cgit v1.2.3-18-g5258 From afaf1591203e4ea12c4c8e0240549ff5e592d7ae Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Mar 2012 18:46:36 +0000 Subject: ASoC: wm8994: Disable JACKDET when disabling detecton Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 17baacbcf2f..6166a578341 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3376,6 +3376,7 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, } else { snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); + wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_NONE); snd_soc_dapm_disable_pin(&codec->dapm, "CLK_SYS"); snd_soc_dapm_sync(&codec->dapm); } -- cgit v1.2.3-18-g5258 From 28e33269a71cb4104c2e0629b6a3ef7344436f93 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Mar 2012 00:10:02 +0000 Subject: ASoC: wm8994: Don't bother updating the jackdet mode needlessly If we're not doing jackdet it's not needed. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6166a578341..bda5ddbeecb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -686,6 +686,9 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + if (!wm8994->jackdet || !wm8994->jack_cb) + return; + if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; -- cgit v1.2.3-18-g5258 From 1defde2a50f9171e665cc8f4c46fe48e86bb364e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Mar 2012 20:02:49 +0000 Subject: ASoC: wm8994: Factor out WM1811A detection mode setting Push everything through one function for active use cases, should be no practical effect. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 48 +++++++++++++++++++++++++++-------------------- 1 file changed, 28 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bda5ddbeecb..2417ef9316e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -685,6 +685,8 @@ SOC_SINGLE_TLV("MIXINL IN1RP Boost Volume", WM8994_INPUT_MIXER_1, 8, 1, 0, static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + u16 old = snd_soc_read(codec, WM8994_ANTIPOP_2) + & WM1811_JACKDET_MODE_MASK; if (!wm8994->jackdet || !wm8994->jack_cb) return; @@ -692,11 +694,28 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; + if (mode == old) + return; + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, WM1811_JACKDET_MODE_MASK, mode); - if (mode == WM1811_JACKDET_MODE_MIC) - msleep(2); + switch (mode) { + case WM1811_JACKDET_MODE_MIC: + case WM1811_JACKDET_MODE_AUDIO: + switch (old) { + case WM1811_JACKDET_MODE_MIC: + case WM1811_JACKDET_MODE_AUDIO: + break; + default: + msleep(2); + break; + } + + default: + break; + } + } static void active_reference(struct snd_soc_codec *codec) @@ -710,15 +729,8 @@ static void active_reference(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Active refcount incremented, now %d\n", wm8994->active_refcount); - if (wm8994->active_refcount == 1) { - /* If we're using jack detection go into audio mode */ - if (wm8994->jackdet && wm8994->jack_cb) { - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, - WM1811_JACKDET_MODE_AUDIO); - msleep(2); - } - } + /* If we're using jack detection go into audio mode */ + wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_AUDIO); mutex_unlock(&wm8994->accdet_lock); } @@ -737,16 +749,12 @@ static void active_dereference(struct snd_soc_codec *codec) if (wm8994->active_refcount == 0) { /* Go into appropriate detection only mode */ - if (wm8994->jackdet && wm8994->jack_cb) { - if (wm8994->jack_mic || wm8994->mic_detecting) - mode = WM1811_JACKDET_MODE_MIC; - else - mode = WM1811_JACKDET_MODE_JACK; + if (wm8994->jack_mic || wm8994->mic_detecting) + mode = WM1811_JACKDET_MODE_MIC; + else + mode = WM1811_JACKDET_MODE_JACK; - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, - mode); - } + wm1811_jackdet_set_mode(codec, mode); } mutex_unlock(&wm8994->accdet_lock); -- cgit v1.2.3-18-g5258 From 9a3a101c1a34b498f466409ce5a1bf0fcbcc476b Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Sat, 3 Mar 2012 23:19:49 +0800 Subject: ASoC: imx: initialize dma_params burstsize just in imx-ssi It's not necessary for imx-pcm-dma-mx2 to access imx_ssi.dma_params for burstsize initialization. Instead, it can just be done in imx-ssi probe function once. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 5 ----- sound/soc/imx/imx-ssi.c | 2 +- 2 files changed, 1 insertion(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 4cd5462049b..471e2218c97 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -149,11 +149,6 @@ static struct snd_soc_platform_driver imx_soc_platform_mx2 = { static int __devinit imx_soc_platform_probe(struct platform_device *pdev) { - struct imx_ssi *ssi = platform_get_drvdata(pdev); - - ssi->dma_params_tx.burstsize = 6; - ssi->dma_params_rx.burstsize = 4; - return snd_soc_register_platform(&pdev->dev, &imx_soc_platform_mx2); } diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index dbf43f5f0ae..25c623115a9 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -668,7 +668,7 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0; ssi->dma_params_tx.dma_addr = res->start + SSI_STX0; - ssi->dma_params_tx.burstsize = 4; + ssi->dma_params_tx.burstsize = 6; ssi->dma_params_rx.burstsize = 4; res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); -- cgit v1.2.3-18-g5258 From 15ca1b19c6ab11fab61512684e987517e3cafad3 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Sat, 3 Mar 2012 23:19:48 +0800 Subject: ASoC: imx: move SND_SOC_AC97_BUS selection down to machine driver SND_SOC_AC97_BUS is selected to enable the AC97 support in soc-core. Rather than selecting the option under SND_IMX_SOC, it's better to leave the selection to individual machine driver which knows if AC97 support is needed or not. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 192861c2b4a..aa4294bf49b 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC - select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to the i.MX SSI interface. @@ -38,6 +37,7 @@ config SND_SOC_MX27VIS_AIC32X4 config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 + select SND_SOC_AC97_BUS select SND_SOC_WM9712 select SND_MXC_SOC_FIQ help -- cgit v1.2.3-18-g5258 From ad20ff920c1fd217578e2c637dd50c1878a21c06 Mon Sep 17 00:00:00 2001 From: Denis 'GNUtoo' Carikli Date: Sun, 26 Feb 2012 19:21:53 +0100 Subject: ASoC: wm8753: fix initialization Without that fix the wm8753 SPI initialization fails, and then produces a kernel panic during boot with the following call trace: Unable to handle kernel paging request at virtual address 37386d9b [] (regmap_get_val_bytes+0x0/0x14) from [] (snd_soc_codec_set_cache_io+0x9c/0xcc) [] (snd_soc_codec_set_cache_io+0x9c/0xcc) from [] (wm8753_probe+0x5c/0x1c4) [] (wm8753_probe+0x5c/0x1c4) from [] (soc_probe_codec+0x174/0x284) [] (soc_probe_codec+0x174/0x284) from [] (snd_soc_instantiate_cards+0x68c/0xe28) [] (snd_soc_instantiate_cards+0x68c/0xe28) from [] (snd_soc_register_card+0x240/0x2d4) [] (snd_soc_register_card+0x240/0x2d4) from [] (soc_probe+0x24/0x40) [] (soc_probe+0x24/0x40) from [] (platform_drv_probe+0x14/0x18) [...] The commit d3398ff05907167f463e119421b053ce043741d1 ( ASoC: Convert WM8753 to direct regmap API usage ) introduced the problem. Thanks to Lars-Peter Clausen for helping me a bit during the debugging. Signed-off-by: Denis 'GNUtoo' Carikli Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 59c28dbc5fc..e27e7b62b36 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1578,6 +1578,9 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret); goto err_regmap; } + + return 0; + err_regmap: regmap_exit(wm8753->regmap); err: @@ -1634,6 +1637,8 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, goto err_regmap; } + return 0; + err_regmap: regmap_exit(wm8753->regmap); err: -- cgit v1.2.3-18-g5258 From db05828aadbcc71aeea1c0b33ffadc8655dec600 Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Fri, 2 Mar 2012 22:09:39 +0100 Subject: ALSA: ice1724 - constrain runtime rates for locked internal rate The driver already defines control "Multi Track Rate Locking" which locks the card at current rate if switched to internal clock. This patch limits the runtime rates to this rate only, allowing proper reporting of the card capabilities, and e.g. automatic rate conversion by the plug plugin to the currently locked rate. Signed-off-by: Pavel Hofman Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 92362973764..812d10e43ae 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1013,6 +1013,25 @@ static int set_rate_constraints(struct snd_ice1712 *ice, ice->hw_rates); } +/* if the card has the internal rate locked (is_pro_locked), limit runtime + hw rates to the current internal rate only. +*/ +static void constrain_rate_if_locked(struct snd_pcm_substream *substream) +{ + struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int rate; + if (is_pro_rate_locked(ice)) { + rate = ice->get_rate(ice); + if (rate >= runtime->hw.rate_min + && rate <= runtime->hw.rate_max) { + runtime->hw.rate_min = rate; + runtime->hw.rate_max = rate; + } + } +} + + /* multi-channel playback needs alignment 8x32bit regardless of the channels * actually used */ @@ -1046,6 +1065,7 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + constrain_rate_if_locked(substream); if (ice->pro_open) ice->pro_open(ice, substream); return 0; @@ -1066,6 +1086,7 @@ static int snd_vt1724_capture_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + constrain_rate_if_locked(substream); if (ice->pro_open) ice->pro_open(ice, substream); return 0; @@ -1215,6 +1236,7 @@ static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + constrain_rate_if_locked(substream); if (ice->spdif.ops.open) ice->spdif.ops.open(ice, substream); return 0; @@ -1251,6 +1273,7 @@ static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + constrain_rate_if_locked(substream); if (ice->spdif.ops.open) ice->spdif.ops.open(ice, substream); return 0; -- cgit v1.2.3-18-g5258 From e21af48583380ed9b5ca07b6dd962dbcd3748e0a Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 5 Mar 2012 11:38:46 +0100 Subject: ALSA: hda - fix broken automute/autoswitch for Realtek The recent addition of volume-knob widget in the auto-parser broke automute/autoswitch for some Realtek devices. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 01179d53edc..7e651682eec 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -686,7 +686,7 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) else res >>= 26; action = snd_hda_jack_get_action(codec, res); - if (res == ALC_DCVOL_EVENT) { + if (action == ALC_DCVOL_EVENT) { /* Execute the dc-vol event here as it requires the NID * but we don't pass NID to alc_exec_unsol_event(). * Once when we convert all static quirks to the auto-parser, -- cgit v1.2.3-18-g5258 From f13ebada17142438ab97afa0421aa5084ce174f2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Mar 2012 18:01:01 +0000 Subject: ASoC: dapm: Show if widgets are forced in debugfs The information was not otherwise visible. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c9b088dab1c..a4d4aa1e6c4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1569,8 +1569,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w); dapm_clear_walk(w->dapm); - ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", - w->name, w->power ? "On" : "Off", in, out); + ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", + w->name, w->power ? "On" : "Off", + w->force ? " (forced)" : "", in, out); if (w->reg >= 0) ret += snprintf(buf + ret, PAGE_SIZE - ret, -- cgit v1.2.3-18-g5258 From 60282ede6b73d6ac7b571df5c65fa6a77db1a4a2 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 5 Mar 2012 22:30:49 +0800 Subject: ASoC: imx: move eukrea audmux call into ASoC machine driver It moves eukrea audmux configuration call from board file into ASoC machine driver eukrea-tlv320, so that it gets aligned wm1133-ev1 and mx27vis-aic32x4, and more importantly it will ease the moving of audmux into sound/soc/imx as a platform driver later. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/imx/eukrea-tlv320.c | 40 ++++++++++++++++++++++++++++++++++++---- 1 file changed, 36 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 1c1fdd10f73..bfcb6d9768b 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -23,6 +23,7 @@ #include #include #include +#include #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" @@ -97,12 +98,43 @@ static struct platform_device *eukrea_tlv320_snd_device; static int __init eukrea_tlv320_init(void) { int ret; - - if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd() - && !machine_is_eukrea_cpuimx35sd() - && !machine_is_eukrea_cpuimx51sd()) + int int_port = 0, ext_port; + + if (machine_is_eukrea_cpuimx27()) { + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_TFSDIR | + MXC_AUDMUX_V1_PCR_TCLKDIR | + MXC_AUDMUX_V1_PCR_RFSDIR | + MXC_AUDMUX_V1_PCR_RCLKDIR | + MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + MXC_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) + ); + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); + } else if (machine_is_eukrea_cpuimx25sd() || + machine_is_eukrea_cpuimx35sd() || + machine_is_eukrea_cpuimx51sd()) { + ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + mxc_audmux_v2_configure_port(int_port, + MXC_AUDMUX_V2_PTCR_SYN | + MXC_AUDMUX_V2_PTCR_TFSDIR | + MXC_AUDMUX_V2_PTCR_TFSEL(ext_port) | + MXC_AUDMUX_V2_PTCR_TCLKDIR | + MXC_AUDMUX_V2_PTCR_TCSEL(ext_port), + MXC_AUDMUX_V2_PDCR_RXDSEL(ext_port) + ); + mxc_audmux_v2_configure_port(ext_port, + MXC_AUDMUX_V2_PTCR_SYN, + MXC_AUDMUX_V2_PDCR_RXDSEL(int_port) + ); + } else { /* return happy. We might run on a totally different machine */ return 0; + } eukrea_tlv320_snd_device = platform_device_alloc("soc-audio", -1); if (!eukrea_tlv320_snd_device) -- cgit v1.2.3-18-g5258 From 17ec38a8b6d95100a585ed66ccc7bada13e09d0d Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 5 Mar 2012 22:30:50 +0800 Subject: ASoC: imx: move phycore audmux call into ASoC machine driver It moves phycore audmux configuration call from board file into ASoC machine driver phycore-ac97 to ease converting audmux into a platform driver later. It moves phycore audmux configuration call from board file into ASoC machine driver phycore-ac97, so that it gets aligned with wm1133-ev1 and mx27vis-aic32x4, and more importantly it will ease the moving of audmux into sound/soc/imx as a platform driver later. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/imx/phycore-ac97.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c index 6ac12111de6..a59692e740b 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/imx/phycore-ac97.c @@ -18,6 +18,7 @@ #include #include #include +#include static struct snd_soc_card imx_phycore; @@ -50,9 +51,32 @@ static int __init imx_phycore_init(void) { int ret; - if (!machine_is_pcm043() && !machine_is_pca100()) + if (machine_is_pca100()) { + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V1_PCR_TFCSEL(3) | + MXC_AUDMUX_V1_PCR_TCLKDIR | /* clock is output */ + MXC_AUDMUX_V1_PCR_RXDSEL(3)); + mxc_audmux_v1_configure_port(3, + MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V1_PCR_TFCSEL(0) | + MXC_AUDMUX_V1_PCR_TFSDIR | + MXC_AUDMUX_V1_PCR_RXDSEL(0)); + } else if (machine_is_pcm043()) { + mxc_audmux_v2_configure_port(3, + MXC_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V2_PTCR_TFSEL(0) | + MXC_AUDMUX_V2_PTCR_TFSDIR, + MXC_AUDMUX_V2_PDCR_RXDSEL(0)); + mxc_audmux_v2_configure_port(0, + MXC_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V2_PTCR_TCSEL(3) | + MXC_AUDMUX_V2_PTCR_TCLKDIR, /* clock is output */ + MXC_AUDMUX_V2_PDCR_RXDSEL(3)); + } else { /* return happy. We might run on a totally different machine */ return 0; + } imx_phycore_snd_ac97_device = platform_device_alloc("soc-audio", -1); if (!imx_phycore_snd_ac97_device) -- cgit v1.2.3-18-g5258 From 3c77c29c49c6213c55ad8dacc687817b3568c0ce Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 5 Mar 2012 22:30:53 +0800 Subject: ASoC: imx: move audmux driver into sound/soc/imx As audmux becomes a platform driver and its callers are all ASoC machine drivers, there is no reason to keep it in arch folder, so move it to sound/soc/imx. One bonus point would be those ASoC machine drivers stop including mach/audmux.h, since it's been moved to sound/soc/imx/imx-audmux.h. This should be a move to the right direction in terms of single kernel image goal. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 7 + sound/soc/imx/Makefile | 2 + sound/soc/imx/eukrea-tlv320.c | 2 +- sound/soc/imx/imx-audmux.c | 300 ++++++++++++++++++++++++++++++++++++++++ sound/soc/imx/imx-audmux.h | 60 ++++++++ sound/soc/imx/mx27vis-aic32x4.c | 2 +- sound/soc/imx/phycore-ac97.c | 3 +- sound/soc/imx/wm1133-ev1.c | 3 +- 8 files changed, 374 insertions(+), 5 deletions(-) create mode 100644 sound/soc/imx/imx-audmux.c create mode 100644 sound/soc/imx/imx-audmux.h (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index aa4294bf49b..d3b716663d1 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -16,11 +16,15 @@ config SND_MXC_SOC_MX2 select SND_SOC_DMAENGINE_PCM tristate +config SND_SOC_IMX_AUDMUX + tristate + config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 select SND_MXC_SOC_FIQ + select SND_SOC_IMX_AUDMUX help Enable support for audio on the i.MX31ADS with the WM1133-EV1 PMIC board with WM8835x fitted. @@ -30,6 +34,7 @@ config SND_SOC_MX27VIS_AIC32X4 depends on MACH_IMX27_VISSTRIM_M10 && I2C select SND_SOC_TLV320AIC32X4 select SND_MXC_SOC_MX2 + select SND_SOC_IMX_AUDMUX help Say Y if you want to add support for SoC audio on Visstrim SM10 board with TLV320AIC32X4 codec. @@ -40,6 +45,7 @@ config SND_SOC_PHYCORE_AC97 select SND_SOC_AC97_BUS select SND_SOC_WM9712 select SND_MXC_SOC_FIQ + select SND_SOC_IMX_AUDMUX help Say Y if you want to add support for SoC audio on Phytec phyCORE and phyCARD boards in AC97 mode @@ -53,6 +59,7 @@ config SND_SOC_EUKREA_TLV320 depends on I2C select SND_SOC_TLV320AIC23 select SND_MXC_SOC_FIQ + select SND_SOC_IMX_AUDMUX help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index d6d609ba7e2..5c40541b831 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -2,10 +2,12 @@ snd-soc-imx-objs := imx-ssi.o snd-soc-imx-fiq-objs := imx-pcm-fiq.o snd-soc-imx-mx2-objs := imx-pcm-dma-mx2.o +snd-soc-imx-audmux-objs := imx-audmux.o obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o obj-$(CONFIG_SND_MXC_SOC_FIQ) += snd-soc-imx-fiq.o obj-$(CONFIG_SND_MXC_SOC_MX2) += snd-soc-imx-mx2.o +obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index bfcb6d9768b..b375ed4541f 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -23,10 +23,10 @@ #include #include #include -#include #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "imx-audmux.h" #define CODEC_CLOCK 12000000 diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c new file mode 100644 index 00000000000..7b162662fe6 --- /dev/null +++ b/sound/soc/imx/imx-audmux.c @@ -0,0 +1,300 @@ +/* + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * Copyright 2009 Pengutronix, Sascha Hauer + * + * Initial development of this code was funded by + * Phytec Messtechnik GmbH, http://www.phytec.de + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include "imx-audmux.h" + +#define DRIVER_NAME "imx-audmux" + +static struct clk *audmux_clk; +static void __iomem *audmux_base; + +#define MXC_AUDMUX_V2_PTCR(x) ((x) * 8) +#define MXC_AUDMUX_V2_PDCR(x) ((x) * 8 + 4) + +#ifdef CONFIG_DEBUG_FS +static struct dentry *audmux_debugfs_root; + +static int audmux_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +/* There is an annoying discontinuity in the SSI numbering with regard + * to the Linux number of the devices */ +static const char *audmux_port_string(int port) +{ + switch (port) { + case MX31_AUDMUX_PORT1_SSI0: + return "imx-ssi.0"; + case MX31_AUDMUX_PORT2_SSI1: + return "imx-ssi.1"; + case MX31_AUDMUX_PORT3_SSI_PINS_3: + return "SSI3"; + case MX31_AUDMUX_PORT4_SSI_PINS_4: + return "SSI4"; + case MX31_AUDMUX_PORT5_SSI_PINS_5: + return "SSI5"; + case MX31_AUDMUX_PORT6_SSI_PINS_6: + return "SSI6"; + default: + return "UNKNOWN"; + } +} + +static ssize_t audmux_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + int port = (int)file->private_data; + u32 pdcr, ptcr; + + if (!buf) + return -ENOMEM; + + if (audmux_clk) + clk_enable(audmux_clk); + + ptcr = readl(audmux_base + MXC_AUDMUX_V2_PTCR(port)); + pdcr = readl(audmux_base + MXC_AUDMUX_V2_PDCR(port)); + + if (audmux_clk) + clk_disable(audmux_clk); + + ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", + pdcr, ptcr); + + if (ptcr & MXC_AUDMUX_V2_PTCR_TFSDIR) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "TxFS output from %s, ", + audmux_port_string((ptcr >> 27) & 0x7)); + else + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "TxFS input, "); + + if (ptcr & MXC_AUDMUX_V2_PTCR_TCLKDIR) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "TxClk output from %s", + audmux_port_string((ptcr >> 22) & 0x7)); + else + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "TxClk input"); + + ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + + if (ptcr & MXC_AUDMUX_V2_PTCR_SYN) { + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "Port is symmetric"); + } else { + if (ptcr & MXC_AUDMUX_V2_PTCR_RFSDIR) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "RxFS output from %s, ", + audmux_port_string((ptcr >> 17) & 0x7)); + else + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "RxFS input, "); + + if (ptcr & MXC_AUDMUX_V2_PTCR_RCLKDIR) + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "RxClk output from %s", + audmux_port_string((ptcr >> 12) & 0x7)); + else + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "RxClk input"); + } + + ret += snprintf(buf + ret, PAGE_SIZE - ret, + "\nData received from %s\n", + audmux_port_string((pdcr >> 13) & 0x7)); + + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + + kfree(buf); + + return ret; +} + +static const struct file_operations audmux_debugfs_fops = { + .open = audmux_open_file, + .read = audmux_read_file, + .llseek = default_llseek, +}; + +static void __init audmux_debugfs_init(void) +{ + int i; + char buf[20]; + + audmux_debugfs_root = debugfs_create_dir("audmux", NULL); + if (!audmux_debugfs_root) { + pr_warning("Failed to create AUDMUX debugfs root\n"); + return; + } + + for (i = 1; i < 8; i++) { + snprintf(buf, sizeof(buf), "ssi%d", i); + if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, + (void *)i, &audmux_debugfs_fops)) + pr_warning("Failed to create AUDMUX port %d debugfs file\n", + i); + } +} + +static void __exit audmux_debugfs_remove(void) +{ + debugfs_remove_recursive(audmux_debugfs_root); +} +#else +static inline void audmux_debugfs_init(void) +{ +} + +static inline void audmux_debugfs_remove(void) +{ +} +#endif + +enum imx_audmux_type { + IMX21_AUDMUX, + IMX31_AUDMUX, +} audmux_type; + +static struct platform_device_id imx_audmux_ids[] = { + { + .name = "imx21-audmux", + .driver_data = IMX21_AUDMUX, + }, { + .name = "imx31-audmux", + .driver_data = IMX31_AUDMUX, + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(platform, imx_audmux_ids); + +static const uint8_t port_mapping[] = { + 0x0, 0x4, 0x8, 0x10, 0x14, 0x1c, +}; + +int mxc_audmux_v1_configure_port(unsigned int port, unsigned int pcr) +{ + if (audmux_type != IMX21_AUDMUX) + return -EINVAL; + + if (!audmux_base) + return -ENOSYS; + + if (port >= ARRAY_SIZE(port_mapping)) + return -EINVAL; + + writel(pcr, audmux_base + port_mapping[port]); + + return 0; +} +EXPORT_SYMBOL_GPL(mxc_audmux_v1_configure_port); + +int mxc_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, + unsigned int pdcr) +{ + if (audmux_type != IMX31_AUDMUX) + return -EINVAL; + + if (!audmux_base) + return -ENOSYS; + + if (audmux_clk) + clk_enable(audmux_clk); + + writel(ptcr, audmux_base + MXC_AUDMUX_V2_PTCR(port)); + writel(pdcr, audmux_base + MXC_AUDMUX_V2_PDCR(port)); + + if (audmux_clk) + clk_disable(audmux_clk); + + return 0; +} +EXPORT_SYMBOL_GPL(mxc_audmux_v2_configure_port); + +static int __init imx_audmux_probe(struct platform_device *pdev) +{ + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + audmux_base = devm_request_and_ioremap(&pdev->dev, res); + if (!audmux_base) + return -EADDRNOTAVAIL; + + audmux_clk = clk_get(&pdev->dev, "audmux"); + if (IS_ERR(audmux_clk)) { + dev_dbg(&pdev->dev, "cannot get clock: %ld\n", + PTR_ERR(audmux_clk)); + audmux_clk = NULL; + } + + audmux_type = pdev->id_entry->driver_data; + if (audmux_type == IMX31_AUDMUX) + audmux_debugfs_init(); + + return 0; +} + +static int __exit imx_audmux_remove(struct platform_device *pdev) +{ + if (audmux_type == IMX31_AUDMUX) + audmux_debugfs_remove(); + clk_put(audmux_clk); + + return 0; +} + +static struct platform_driver imx_audmux_driver = { + .probe = imx_audmux_probe, + .remove = __exit_p(imx_audmux_remove), + .id_table = imx_audmux_ids, + .driver = { + .name = DRIVER_NAME, + .owner = THIS_MODULE, + } +}; + +static int __init imx_audmux_init(void) +{ + return platform_driver_register(&imx_audmux_driver); +} +subsys_initcall(imx_audmux_init); + +static void __exit imx_audmux_exit(void) +{ + platform_driver_unregister(&imx_audmux_driver); +} +module_exit(imx_audmux_exit); + +MODULE_DESCRIPTION("Freescale i.MX AUDMUX driver"); +MODULE_AUTHOR("Sascha Hauer "); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRIVER_NAME); diff --git a/sound/soc/imx/imx-audmux.h b/sound/soc/imx/imx-audmux.h new file mode 100644 index 00000000000..5136d9483f4 --- /dev/null +++ b/sound/soc/imx/imx-audmux.h @@ -0,0 +1,60 @@ +#ifndef __IMX_AUDMUX_H +#define __IMX_AUDMUX_H + +#define MX27_AUDMUX_HPCR1_SSI0 0 +#define MX27_AUDMUX_HPCR2_SSI1 1 +#define MX27_AUDMUX_HPCR3_SSI_PINS_4 2 +#define MX27_AUDMUX_PPCR1_SSI_PINS_1 3 +#define MX27_AUDMUX_PPCR2_SSI_PINS_2 4 +#define MX27_AUDMUX_PPCR3_SSI_PINS_3 5 + +#define MX31_AUDMUX_PORT1_SSI0 0 +#define MX31_AUDMUX_PORT2_SSI1 1 +#define MX31_AUDMUX_PORT3_SSI_PINS_3 2 +#define MX31_AUDMUX_PORT4_SSI_PINS_4 3 +#define MX31_AUDMUX_PORT5_SSI_PINS_5 4 +#define MX31_AUDMUX_PORT6_SSI_PINS_6 5 + +#define MX51_AUDMUX_PORT1_SSI0 0 +#define MX51_AUDMUX_PORT2_SSI1 1 +#define MX51_AUDMUX_PORT3 2 +#define MX51_AUDMUX_PORT4 3 +#define MX51_AUDMUX_PORT5 4 +#define MX51_AUDMUX_PORT6 5 +#define MX51_AUDMUX_PORT7 6 + +/* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ +#define MXC_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) +#define MXC_AUDMUX_V1_PCR_INMEN (1 << 8) +#define MXC_AUDMUX_V1_PCR_TXRXEN (1 << 10) +#define MXC_AUDMUX_V1_PCR_SYN (1 << 12) +#define MXC_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) +#define MXC_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) +#define MXC_AUDMUX_V1_PCR_RCLKDIR (1 << 24) +#define MXC_AUDMUX_V1_PCR_RFSDIR (1 << 25) +#define MXC_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) +#define MXC_AUDMUX_V1_PCR_TCLKDIR (1 << 30) +#define MXC_AUDMUX_V1_PCR_TFSDIR (1 << 31) + +/* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ +#define MXC_AUDMUX_V2_PTCR_TFSDIR (1 << 31) +#define MXC_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) +#define MXC_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) +#define MXC_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) +#define MXC_AUDMUX_V2_PTCR_RFSDIR (1 << 21) +#define MXC_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) +#define MXC_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) +#define MXC_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) +#define MXC_AUDMUX_V2_PTCR_SYN (1 << 11) + +#define MXC_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) +#define MXC_AUDMUX_V2_PDCR_TXRXEN (1 << 12) +#define MXC_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) +#define MXC_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) + +int mxc_audmux_v1_configure_port(unsigned int port, unsigned int pcr); + +int mxc_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, + unsigned int pdcr); + +#endif /* __IMX_AUDMUX_H */ diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 155899c08c0..dbfad0f6251 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -32,11 +32,11 @@ #include #include #include -#include #include #include "../codecs/tlv320aic32x4.h" #include "imx-ssi.h" +#include "imx-audmux.h" #define MX27VIS_AMP_GAIN 0 #define MX27VIS_AMP_MUTE 1 diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c index a59692e740b..7dab077f9c3 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/imx/phycore-ac97.c @@ -18,7 +18,8 @@ #include #include #include -#include + +#include "imx-audmux.h" static struct snd_soc_card imx_phycore; diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c index 37480c90e99..15056d6a164 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/imx/wm1133-ev1.c @@ -21,10 +21,9 @@ #include #include -#include - #include "imx-ssi.h" #include "../codecs/wm8350.h" +#include "imx-audmux.h" /* There is a silicon mic on the board optionally connected via a solder pad * SP1. Define this to enable it. -- cgit v1.2.3-18-g5258 From af4872fb39301bbe196d0778f80d22ec51d8884b Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 5 Mar 2012 22:30:54 +0800 Subject: ASoC: imx: rename audmux prefix mxc to imx It renames the legacy name mxc used in audmux function and macro to imx. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/imx/eukrea-tlv320.c | 44 +++++++++++++++---------------- sound/soc/imx/imx-audmux.c | 30 ++++++++++----------- sound/soc/imx/imx-audmux.h | 58 ++++++++++++++++++++--------------------- sound/soc/imx/mx27vis-aic32x4.c | 18 ++++++------- sound/soc/imx/phycore-ac97.c | 40 ++++++++++++++-------------- sound/soc/imx/wm1133-ev1.c | 22 ++++++++-------- 6 files changed, 106 insertions(+), 106 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index b375ed4541f..7d4475cfdb2 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -101,35 +101,35 @@ static int __init eukrea_tlv320_init(void) int int_port = 0, ext_port; if (machine_is_eukrea_cpuimx27()) { - mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, - MXC_AUDMUX_V1_PCR_SYN | - MXC_AUDMUX_V1_PCR_TFSDIR | - MXC_AUDMUX_V1_PCR_TCLKDIR | - MXC_AUDMUX_V1_PCR_RFSDIR | - MXC_AUDMUX_V1_PCR_RCLKDIR | - MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | - MXC_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | - MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_TFSDIR | + IMX_AUDMUX_V1_PCR_TCLKDIR | + IMX_AUDMUX_V1_PCR_RFSDIR | + IMX_AUDMUX_V1_PCR_RCLKDIR | + IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) ); - mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, - MXC_AUDMUX_V1_PCR_SYN | - MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || machine_is_eukrea_cpuimx51sd()) { ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; - mxc_audmux_v2_configure_port(int_port, - MXC_AUDMUX_V2_PTCR_SYN | - MXC_AUDMUX_V2_PTCR_TFSDIR | - MXC_AUDMUX_V2_PTCR_TFSEL(ext_port) | - MXC_AUDMUX_V2_PTCR_TCLKDIR | - MXC_AUDMUX_V2_PTCR_TCSEL(ext_port), - MXC_AUDMUX_V2_PDCR_RXDSEL(ext_port) + imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port), + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port) ); - mxc_audmux_v2_configure_port(ext_port, - MXC_AUDMUX_V2_PTCR_SYN, - MXC_AUDMUX_V2_PDCR_RXDSEL(int_port) + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { /* return happy. We might run on a totally different machine */ diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index 7b162662fe6..87f8768e1cd 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -32,8 +32,8 @@ static struct clk *audmux_clk; static void __iomem *audmux_base; -#define MXC_AUDMUX_V2_PTCR(x) ((x) * 8) -#define MXC_AUDMUX_V2_PDCR(x) ((x) * 8 + 4) +#define IMX_AUDMUX_V2_PTCR(x) ((x) * 8) +#define IMX_AUDMUX_V2_PDCR(x) ((x) * 8 + 4) #ifdef CONFIG_DEBUG_FS static struct dentry *audmux_debugfs_root; @@ -80,8 +80,8 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (audmux_clk) clk_enable(audmux_clk); - ptcr = readl(audmux_base + MXC_AUDMUX_V2_PTCR(port)); - pdcr = readl(audmux_base + MXC_AUDMUX_V2_PDCR(port)); + ptcr = readl(audmux_base + IMX_AUDMUX_V2_PTCR(port)); + pdcr = readl(audmux_base + IMX_AUDMUX_V2_PDCR(port)); if (audmux_clk) clk_disable(audmux_clk); @@ -89,7 +89,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); - if (ptcr & MXC_AUDMUX_V2_PTCR_TFSDIR) + if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) ret += snprintf(buf + ret, PAGE_SIZE - ret, "TxFS output from %s, ", audmux_port_string((ptcr >> 27) & 0x7)); @@ -97,7 +97,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, ret += snprintf(buf + ret, PAGE_SIZE - ret, "TxFS input, "); - if (ptcr & MXC_AUDMUX_V2_PTCR_TCLKDIR) + if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) ret += snprintf(buf + ret, PAGE_SIZE - ret, "TxClk output from %s", audmux_port_string((ptcr >> 22) & 0x7)); @@ -107,11 +107,11 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); - if (ptcr & MXC_AUDMUX_V2_PTCR_SYN) { + if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { ret += snprintf(buf + ret, PAGE_SIZE - ret, "Port is symmetric"); } else { - if (ptcr & MXC_AUDMUX_V2_PTCR_RFSDIR) + if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) ret += snprintf(buf + ret, PAGE_SIZE - ret, "RxFS output from %s, ", audmux_port_string((ptcr >> 17) & 0x7)); @@ -119,7 +119,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, ret += snprintf(buf + ret, PAGE_SIZE - ret, "RxFS input, "); - if (ptcr & MXC_AUDMUX_V2_PTCR_RCLKDIR) + if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) ret += snprintf(buf + ret, PAGE_SIZE - ret, "RxClk output from %s", audmux_port_string((ptcr >> 12) & 0x7)); @@ -201,7 +201,7 @@ static const uint8_t port_mapping[] = { 0x0, 0x4, 0x8, 0x10, 0x14, 0x1c, }; -int mxc_audmux_v1_configure_port(unsigned int port, unsigned int pcr) +int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr) { if (audmux_type != IMX21_AUDMUX) return -EINVAL; @@ -216,9 +216,9 @@ int mxc_audmux_v1_configure_port(unsigned int port, unsigned int pcr) return 0; } -EXPORT_SYMBOL_GPL(mxc_audmux_v1_configure_port); +EXPORT_SYMBOL_GPL(imx_audmux_v1_configure_port); -int mxc_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, +int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, unsigned int pdcr) { if (audmux_type != IMX31_AUDMUX) @@ -230,15 +230,15 @@ int mxc_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, if (audmux_clk) clk_enable(audmux_clk); - writel(ptcr, audmux_base + MXC_AUDMUX_V2_PTCR(port)); - writel(pdcr, audmux_base + MXC_AUDMUX_V2_PDCR(port)); + writel(ptcr, audmux_base + IMX_AUDMUX_V2_PTCR(port)); + writel(pdcr, audmux_base + IMX_AUDMUX_V2_PDCR(port)); if (audmux_clk) clk_disable(audmux_clk); return 0; } -EXPORT_SYMBOL_GPL(mxc_audmux_v2_configure_port); +EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); static int __init imx_audmux_probe(struct platform_device *pdev) { diff --git a/sound/soc/imx/imx-audmux.h b/sound/soc/imx/imx-audmux.h index 5136d9483f4..04ebbab8d7b 100644 --- a/sound/soc/imx/imx-audmux.h +++ b/sound/soc/imx/imx-audmux.h @@ -24,37 +24,37 @@ #define MX51_AUDMUX_PORT7 6 /* Register definitions for the i.MX21/27 Digital Audio Multiplexer */ -#define MXC_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) -#define MXC_AUDMUX_V1_PCR_INMEN (1 << 8) -#define MXC_AUDMUX_V1_PCR_TXRXEN (1 << 10) -#define MXC_AUDMUX_V1_PCR_SYN (1 << 12) -#define MXC_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) -#define MXC_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) -#define MXC_AUDMUX_V1_PCR_RCLKDIR (1 << 24) -#define MXC_AUDMUX_V1_PCR_RFSDIR (1 << 25) -#define MXC_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) -#define MXC_AUDMUX_V1_PCR_TCLKDIR (1 << 30) -#define MXC_AUDMUX_V1_PCR_TFSDIR (1 << 31) +#define IMX_AUDMUX_V1_PCR_INMMASK(x) ((x) & 0xff) +#define IMX_AUDMUX_V1_PCR_INMEN (1 << 8) +#define IMX_AUDMUX_V1_PCR_TXRXEN (1 << 10) +#define IMX_AUDMUX_V1_PCR_SYN (1 << 12) +#define IMX_AUDMUX_V1_PCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V1_PCR_RFCSEL(x) (((x) & 0xf) << 20) +#define IMX_AUDMUX_V1_PCR_RCLKDIR (1 << 24) +#define IMX_AUDMUX_V1_PCR_RFSDIR (1 << 25) +#define IMX_AUDMUX_V1_PCR_TFCSEL(x) (((x) & 0xf) << 26) +#define IMX_AUDMUX_V1_PCR_TCLKDIR (1 << 30) +#define IMX_AUDMUX_V1_PCR_TFSDIR (1 << 31) /* Register definitions for the i.MX25/31/35/51 Digital Audio Multiplexer */ -#define MXC_AUDMUX_V2_PTCR_TFSDIR (1 << 31) -#define MXC_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) -#define MXC_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) -#define MXC_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) -#define MXC_AUDMUX_V2_PTCR_RFSDIR (1 << 21) -#define MXC_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) -#define MXC_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) -#define MXC_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) -#define MXC_AUDMUX_V2_PTCR_SYN (1 << 11) - -#define MXC_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) -#define MXC_AUDMUX_V2_PDCR_TXRXEN (1 << 12) -#define MXC_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) -#define MXC_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) - -int mxc_audmux_v1_configure_port(unsigned int port, unsigned int pcr); - -int mxc_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, +#define IMX_AUDMUX_V2_PTCR_TFSDIR (1 << 31) +#define IMX_AUDMUX_V2_PTCR_TFSEL(x) (((x) & 0xf) << 27) +#define IMX_AUDMUX_V2_PTCR_TCLKDIR (1 << 26) +#define IMX_AUDMUX_V2_PTCR_TCSEL(x) (((x) & 0xf) << 22) +#define IMX_AUDMUX_V2_PTCR_RFSDIR (1 << 21) +#define IMX_AUDMUX_V2_PTCR_RFSEL(x) (((x) & 0xf) << 17) +#define IMX_AUDMUX_V2_PTCR_RCLKDIR (1 << 16) +#define IMX_AUDMUX_V2_PTCR_RCSEL(x) (((x) & 0xf) << 12) +#define IMX_AUDMUX_V2_PTCR_SYN (1 << 11) + +#define IMX_AUDMUX_V2_PDCR_RXDSEL(x) (((x) & 0x7) << 13) +#define IMX_AUDMUX_V2_PDCR_TXRXEN (1 << 12) +#define IMX_AUDMUX_V2_PDCR_MODE(x) (((x) & 0x3) << 8) +#define IMX_AUDMUX_V2_PDCR_INMMASK(x) ((x) & 0xff) + +int imx_audmux_v1_configure_port(unsigned int port, unsigned int pcr); + +int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, unsigned int pdcr); #endif /* __IMX_AUDMUX_H */ diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index dbfad0f6251..976f857151f 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -207,16 +207,16 @@ static int __init mx27vis_aic32x4_init(void) } /* Connect SSI0 as clock slave to SSI1 external pins */ - mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, - MXC_AUDMUX_V1_PCR_SYN | - MXC_AUDMUX_V1_PCR_TFSDIR | - MXC_AUDMUX_V1_PCR_TCLKDIR | - MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) | - MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_TFSDIR | + IMX_AUDMUX_V1_PCR_TCLKDIR | + IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) ); - mxc_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1, - MXC_AUDMUX_V1_PCR_SYN | - MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + imx_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) ); ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c index 7dab077f9c3..f8da6dd115e 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/imx/phycore-ac97.c @@ -53,27 +53,27 @@ static int __init imx_phycore_init(void) int ret; if (machine_is_pca100()) { - mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, - MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ - MXC_AUDMUX_V1_PCR_TFCSEL(3) | - MXC_AUDMUX_V1_PCR_TCLKDIR | /* clock is output */ - MXC_AUDMUX_V1_PCR_RXDSEL(3)); - mxc_audmux_v1_configure_port(3, - MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ - MXC_AUDMUX_V1_PCR_TFCSEL(0) | - MXC_AUDMUX_V1_PCR_TFSDIR | - MXC_AUDMUX_V1_PCR_RXDSEL(0)); + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + IMX_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + IMX_AUDMUX_V1_PCR_TFCSEL(3) | + IMX_AUDMUX_V1_PCR_TCLKDIR | /* clock is output */ + IMX_AUDMUX_V1_PCR_RXDSEL(3)); + imx_audmux_v1_configure_port(3, + IMX_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + IMX_AUDMUX_V1_PCR_TFCSEL(0) | + IMX_AUDMUX_V1_PCR_TFSDIR | + IMX_AUDMUX_V1_PCR_RXDSEL(0)); } else if (machine_is_pcm043()) { - mxc_audmux_v2_configure_port(3, - MXC_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ - MXC_AUDMUX_V2_PTCR_TFSEL(0) | - MXC_AUDMUX_V2_PTCR_TFSDIR, - MXC_AUDMUX_V2_PDCR_RXDSEL(0)); - mxc_audmux_v2_configure_port(0, - MXC_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ - MXC_AUDMUX_V2_PTCR_TCSEL(3) | - MXC_AUDMUX_V2_PTCR_TCLKDIR, /* clock is output */ - MXC_AUDMUX_V2_PDCR_RXDSEL(3)); + imx_audmux_v2_configure_port(3, + IMX_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ + IMX_AUDMUX_V2_PTCR_TFSEL(0) | + IMX_AUDMUX_V2_PTCR_TFSDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(0)); + imx_audmux_v2_configure_port(0, + IMX_AUDMUX_V2_PTCR_SYN | /* 4wire mode */ + IMX_AUDMUX_V2_PTCR_TCSEL(3) | + IMX_AUDMUX_V2_PTCR_TCLKDIR, /* clock is output */ + IMX_AUDMUX_V2_PDCR_RXDSEL(3)); } else { /* return happy. We might run on a totally different machine */ return 0; diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c index 15056d6a164..fe54a69073e 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/imx/wm1133-ev1.c @@ -267,17 +267,17 @@ static int __init wm1133_ev1_audio_init(void) unsigned int ptcr, pdcr; /* SSI0 mastered by port 5 */ - ptcr = MXC_AUDMUX_V2_PTCR_SYN | - MXC_AUDMUX_V2_PTCR_TFSDIR | - MXC_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) | - MXC_AUDMUX_V2_PTCR_TCLKDIR | - MXC_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); - pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); - mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr); - - ptcr = MXC_AUDMUX_V2_PTCR_SYN; - pdcr = MXC_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0); - mxc_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr); + ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr); + + ptcr = IMX_AUDMUX_V2_PTCR_SYN; + pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0); + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr); wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1); if (!wm1133_ev1_snd_device) -- cgit v1.2.3-18-g5258 From 4762fbab0b1cf1d4f3e02b78351dc6fa59ca564e Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 5 Mar 2012 22:30:55 +0800 Subject: ASoC: imx: separate imx-pcm bits from imx-ssi driver Currently the imx-ssi.c[h] accommodates the imx-pcm common bits which are shared between imx-pcm-dma-mx2 and imx-pcm-fiq drivers. It assumes that imx-pcm-dma-mx2 and imx-pcm-fiq will always be used together with imx-ssi driver. However this becomes untrue when we see that driver sound/soc/fsl/fsl_ssi could possibly work with imx-pcm-dma-mx2 too. The patch moves the imx-pcm common bits from imx-ssi.c[h] into new files imx-pcm.c[h], and let imx-pcm-dma-mx2 and imx-pcm-fiq drivers build it in, so that imx-pcm-dma-mx2 can work with no dependency on imx-ssi driver. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 7 ++- sound/soc/imx/Makefile | 9 ++-- sound/soc/imx/imx-pcm-dma-mx2.c | 2 +- sound/soc/imx/imx-pcm.c | 105 ++++++++++++++++++++++++++++++++++++++++ sound/soc/imx/imx-pcm.h | 32 ++++++++++++ sound/soc/imx/imx-ssi.c | 88 --------------------------------- sound/soc/imx/imx-ssi.h | 16 +----- 7 files changed, 150 insertions(+), 109 deletions(-) create mode 100644 sound/soc/imx/imx-pcm.c create mode 100644 sound/soc/imx/imx-pcm.h (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index d3b716663d1..2566032a490 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -8,13 +8,18 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC +config SND_SOC_IMX_PCM + tristate + config SND_MXC_SOC_FIQ - select FIQ tristate + select FIQ + select SND_SOC_IMX_PCM config SND_MXC_SOC_MX2 select SND_SOC_DMAENGINE_PCM tristate + select SND_SOC_IMX_PCM config SND_SOC_IMX_AUDMUX tristate diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index 5c40541b831..e9ed362636e 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,14 +1,15 @@ # i.MX Platform Support snd-soc-imx-objs := imx-ssi.o -snd-soc-imx-fiq-objs := imx-pcm-fiq.o -snd-soc-imx-mx2-objs := imx-pcm-dma-mx2.o snd-soc-imx-audmux-objs := imx-audmux.o obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o -obj-$(CONFIG_SND_MXC_SOC_FIQ) += snd-soc-imx-fiq.o -obj-$(CONFIG_SND_MXC_SOC_MX2) += snd-soc-imx-mx2.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o +obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o +snd-soc-imx-pcm-y := imx-pcm.o +snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_FIQ) += imx-pcm-fiq.o +snd-soc-imx-pcm-$(CONFIG_SND_MXC_SOC_MX2) += imx-pcm-dma-mx2.o + # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 471e2218c97..e43c8fa2788 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -31,7 +31,7 @@ #include -#include "imx-ssi.h" +#include "imx-pcm.h" static bool filter(struct dma_chan *chan, void *param) { diff --git a/sound/soc/imx/imx-pcm.c b/sound/soc/imx/imx-pcm.c new file mode 100644 index 00000000000..93dc360b177 --- /dev/null +++ b/sound/soc/imx/imx-pcm.c @@ -0,0 +1,105 @@ +/* + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include +#include +#include +#include +#include "imx-pcm.h" + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); + + pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + return ret; +} +EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); + +static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = IMX_SSI_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); + +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &imx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} +EXPORT_SYMBOL_GPL(imx_pcm_new); + +void imx_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} +EXPORT_SYMBOL_GPL(imx_pcm_free); diff --git a/sound/soc/imx/imx-pcm.h b/sound/soc/imx/imx-pcm.h new file mode 100644 index 00000000000..b5f5c3acf34 --- /dev/null +++ b/sound/soc/imx/imx-pcm.h @@ -0,0 +1,32 @@ +/* + * Copyright 2009 Sascha Hauer + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _IMX_PCM_H +#define _IMX_PCM_H + +/* + * Do not change this as the FIQ handler depends on this size + */ +#define IMX_SSI_DMABUF_SIZE (64 * 1024) + +struct imx_pcm_dma_params { + int dma; + unsigned long dma_addr; + int burstsize; +}; + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma); +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); +void imx_pcm_free(struct snd_pcm *pcm); + +#endif /* _IMX_PCM_H */ diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 25c623115a9..9203cdd0a15 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -363,94 +363,6 @@ static const struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { .trigger = imx_ssi_trigger, }; -int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - ret = dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - - pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); - return ret; -} -EXPORT_SYMBOL_GPL(snd_imx_pcm_mmap); - -static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = IMX_SSI_DMABUF_SIZE; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - - return 0; -} - -static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); - -int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &imx_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = imx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = imx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - -out: - return ret; -} -EXPORT_SYMBOL_GPL(imx_pcm_new); - -void imx_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} -EXPORT_SYMBOL_GPL(imx_pcm_free); - static int imx_ssi_dai_probe(struct snd_soc_dai *dai) { struct imx_ssi *ssi = dev_get_drvdata(dai->dev); diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index 1072dfb53e4..5744e86ca87 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -187,12 +187,7 @@ #include #include - -struct imx_pcm_dma_params { - int dma; - unsigned long dma_addr; - int burstsize; -}; +#include "imx-pcm.h" struct imx_ssi { struct platform_device *ac97_dev; @@ -218,13 +213,4 @@ struct imx_ssi { struct platform_device *soc_platform_pdev_fiq; }; -int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); -int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); -void imx_pcm_free(struct snd_pcm *pcm); - -/* - * Do not change this as the FIQ handler depends on this size - */ -#define IMX_SSI_DMABUF_SIZE (64 * 1024) - #endif /* _IMX_SSI_H */ -- cgit v1.2.3-18-g5258 From 56cea3f1e7db0ccde9e2ac66df2f920c73c419ef Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 5 Mar 2012 22:30:56 +0800 Subject: ASoC: imx: add an explicit Kconfig option for imx-ssi driver Currently ASoC:imx uses menuconfig option SND_IMX_SOC selects imx-ssi driver, and it works because all the machine driver covered by the menuconfig need to build imx-ssi driver in. However, it will not work any more if we have a imx based machine driver going into the menuconfig while working with fsl_ssi driver (sound/soc/fsl/fsl_ssi.c) rather than imx-ssi one. The patch adds an explicit Kconfig option SND_SOC_IMX_SSI for imx-ssi driver, so that it can be selected independently from the menuconfig option SND_IMX_SOC. Signed-off-by: Shawn Guo Acked-by: Sascha Hauer Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 7 +++++++ sound/soc/imx/Makefile | 4 ++-- 2 files changed, 9 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 2566032a490..810acaa0900 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -8,6 +8,9 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC +config SND_SOC_IMX_SSI + tristate + config SND_SOC_IMX_PCM tristate @@ -30,6 +33,7 @@ config SND_MXC_SOC_WM1133_EV1 select SND_SOC_WM8350 select SND_MXC_SOC_FIQ select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI help Enable support for audio on the i.MX31ADS with the WM1133-EV1 PMIC board with WM8835x fitted. @@ -40,6 +44,7 @@ config SND_SOC_MX27VIS_AIC32X4 select SND_SOC_TLV320AIC32X4 select SND_MXC_SOC_MX2 select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Visstrim SM10 board with TLV320AIC32X4 codec. @@ -51,6 +56,7 @@ config SND_SOC_PHYCORE_AC97 select SND_SOC_WM9712 select SND_MXC_SOC_FIQ select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI help Say Y if you want to add support for SoC audio on Phytec phyCORE and phyCARD boards in AC97 mode @@ -65,6 +71,7 @@ config SND_SOC_EUKREA_TLV320 select SND_SOC_TLV320AIC23 select SND_MXC_SOC_FIQ select SND_SOC_IMX_AUDMUX + select SND_SOC_IMX_SSI help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index e9ed362636e..f5db3e92d0d 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,8 +1,8 @@ # i.MX Platform Support -snd-soc-imx-objs := imx-ssi.o +snd-soc-imx-ssi-objs := imx-ssi.o snd-soc-imx-audmux-objs := imx-audmux.o -obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o +obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o -- cgit v1.2.3-18-g5258 From 4752a887190ff38175be47aae26a821e8941b96e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 4 Mar 2012 02:16:01 +0000 Subject: ASoC: wm8994: Use audio mode for jack detection when system is active When we are out of system sleep always use audio mode for jack detection in order to avoid potential performance issues handing off between modes. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 77 +++++++++++++++++++++++++++++------------------ sound/soc/codecs/wm8994.h | 1 + 2 files changed, 49 insertions(+), 29 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2417ef9316e..bc12d097ef0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -685,8 +685,6 @@ SOC_SINGLE_TLV("MIXINL IN1RP Boost Volume", WM8994_INPUT_MIXER_1, 8, 1, 0, static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - u16 old = snd_soc_read(codec, WM8994_ANTIPOP_2) - & WM1811_JACKDET_MODE_MASK; if (!wm8994->jackdet || !wm8994->jack_cb) return; @@ -694,28 +692,17 @@ static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode) if (wm8994->active_refcount) mode = WM1811_JACKDET_MODE_AUDIO; - if (mode == old) + if (mode == wm8994->jackdet_mode) return; - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, mode); - - switch (mode) { - case WM1811_JACKDET_MODE_MIC: - case WM1811_JACKDET_MODE_AUDIO: - switch (old) { - case WM1811_JACKDET_MODE_MIC: - case WM1811_JACKDET_MODE_AUDIO: - break; - default: - msleep(2); - break; - } + wm8994->jackdet_mode = mode; - default: - break; - } + /* Always use audio mode to detect while the system is active */ + if (mode != WM1811_JACKDET_MODE_NONE) + mode = WM1811_JACKDET_MODE_AUDIO; + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, mode); } static void active_reference(struct snd_soc_codec *codec) @@ -2749,7 +2736,7 @@ static struct snd_soc_dai_driver wm8994_dai[] = { }; #ifdef CONFIG_PM -static int wm8994_suspend(struct snd_soc_codec *codec) +static int wm8994_codec_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -2783,7 +2770,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec) return 0; } -static int wm8994_resume(struct snd_soc_codec *codec) +static int wm8994_codec_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; @@ -2842,8 +2829,8 @@ static int wm8994_resume(struct snd_soc_codec *codec) return 0; } #else -#define wm8994_suspend NULL -#define wm8994_resume NULL +#define wm8994_codec_suspend NULL +#define wm8994_codec_resume NULL #endif static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) @@ -3955,8 +3942,8 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { .probe = wm8994_codec_probe, .remove = wm8994_codec_remove, - .suspend = wm8994_suspend, - .resume = wm8994_resume, + .suspend = wm8994_codec_suspend, + .resume = wm8994_codec_resume, .set_bias_level = wm8994_set_bias_level, }; @@ -3983,11 +3970,43 @@ static int __devexit wm8994_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int wm8994_suspend(struct device *dev) +{ + struct wm8994_priv *wm8994 = dev_get_drvdata(dev); + + /* Drop down to power saving mode when system is suspended */ + if (wm8994->jackdet && !wm8994->active_refcount) + regmap_update_bits(wm8994->wm8994->regmap, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, + wm8994->jackdet_mode); + + return 0; +} + +static int wm8994_resume(struct device *dev) +{ + struct wm8994_priv *wm8994 = dev_get_drvdata(dev); + + if (wm8994->jackdet && wm8994->jack_cb) + regmap_update_bits(wm8994->wm8994->regmap, WM8994_ANTIPOP_2, + WM1811_JACKDET_MODE_MASK, + WM1811_JACKDET_MODE_AUDIO); + + return 0; +} +#endif + +static const struct dev_pm_ops wm8994_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(wm8994_suspend, wm8994_resume) +}; + static struct platform_driver wm8994_codec_driver = { .driver = { - .name = "wm8994-codec", - .owner = THIS_MODULE, - }, + .name = "wm8994-codec", + .owner = THIS_MODULE, + .pm = &wm8994_pm_ops, + }, .probe = wm8994_probe, .remove = __devexit_p(wm8994_remove), }; diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index f996d14766d..2f4d2d12a45 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -122,6 +122,7 @@ struct wm8994_priv { bool jack_mic; int btn_mask; bool jackdet; + int jackdet_mode; wm8958_micdet_cb jack_cb; void *jack_cb_data; -- cgit v1.2.3-18-g5258 From fbe4ff795f3c081e2cc21507b804b5ddc78cd362 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Feb 2012 21:50:03 +0000 Subject: ASoC: wm8962: Remove unneeded pm_runtime_set_active() The default pm_runtime status is enabled which is what we want. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 445d2090661..b4d472f7f8e 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3673,7 +3673,6 @@ static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, ret); } - pm_runtime_set_active(&i2c->dev); pm_runtime_enable(&i2c->dev); pm_request_idle(&i2c->dev); -- cgit v1.2.3-18-g5258 From ba106ce3d04db9085d32b47aee545c35b586827a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Mar 2012 00:25:28 +0000 Subject: ASoC: wm8962: Remove defaults for volatile registers Save a little RAM. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 17 +++-------------- 1 file changed, 3 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b4d472f7f8e..5bcb350bacc 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -116,11 +116,11 @@ static struct reg_default wm8962_reg[] = { { 1, 0x049F }, /* R1 - Right Input volume */ { 2, 0x0000 }, /* R2 - HPOUTL volume */ { 3, 0x0000 }, /* R3 - HPOUTR volume */ - { 4, 0x0020 }, /* R4 - Clocking1 */ + { 5, 0x0018 }, /* R5 - ADC & DAC Control 1 */ { 6, 0x2008 }, /* R6 - ADC & DAC Control 2 */ { 7, 0x000A }, /* R7 - Audio Interface 0 */ - { 8, 0x01E4 }, /* R8 - Clocking2 */ + { 9, 0x0300 }, /* R9 - Audio Interface 1 */ { 10, 0x00C0 }, /* R10 - Left DAC volume */ { 11, 0x00C0 }, /* R11 - Right DAC volume */ @@ -129,7 +129,7 @@ static struct reg_default wm8962_reg[] = { { 15, 0x6243 }, /* R15 - Software Reset */ { 17, 0x007B }, /* R17 - ALC1 */ - { 18, 0x0000 }, /* R18 - ALC2 */ + { 19, 0x1C32 }, /* R19 - ALC3 */ { 20, 0x3200 }, /* R20 - Noise Gate */ { 21, 0x00C0 }, /* R21 - Left ADC volume */ @@ -153,10 +153,6 @@ static struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ - { 47, 0x0000 }, /* R47 - Thermal Shutdown Status */ - { 48, 0x8027 }, /* R48 - Additional Control (4) */ - { 49, 0x0010 }, /* R49 - Class D Control 1 */ - { 51, 0x0003 }, /* R51 - Class D Control 2 */ { 56, 0x0506 }, /* R56 - Clocking 4 */ @@ -168,8 +164,6 @@ static struct reg_default wm8962_reg[] = { { 64, 0x0810 }, /* R64 - DC Servo 4 */ - { 66, 0x0000 }, /* R66 - DC Servo 6 */ - { 68, 0x001B }, /* R68 - Analogue PGA Bias */ { 69, 0x0000 }, /* R69 - Analogue HP 0 */ @@ -302,9 +296,6 @@ static struct reg_default wm8962_reg[] = { { 516, 0x8100 }, /* R516 - GPIO 5 */ { 517, 0x8100 }, /* R517 - GPIO 6 */ - { 560, 0x0000 }, /* R560 - Interrupt Status 1 */ - { 561, 0x0000 }, /* R561 - Interrupt Status 2 */ - { 568, 0x0030 }, /* R568 - Interrupt Status 1 Mask */ { 569, 0xFFED }, /* R569 - Interrupt Status 2 Mask */ @@ -316,8 +307,6 @@ static struct reg_default wm8962_reg[] = { { 768, 0x1C00 }, /* R768 - DSP2 Power Management */ - { 1037, 0x0000 }, /* R1037 - DSP2_ExecControl */ - { 8192, 0x0000 }, /* R8192 - DSP2 Instruction RAM 0 */ { 9216, 0x0030 }, /* R9216 - DSP2 Address RAM 2 */ -- cgit v1.2.3-18-g5258 From f320515a589eeb9bfbc317801e60b87a12f9eae1 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Tue, 6 Mar 2012 15:06:53 +0800 Subject: ASoC: Add missing regmap_init_i2c in wm8804_i2c_probe commit 891271c "ASoC: Convert wm8804 to direct regmap API usage" only converts wm8804_spi_probe to use regmap_init_spi. This patch adds missing regmap_init_i2c in wm8804_i2c_probe. Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 7ee8dcf1fe3..6bd1b767b13 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -755,6 +755,12 @@ static __devinit int wm8804_i2c_probe(struct i2c_client *i2c, if (!wm8804) return -ENOMEM; + wm8804->regmap = regmap_init_i2c(i2c, &wm8804_regmap_config); + if (IS_ERR(wm8804->regmap)) { + ret = PTR_ERR(wm8804->regmap); + return ret; + } + i2c_set_clientdata(i2c, wm8804); ret = snd_soc_register_codec(&i2c->dev, -- cgit v1.2.3-18-g5258 From 9d5ef2663fe220a88412a7190942b7d933da0333 Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Mon, 5 Mar 2012 22:31:04 +0800 Subject: ASoC: fsl: add dt support for imx-audmux It adds device tree probe support for imx-audmux driver. Signed-off-by: Richard Zhao Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/imx/imx-audmux.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound') diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index 87f8768e1cd..b83699d905b 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -22,6 +22,8 @@ #include #include #include +#include +#include #include #include @@ -197,6 +199,13 @@ static struct platform_device_id imx_audmux_ids[] = { }; MODULE_DEVICE_TABLE(platform, imx_audmux_ids); +static const struct of_device_id imx_audmux_dt_ids[] = { + { .compatible = "fsl,imx21-audmux", .data = &imx_audmux_ids[0], }, + { .compatible = "fsl,imx31-audmux", .data = &imx_audmux_ids[1], }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_audmux_dt_ids); + static const uint8_t port_mapping[] = { 0x0, 0x4, 0x8, 0x10, 0x14, 0x1c, }; @@ -243,6 +252,8 @@ EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); static int __init imx_audmux_probe(struct platform_device *pdev) { struct resource *res; + const struct of_device_id *of_id = + of_match_device(imx_audmux_dt_ids, &pdev->dev); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); audmux_base = devm_request_and_ioremap(&pdev->dev, res); @@ -256,6 +267,8 @@ static int __init imx_audmux_probe(struct platform_device *pdev) audmux_clk = NULL; } + if (of_id) + pdev->id_entry = of_id->data; audmux_type = pdev->id_entry->driver_data; if (audmux_type == IMX31_AUDMUX) audmux_debugfs_init(); @@ -279,6 +292,7 @@ static struct platform_driver imx_audmux_driver = { .driver = { .name = DRIVER_NAME, .owner = THIS_MODULE, + .of_match_table = imx_audmux_dt_ids, } }; -- cgit v1.2.3-18-g5258 From 78f8baf138311be3e170526388b0530a172bdbff Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Mar 2012 14:02:32 +0100 Subject: ALSA: hda - Add Gigabyte GA-MA790X to the beep whitelist Its BIOS suppresses the PC beep although it's implemented. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7e651682eec..2ae6bfbc678 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4304,6 +4304,7 @@ static const struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1), SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1), + SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; -- cgit v1.2.3-18-g5258 From f1e90af2b55ee13a3ed5ee1b9229d0edefeff27c Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 6 Mar 2012 18:13:25 +0000 Subject: ASoC: dapm: Use dev_warn for debugfs warning message Remove printk(KERN_WARNING) and use dev_warn() instead. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a4d4aa1e6c4..dcd11609f93 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1667,7 +1667,7 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); if (!dapm->debugfs_dapm) { - printk(KERN_WARNING + dev_warn(dapm->dev, "Failed to create DAPM debugfs directory\n"); return; } -- cgit v1.2.3-18-g5258 From c25cd1543986e7c16c7ddf738748ccd530a18268 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 6 Mar 2012 12:13:07 -0700 Subject: ASoC: tegra: Remove unused variable Fixes the following warning: sound/soc/tegra/tegra_alc5632.c: In function 'tegra_alc5632_asoc_init': sound/soc/tegra/tegra_alc5632.c:118:6: warning: unused variable 'ret' [-Wunused-variable] Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_alc5632.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 2a27725cc9b..e45ccd851f6 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -115,7 +115,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dapm_context *dapm = &codec->dapm; struct device_node *np = codec->card->dev->of_node; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); - int ret; snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, &tegra_alc5632_hs_jack); -- cgit v1.2.3-18-g5258 From cc22d37e7f5e1745658760660f03793913f43e49 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 6 Mar 2012 18:16:18 +0000 Subject: ASoC: core: Add platform component mutex Add mutex support for platform IO operations. e.g. can be used for platform DAPM widget IO ops. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7978f6c01ef..c90bb0110bd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3382,6 +3382,7 @@ int snd_soc_register_platform(struct device *dev, platform->dapm.dev = dev; platform->dapm.platform = platform; platform->dapm.stream_event = platform_drv->stream_event; + mutex_init(&platform->mutex); mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); -- cgit v1.2.3-18-g5258 From 96acc357bedad69fbc94d1b923a960af5a411c6f Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 6 Mar 2012 18:16:19 +0000 Subject: ASoC: DAPM: Make sure DAPM widget IO ops hold the component mutex Currently not all DAPM widget IO ops are holding their component mutex (codec or platform). Make sure this is now held for DAPM widget IO operations. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcd11609f93..a837977f0ac 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -206,7 +206,23 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) return -1; } -static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, +static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) +{ + if (w->codec) + mutex_lock(&w->codec->mutex); + else if (w->platform) + mutex_lock(&w->platform->mutex); +} + +static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) +{ + if (w->codec) + mutex_unlock(&w->codec->mutex); + else if (w->platform) + mutex_unlock(&w->platform->mutex); +} + +static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { bool change; @@ -219,18 +235,24 @@ static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, if (ret != 0) return ret; } else { + soc_widget_lock(w); ret = soc_widget_read(w, reg); - if (ret < 0) + if (ret < 0) { + soc_widget_unlock(w); return ret; + } old = ret; new = (old & ~mask) | (value & mask); change = old != new; if (change) { ret = soc_widget_write(w, reg, new); - if (ret < 0) + if (ret < 0) { + soc_widget_unlock(w); return ret; + } } + soc_widget_unlock(w); } return change; @@ -847,7 +869,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - soc_widget_update_bits(w, -(w->reg + 1), + soc_widget_update_bits_locked(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -1105,7 +1127,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - soc_widget_update_bits(w, reg, mask, value); + soc_widget_update_bits_locked(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1235,7 +1257,7 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm) w->name, ret); } - ret = snd_soc_update_bits(w->codec, update->reg, update->mask, + ret = soc_widget_update_bits_locked(w, update->reg, update->mask, update->val); if (ret < 0) pr_err("%s DAPM update failed: %d\n", w->name, ret); -- cgit v1.2.3-18-g5258 From 546bb6785265f3413fa76e06b9fdce58ee15ea87 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Mar 2012 08:37:19 +0100 Subject: ALSA: hda/realtek - Reuse init_hook for ALC269VB coef setup Move the currently unused spec->init_hook at the beginning of the init sequence so that the recently added ALC269VB coef setup can be put there. The alc_init() is again clean without an ugly check. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 65955dabc15..1de0c1629ba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2052,15 +2052,14 @@ static int alc_build_controls(struct hda_codec *codec) static void alc_init_special_input_src(struct hda_codec *codec); static void alc_auto_init_std(struct hda_codec *codec); -static int alc269_fill_coef(struct hda_codec *codec); static int alc_init(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; unsigned int i; - if (codec->vendor_id == 0x10ec0269) - alc269_fill_coef(codec); + if (spec->init_hook) + spec->init_hook(codec); alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); @@ -2070,9 +2069,6 @@ static int alc_init(struct hda_codec *codec) alc_init_special_input_src(codec); alc_auto_init_std(codec); - if (spec->init_hook) - spec->init_hook(codec); - alc_apply_fixup(codec, ALC_FIXUP_ACT_INIT); snd_hda_jack_report_sync(codec); @@ -6124,13 +6120,13 @@ static const struct alc_model_fixup alc269_fixup_models[] = { }; -static int alc269_fill_coef(struct hda_codec *codec) +static void alc269_fill_coef(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int val; if (spec->codec_variant != ALC269_TYPE_ALC269VB) - return 0; + return; if ((alc_get_coef0(codec) & 0x00ff) < 0x015) { alc_write_coef_idx(codec, 0xf, 0x960b); @@ -6166,8 +6162,6 @@ static int alc269_fill_coef(struct hda_codec *codec) val = alc_read_coef_idx(codec, 0x4); /* HP */ alc_write_coef_idx(codec, 0x4, val | (1<<11)); - - return 0; } /* @@ -6211,6 +6205,7 @@ static int patch_alc269(struct hda_codec *codec) } if (err < 0) goto error; + spec->init_hook = alc269_fill_coef; alc269_fill_coef(codec); } -- cgit v1.2.3-18-g5258 From 785f857d1cb0856b612b46a0545b74aa2596e44a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Mar 2012 10:58:39 +0100 Subject: ALSA: hda - Set codec to D3 forcibly even if not used We've seen a problem with a pop-noise at suspend/resume on a HP machine with ALC269, and it turned out to be an issue that the controller going to D3 while the codec is unused. When the device is once suspended and resumed and kept unused, the driver doesn't initialize the codecs. Instead, the codec chips are set up dynamically at the first usage. Now, suppose the device going to suspend again before the codec is set up. The controller is turned off to D3 while the codec chips are untouched. This caused a pop noise because the codec chip might have been turned on implicitly by the hardware. As a workaround, the codec chip needs to be set to D3 when going to suspend no matter whether it was used or not. Also, for making it happening, the controller has to be always set up in the resume path. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 4 ++++ sound/pci/hda/hda_intel.c | 14 +------------- 2 files changed, 5 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 76bac4fc047..0527ae1ab96 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5281,6 +5281,10 @@ int snd_hda_suspend(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); + else /* forcibly change the power to D3 even if not used */ + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D3); if (codec->patch_ops.post_suspend) codec->patch_ops.post_suspend(codec); } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e958bf9419..c19e71a94e1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2351,17 +2351,6 @@ static void azx_power_notify(struct hda_bus *bus) * power management */ -static int snd_hda_codecs_inuse(struct hda_bus *bus) -{ - struct hda_codec *codec; - - list_for_each_entry(codec, &bus->codec_list, list) { - if (snd_hda_codec_needs_resume(codec)) - return 1; - } - return 0; -} - static int azx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); @@ -2408,8 +2397,7 @@ static int azx_resume(struct pci_dev *pci) return -EIO; azx_init_pci(chip); - if (snd_hda_codecs_inuse(chip->bus)) - azx_init_chip(chip, 1); + azx_init_chip(chip, 1); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); -- cgit v1.2.3-18-g5258 From 66bf93212f19548f5ed221356b2d70189cc18254 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Mar 2012 23:58:22 +0000 Subject: ASoC: dapm: Only lock CODEC for I/O if not using regmap If we do use regmap then regmap will take care of things for us. We actually already have this check at a higher level for the current users but this makes sure we do the right thing in the future too if we need to. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a837977f0ac..1ba2a711b54 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -208,7 +208,7 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) { - if (w->codec) + if (w->codec && !w->codec->using_regmap) mutex_lock(&w->codec->mutex); else if (w->platform) mutex_lock(&w->platform->mutex); @@ -216,7 +216,7 @@ static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) { - if (w->codec) + if (w->codec && !w->codec->using_regmap) mutex_unlock(&w->codec->mutex); else if (w->platform) mutex_unlock(&w->platform->mutex); -- cgit v1.2.3-18-g5258 From fb97624ad61f734837998b316cc4eb1cf899900f Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 7 Mar 2012 15:19:57 -0300 Subject: ASoC: imx: imx-audmux: Fix section mismatch Fix the following section mismatch warning: WARNING: vmlinux.o(.data+0x35be8): Section mismatch in reference from the variable imx_audmux_driver to the function .init.text:imx_audmux_probe() The variable imx_audmux_driver references the function __init imx_audmux_probe() If the reference is valid then annotate the variable with __init* or __refdata (see linux/init.h) or name the variable: *_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/imx/imx-audmux.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index b83699d905b..a839494c5ea 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -167,7 +167,7 @@ static void __init audmux_debugfs_init(void) } } -static void __exit audmux_debugfs_remove(void) +static void __devexit audmux_debugfs_remove(void) { debugfs_remove_recursive(audmux_debugfs_root); } @@ -249,7 +249,7 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); -static int __init imx_audmux_probe(struct platform_device *pdev) +static int __devinit imx_audmux_probe(struct platform_device *pdev) { struct resource *res; const struct of_device_id *of_id = @@ -276,7 +276,7 @@ static int __init imx_audmux_probe(struct platform_device *pdev) return 0; } -static int __exit imx_audmux_remove(struct platform_device *pdev) +static int __devexit imx_audmux_remove(struct platform_device *pdev) { if (audmux_type == IMX31_AUDMUX) audmux_debugfs_remove(); @@ -287,7 +287,7 @@ static int __exit imx_audmux_remove(struct platform_device *pdev) static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, - .remove = __exit_p(imx_audmux_remove), + .remove = __devexit_p(imx_audmux_remove), .id_table = imx_audmux_ids, .driver = { .name = DRIVER_NAME, -- cgit v1.2.3-18-g5258 From 8d8bf58b06d1c4d49d08b2958cbd84145bf8b1bd Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Thu, 8 Mar 2012 10:02:36 +0800 Subject: ASoC: add more sample rate for pxa-ssp add more sample rate for pxa-ssp, which are supported, such as 32KHz, 64KHz. Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown --- sound/soc/pxa/pxa-ssp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a57cfbc038e..a16df0fa6ef 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -764,7 +764,8 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai) #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ -- cgit v1.2.3-18-g5258 From 33593b52ebb0d6d37d96bd5e01a31951fc3b8ddf Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Thu, 8 Mar 2012 19:04:56 +0530 Subject: ASoC: da7210: Update for using I2C regmap Current DA7210 driver has I2C support using older register cache methods. This patch updates it for latest regmap framework. This has been tested on DA7210 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 234 ++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 196 insertions(+), 38 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ab38e93c354..0c23f19dfca 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -17,6 +17,7 @@ #include #include +#include #include #include #include @@ -626,41 +627,170 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; }; -/* - * Register cache - */ -static const u8 da7210_reg[] = { - 0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R0 - R7 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, /* R8 - RF */ - 0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54, /* R10 - R17 */ - 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R18 - R1F */ - 0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00, /* R20 - R27 */ - 0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00, /* R28 - R2F */ - 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, /* R30 - R37 */ - 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00, /* R38 - R3F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R40 - R4F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R48 - R4F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R50 - R57 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R58 - R5F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R60 - R67 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R68 - R6F */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R70 - R77 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00, /* R78 - R7F */ - 0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00, /* R80 - R87 */ - 0x00, /* R88 */ +static struct reg_default da7210_reg_defaults[] = { + { 0x00, 0x00 }, + { 0x01, 0x11 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0a, 0x00 }, + { 0x0b, 0x00 }, + { 0x0c, 0x00 }, + { 0x0d, 0x00 }, + { 0x0e, 0x00 }, + { 0x0f, 0x08 }, + { 0x10, 0x00 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x08 }, + { 0x15, 0x10 }, + { 0x16, 0x10 }, + { 0x17, 0x54 }, + { 0x18, 0x40 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x02 }, + { 0x24, 0x00 }, + { 0x25, 0x76 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x04 }, + { 0x29, 0x00 }, + { 0x2a, 0x00 }, + { 0x2b, 0x30 }, + { 0x2c, 0x2A }, + { 0x2d, 0x00 }, + { 0x2e, 0x40 }, + { 0x2f, 0x00 }, + { 0x30, 0x40 }, + { 0x31, 0x00 }, + { 0x32, 0x40 }, + { 0x33, 0x00 }, + { 0x34, 0x40 }, + { 0x35, 0x00 }, + { 0x36, 0x40 }, + { 0x37, 0x00 }, + { 0x38, 0x40 }, + { 0x39, 0x00 }, + { 0x3a, 0x40 }, + { 0x3b, 0x00 }, + { 0x3c, 0x40 }, + { 0x3d, 0x00 }, + { 0x3e, 0x00 }, + { 0x3f, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x44, 0x00 }, + { 0x45, 0x00 }, + { 0x46, 0x00 }, + { 0x47, 0x00 }, + { 0x48, 0x00 }, + { 0x49, 0x00 }, + { 0x4a, 0x00 }, + { 0x4b, 0x00 }, + { 0x4c, 0x00 }, + { 0x4d, 0x00 }, + { 0x4e, 0x00 }, + { 0x4f, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x52, 0x00 }, + { 0x53, 0x00 }, + { 0x54, 0x00 }, + { 0x55, 0x00 }, + { 0x56, 0x00 }, + { 0x57, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5a, 0x00 }, + { 0x5b, 0x00 }, + { 0x5c, 0x00 }, + { 0x5d, 0x00 }, + { 0x5e, 0x00 }, + { 0x5f, 0x00 }, + { 0x60, 0x00 }, + { 0x61, 0x00 }, + { 0x62, 0x00 }, + { 0x63, 0x00 }, + { 0x64, 0x00 }, + { 0x65, 0x00 }, + { 0x66, 0x00 }, + { 0x67, 0x00 }, + { 0x68, 0x00 }, + { 0x69, 0x00 }, + { 0x6a, 0x00 }, + { 0x6b, 0x00 }, + { 0x6c, 0x00 }, + { 0x6d, 0x00 }, + { 0x6e, 0x00 }, + { 0x6f, 0x00 }, + { 0x70, 0x00 }, + { 0x71, 0x00 }, + { 0x72, 0x00 }, + { 0x73, 0x00 }, + { 0x74, 0x00 }, + { 0x75, 0x00 }, + { 0x76, 0x00 }, + { 0x77, 0x00 }, + { 0x78, 0x00 }, + { 0x79, 0x00 }, + { 0x7a, 0x00 }, + { 0x7b, 0x00 }, + { 0x7c, 0x00 }, + { 0x7d, 0x54 }, + { 0x7e, 0x54 }, + { 0x7f, 0x00 }, + { 0x80, 0x00 }, + { 0x81, 0x00 }, + { 0x82, 0x2c }, + { 0x83, 0x00 }, + { 0x84, 0x00 }, + { 0x85, 0x00 }, + { 0x86, 0x00 }, + { 0x87, 0x00 }, + { 0x88, 0x00 }, }; -static int da7210_volatile_register(struct snd_soc_codec *codec, +static bool da7210_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA7210_A_HID_UNLOCK: + case DA7210_A_TEST_UNLOCK: + case DA7210_A_PLL1: + case DA7210_A_CP_MODE: + return false; + default: + return true; + } +} + +static bool da7210_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case DA7210_STATUS: - return 1; + return true; default: - return 0; + return false; } } @@ -866,7 +996,8 @@ static int da7210_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); + codec->control_data = da7210->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -983,12 +1114,14 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); /* As suggested by Dialog */ - snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ - snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); - snd_soc_write(codec, DA7210_A_PLL1, 0x01); - snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); - snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ - snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); + /* unlock */ + regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B); + regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4); + regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01); + regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C); + /* re-lock */ + regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00); + regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00); /* Activate all enabled subsystem */ snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); @@ -1000,10 +1133,6 @@ static int da7210_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .probe = da7210_probe, - .reg_cache_size = ARRAY_SIZE(da7210_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = da7210_reg, - .volatile_register = da7210_volatile_register, .controls = da7210_snd_controls, .num_controls = ARRAY_SIZE(da7210_snd_controls), @@ -1014,6 +1143,17 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = { .num_dapm_routes = ARRAY_SIZE(da7210_audio_map), }; +static struct regmap_config da7210_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = da7210_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults), + .volatile_reg = da7210_volatile_register, + .readable_reg = da7210_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static int __devinit da7210_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -1027,16 +1167,34 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, da7210); - da7210->control_type = SND_SOC_I2C; + + da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap); + if (IS_ERR(da7210->regmap)) { + ret = PTR_ERR(da7210->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + return ret; + } ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7210, &da7210_dai, 1); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + goto err_regmap; + } + return ret; + +err_regmap: + regmap_exit(da7210->regmap); + return ret; } static int __devexit da7210_i2c_remove(struct i2c_client *client) { + struct da7210_priv *da7210 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + regmap_exit(da7210->regmap); return 0; } -- cgit v1.2.3-18-g5258 From 5e4ba569a5aa631852ec8240f11142392116633d Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 9 Mar 2012 00:59:40 +0800 Subject: ASoC: core: missing set_fmt should not be complaint Not having a DAI link set_fmt operation is perfectly normal and should not be complaint. Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c90bb0110bd..93a0daac508 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1531,14 +1531,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (dai_link->dai_fmt) { ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, dai_link->dai_fmt); - if (ret != 0) + if (ret != 0 && ret != -ENOTSUPP) dev_warn(card->rtd[i].codec_dai->dev, "Failed to set DAI format: %d\n", ret); ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, dai_link->dai_fmt); - if (ret != 0) + if (ret != 0 && ret != -ENOTSUPP) dev_warn(card->rtd[i].cpu_dai->dev, "Failed to set DAI format: %d\n", ret); @@ -2971,10 +2971,11 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->driver && dai->driver->ops->set_fmt) - return dai->driver->ops->set_fmt(dai, fmt); - else + if (dai->driver == NULL) return -EINVAL; + if (dai->driver->ops->set_fmt == NULL) + return -ENOTSUPP; + return dai->driver->ops->set_fmt(dai, fmt); } EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); -- cgit v1.2.3-18-g5258 From da7f910bd0d3da2355b29dd19f4dcd4dfe3563e4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Mar 2012 15:59:50 +0000 Subject: ASoC: wm8996: Remove some volatile regisers from the defaults table Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 9376b19941b..dd94cd035ac 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -118,7 +118,6 @@ WM8996_REGULATOR_EVENT(1) WM8996_REGULATOR_EVENT(2) static struct reg_default wm8996_reg[] = { - { WM8996_SOFTWARE_RESET, 0x8996 }, { WM8996_POWER_MANAGEMENT_1, 0x0 }, { WM8996_POWER_MANAGEMENT_2, 0x0 }, { WM8996_POWER_MANAGEMENT_3, 0x0 }, @@ -153,7 +152,6 @@ static struct reg_default wm8996_reg[] = { { WM8996_CHARGE_PUMP_1, 0x1f25 }, { WM8996_CHARGE_PUMP_2, 0xab19 }, { WM8996_DC_SERVO_1, 0x0 }, - { WM8996_DC_SERVO_2, 0x0 }, { WM8996_DC_SERVO_3, 0x0 }, { WM8996_DC_SERVO_5, 0x2a2a }, { WM8996_DC_SERVO_6, 0x0 }, -- cgit v1.2.3-18-g5258 From 5b596483936230b926898efe10f9cc258d5ed092 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Mar 2012 17:00:57 +0000 Subject: ASoC: wm8996: Remove separate output stage enable step Marginally improve performance during startup. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 32 ++++++++++++-------------------- 1 file changed, 12 insertions(+), 20 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index dd94cd035ac..40a124c9f15 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -890,8 +890,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, val = 0; mask = 0; if (wm8996->hpout_pending & HPOUT1L) { - val |= WM8996_HPOUT1L_RMV_SHORT; - mask |= WM8996_HPOUT1L_RMV_SHORT; + val |= WM8996_HPOUT1L_RMV_SHORT | WM8996_HPOUT1L_OUTP; + mask |= WM8996_HPOUT1L_RMV_SHORT | WM8996_HPOUT1L_OUTP; } else { mask |= WM8996_HPOUT1L_RMV_SHORT | WM8996_HPOUT1L_OUTP | @@ -899,8 +899,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, } if (wm8996->hpout_pending & HPOUT1R) { - val |= WM8996_HPOUT1R_RMV_SHORT; - mask |= WM8996_HPOUT1R_RMV_SHORT; + val |= WM8996_HPOUT1R_RMV_SHORT | WM8996_HPOUT1R_OUTP; + mask |= WM8996_HPOUT1R_RMV_SHORT | WM8996_HPOUT1R_OUTP; } else { mask |= WM8996_HPOUT1R_RMV_SHORT | WM8996_HPOUT1R_OUTP | @@ -912,8 +912,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, val = 0; mask = 0; if (wm8996->hpout_pending & HPOUT2L) { - val |= WM8996_HPOUT2L_RMV_SHORT; - mask |= WM8996_HPOUT2L_RMV_SHORT; + val |= WM8996_HPOUT2L_RMV_SHORT | WM8996_HPOUT2L_OUTP; + mask |= WM8996_HPOUT2L_RMV_SHORT | WM8996_HPOUT2L_OUTP; } else { mask |= WM8996_HPOUT2L_RMV_SHORT | WM8996_HPOUT2L_OUTP | @@ -921,8 +921,8 @@ static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, } if (wm8996->hpout_pending & HPOUT2R) { - val |= WM8996_HPOUT2R_RMV_SHORT; - mask |= WM8996_HPOUT2R_RMV_SHORT; + val |= WM8996_HPOUT2R_RMV_SHORT | WM8996_HPOUT2R_OUTP; + mask |= WM8996_HPOUT2R_RMV_SHORT | WM8996_HPOUT2R_OUTP; } else { mask |= WM8996_HPOUT2R_RMV_SHORT | WM8996_HPOUT2R_OUTP | @@ -1214,7 +1214,6 @@ SND_SOC_DAPM_PGA_S("HPOUT2L PGA", 0, WM8996_POWER_MANAGEMENT_1, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2L_DLY", 1, WM8996_ANALOGUE_HP_2, 5, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2L_DCS", 2, WM8996_DC_SERVO_1, 2, 0, dcs_start, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_S("HPOUT2L_OUTP", 3, WM8996_ANALOGUE_HP_2, 6, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2L_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT2L, 0, rmv_short_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1223,7 +1222,6 @@ SND_SOC_DAPM_PGA_S("HPOUT2R PGA", 0, WM8996_POWER_MANAGEMENT_1, 6, 0,NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2R_DLY", 1, WM8996_ANALOGUE_HP_2, 1, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2R_DCS", 2, WM8996_DC_SERVO_1, 3, 0, dcs_start, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_S("HPOUT2R_OUTP", 3, WM8996_ANALOGUE_HP_2, 2, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT2R_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT2R, 0, rmv_short_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1232,7 +1230,6 @@ SND_SOC_DAPM_PGA_S("HPOUT1L PGA", 0, WM8996_POWER_MANAGEMENT_1, 5, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1L_DLY", 1, WM8996_ANALOGUE_HP_1, 5, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1L_DCS", 2, WM8996_DC_SERVO_1, 0, 0, dcs_start, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_S("HPOUT1L_OUTP", 3, WM8996_ANALOGUE_HP_1, 6, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1L_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT1L, 0, rmv_short_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1241,7 +1238,6 @@ SND_SOC_DAPM_PGA_S("HPOUT1R PGA", 0, WM8996_POWER_MANAGEMENT_1, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1R_DLY", 1, WM8996_ANALOGUE_HP_1, 1, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1R_DCS", 2, WM8996_DC_SERVO_1, 1, 0, dcs_start, SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_S("HPOUT1R_OUTP", 3, WM8996_ANALOGUE_HP_1, 2, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPOUT1R_RMV_SHORT", 3, SND_SOC_NOPM, HPOUT1R, 0, rmv_short_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), @@ -1434,32 +1430,28 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2L PGA", NULL, "DAC2L" }, { "HPOUT2L_DLY", NULL, "HPOUT2L PGA" }, { "HPOUT2L_DCS", NULL, "HPOUT2L_DLY" }, - { "HPOUT2L_OUTP", NULL, "HPOUT2L_DCS" }, - { "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_OUTP" }, + { "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_DCS" }, { "HPOUT2R PGA", NULL, "Charge Pump" }, { "HPOUT2R PGA", NULL, "Bandgap" }, { "HPOUT2R PGA", NULL, "DAC2R" }, { "HPOUT2R_DLY", NULL, "HPOUT2R PGA" }, { "HPOUT2R_DCS", NULL, "HPOUT2R_DLY" }, - { "HPOUT2R_OUTP", NULL, "HPOUT2R_DCS" }, - { "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_OUTP" }, + { "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_DCS" }, { "HPOUT1L PGA", NULL, "Charge Pump" }, { "HPOUT1L PGA", NULL, "Bandgap" }, { "HPOUT1L PGA", NULL, "DAC1L" }, { "HPOUT1L_DLY", NULL, "HPOUT1L PGA" }, { "HPOUT1L_DCS", NULL, "HPOUT1L_DLY" }, - { "HPOUT1L_OUTP", NULL, "HPOUT1L_DCS" }, - { "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_OUTP" }, + { "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_DCS" }, { "HPOUT1R PGA", NULL, "Charge Pump" }, { "HPOUT1R PGA", NULL, "Bandgap" }, { "HPOUT1R PGA", NULL, "DAC1R" }, { "HPOUT1R_DLY", NULL, "HPOUT1R PGA" }, { "HPOUT1R_DCS", NULL, "HPOUT1R_DLY" }, - { "HPOUT1R_OUTP", NULL, "HPOUT1R_DCS" }, - { "HPOUT1R_RMV_SHORT", NULL, "HPOUT1R_OUTP" }, + { "HPOUT1R_RMV_SHORT", NULL, "HPOUT1R_DCS" }, { "HPOUT2L", NULL, "HPOUT2L_RMV_SHORT" }, { "HPOUT2R", NULL, "HPOUT2R_RMV_SHORT" }, -- cgit v1.2.3-18-g5258 From 2b81ec69144de93f29fa258d3435557a5773ffb5 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Fri, 9 Mar 2012 00:59:46 +0800 Subject: ASoC: fsl: check property 'compatible' for the machine name Check /compatible rather than /model to determine the machine name. The p1022ds older device trees get a different /model from the new ones, while /compatible is consistent there, so checking /compatible will save the bother of detecting older p1022ds device trees. Signed-off-by: Shawn Guo Acked-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 6 +++--- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 32 +++++--------------------------- 3 files changed, 9 insertions(+), 31 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3e066966d87..2eb407fa3b4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -716,12 +716,12 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) } /* Trigger the machine driver's probe function. The platform driver - * name of the machine driver is taken from the /model property of the + * name of the machine driver is taken from /compatible property of the * device tree. We also pass the address of the CPU DAI driver * structure. */ - sprop = of_get_property(of_find_node_by_path("/"), "model", NULL); - /* Sometimes the model name has a "fsl," prefix, so we strip that. */ + sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL); + /* Sometimes the compatible name has a "fsl," prefix, so we strip it. */ p = strrchr(sprop, ','); if (p) sprop = p + 1; diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index fcf9302f59b..afbabf427f2 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -546,7 +546,7 @@ static struct platform_driver mpc8610_hpcd_driver = { .probe = mpc8610_hpcd_probe, .remove = __devexit_p(mpc8610_hpcd_remove), .driver = { - /* The name must match the 'model' property in the device tree, + /* The name must match 'compatible' property in the device tree, * in lowercase letters. */ .name = "snd-soc-mpc8610hpcd", diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index d32ec4646d2..b8898708347 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -543,6 +543,11 @@ static struct platform_driver p1022_ds_driver = { .probe = p1022_ds_probe, .remove = __devexit_p(p1022_ds_remove), .driver = { + /* + * The name must match 'compatible' property in the device tree, + * in lowercase letters. + */ + .name = "snd-soc-p1022ds", .owner = THIS_MODULE, }, }; @@ -556,33 +561,6 @@ static int __init p1022_ds_init(void) { struct device_node *guts_np; struct resource res; - const char *sprop; - - /* - * Check if we're actually running on a P1022DS. Older device trees - * have a model of "fsl,P1022" and newer ones use "fsl,P1022DS", so we - * need to support both. The SSI driver uses that property to link to - * the machine driver, so have to match it. - */ - sprop = of_get_property(of_find_node_by_path("/"), "model", NULL); - if (!sprop) { - pr_err("snd-soc-p1022ds: missing /model node"); - return -ENODEV; - } - - pr_debug("snd-soc-p1022ds: board model name is %s\n", sprop); - - /* - * The name of this board, taken from the device tree. Normally, this is a* - * fixed string, but some P1022DS device trees have a /model property of - * "fsl,P1022", and others have "fsl,P1022DS". - */ - if (strcasecmp(sprop, "fsl,p1022ds") == 0) - p1022_ds_driver.driver.name = "snd-soc-p1022ds"; - else if (strcasecmp(sprop, "fsl,p1022") == 0) - p1022_ds_driver.driver.name = "snd-soc-p1022"; - else - return -ENODEV; /* Get the physical address of the global utilities registers */ guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts"); -- cgit v1.2.3-18-g5258 From aeb87073f00524dbc4017aba1de5469948d134d4 Mon Sep 17 00:00:00 2001 From: Ashish Chavan Date: Fri, 9 Mar 2012 13:51:30 +0530 Subject: ASoC: da7210: Remove extra registers from defaults list This patch removes following registers from reg map defaults, - Registers which are currently not used by driver - Non existing registers - Volatile registers Signed-off-by: Ashish Chavan Signed-off-by: David Dajun Chen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 88 ----------------------------------------------- 1 file changed, 88 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0c23f19dfca..7843711729b 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -631,9 +631,7 @@ struct da7210_priv { }; static struct reg_default da7210_reg_defaults[] = { - { 0x00, 0x00 }, { 0x01, 0x11 }, - { 0x02, 0x00 }, { 0x03, 0x00 }, { 0x04, 0x00 }, { 0x05, 0x00 }, @@ -676,92 +674,6 @@ static struct reg_default da7210_reg_defaults[] = { { 0x2a, 0x00 }, { 0x2b, 0x30 }, { 0x2c, 0x2A }, - { 0x2d, 0x00 }, - { 0x2e, 0x40 }, - { 0x2f, 0x00 }, - { 0x30, 0x40 }, - { 0x31, 0x00 }, - { 0x32, 0x40 }, - { 0x33, 0x00 }, - { 0x34, 0x40 }, - { 0x35, 0x00 }, - { 0x36, 0x40 }, - { 0x37, 0x00 }, - { 0x38, 0x40 }, - { 0x39, 0x00 }, - { 0x3a, 0x40 }, - { 0x3b, 0x00 }, - { 0x3c, 0x40 }, - { 0x3d, 0x00 }, - { 0x3e, 0x00 }, - { 0x3f, 0x00 }, - { 0x40, 0x00 }, - { 0x41, 0x00 }, - { 0x42, 0x00 }, - { 0x43, 0x00 }, - { 0x44, 0x00 }, - { 0x45, 0x00 }, - { 0x46, 0x00 }, - { 0x47, 0x00 }, - { 0x48, 0x00 }, - { 0x49, 0x00 }, - { 0x4a, 0x00 }, - { 0x4b, 0x00 }, - { 0x4c, 0x00 }, - { 0x4d, 0x00 }, - { 0x4e, 0x00 }, - { 0x4f, 0x00 }, - { 0x50, 0x00 }, - { 0x51, 0x00 }, - { 0x52, 0x00 }, - { 0x53, 0x00 }, - { 0x54, 0x00 }, - { 0x55, 0x00 }, - { 0x56, 0x00 }, - { 0x57, 0x00 }, - { 0x58, 0x00 }, - { 0x59, 0x00 }, - { 0x5a, 0x00 }, - { 0x5b, 0x00 }, - { 0x5c, 0x00 }, - { 0x5d, 0x00 }, - { 0x5e, 0x00 }, - { 0x5f, 0x00 }, - { 0x60, 0x00 }, - { 0x61, 0x00 }, - { 0x62, 0x00 }, - { 0x63, 0x00 }, - { 0x64, 0x00 }, - { 0x65, 0x00 }, - { 0x66, 0x00 }, - { 0x67, 0x00 }, - { 0x68, 0x00 }, - { 0x69, 0x00 }, - { 0x6a, 0x00 }, - { 0x6b, 0x00 }, - { 0x6c, 0x00 }, - { 0x6d, 0x00 }, - { 0x6e, 0x00 }, - { 0x6f, 0x00 }, - { 0x70, 0x00 }, - { 0x71, 0x00 }, - { 0x72, 0x00 }, - { 0x73, 0x00 }, - { 0x74, 0x00 }, - { 0x75, 0x00 }, - { 0x76, 0x00 }, - { 0x77, 0x00 }, - { 0x78, 0x00 }, - { 0x79, 0x00 }, - { 0x7a, 0x00 }, - { 0x7b, 0x00 }, - { 0x7c, 0x00 }, - { 0x7d, 0x54 }, - { 0x7e, 0x54 }, - { 0x7f, 0x00 }, - { 0x80, 0x00 }, - { 0x81, 0x00 }, - { 0x82, 0x2c }, { 0x83, 0x00 }, { 0x84, 0x00 }, { 0x85, 0x00 }, -- cgit v1.2.3-18-g5258 From e7df2a3ae569ed6d178510f58b22308edac7a4c7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Mar 2012 17:41:53 +0100 Subject: ALSA: core - Refactor card id string creation code The code to handle the card id string is fairly messy, so here is a tidy up. Signed-off-by: Takashi Iwai --- sound/core/init.c | 169 ++++++++++++++++++++++++++++++++---------------------- 1 file changed, 100 insertions(+), 69 deletions(-) (limited to 'sound') diff --git a/sound/core/init.c b/sound/core/init.c index 3ac49b1b7cb..068cf08d3ff 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -480,74 +480,104 @@ int snd_card_free(struct snd_card *card) EXPORT_SYMBOL(snd_card_free); -static void snd_card_set_id_no_lock(struct snd_card *card, const char *nid) +/* retrieve the last word of shortname or longname */ +static const char *retrieve_id_from_card_name(const char *name) { - int i, len, idx_flag = 0, loops = SNDRV_CARDS; - const char *spos, *src; - char *id; - - if (nid == NULL) { - id = card->shortname; - spos = src = id; - while (*id != '\0') { - if (*id == ' ') - spos = id + 1; - id++; - } - } else { - spos = src = nid; + const char *spos = name; + + while (*name) { + if (isspace(*name) && isalnum(name[1])) + spos = name + 1; + name++; } - id = card->id; - while (*spos != '\0' && !isalnum(*spos)) - spos++; - if (isdigit(*spos)) - *id++ = isalpha(src[0]) ? src[0] : 'D'; - while (*spos != '\0' && (size_t)(id - card->id) < sizeof(card->id) - 1) { - if (isalnum(*spos)) - *id++ = *spos; - spos++; + return spos; +} + +/* return true if the given id string doesn't conflict any other card ids */ +static bool card_id_ok(struct snd_card *card, const char *id) +{ + int i; + if (!snd_info_check_reserved_words(id)) + return false; + for (i = 0; i < snd_ecards_limit; i++) { + if (snd_cards[i] && snd_cards[i] != card && + !strcmp(snd_cards[i]->id, id)) + return false; } - *id = '\0'; + return true; +} - id = card->id; +/* copy to card->id only with valid letters from nid */ +static void copy_valid_id_string(struct snd_card *card, const char *src, + const char *nid) +{ + char *id = card->id; + + while (*nid && !isalnum(*nid)) + nid++; + if (isdigit(*nid)) + *id++ = isalpha(*src) ? *src : 'D'; + while (*nid && (size_t)(id - card->id) < sizeof(card->id) - 1) { + if (isalnum(*nid)) + *id++ = *nid; + nid++; + } + *id = 0; +} + +/* Set card->id from the given string + * If the string conflicts with other ids, add a suffix to make it unique. + */ +static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, + const char *nid) +{ + int len, loops; + bool with_suffix; + bool is_default = false; + char *id; - if (*id == '\0') + copy_valid_id_string(card, src, nid); + id = card->id; + + again: + /* use "Default" for obviously invalid strings + * ("card" conflicts with proc directories) + */ + if (!*id || !strncmp(id, "card", 4)) { strcpy(id, "Default"); + is_default = true; + } - while (1) { - if (loops-- == 0) { - snd_printk(KERN_ERR "unable to set card id (%s)\n", id); - strcpy(card->id, card->proc_root->name); - return; - } - if (!snd_info_check_reserved_words(id)) - goto __change; - for (i = 0; i < snd_ecards_limit; i++) { - if (snd_cards[i] && !strcmp(snd_cards[i]->id, id)) - goto __change; - } - break; + with_suffix = false; + for (loops = 0; loops < SNDRV_CARDS; loops++) { + if (card_id_ok(card, id)) + return; /* OK */ - __change: len = strlen(id); - if (idx_flag) { - if (id[len-1] != '9') - id[len-1]++; - else - id[len-1] = 'A'; - } else if ((size_t)len <= sizeof(card->id) - 3) { - strcat(id, "_1"); - idx_flag++; + if (!with_suffix) { + /* add the "_X" suffix */ + char *spos = id + len; + if (len > sizeof(card->id) - 3) + spos = id + sizeof(card->id) - 3; + strcpy(spos, "_1"); + with_suffix = true; } else { - spos = id + len - 2; - if ((size_t)len <= sizeof(card->id) - 2) - spos++; - *(char *)spos++ = '_'; - *(char *)spos++ = '1'; - *(char *)spos++ = '\0'; - idx_flag++; + /* modify the existing suffix */ + if (id[len - 1] != '9') + id[len - 1]++; + else + id[len - 1] = 'A'; } } + /* fallback to the default id */ + if (!is_default) { + *id = 0; + goto again; + } + /* last resort... */ + snd_printk(KERN_ERR "unable to set card id (%s)\n", id); + if (card->proc_root->name) + strcpy(card->id, card->proc_root->name); } /** @@ -564,7 +594,7 @@ void snd_card_set_id(struct snd_card *card, const char *nid) if (card->id[0] != '\0') return; mutex_lock(&snd_card_mutex); - snd_card_set_id_no_lock(card, nid); + snd_card_set_id_no_lock(card, nid, nid); mutex_unlock(&snd_card_mutex); } EXPORT_SYMBOL(snd_card_set_id); @@ -596,22 +626,12 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, memcpy(buf1, buf, copy); buf1[copy] = '\0'; mutex_lock(&snd_card_mutex); - if (!snd_info_check_reserved_words(buf1)) { - __exist: + if (!card_id_ok(NULL, buf1)) { mutex_unlock(&snd_card_mutex); return -EEXIST; } - for (idx = 0; idx < snd_ecards_limit; idx++) { - if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) { - if (card == snd_cards[idx]) - goto __ok; - else - goto __exist; - } - } strcpy(card->id, buf1); snd_info_card_id_change(card); -__ok: mutex_unlock(&snd_card_mutex); return count; @@ -665,7 +685,18 @@ int snd_card_register(struct snd_card *card) mutex_unlock(&snd_card_mutex); return 0; } - snd_card_set_id_no_lock(card, card->id[0] == '\0' ? NULL : card->id); + if (*card->id) { + /* make a unique id name from the given string */ + char tmpid[sizeof(card->id)]; + memcpy(tmpid, card->id, sizeof(card->id)); + snd_card_set_id_no_lock(card, tmpid, tmpid); + } else { + /* create an id from either shortname or longname */ + const char *src; + src = *card->shortname ? card->shortname : card->longname; + snd_card_set_id_no_lock(card, src, + retrieve_id_from_card_name(src)); + } snd_cards[card->number] = card; mutex_unlock(&snd_card_mutex); init_info_for_card(card); -- cgit v1.2.3-18-g5258 From 18478e8b626edc2d181dcb1b93e1f99ad72095e9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Mar 2012 17:51:10 +0100 Subject: ALSA: hda - Initialize vmaster slave volumes When the driver is changed to use vmaster or a new slave element is added by the improvement of the parser code, user may face often the silent output because of the muted slave mixer although Master volume is properly set. And they complain. And I get upset. Although such a mixer element should be initialized via "alsactl init", it'd be more user-friendly if the known output slaves are unmuted and set to 0dB so that user can control the volume only with Master as default. Since Master is still set muted as default even with this change, no risk of the speaker blow up, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 65 ++++++++++++++++++++++++++++++++++++++++++-- sound/pci/hda/hda_local.h | 6 ++-- sound/pci/hda/patch_analog.c | 8 ++++-- 3 files changed, 72 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0527ae1ab96..0c0ac0e1d50 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -19,6 +19,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include #include #include #include @@ -2340,6 +2341,56 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) return 1; } +/* guess the value corresponding to 0dB */ +static int get_kctl_0dB_offset(struct snd_kcontrol *kctl) +{ + int _tlv[4]; + const int *tlv = NULL; + int val = -1; + + if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + /* FIXME: set_fs() hack for obtaining user-space TLV data */ + mm_segment_t fs = get_fs(); + set_fs(get_ds()); + if (!kctl->tlv.c(kctl, 0, sizeof(_tlv), _tlv)) + tlv = _tlv; + set_fs(fs); + } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ) + tlv = kctl->tlv.p; + if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE) + val = -tlv[2] / tlv[3]; + return val; +} + +/* call kctl->put with the given value(s) */ +static int put_kctl_with_value(struct snd_kcontrol *kctl, int val) +{ + struct snd_ctl_elem_value *ucontrol; + ucontrol = kzalloc(sizeof(*ucontrol), GFP_KERNEL); + if (!ucontrol) + return -ENOMEM; + ucontrol->value.integer.value[0] = val; + ucontrol->value.integer.value[1] = val; + kctl->put(kctl, ucontrol); + kfree(ucontrol); + return 0; +} + +/* initialize the slave volume with 0dB */ +static int init_slave_0dB(void *data, struct snd_kcontrol *slave) +{ + int offset = get_kctl_0dB_offset(slave); + if (offset > 0) + put_kctl_with_value(slave, offset); + return 0; +} + +/* unmute the slave */ +static int init_slave_unmute(void *data, struct snd_kcontrol *slave) +{ + return put_kctl_with_value(slave, 1); +} + /** * snd_hda_add_vmaster - create a virtual master control and add slaves * @codec: HD-audio codec @@ -2347,6 +2398,7 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) * @tlv: TLV data (optional) * @slaves: slave control names (optional) * @suffix: suffix string to each slave name (optional) + * @init_slave_vol: initialize slaves to unmute/0dB * * Create a virtual master control with the given name. The TLV data * must be either NULL or a valid data. @@ -2357,9 +2409,9 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl) * * This function returns zero if successful or a negative error code. */ -int snd_hda_add_vmaster(struct hda_codec *codec, char *name, +int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char * const *slaves, - const char *suffix) + const char *suffix, bool init_slave_vol) { struct snd_kcontrol *kctl; int err; @@ -2380,9 +2432,16 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, (map_slave_func_t)snd_ctl_add_slave, kctl); if (err < 0) return err; + + /* init with master mute & zero volume */ + put_kctl_with_value(kctl, 0); + if (init_slave_vol) + map_slaves(codec, slaves, suffix, + tlv ? init_slave_0dB : init_slave_unmute, kctl); + return 0; } -EXPORT_SYMBOL_HDA(snd_hda_add_vmaster); +EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster); /** * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 6094dea82bc..caa64686267 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -139,9 +139,11 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int *tlv); struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); -int snd_hda_add_vmaster(struct hda_codec *codec, char *name, +int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char * const *slaves, - const char *suffix); + const char *suffix, bool init_slave_vol); +#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ + __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true) int snd_hda_codec_reset(struct hda_codec *codec); /* amp value bits */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 9771b070245..fa97a0c5ced 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -82,6 +82,7 @@ struct ad198x_spec { unsigned int inv_jack_detect: 1;/* inverted jack-detection */ unsigned int inv_eapd: 1; /* inverted EAPD implementation */ unsigned int analog_beep: 1; /* analog beep input present */ + unsigned int avoid_init_slave_vol:1; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; @@ -223,11 +224,12 @@ static int ad198x_build_controls(struct hda_codec *codec) unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); - err = snd_hda_add_vmaster(codec, "Master Playback Volume", + err = __snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, (spec->slave_vols ? spec->slave_vols : ad_slave_pfxs), - "Playback Volume"); + "Playback Volume", + !spec->avoid_init_slave_vol); if (err < 0) return err; } @@ -3604,6 +3606,8 @@ static int patch_ad1884(struct hda_codec *codec) spec->vmaster_nid = 0x04; /* we need to cover all playback volumes */ spec->slave_vols = ad1884_slave_vols; + /* slaves may contain input volumes, so we can't raise to 0dB blindly */ + spec->avoid_init_slave_vol = 1; codec->patch_ops = ad198x_patch_ops; -- cgit v1.2.3-18-g5258 From 80f48143ffde97c48c5e550e2fcd2c9f8e77e554 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Mar 2012 10:33:34 +0000 Subject: ASoC: Revert widget I/O locking for 3.4 The widget locking depends on some of the other locking changes which are queued up for 3.5 not 3.4 so revert the locking changes and reapply them in 3.5. This reverts commit 66bf93212f19548f5ed221356b2d70189cc18254 and 96acc357bedad69fbc94d1b923a960af5a411c6f. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-dapm.c | 34 ++++++---------------------------- 1 file changed, 6 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1ba2a711b54..dcd11609f93 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -206,23 +206,7 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) return -1; } -static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) -{ - if (w->codec && !w->codec->using_regmap) - mutex_lock(&w->codec->mutex); - else if (w->platform) - mutex_lock(&w->platform->mutex); -} - -static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) -{ - if (w->codec && !w->codec->using_regmap) - mutex_unlock(&w->codec->mutex); - else if (w->platform) - mutex_unlock(&w->platform->mutex); -} - -static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, +static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { bool change; @@ -235,24 +219,18 @@ static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, if (ret != 0) return ret; } else { - soc_widget_lock(w); ret = soc_widget_read(w, reg); - if (ret < 0) { - soc_widget_unlock(w); + if (ret < 0) return ret; - } old = ret; new = (old & ~mask) | (value & mask); change = old != new; if (change) { ret = soc_widget_write(w, reg, new); - if (ret < 0) { - soc_widget_unlock(w); + if (ret < 0) return ret; - } } - soc_widget_unlock(w); } return change; @@ -869,7 +847,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, else val = w->off_val; - soc_widget_update_bits_locked(w, -(w->reg + 1), + soc_widget_update_bits(w, -(w->reg + 1), w->mask << w->shift, val << w->shift); return 0; @@ -1127,7 +1105,7 @@ static void dapm_seq_run_coalesced(struct snd_soc_dapm_context *dapm, "pop test : Applying 0x%x/0x%x to %x in %dms\n", value, mask, reg, card->pop_time); pop_wait(card->pop_time); - soc_widget_update_bits_locked(w, reg, mask, value); + soc_widget_update_bits(w, reg, mask, value); } list_for_each_entry(w, pending, power_list) { @@ -1257,7 +1235,7 @@ static void dapm_widget_update(struct snd_soc_dapm_context *dapm) w->name, ret); } - ret = soc_widget_update_bits_locked(w, update->reg, update->mask, + ret = snd_soc_update_bits(w->codec, update->reg, update->mask, update->val); if (ret < 0) pr_err("%s DAPM update failed: %d\n", w->name, ret); -- cgit v1.2.3-18-g5258 From 2ad787e9aae8bfac14fa96748c0f2b034577be6a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Mar 2012 12:18:37 +0100 Subject: ALSA: Add a hook capability to vmaster controls This patch adds a hook to vmaster control to be called at each time when the master value is changed. It'd be handy for an additional mute LED control following the Master switch, for example. Signed-off-by: Takashi Iwai --- sound/core/vmaster.c | 46 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 45 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 130cfe677d6..14a286a7bf2 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -37,6 +37,8 @@ struct link_master { struct link_ctl_info info; int val; /* the master value */ unsigned int tlv[4]; + void (*hook)(void *private_data, int); + void *hook_private_data; }; /* @@ -126,7 +128,9 @@ static int master_init(struct link_master *master) master->info.count = 1; /* always mono */ /* set full volume as default (= no attenuation) */ master->val = master->info.max_val; - return 0; + if (master->hook) + master->hook(master->hook_private_data, master->val); + return 1; } return -ENOENT; } @@ -329,6 +333,8 @@ static int master_put(struct snd_kcontrol *kcontrol, slave_put_val(slave, uval); } kfree(uval); + if (master->hook && !err) + master->hook(master->hook_private_data, master->val); return 1; } @@ -408,3 +414,41 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, return kctl; } EXPORT_SYMBOL(snd_ctl_make_virtual_master); + +/** + * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control + * @kcontrol: vmaster kctl element + * @hook: the hook function + * + * Adds the given hook to the vmaster control element so that it's called + * at each time when the value is changed. + */ +int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol, + void (*hook)(void *private_data, int), + void *private_data) +{ + struct link_master *master = snd_kcontrol_chip(kcontrol); + master->hook = hook; + master->hook_private_data = private_data; + return 0; +} +EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook); + +/** + * snd_ctl_sync_vmaster_hook - Sync the vmaster hook + * @kcontrol: vmaster kctl element + * + * Call the hook function to synchronize with the current value of the given + * vmaster element. NOP when NULL is passed to @kcontrol or the hook doesn't + * exist. + */ +void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kcontrol) +{ + struct link_master *master; + if (!kcontrol) + return; + master = snd_kcontrol_chip(kcontrol); + if (master->hook) + master->hook(master->hook_private_data, master->val); +} +EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster_hook); -- cgit v1.2.3-18-g5258 From 71e822e9dcd80923813705e5843eb39e065e8250 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 26 Jan 2012 12:47:22 +0200 Subject: OMAP: mcbsp: Move core driver under sound/soc/omap In order to consolidate the McBSP driver move it out from arch/arm/plat-omap directory under sound/soc/omap/ Signed-off-by: Peter Ujfalusi Acked-by: Tony Lindgren Tested-by: Grazvydas Ignotas Tested-by: Janusz Krzysztofik Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/Kconfig | 4 + sound/soc/omap/Makefile | 1 + sound/soc/omap/mcbsp.c | 1364 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 1369 insertions(+) create mode 100644 sound/soc/omap/mcbsp.c (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 47b23fea20c..27a3a29f7cd 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -5,6 +5,10 @@ config SND_OMAP_SOC config SND_OMAP_SOC_DMIC tristate +config OMAP_MCBSP + tristate + depends on ARCH_OMAP + config SND_OMAP_SOC_MCBSP tristate select OMAP_MCBSP diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 123ac18303e..9f8fbd554eb 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -7,6 +7,7 @@ snd-soc-omap-hdmi-objs := omap-hdmi.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_DMIC) += snd-soc-omap-dmic.o +obj-$(CONFIG_OMAP_MCBSP) += mcbsp.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o obj-$(CONFIG_SND_OMAP_SOC_HDMI) += snd-soc-omap-hdmi.o diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c new file mode 100644 index 00000000000..36d83b0c9f0 --- /dev/null +++ b/sound/soc/omap/mcbsp.c @@ -0,0 +1,1364 @@ +/* + * sound/soc/omap/mcbsp.c + * + * Copyright (C) 2004 Nokia Corporation + * Author: Samuel Ortiz + * + * Contact: Jarkko Nikula + * Peter Ujfalusi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Multichannel mode not supported. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +struct omap_mcbsp **mcbsp_ptr; +int omap_mcbsp_count; + +#define omap_mcbsp_check_valid_id(id) (id < omap_mcbsp_count) +#define id_to_mcbsp_ptr(id) mcbsp_ptr[id]; + +static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) +{ + void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; + + if (mcbsp->pdata->reg_size == 2) { + ((u16 *)mcbsp->reg_cache)[reg] = (u16)val; + __raw_writew((u16)val, addr); + } else { + ((u32 *)mcbsp->reg_cache)[reg] = val; + __raw_writel(val, addr); + } +} + +static int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, bool from_cache) +{ + void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; + + if (mcbsp->pdata->reg_size == 2) { + return !from_cache ? __raw_readw(addr) : + ((u16 *)mcbsp->reg_cache)[reg]; + } else { + return !from_cache ? __raw_readl(addr) : + ((u32 *)mcbsp->reg_cache)[reg]; + } +} + +static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) +{ + __raw_writel(val, mcbsp->st_data->io_base_st + reg); +} + +static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) +{ + return __raw_readl(mcbsp->st_data->io_base_st + reg); +} + +#define MCBSP_READ(mcbsp, reg) \ + omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 0) +#define MCBSP_WRITE(mcbsp, reg, val) \ + omap_mcbsp_write(mcbsp, OMAP_MCBSP_REG_##reg, val) +#define MCBSP_READ_CACHE(mcbsp, reg) \ + omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 1) + +#define MCBSP_ST_READ(mcbsp, reg) \ + omap_mcbsp_st_read(mcbsp, OMAP_ST_REG_##reg) +#define MCBSP_ST_WRITE(mcbsp, reg, val) \ + omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val) + +static void omap_mcbsp_dump_reg(u8 id) +{ + struct omap_mcbsp *mcbsp = id_to_mcbsp_ptr(id); + + dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id); + dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", + MCBSP_READ(mcbsp, DRR2)); + dev_dbg(mcbsp->dev, "DRR1: 0x%04x\n", + MCBSP_READ(mcbsp, DRR1)); + dev_dbg(mcbsp->dev, "DXR2: 0x%04x\n", + MCBSP_READ(mcbsp, DXR2)); + dev_dbg(mcbsp->dev, "DXR1: 0x%04x\n", + MCBSP_READ(mcbsp, DXR1)); + dev_dbg(mcbsp->dev, "SPCR2: 0x%04x\n", + MCBSP_READ(mcbsp, SPCR2)); + dev_dbg(mcbsp->dev, "SPCR1: 0x%04x\n", + MCBSP_READ(mcbsp, SPCR1)); + dev_dbg(mcbsp->dev, "RCR2: 0x%04x\n", + MCBSP_READ(mcbsp, RCR2)); + dev_dbg(mcbsp->dev, "RCR1: 0x%04x\n", + MCBSP_READ(mcbsp, RCR1)); + dev_dbg(mcbsp->dev, "XCR2: 0x%04x\n", + MCBSP_READ(mcbsp, XCR2)); + dev_dbg(mcbsp->dev, "XCR1: 0x%04x\n", + MCBSP_READ(mcbsp, XCR1)); + dev_dbg(mcbsp->dev, "SRGR2: 0x%04x\n", + MCBSP_READ(mcbsp, SRGR2)); + dev_dbg(mcbsp->dev, "SRGR1: 0x%04x\n", + MCBSP_READ(mcbsp, SRGR1)); + dev_dbg(mcbsp->dev, "PCR0: 0x%04x\n", + MCBSP_READ(mcbsp, PCR0)); + dev_dbg(mcbsp->dev, "***********************\n"); +} + +static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *dev_id) +{ + struct omap_mcbsp *mcbsp_tx = dev_id; + u16 irqst_spcr2; + + irqst_spcr2 = MCBSP_READ(mcbsp_tx, SPCR2); + dev_dbg(mcbsp_tx->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2); + + if (irqst_spcr2 & XSYNC_ERR) { + dev_err(mcbsp_tx->dev, "TX Frame Sync Error! : 0x%x\n", + irqst_spcr2); + /* Writing zero to XSYNC_ERR clears the IRQ */ + MCBSP_WRITE(mcbsp_tx, SPCR2, MCBSP_READ_CACHE(mcbsp_tx, SPCR2)); + } + + return IRQ_HANDLED; +} + +static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *dev_id) +{ + struct omap_mcbsp *mcbsp_rx = dev_id; + u16 irqst_spcr1; + + irqst_spcr1 = MCBSP_READ(mcbsp_rx, SPCR1); + dev_dbg(mcbsp_rx->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1); + + if (irqst_spcr1 & RSYNC_ERR) { + dev_err(mcbsp_rx->dev, "RX Frame Sync Error! : 0x%x\n", + irqst_spcr1); + /* Writing zero to RSYNC_ERR clears the IRQ */ + MCBSP_WRITE(mcbsp_rx, SPCR1, MCBSP_READ_CACHE(mcbsp_rx, SPCR1)); + } + + return IRQ_HANDLED; +} + +/* + * omap_mcbsp_config simply write a config to the + * appropriate McBSP. + * You either call this function or set the McBSP registers + * by yourself before calling omap_mcbsp_start(). + */ +void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg *config) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + + dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", + mcbsp->id, mcbsp->phys_base); + + /* We write the given config */ + MCBSP_WRITE(mcbsp, SPCR2, config->spcr2); + MCBSP_WRITE(mcbsp, SPCR1, config->spcr1); + MCBSP_WRITE(mcbsp, RCR2, config->rcr2); + MCBSP_WRITE(mcbsp, RCR1, config->rcr1); + MCBSP_WRITE(mcbsp, XCR2, config->xcr2); + MCBSP_WRITE(mcbsp, XCR1, config->xcr1); + MCBSP_WRITE(mcbsp, SRGR2, config->srgr2); + MCBSP_WRITE(mcbsp, SRGR1, config->srgr1); + MCBSP_WRITE(mcbsp, MCR2, config->mcr2); + MCBSP_WRITE(mcbsp, MCR1, config->mcr1); + MCBSP_WRITE(mcbsp, PCR0, config->pcr0); + if (mcbsp->pdata->has_ccr) { + MCBSP_WRITE(mcbsp, XCCR, config->xccr); + MCBSP_WRITE(mcbsp, RCCR, config->rccr); + } +} +EXPORT_SYMBOL(omap_mcbsp_config); + +/** + * omap_mcbsp_dma_params - returns the dma channel number + * @id - mcbsp id + * @stream - indicates the direction of data flow (rx or tx) + * + * Returns the dma channel number for the rx channel or tx channel + * based on the value of @stream for the requested mcbsp given by @id + */ +int omap_mcbsp_dma_ch_params(unsigned int id, unsigned int stream) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + if (stream) + return mcbsp->dma_rx_sync; + else + return mcbsp->dma_tx_sync; +} +EXPORT_SYMBOL(omap_mcbsp_dma_ch_params); + +/** + * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register + * @id - mcbsp id + * @stream - indicates the direction of data flow (rx or tx) + * + * Returns the address of mcbsp data transmit register or data receive register + * to be used by DMA for transferring/receiving data based on the value of + * @stream for the requested mcbsp given by @id + */ +int omap_mcbsp_dma_reg_params(unsigned int id, unsigned int stream) +{ + struct omap_mcbsp *mcbsp; + int data_reg; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + if (mcbsp->pdata->reg_size == 2) { + if (stream) + data_reg = OMAP_MCBSP_REG_DRR1; + else + data_reg = OMAP_MCBSP_REG_DXR1; + } else { + if (stream) + data_reg = OMAP_MCBSP_REG_DRR; + else + data_reg = OMAP_MCBSP_REG_DXR; + } + + return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step; +} +EXPORT_SYMBOL(omap_mcbsp_dma_reg_params); + +static void omap_st_on(struct omap_mcbsp *mcbsp) +{ + unsigned int w; + + if (mcbsp->pdata->enable_st_clock) + mcbsp->pdata->enable_st_clock(mcbsp->id, 1); + + /* Enable McBSP Sidetone */ + w = MCBSP_READ(mcbsp, SSELCR); + MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN); + + /* Enable Sidetone from Sidetone Core */ + w = MCBSP_ST_READ(mcbsp, SSELCR); + MCBSP_ST_WRITE(mcbsp, SSELCR, w | ST_SIDETONEEN); +} + +static void omap_st_off(struct omap_mcbsp *mcbsp) +{ + unsigned int w; + + w = MCBSP_ST_READ(mcbsp, SSELCR); + MCBSP_ST_WRITE(mcbsp, SSELCR, w & ~(ST_SIDETONEEN)); + + w = MCBSP_READ(mcbsp, SSELCR); + MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN)); + + if (mcbsp->pdata->enable_st_clock) + mcbsp->pdata->enable_st_clock(mcbsp->id, 0); +} + +static void omap_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir) +{ + u16 val, i; + + val = MCBSP_ST_READ(mcbsp, SSELCR); + + if (val & ST_COEFFWREN) + MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); + + MCBSP_ST_WRITE(mcbsp, SSELCR, val | ST_COEFFWREN); + + for (i = 0; i < 128; i++) + MCBSP_ST_WRITE(mcbsp, SFIRCR, fir[i]); + + i = 0; + + val = MCBSP_ST_READ(mcbsp, SSELCR); + while (!(val & ST_COEFFWRDONE) && (++i < 1000)) + val = MCBSP_ST_READ(mcbsp, SSELCR); + + MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN)); + + if (i == 1000) + dev_err(mcbsp->dev, "McBSP FIR load error!\n"); +} + +static void omap_st_chgain(struct omap_mcbsp *mcbsp) +{ + u16 w; + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + w = MCBSP_ST_READ(mcbsp, SSELCR); + + MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) | \ + ST_CH1GAIN(st_data->ch1gain)); +} + +int omap_st_set_chgain(unsigned int id, int channel, s16 chgain) +{ + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; + int ret = 0; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; + + if (!st_data) + return -ENOENT; + + spin_lock_irq(&mcbsp->lock); + if (channel == 0) + st_data->ch0gain = chgain; + else if (channel == 1) + st_data->ch1gain = chgain; + else + ret = -EINVAL; + + if (st_data->enabled) + omap_st_chgain(mcbsp); + spin_unlock_irq(&mcbsp->lock); + + return ret; +} +EXPORT_SYMBOL(omap_st_set_chgain); + +int omap_st_get_chgain(unsigned int id, int channel, s16 *chgain) +{ + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; + int ret = 0; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; + + if (!st_data) + return -ENOENT; + + spin_lock_irq(&mcbsp->lock); + if (channel == 0) + *chgain = st_data->ch0gain; + else if (channel == 1) + *chgain = st_data->ch1gain; + else + ret = -EINVAL; + spin_unlock_irq(&mcbsp->lock); + + return ret; +} +EXPORT_SYMBOL(omap_st_get_chgain); + +static int omap_st_start(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (st_data && st_data->enabled && !st_data->running) { + omap_st_fir_write(mcbsp, st_data->taps); + omap_st_chgain(mcbsp); + + if (!mcbsp->free) { + omap_st_on(mcbsp); + st_data->running = 1; + } + } + + return 0; +} + +int omap_st_enable(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; + + if (!st_data) + return -ENODEV; + + spin_lock_irq(&mcbsp->lock); + st_data->enabled = 1; + omap_st_start(mcbsp); + spin_unlock_irq(&mcbsp->lock); + + return 0; +} +EXPORT_SYMBOL(omap_st_enable); + +static int omap_st_stop(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + if (st_data && st_data->running) { + if (!mcbsp->free) { + omap_st_off(mcbsp); + st_data->running = 0; + } + } + + return 0; +} + +int omap_st_disable(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; + int ret = 0; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; + + if (!st_data) + return -ENODEV; + + spin_lock_irq(&mcbsp->lock); + omap_st_stop(mcbsp); + st_data->enabled = 0; + spin_unlock_irq(&mcbsp->lock); + + return ret; +} +EXPORT_SYMBOL(omap_st_disable); + +int omap_st_is_enabled(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + struct omap_mcbsp_st_data *st_data; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + + mcbsp = id_to_mcbsp_ptr(id); + st_data = mcbsp->st_data; + + if (!st_data) + return -ENODEV; + + + return st_data->enabled; +} +EXPORT_SYMBOL(omap_st_is_enabled); + +/* + * omap_mcbsp_set_rx_threshold configures the transmit threshold in words. + * The threshold parameter is 1 based, and it is converted (threshold - 1) + * for the THRSH2 register. + */ +void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + if (mcbsp->pdata->buffer_size == 0) + return; + + if (threshold && threshold <= mcbsp->max_tx_thres) + MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); +} +EXPORT_SYMBOL(omap_mcbsp_set_tx_threshold); + +/* + * omap_mcbsp_set_rx_threshold configures the receive threshold in words. + * The threshold parameter is 1 based, and it is converted (threshold - 1) + * for the THRSH1 register. + */ +void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + if (mcbsp->pdata->buffer_size == 0) + return; + + if (threshold && threshold <= mcbsp->max_rx_thres) + MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); +} +EXPORT_SYMBOL(omap_mcbsp_set_rx_threshold); + +/* + * omap_mcbsp_get_max_tx_thres just return the current configured + * maximum threshold for transmission + */ +u16 omap_mcbsp_get_max_tx_threshold(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + return mcbsp->max_tx_thres; +} +EXPORT_SYMBOL(omap_mcbsp_get_max_tx_threshold); + +/* + * omap_mcbsp_get_max_rx_thres just return the current configured + * maximum threshold for reception + */ +u16 omap_mcbsp_get_max_rx_threshold(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + return mcbsp->max_rx_thres; +} +EXPORT_SYMBOL(omap_mcbsp_get_max_rx_threshold); + +u16 omap_mcbsp_get_fifo_size(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + return mcbsp->pdata->buffer_size; +} +EXPORT_SYMBOL(omap_mcbsp_get_fifo_size); + +/* + * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO + */ +u16 omap_mcbsp_get_tx_delay(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + u16 buffstat; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + if (mcbsp->pdata->buffer_size == 0) + return 0; + + /* Returns the number of free locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, XBUFFSTAT); + + /* Number of slots are different in McBSP ports */ + return mcbsp->pdata->buffer_size - buffstat; +} +EXPORT_SYMBOL(omap_mcbsp_get_tx_delay); + +/* + * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO + * to reach the threshold value (when the DMA will be triggered to read it) + */ +u16 omap_mcbsp_get_rx_delay(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + u16 buffstat, threshold; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + if (mcbsp->pdata->buffer_size == 0) + return 0; + + /* Returns the number of used locations in the buffer */ + buffstat = MCBSP_READ(mcbsp, RBUFFSTAT); + /* RX threshold */ + threshold = MCBSP_READ(mcbsp, THRSH1); + + /* Return the number of location till we reach the threshold limit */ + if (threshold <= buffstat) + return 0; + else + return threshold - buffstat; +} +EXPORT_SYMBOL(omap_mcbsp_get_rx_delay); + +/* + * omap_mcbsp_get_dma_op_mode just return the current configured + * operating mode for the mcbsp channel + */ +int omap_mcbsp_get_dma_op_mode(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + int dma_op_mode; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%u)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + dma_op_mode = mcbsp->dma_op_mode; + + return dma_op_mode; +} +EXPORT_SYMBOL(omap_mcbsp_get_dma_op_mode); + +int omap_mcbsp_request(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + void *reg_cache; + int err; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return -ENODEV; + } + mcbsp = id_to_mcbsp_ptr(id); + + reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL); + if (!reg_cache) { + return -ENOMEM; + } + + spin_lock(&mcbsp->lock); + if (!mcbsp->free) { + dev_err(mcbsp->dev, "McBSP%d is currently in use\n", + mcbsp->id); + err = -EBUSY; + goto err_kfree; + } + + mcbsp->free = false; + mcbsp->reg_cache = reg_cache; + spin_unlock(&mcbsp->lock); + + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) + mcbsp->pdata->ops->request(id); + + pm_runtime_get_sync(mcbsp->dev); + + /* Enable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); + + /* + * Make sure that transmitter, receiver and sample-rate generator are + * not running before activating IRQs. + */ + MCBSP_WRITE(mcbsp, SPCR1, 0); + MCBSP_WRITE(mcbsp, SPCR2, 0); + + err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, + 0, "McBSP", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request TX IRQ %d " + "for McBSP%d\n", mcbsp->tx_irq, + mcbsp->id); + goto err_clk_disable; + } + + if (mcbsp->rx_irq) { + err = request_irq(mcbsp->rx_irq, + omap_mcbsp_rx_irq_handler, + 0, "McBSP", (void *)mcbsp); + if (err != 0) { + dev_err(mcbsp->dev, "Unable to request RX IRQ %d " + "for McBSP%d\n", mcbsp->rx_irq, + mcbsp->id); + goto err_free_irq; + } + } + + return 0; +err_free_irq: + free_irq(mcbsp->tx_irq, (void *)mcbsp); +err_clk_disable: + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(id); + + /* Disable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + + pm_runtime_put_sync(mcbsp->dev); + + spin_lock(&mcbsp->lock); + mcbsp->free = true; + mcbsp->reg_cache = NULL; +err_kfree: + spin_unlock(&mcbsp->lock); + kfree(reg_cache); + + return err; +} +EXPORT_SYMBOL(omap_mcbsp_request); + +void omap_mcbsp_free(unsigned int id) +{ + struct omap_mcbsp *mcbsp; + void *reg_cache; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + + if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(id); + + /* Disable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, 0); + + pm_runtime_put_sync(mcbsp->dev); + + if (mcbsp->rx_irq) + free_irq(mcbsp->rx_irq, (void *)mcbsp); + free_irq(mcbsp->tx_irq, (void *)mcbsp); + + reg_cache = mcbsp->reg_cache; + + spin_lock(&mcbsp->lock); + if (mcbsp->free) + dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); + else + mcbsp->free = true; + mcbsp->reg_cache = NULL; + spin_unlock(&mcbsp->lock); + + if (reg_cache) + kfree(reg_cache); +} +EXPORT_SYMBOL(omap_mcbsp_free); + +/* + * Here we start the McBSP, by enabling transmitter, receiver or both. + * If no transmitter or receiver is active prior calling, then sample-rate + * generator and frame sync are started. + */ +void omap_mcbsp_start(unsigned int id, int tx, int rx) +{ + struct omap_mcbsp *mcbsp; + int enable_srg = 0; + u16 w; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + mcbsp = id_to_mcbsp_ptr(id); + + if (mcbsp->st_data) + omap_st_start(mcbsp); + + /* Only enable SRG, if McBSP is master */ + w = MCBSP_READ_CACHE(mcbsp, PCR0); + if (w & (FSXM | FSRM | CLKXM | CLKRM)) + enable_srg = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | + MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); + + if (enable_srg) { + /* Start the sample generator */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 6)); + } + + /* Enable transmitter and receiver */ + tx &= 1; + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | tx); + + rx &= 1; + w = MCBSP_READ_CACHE(mcbsp, SPCR1); + MCBSP_WRITE(mcbsp, SPCR1, w | rx); + + /* + * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec + * REVISIT: 100us may give enough time for two CLKSRG, however + * due to some unknown PM related, clock gating etc. reason it + * is now at 500us. + */ + udelay(500); + + if (enable_srg) { + /* Start frame sync */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 7)); + } + + if (mcbsp->pdata->has_ccr) { + /* Release the transmitter and receiver */ + w = MCBSP_READ_CACHE(mcbsp, XCCR); + w &= ~(tx ? XDISABLE : 0); + MCBSP_WRITE(mcbsp, XCCR, w); + w = MCBSP_READ_CACHE(mcbsp, RCCR); + w &= ~(rx ? RDISABLE : 0); + MCBSP_WRITE(mcbsp, RCCR, w); + } + + /* Dump McBSP Regs */ + omap_mcbsp_dump_reg(id); +} +EXPORT_SYMBOL(omap_mcbsp_start); + +void omap_mcbsp_stop(unsigned int id, int tx, int rx) +{ + struct omap_mcbsp *mcbsp; + int idle; + u16 w; + + if (!omap_mcbsp_check_valid_id(id)) { + printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); + return; + } + + mcbsp = id_to_mcbsp_ptr(id); + + /* Reset transmitter */ + tx &= 1; + if (mcbsp->pdata->has_ccr) { + w = MCBSP_READ_CACHE(mcbsp, XCCR); + w |= (tx ? XDISABLE : 0); + MCBSP_WRITE(mcbsp, XCCR, w); + } + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w & ~tx); + + /* Reset receiver */ + rx &= 1; + if (mcbsp->pdata->has_ccr) { + w = MCBSP_READ_CACHE(mcbsp, RCCR); + w |= (rx ? RDISABLE : 0); + MCBSP_WRITE(mcbsp, RCCR, w); + } + w = MCBSP_READ_CACHE(mcbsp, SPCR1); + MCBSP_WRITE(mcbsp, SPCR1, w & ~rx); + + idle = !((MCBSP_READ_CACHE(mcbsp, SPCR2) | + MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1); + + if (idle) { + /* Reset the sample rate generator */ + w = MCBSP_READ_CACHE(mcbsp, SPCR2); + MCBSP_WRITE(mcbsp, SPCR2, w & ~(1 << 6)); + } + + if (mcbsp->st_data) + omap_st_stop(mcbsp); +} +EXPORT_SYMBOL(omap_mcbsp_stop); + +int omap2_mcbsp_set_clks_src(u8 id, u8 fck_src_id) +{ + struct omap_mcbsp *mcbsp; + const char *src; + + if (!omap_mcbsp_check_valid_id(id)) { + pr_err("%s: Invalid id (%d)\n", __func__, id + 1); + return -EINVAL; + } + mcbsp = id_to_mcbsp_ptr(id); + + if (fck_src_id == MCBSP_CLKS_PAD_SRC) + src = "clks_ext"; + else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) + src = "clks_fclk"; + else + return -EINVAL; + + if (mcbsp->pdata->set_clk_src) + return mcbsp->pdata->set_clk_src(mcbsp->dev, mcbsp->fclk, src); + else + return -EINVAL; +} +EXPORT_SYMBOL(omap2_mcbsp_set_clks_src); + +void omap2_mcbsp1_mux_clkr_src(u8 mux) +{ + struct omap_mcbsp *mcbsp; + const char *src; + + if (mux == CLKR_SRC_CLKR) + src = "clkr"; + else if (mux == CLKR_SRC_CLKX) + src = "clkx"; + else + return; + + mcbsp = id_to_mcbsp_ptr(0); + if (mcbsp->pdata->mux_signal) + mcbsp->pdata->mux_signal(mcbsp->dev, "clkr", src); +} +EXPORT_SYMBOL(omap2_mcbsp1_mux_clkr_src); + +void omap2_mcbsp1_mux_fsr_src(u8 mux) +{ + struct omap_mcbsp *mcbsp; + const char *src; + + if (mux == FSR_SRC_FSR) + src = "fsr"; + else if (mux == FSR_SRC_FSX) + src = "fsx"; + else + return; + + mcbsp = id_to_mcbsp_ptr(0); + if (mcbsp->pdata->mux_signal) + mcbsp->pdata->mux_signal(mcbsp->dev, "fsr", src); +} +EXPORT_SYMBOL(omap2_mcbsp1_mux_fsr_src); + +#define max_thres(m) (mcbsp->pdata->buffer_size) +#define valid_threshold(m, val) ((val) <= max_thres(m)) +#define THRESHOLD_PROP_BUILDER(prop) \ +static ssize_t prop##_show(struct device *dev, \ + struct device_attribute *attr, char *buf) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + \ + return sprintf(buf, "%u\n", mcbsp->prop); \ +} \ + \ +static ssize_t prop##_store(struct device *dev, \ + struct device_attribute *attr, \ + const char *buf, size_t size) \ +{ \ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); \ + unsigned long val; \ + int status; \ + \ + status = strict_strtoul(buf, 0, &val); \ + if (status) \ + return status; \ + \ + if (!valid_threshold(mcbsp, val)) \ + return -EDOM; \ + \ + mcbsp->prop = val; \ + return size; \ +} \ + \ +static DEVICE_ATTR(prop, 0644, prop##_show, prop##_store); + +THRESHOLD_PROP_BUILDER(max_tx_thres); +THRESHOLD_PROP_BUILDER(max_rx_thres); + +static const char *dma_op_modes[] = { + "element", "threshold", "frame", +}; + +static ssize_t dma_op_mode_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + int dma_op_mode, i = 0; + ssize_t len = 0; + const char * const *s; + + dma_op_mode = mcbsp->dma_op_mode; + + for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) { + if (dma_op_mode == i) + len += sprintf(buf + len, "[%s] ", *s); + else + len += sprintf(buf + len, "%s ", *s); + } + len += sprintf(buf + len, "\n"); + + return len; +} + +static ssize_t dma_op_mode_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t size) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + const char * const *s; + int i = 0; + + for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) + if (sysfs_streq(buf, *s)) + break; + + if (i == ARRAY_SIZE(dma_op_modes)) + return -EINVAL; + + spin_lock_irq(&mcbsp->lock); + if (!mcbsp->free) { + size = -EBUSY; + goto unlock; + } + mcbsp->dma_op_mode = i; + +unlock: + spin_unlock_irq(&mcbsp->lock); + + return size; +} + +static DEVICE_ATTR(dma_op_mode, 0644, dma_op_mode_show, dma_op_mode_store); + +static const struct attribute *additional_attrs[] = { + &dev_attr_max_tx_thres.attr, + &dev_attr_max_rx_thres.attr, + &dev_attr_dma_op_mode.attr, + NULL, +}; + +static const struct attribute_group additional_attr_group = { + .attrs = (struct attribute **)additional_attrs, +}; + +static ssize_t st_taps_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + ssize_t status = 0; + int i; + + spin_lock_irq(&mcbsp->lock); + for (i = 0; i < st_data->nr_taps; i++) + status += sprintf(&buf[status], (i ? ", %d" : "%d"), + st_data->taps[i]); + if (i) + status += sprintf(&buf[status], "\n"); + spin_unlock_irq(&mcbsp->lock); + + return status; +} + +static ssize_t st_taps_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t size) +{ + struct omap_mcbsp *mcbsp = dev_get_drvdata(dev); + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + int val, tmp, status, i = 0; + + spin_lock_irq(&mcbsp->lock); + memset(st_data->taps, 0, sizeof(st_data->taps)); + st_data->nr_taps = 0; + + do { + status = sscanf(buf, "%d%n", &val, &tmp); + if (status < 0 || status == 0) { + size = -EINVAL; + goto out; + } + if (val < -32768 || val > 32767) { + size = -EINVAL; + goto out; + } + st_data->taps[i++] = val; + buf += tmp; + if (*buf != ',') + break; + buf++; + } while (1); + + st_data->nr_taps = i; + +out: + spin_unlock_irq(&mcbsp->lock); + + return size; +} + +static DEVICE_ATTR(st_taps, 0644, st_taps_show, st_taps_store); + +static const struct attribute *sidetone_attrs[] = { + &dev_attr_st_taps.attr, + NULL, +}; + +static const struct attribute_group sidetone_attr_group = { + .attrs = (struct attribute **)sidetone_attrs, +}; + +static int __devinit omap_st_add(struct omap_mcbsp *mcbsp, + struct resource *res) +{ + struct omap_mcbsp_st_data *st_data; + int err; + + st_data = kzalloc(sizeof(*mcbsp->st_data), GFP_KERNEL); + if (!st_data) { + err = -ENOMEM; + goto err1; + } + + st_data->io_base_st = ioremap(res->start, resource_size(res)); + if (!st_data->io_base_st) { + err = -ENOMEM; + goto err2; + } + + err = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + if (err) + goto err3; + + mcbsp->st_data = st_data; + return 0; + +err3: + iounmap(st_data->io_base_st); +err2: + kfree(st_data); +err1: + return err; + +} + +static void __devexit omap_st_remove(struct omap_mcbsp *mcbsp) +{ + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; + + sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); + iounmap(st_data->io_base_st); + kfree(st_data); +} + +/* + * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. + * 730 has only 2 McBSP, and both of them are MPU peripherals. + */ +static int __devinit omap_mcbsp_probe(struct platform_device *pdev) +{ + struct omap_mcbsp_platform_data *pdata = pdev->dev.platform_data; + struct omap_mcbsp *mcbsp; + int id = pdev->id - 1; + struct resource *res; + int ret = 0; + + if (!pdata) { + dev_err(&pdev->dev, "McBSP device initialized without" + "platform data\n"); + ret = -EINVAL; + goto exit; + } + + dev_dbg(&pdev->dev, "Initializing OMAP McBSP (%d).\n", pdev->id); + + if (id >= omap_mcbsp_count) { + dev_err(&pdev->dev, "Invalid McBSP device id (%d)\n", id); + ret = -EINVAL; + goto exit; + } + + mcbsp = kzalloc(sizeof(struct omap_mcbsp), GFP_KERNEL); + if (!mcbsp) { + ret = -ENOMEM; + goto exit; + } + + spin_lock_init(&mcbsp->lock); + mcbsp->id = id + 1; + mcbsp->free = true; + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); + if (!res) { + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "%s:mcbsp%d has invalid memory" + "resource\n", __func__, pdev->id); + ret = -ENOMEM; + goto exit; + } + } + mcbsp->phys_base = res->start; + mcbsp->reg_cache_size = resource_size(res); + mcbsp->io_base = ioremap(res->start, resource_size(res)); + if (!mcbsp->io_base) { + ret = -ENOMEM; + goto err_ioremap; + } + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); + if (!res) + mcbsp->phys_dma_base = mcbsp->phys_base; + else + mcbsp->phys_dma_base = res->start; + + mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx"); + mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); + + /* From OMAP4 there will be a single irq line */ + if (mcbsp->tx_irq == -ENXIO) + mcbsp->tx_irq = platform_get_irq(pdev, 0); + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); + if (!res) { + dev_err(&pdev->dev, "%s:mcbsp%d has invalid rx DMA channel\n", + __func__, pdev->id); + ret = -ENODEV; + goto err_res; + } + mcbsp->dma_rx_sync = res->start; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); + if (!res) { + dev_err(&pdev->dev, "%s:mcbsp%d has invalid tx DMA channel\n", + __func__, pdev->id); + ret = -ENODEV; + goto err_res; + } + mcbsp->dma_tx_sync = res->start; + + mcbsp->fclk = clk_get(&pdev->dev, "fck"); + if (IS_ERR(mcbsp->fclk)) { + ret = PTR_ERR(mcbsp->fclk); + dev_err(&pdev->dev, "unable to get fck: %d\n", ret); + goto err_res; + } + + mcbsp->pdata = pdata; + mcbsp->dev = &pdev->dev; + mcbsp_ptr[id] = mcbsp; + platform_set_drvdata(pdev, mcbsp); + pm_runtime_enable(mcbsp->dev); + + mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; + if (mcbsp->pdata->buffer_size) { + /* + * Initially configure the maximum thresholds to a safe value. + * The McBSP FIFO usage with these values should not go under + * 16 locations. + * If the whole FIFO without safety buffer is used, than there + * is a possibility that the DMA will be not able to push the + * new data on time, causing channel shifts in runtime. + */ + mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; + mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; + + ret = sysfs_create_group(&mcbsp->dev->kobj, + &additional_attr_group); + if (ret) { + dev_err(mcbsp->dev, + "Unable to create additional controls\n"); + goto err_thres; + } + } else { + mcbsp->max_tx_thres = -EINVAL; + mcbsp->max_rx_thres = -EINVAL; + } + + res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "sidetone"); + if (res) { + ret = omap_st_add(mcbsp, res); + if (ret) { + dev_err(mcbsp->dev, + "Unable to create sidetone controls\n"); + goto err_st; + } + } + + return 0; + +err_st: + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, + &additional_attr_group); +err_thres: + clk_put(mcbsp->fclk); +err_res: + iounmap(mcbsp->io_base); +err_ioremap: + kfree(mcbsp); +exit: + return ret; +} + +static int __devexit omap_mcbsp_remove(struct platform_device *pdev) +{ + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + + platform_set_drvdata(pdev, NULL); + if (mcbsp) { + + if (mcbsp->pdata && mcbsp->pdata->ops && + mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(mcbsp->id); + + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, + &additional_attr_group); + + if (mcbsp->st_data) + omap_st_remove(mcbsp); + + clk_put(mcbsp->fclk); + + iounmap(mcbsp->io_base); + kfree(mcbsp); + } + + return 0; +} + +static struct platform_driver omap_mcbsp_driver = { + .probe = omap_mcbsp_probe, + .remove = __devexit_p(omap_mcbsp_remove), + .driver = { + .name = "omap-mcbsp", + }, +}; + +module_platform_driver(omap_mcbsp_driver); + +MODULE_AUTHOR("Samuel Ortiz "); +MODULE_DESCRIPTION("OMAP McBSP core driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap-mcbsp"); -- cgit v1.2.3-18-g5258 From 219f43164e8c611c6b8e7b628def9183098b430b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 3 Feb 2012 13:11:47 +0200 Subject: ASoC: OMAP: McBSP: Consolidate plat/mcbsp.h content Move most of the content of the plat/mcbsp.h header file under sound/soc/omap/ to help further cleanups. Signed-off-by: Peter Ujfalusi Tested-by: Grazvydas Ignotas Tested-by: Janusz Krzysztofik Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 2 + sound/soc/omap/mcbsp.h | 292 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcbsp.c | 10 ++ 3 files changed, 304 insertions(+) create mode 100644 sound/soc/omap/mcbsp.h (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 36d83b0c9f0..20d46bf3626 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -28,6 +28,8 @@ #include #include +#include "mcbsp.h" + struct omap_mcbsp **mcbsp_ptr; int omap_mcbsp_count; diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h new file mode 100644 index 00000000000..5590ab271ee --- /dev/null +++ b/sound/soc/omap/mcbsp.h @@ -0,0 +1,292 @@ +/* + * sound/soc/omap/mcbsp.h + * + * OMAP Multi-Channel Buffered Serial Port + * + * Contact: Jarkko Nikula + * Peter Ujfalusi + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef __ASOC_MCBSP_H +#define __ASOC_MCBSP_H + +/* McBSP register numbers. Register address offset = num * reg_step */ +enum { + /* Common registers */ + OMAP_MCBSP_REG_SPCR2 = 4, + OMAP_MCBSP_REG_SPCR1, + OMAP_MCBSP_REG_RCR2, + OMAP_MCBSP_REG_RCR1, + OMAP_MCBSP_REG_XCR2, + OMAP_MCBSP_REG_XCR1, + OMAP_MCBSP_REG_SRGR2, + OMAP_MCBSP_REG_SRGR1, + OMAP_MCBSP_REG_MCR2, + OMAP_MCBSP_REG_MCR1, + OMAP_MCBSP_REG_RCERA, + OMAP_MCBSP_REG_RCERB, + OMAP_MCBSP_REG_XCERA, + OMAP_MCBSP_REG_XCERB, + OMAP_MCBSP_REG_PCR0, + OMAP_MCBSP_REG_RCERC, + OMAP_MCBSP_REG_RCERD, + OMAP_MCBSP_REG_XCERC, + OMAP_MCBSP_REG_XCERD, + OMAP_MCBSP_REG_RCERE, + OMAP_MCBSP_REG_RCERF, + OMAP_MCBSP_REG_XCERE, + OMAP_MCBSP_REG_XCERF, + OMAP_MCBSP_REG_RCERG, + OMAP_MCBSP_REG_RCERH, + OMAP_MCBSP_REG_XCERG, + OMAP_MCBSP_REG_XCERH, + + /* OMAP1-OMAP2420 registers */ + OMAP_MCBSP_REG_DRR2 = 0, + OMAP_MCBSP_REG_DRR1, + OMAP_MCBSP_REG_DXR2, + OMAP_MCBSP_REG_DXR1, + + /* OMAP2430 and onwards */ + OMAP_MCBSP_REG_DRR = 0, + OMAP_MCBSP_REG_DXR = 2, + OMAP_MCBSP_REG_SYSCON = 35, + OMAP_MCBSP_REG_THRSH2, + OMAP_MCBSP_REG_THRSH1, + OMAP_MCBSP_REG_IRQST = 40, + OMAP_MCBSP_REG_IRQEN, + OMAP_MCBSP_REG_WAKEUPEN, + OMAP_MCBSP_REG_XCCR, + OMAP_MCBSP_REG_RCCR, + OMAP_MCBSP_REG_XBUFFSTAT, + OMAP_MCBSP_REG_RBUFFSTAT, + OMAP_MCBSP_REG_SSELCR, +}; + +/* OMAP3 sidetone control registers */ +#define OMAP_ST_REG_REV 0x00 +#define OMAP_ST_REG_SYSCONFIG 0x10 +#define OMAP_ST_REG_IRQSTATUS 0x18 +#define OMAP_ST_REG_IRQENABLE 0x1C +#define OMAP_ST_REG_SGAINCR 0x24 +#define OMAP_ST_REG_SFIRCR 0x28 +#define OMAP_ST_REG_SSELCR 0x2C + +/************************** McBSP SPCR1 bit definitions ***********************/ +#define RRST 0x0001 +#define RRDY 0x0002 +#define RFULL 0x0004 +#define RSYNC_ERR 0x0008 +#define RINTM(value) ((value)<<4) /* bits 4:5 */ +#define ABIS 0x0040 +#define DXENA 0x0080 +#define CLKSTP(value) ((value)<<11) /* bits 11:12 */ +#define RJUST(value) ((value)<<13) /* bits 13:14 */ +#define ALB 0x8000 +#define DLB 0x8000 + +/************************** McBSP SPCR2 bit definitions ***********************/ +#define XRST 0x0001 +#define XRDY 0x0002 +#define XEMPTY 0x0004 +#define XSYNC_ERR 0x0008 +#define XINTM(value) ((value)<<4) /* bits 4:5 */ +#define GRST 0x0040 +#define FRST 0x0080 +#define SOFT 0x0100 +#define FREE 0x0200 + +/************************** McBSP PCR bit definitions *************************/ +#define CLKRP 0x0001 +#define CLKXP 0x0002 +#define FSRP 0x0004 +#define FSXP 0x0008 +#define DR_STAT 0x0010 +#define DX_STAT 0x0020 +#define CLKS_STAT 0x0040 +#define SCLKME 0x0080 +#define CLKRM 0x0100 +#define CLKXM 0x0200 +#define FSRM 0x0400 +#define FSXM 0x0800 +#define RIOEN 0x1000 +#define XIOEN 0x2000 +#define IDLE_EN 0x4000 + +/************************** McBSP RCR1 bit definitions ************************/ +#define RWDLEN1(value) ((value)<<5) /* Bits 5:7 */ +#define RFRLEN1(value) ((value)<<8) /* Bits 8:14 */ + +/************************** McBSP XCR1 bit definitions ************************/ +#define XWDLEN1(value) ((value)<<5) /* Bits 5:7 */ +#define XFRLEN1(value) ((value)<<8) /* Bits 8:14 */ + +/*************************** McBSP RCR2 bit definitions ***********************/ +#define RDATDLY(value) (value) /* Bits 0:1 */ +#define RFIG 0x0004 +#define RCOMPAND(value) ((value)<<3) /* Bits 3:4 */ +#define RWDLEN2(value) ((value)<<5) /* Bits 5:7 */ +#define RFRLEN2(value) ((value)<<8) /* Bits 8:14 */ +#define RPHASE 0x8000 + +/*************************** McBSP XCR2 bit definitions ***********************/ +#define XDATDLY(value) (value) /* Bits 0:1 */ +#define XFIG 0x0004 +#define XCOMPAND(value) ((value)<<3) /* Bits 3:4 */ +#define XWDLEN2(value) ((value)<<5) /* Bits 5:7 */ +#define XFRLEN2(value) ((value)<<8) /* Bits 8:14 */ +#define XPHASE 0x8000 + +/************************* McBSP SRGR1 bit definitions ************************/ +#define CLKGDV(value) (value) /* Bits 0:7 */ +#define FWID(value) ((value)<<8) /* Bits 8:15 */ + +/************************* McBSP SRGR2 bit definitions ************************/ +#define FPER(value) (value) /* Bits 0:11 */ +#define FSGM 0x1000 +#define CLKSM 0x2000 +#define CLKSP 0x4000 +#define GSYNC 0x8000 + +/************************* McBSP MCR1 bit definitions *************************/ +#define RMCM 0x0001 +#define RCBLK(value) ((value)<<2) /* Bits 2:4 */ +#define RPABLK(value) ((value)<<5) /* Bits 5:6 */ +#define RPBBLK(value) ((value)<<7) /* Bits 7:8 */ + +/************************* McBSP MCR2 bit definitions *************************/ +#define XMCM(value) (value) /* Bits 0:1 */ +#define XCBLK(value) ((value)<<2) /* Bits 2:4 */ +#define XPABLK(value) ((value)<<5) /* Bits 5:6 */ +#define XPBBLK(value) ((value)<<7) /* Bits 7:8 */ + +/*********************** McBSP XCCR bit definitions *************************/ +#define EXTCLKGATE 0x8000 +#define PPCONNECT 0x4000 +#define DXENDLY(value) ((value)<<12) /* Bits 12:13 */ +#define XFULL_CYCLE 0x0800 +#define DILB 0x0020 +#define XDMAEN 0x0008 +#define XDISABLE 0x0001 + +/********************** McBSP RCCR bit definitions *************************/ +#define RFULL_CYCLE 0x0800 +#define RDMAEN 0x0008 +#define RDISABLE 0x0001 + +/********************** McBSP SYSCONFIG bit definitions ********************/ +#define CLOCKACTIVITY(value) ((value)<<8) +#define SIDLEMODE(value) ((value)<<3) +#define ENAWAKEUP 0x0004 +#define SOFTRST 0x0002 + +/********************** McBSP SSELCR bit definitions ***********************/ +#define SIDETONEEN 0x0400 + +/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ +#define ST_AUTOIDLE 0x0001 + +/********************** McBSP Sidetone SGAINCR bit definitions *************/ +#define ST_CH1GAIN(value) ((value<<16)) /* Bits 16:31 */ +#define ST_CH0GAIN(value) (value) /* Bits 0:15 */ + +/********************** McBSP Sidetone SFIRCR bit definitions **************/ +#define ST_FIRCOEFF(value) (value) /* Bits 0:15 */ + +/********************** McBSP Sidetone SSELCR bit definitions **************/ +#define ST_COEFFWRDONE 0x0004 +#define ST_COEFFWREN 0x0002 +#define ST_SIDETONEEN 0x0001 + +/********************** McBSP DMA operating modes **************************/ +#define MCBSP_DMA_MODE_ELEMENT 0 +#define MCBSP_DMA_MODE_THRESHOLD 1 +#define MCBSP_DMA_MODE_FRAME 2 + +/********************** McBSP WAKEUPEN bit definitions *********************/ +#define XEMPTYEOFEN 0x4000 +#define XRDYEN 0x0400 +#define XEOFEN 0x0200 +#define XFSXEN 0x0100 +#define XSYNCERREN 0x0080 +#define RRDYEN 0x0008 +#define REOFEN 0x0004 +#define RFSREN 0x0002 +#define RSYNCERREN 0x0001 + +/* we don't do multichannel for now */ +struct omap_mcbsp_reg_cfg { + u16 spcr2; + u16 spcr1; + u16 rcr2; + u16 rcr1; + u16 xcr2; + u16 xcr1; + u16 srgr2; + u16 srgr1; + u16 mcr2; + u16 mcr1; + u16 pcr0; + u16 rcerc; + u16 rcerd; + u16 xcerc; + u16 xcerd; + u16 rcere; + u16 rcerf; + u16 xcere; + u16 xcerf; + u16 rcerg; + u16 rcerh; + u16 xcerg; + u16 xcerh; + u16 xccr; + u16 rccr; +}; + +void omap_mcbsp_config(unsigned int id, + const struct omap_mcbsp_reg_cfg *config); +void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold); +void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold); +u16 omap_mcbsp_get_max_tx_threshold(unsigned int id); +u16 omap_mcbsp_get_max_rx_threshold(unsigned int id); +u16 omap_mcbsp_get_fifo_size(unsigned int id); +u16 omap_mcbsp_get_tx_delay(unsigned int id); +u16 omap_mcbsp_get_rx_delay(unsigned int id); +int omap_mcbsp_get_dma_op_mode(unsigned int id); +int omap_mcbsp_request(unsigned int id); +void omap_mcbsp_free(unsigned int id); +void omap_mcbsp_start(unsigned int id, int tx, int rx); +void omap_mcbsp_stop(unsigned int id, int tx, int rx); + +/* McBSP functional clock source changing function */ +int omap2_mcbsp_set_clks_src(u8 id, u8 fck_src_id); + +/* McBSP signal muxing API */ +void omap2_mcbsp1_mux_clkr_src(u8 mux); +void omap2_mcbsp1_mux_fsr_src(u8 mux); + +int omap_mcbsp_dma_ch_params(unsigned int id, unsigned int stream); +int omap_mcbsp_dma_reg_params(unsigned int id, unsigned int stream); + +/* Sidetone specific API */ +int omap_st_set_chgain(unsigned int id, int channel, s16 chgain); +int omap_st_get_chgain(unsigned int id, int channel, s16 *chgain); +int omap_st_enable(unsigned int id); +int omap_st_disable(unsigned int id); +int omap_st_is_enabled(unsigned int id); + +#endif /* __ASOC_MCBSP_H */ diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 1287b870f22..f1318c1d4e1 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -33,6 +33,7 @@ #include #include +#include "mcbsp.h" #include "omap-mcbsp.h" #include "omap-pcm.h" @@ -46,6 +47,15 @@ .private_value = (unsigned long) &(struct soc_mixer_control) \ {.min = xmin, .max = xmax} } +enum { + OMAP_MCBSP_WORD_8 = 0, + OMAP_MCBSP_WORD_12, + OMAP_MCBSP_WORD_16, + OMAP_MCBSP_WORD_20, + OMAP_MCBSP_WORD_24, + OMAP_MCBSP_WORD_32, +}; + struct omap_mcbsp_data { unsigned int bus_id; struct omap_mcbsp_reg_cfg regs; -- cgit v1.2.3-18-g5258 From 45656b44f6d1968d838f3abcf3a264ee9fa2fc62 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 14 Feb 2012 18:20:58 +0200 Subject: ASoC: omap-mcbsp: Create a single driver for McBSP The OMAP McBSP driver stack used to contain two different drivers. One of them was used as kind low-level access to the IP, while the other driver was the ASoC DAI driver. There were global, shared structures, in different places, the McBSP instances are reffered with id numbers (sometimes 0 based, in other cases 1 based id numbers). Create one single driver for OMAP McBSP with name: omap-mcbsp. Convert the old omap-mcbsp driver initially to be a library for the omap-mcbsp DAI driver. With this change we can get rid of all global variables, structures. Further cleanup is coming... Signed-off-by: Peter Ujfalusi Tested-by: Grazvydas Ignotas Tested-by: Janusz Krzysztofik Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/igep0020.c | 2 +- sound/soc/omap/mcbsp.c | 284 ++++++------------------------------------ sound/soc/omap/mcbsp.h | 104 ++++++++++++---- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-mcbsp.c | 201 ++++++++++++++---------------- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.h | 2 + sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3evm.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/rx51.c | 4 +- sound/soc/omap/sdp3430.c | 4 +- sound/soc/omap/zoom2.c | 4 +- 17 files changed, 234 insertions(+), 391 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index add4866d7e6..009533ab8d1 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -95,7 +95,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_dai_link am3517evm_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", - .cpu_dai_name ="omap-mcbsp-dai.0", + .cpu_dai_name = "omap-mcbsp.1", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic23-codec.2-001a", diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 78563bbbbf0..49fe63ce51f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -584,7 +584,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link ams_delta_dai_link = { .name = "CX20442", .stream_name = "CX20442", - .cpu_dai_name ="omap-mcbsp-dai.0", + .cpu_dai_name = "omap-mcbsp.1", .codec_dai_name = "cx20442-voice", .init = ams_delta_cx20442_init, .platform_name = "omap-pcm-audio", diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index ccae58a1339..e8357819175 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -60,7 +60,7 @@ static struct snd_soc_ops igep2_ops = { static struct snd_soc_dai_link igep2_dai = { .name = "TWL4030", .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp-dai.1", + .cpu_dai_name = "omap-mcbsp.2", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 20d46bf3626..be92a28e19e 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -30,12 +30,6 @@ #include "mcbsp.h" -struct omap_mcbsp **mcbsp_ptr; -int omap_mcbsp_count; - -#define omap_mcbsp_check_valid_id(id) (id < omap_mcbsp_count) -#define id_to_mcbsp_ptr(id) mcbsp_ptr[id]; - static void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val) { void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step; @@ -84,10 +78,8 @@ static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg) #define MCBSP_ST_WRITE(mcbsp, reg, val) \ omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val) -static void omap_mcbsp_dump_reg(u8 id) +static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp = id_to_mcbsp_ptr(id); - dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id); dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", MCBSP_READ(mcbsp, DRR2)); @@ -160,16 +152,9 @@ static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *dev_id) * You either call this function or set the McBSP registers * by yourself before calling omap_mcbsp_start(). */ -void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg *config) +void omap_mcbsp_config(struct omap_mcbsp *mcbsp, + const struct omap_mcbsp_reg_cfg *config) { - struct omap_mcbsp *mcbsp; - - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return; - } - mcbsp = id_to_mcbsp_ptr(id); - dev_dbg(mcbsp->dev, "Configuring McBSP%d phys_base: 0x%08lx\n", mcbsp->id, mcbsp->phys_base); @@ -190,7 +175,6 @@ void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg *config) MCBSP_WRITE(mcbsp, RCCR, config->rccr); } } -EXPORT_SYMBOL(omap_mcbsp_config); /** * omap_mcbsp_dma_params - returns the dma channel number @@ -200,22 +184,13 @@ EXPORT_SYMBOL(omap_mcbsp_config); * Returns the dma channel number for the rx channel or tx channel * based on the value of @stream for the requested mcbsp given by @id */ -int omap_mcbsp_dma_ch_params(unsigned int id, unsigned int stream) +int omap_mcbsp_dma_ch_params(struct omap_mcbsp *mcbsp, unsigned int stream) { - struct omap_mcbsp *mcbsp; - - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - mcbsp = id_to_mcbsp_ptr(id); - if (stream) return mcbsp->dma_rx_sync; else return mcbsp->dma_tx_sync; } -EXPORT_SYMBOL(omap_mcbsp_dma_ch_params); /** * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register @@ -226,17 +201,10 @@ EXPORT_SYMBOL(omap_mcbsp_dma_ch_params); * to be used by DMA for transferring/receiving data based on the value of * @stream for the requested mcbsp given by @id */ -int omap_mcbsp_dma_reg_params(unsigned int id, unsigned int stream) +int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, unsigned int stream) { - struct omap_mcbsp *mcbsp; int data_reg; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - mcbsp = id_to_mcbsp_ptr(id); - if (mcbsp->pdata->reg_size == 2) { if (stream) data_reg = OMAP_MCBSP_REG_DRR1; @@ -251,7 +219,6 @@ int omap_mcbsp_dma_reg_params(unsigned int id, unsigned int stream) return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step; } -EXPORT_SYMBOL(omap_mcbsp_dma_reg_params); static void omap_st_on(struct omap_mcbsp *mcbsp) { @@ -320,18 +287,11 @@ static void omap_st_chgain(struct omap_mcbsp *mcbsp) ST_CH1GAIN(st_data->ch1gain)); } -int omap_st_set_chgain(unsigned int id, int channel, s16 chgain) +int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain) { - struct omap_mcbsp *mcbsp; struct omap_mcbsp_st_data *st_data; int ret = 0; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - - mcbsp = id_to_mcbsp_ptr(id); st_data = mcbsp->st_data; if (!st_data) @@ -351,20 +311,12 @@ int omap_st_set_chgain(unsigned int id, int channel, s16 chgain) return ret; } -EXPORT_SYMBOL(omap_st_set_chgain); -int omap_st_get_chgain(unsigned int id, int channel, s16 *chgain) +int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain) { - struct omap_mcbsp *mcbsp; struct omap_mcbsp_st_data *st_data; int ret = 0; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - - mcbsp = id_to_mcbsp_ptr(id); st_data = mcbsp->st_data; if (!st_data) @@ -381,7 +333,6 @@ int omap_st_get_chgain(unsigned int id, int channel, s16 *chgain) return ret; } -EXPORT_SYMBOL(omap_st_get_chgain); static int omap_st_start(struct omap_mcbsp *mcbsp) { @@ -400,17 +351,10 @@ static int omap_st_start(struct omap_mcbsp *mcbsp) return 0; } -int omap_st_enable(unsigned int id) +int omap_st_enable(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; struct omap_mcbsp_st_data *st_data; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - - mcbsp = id_to_mcbsp_ptr(id); st_data = mcbsp->st_data; if (!st_data) @@ -423,7 +367,6 @@ int omap_st_enable(unsigned int id) return 0; } -EXPORT_SYMBOL(omap_st_enable); static int omap_st_stop(struct omap_mcbsp *mcbsp) { @@ -439,18 +382,11 @@ static int omap_st_stop(struct omap_mcbsp *mcbsp) return 0; } -int omap_st_disable(unsigned int id) +int omap_st_disable(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; struct omap_mcbsp_st_data *st_data; int ret = 0; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - - mcbsp = id_to_mcbsp_ptr(id); st_data = mcbsp->st_data; if (!st_data) @@ -463,19 +399,11 @@ int omap_st_disable(unsigned int id) return ret; } -EXPORT_SYMBOL(omap_st_disable); -int omap_st_is_enabled(unsigned int id) +int omap_st_is_enabled(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; struct omap_mcbsp_st_data *st_data; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - - mcbsp = id_to_mcbsp_ptr(id); st_data = mcbsp->st_data; if (!st_data) @@ -484,115 +412,65 @@ int omap_st_is_enabled(unsigned int id) return st_data->enabled; } -EXPORT_SYMBOL(omap_st_is_enabled); /* * omap_mcbsp_set_rx_threshold configures the transmit threshold in words. * The threshold parameter is 1 based, and it is converted (threshold - 1) * for the THRSH2 register. */ -void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold) +void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) { - struct omap_mcbsp *mcbsp; - - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return; - } - mcbsp = id_to_mcbsp_ptr(id); if (mcbsp->pdata->buffer_size == 0) return; if (threshold && threshold <= mcbsp->max_tx_thres) MCBSP_WRITE(mcbsp, THRSH2, threshold - 1); } -EXPORT_SYMBOL(omap_mcbsp_set_tx_threshold); /* * omap_mcbsp_set_rx_threshold configures the receive threshold in words. * The threshold parameter is 1 based, and it is converted (threshold - 1) * for the THRSH1 register. */ -void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold) +void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) { - struct omap_mcbsp *mcbsp; - - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return; - } - mcbsp = id_to_mcbsp_ptr(id); if (mcbsp->pdata->buffer_size == 0) return; if (threshold && threshold <= mcbsp->max_rx_thres) MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); } -EXPORT_SYMBOL(omap_mcbsp_set_rx_threshold); /* * omap_mcbsp_get_max_tx_thres just return the current configured * maximum threshold for transmission */ -u16 omap_mcbsp_get_max_tx_threshold(unsigned int id) +u16 omap_mcbsp_get_max_tx_threshold(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; - - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - mcbsp = id_to_mcbsp_ptr(id); - return mcbsp->max_tx_thres; } -EXPORT_SYMBOL(omap_mcbsp_get_max_tx_threshold); /* * omap_mcbsp_get_max_rx_thres just return the current configured * maximum threshold for reception */ -u16 omap_mcbsp_get_max_rx_threshold(unsigned int id) +u16 omap_mcbsp_get_max_rx_threshold(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; - - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - mcbsp = id_to_mcbsp_ptr(id); - return mcbsp->max_rx_thres; } -EXPORT_SYMBOL(omap_mcbsp_get_max_rx_threshold); -u16 omap_mcbsp_get_fifo_size(unsigned int id) +u16 omap_mcbsp_get_fifo_size(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; - - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - mcbsp = id_to_mcbsp_ptr(id); - return mcbsp->pdata->buffer_size; } -EXPORT_SYMBOL(omap_mcbsp_get_fifo_size); /* * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO */ -u16 omap_mcbsp_get_tx_delay(unsigned int id) +u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; u16 buffstat; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - mcbsp = id_to_mcbsp_ptr(id); if (mcbsp->pdata->buffer_size == 0) return 0; @@ -602,22 +480,15 @@ u16 omap_mcbsp_get_tx_delay(unsigned int id) /* Number of slots are different in McBSP ports */ return mcbsp->pdata->buffer_size - buffstat; } -EXPORT_SYMBOL(omap_mcbsp_get_tx_delay); /* * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO * to reach the threshold value (when the DMA will be triggered to read it) */ -u16 omap_mcbsp_get_rx_delay(unsigned int id) +u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; u16 buffstat, threshold; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - mcbsp = id_to_mcbsp_ptr(id); if (mcbsp->pdata->buffer_size == 0) return 0; @@ -632,41 +503,25 @@ u16 omap_mcbsp_get_rx_delay(unsigned int id) else return threshold - buffstat; } -EXPORT_SYMBOL(omap_mcbsp_get_rx_delay); /* * omap_mcbsp_get_dma_op_mode just return the current configured * operating mode for the mcbsp channel */ -int omap_mcbsp_get_dma_op_mode(unsigned int id) +int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; int dma_op_mode; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%u)\n", __func__, id + 1); - return -ENODEV; - } - mcbsp = id_to_mcbsp_ptr(id); - dma_op_mode = mcbsp->dma_op_mode; return dma_op_mode; } -EXPORT_SYMBOL(omap_mcbsp_get_dma_op_mode); -int omap_mcbsp_request(unsigned int id) +int omap_mcbsp_request(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; void *reg_cache; int err; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return -ENODEV; - } - mcbsp = id_to_mcbsp_ptr(id); - reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL); if (!reg_cache) { return -ENOMEM; @@ -685,9 +540,7 @@ int omap_mcbsp_request(unsigned int id) spin_unlock(&mcbsp->lock); if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) - mcbsp->pdata->ops->request(id); - - pm_runtime_get_sync(mcbsp->dev); + mcbsp->pdata->ops->request(mcbsp->id - 1); /* Enable wakeup behavior */ if (mcbsp->pdata->has_wakeup) @@ -726,14 +579,12 @@ err_free_irq: free_irq(mcbsp->tx_irq, (void *)mcbsp); err_clk_disable: if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(id); + mcbsp->pdata->ops->free(mcbsp->id - 1); /* Disable wakeup behavior */ if (mcbsp->pdata->has_wakeup) MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - pm_runtime_put_sync(mcbsp->dev); - spin_lock(&mcbsp->lock); mcbsp->free = true; mcbsp->reg_cache = NULL; @@ -743,28 +594,18 @@ err_kfree: return err; } -EXPORT_SYMBOL(omap_mcbsp_request); -void omap_mcbsp_free(unsigned int id) +void omap_mcbsp_free(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp; void *reg_cache; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return; - } - mcbsp = id_to_mcbsp_ptr(id); - if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(id); + mcbsp->pdata->ops->free(mcbsp->id - 1); /* Disable wakeup behavior */ if (mcbsp->pdata->has_wakeup) MCBSP_WRITE(mcbsp, WAKEUPEN, 0); - pm_runtime_put_sync(mcbsp->dev); - if (mcbsp->rx_irq) free_irq(mcbsp->rx_irq, (void *)mcbsp); free_irq(mcbsp->tx_irq, (void *)mcbsp); @@ -782,25 +623,17 @@ void omap_mcbsp_free(unsigned int id) if (reg_cache) kfree(reg_cache); } -EXPORT_SYMBOL(omap_mcbsp_free); /* * Here we start the McBSP, by enabling transmitter, receiver or both. * If no transmitter or receiver is active prior calling, then sample-rate * generator and frame sync are started. */ -void omap_mcbsp_start(unsigned int id, int tx, int rx) +void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx) { - struct omap_mcbsp *mcbsp; int enable_srg = 0; u16 w; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return; - } - mcbsp = id_to_mcbsp_ptr(id); - if (mcbsp->st_data) omap_st_start(mcbsp); @@ -850,23 +683,14 @@ void omap_mcbsp_start(unsigned int id, int tx, int rx) } /* Dump McBSP Regs */ - omap_mcbsp_dump_reg(id); + omap_mcbsp_dump_reg(mcbsp); } -EXPORT_SYMBOL(omap_mcbsp_start); -void omap_mcbsp_stop(unsigned int id, int tx, int rx) +void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx) { - struct omap_mcbsp *mcbsp; int idle; u16 w; - if (!omap_mcbsp_check_valid_id(id)) { - printk(KERN_ERR "%s: Invalid id (%d)\n", __func__, id + 1); - return; - } - - mcbsp = id_to_mcbsp_ptr(id); - /* Reset transmitter */ tx &= 1; if (mcbsp->pdata->has_ccr) { @@ -899,19 +723,11 @@ void omap_mcbsp_stop(unsigned int id, int tx, int rx) if (mcbsp->st_data) omap_st_stop(mcbsp); } -EXPORT_SYMBOL(omap_mcbsp_stop); -int omap2_mcbsp_set_clks_src(u8 id, u8 fck_src_id) +int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) { - struct omap_mcbsp *mcbsp; const char *src; - if (!omap_mcbsp_check_valid_id(id)) { - pr_err("%s: Invalid id (%d)\n", __func__, id + 1); - return -EINVAL; - } - mcbsp = id_to_mcbsp_ptr(id); - if (fck_src_id == MCBSP_CLKS_PAD_SRC) src = "clks_ext"; else if (fck_src_id == MCBSP_CLKS_PRCM_SRC) @@ -924,13 +740,14 @@ int omap2_mcbsp_set_clks_src(u8 id, u8 fck_src_id) else return -EINVAL; } -EXPORT_SYMBOL(omap2_mcbsp_set_clks_src); -void omap2_mcbsp1_mux_clkr_src(u8 mux) +void omap2_mcbsp1_mux_clkr_src(struct omap_mcbsp *mcbsp, u8 mux) { - struct omap_mcbsp *mcbsp; const char *src; + if (mcbsp->id != 1) + return; + if (mux == CLKR_SRC_CLKR) src = "clkr"; else if (mux == CLKR_SRC_CLKX) @@ -938,17 +755,17 @@ void omap2_mcbsp1_mux_clkr_src(u8 mux) else return; - mcbsp = id_to_mcbsp_ptr(0); if (mcbsp->pdata->mux_signal) mcbsp->pdata->mux_signal(mcbsp->dev, "clkr", src); } -EXPORT_SYMBOL(omap2_mcbsp1_mux_clkr_src); -void omap2_mcbsp1_mux_fsr_src(u8 mux) +void omap2_mcbsp1_mux_fsr_src(struct omap_mcbsp *mcbsp, u8 mux) { - struct omap_mcbsp *mcbsp; const char *src; + if (mcbsp->id != 1) + return; + if (mux == FSR_SRC_FSR) src = "fsr"; else if (mux == FSR_SRC_FSX) @@ -956,11 +773,9 @@ void omap2_mcbsp1_mux_fsr_src(u8 mux) else return; - mcbsp = id_to_mcbsp_ptr(0); if (mcbsp->pdata->mux_signal) mcbsp->pdata->mux_signal(mcbsp->dev, "fsr", src); } -EXPORT_SYMBOL(omap2_mcbsp1_mux_fsr_src); #define max_thres(m) (mcbsp->pdata->buffer_size) #define valid_threshold(m, val) ((val) <= max_thres(m)) @@ -1177,11 +992,10 @@ static void __devexit omap_st_remove(struct omap_mcbsp *mcbsp) * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. * 730 has only 2 McBSP, and both of them are MPU peripherals. */ -static int __devinit omap_mcbsp_probe(struct platform_device *pdev) +int __devinit omap_mcbsp_probe(struct platform_device *pdev) { struct omap_mcbsp_platform_data *pdata = pdev->dev.platform_data; struct omap_mcbsp *mcbsp; - int id = pdev->id - 1; struct resource *res; int ret = 0; @@ -1194,12 +1008,6 @@ static int __devinit omap_mcbsp_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "Initializing OMAP McBSP (%d).\n", pdev->id); - if (id >= omap_mcbsp_count) { - dev_err(&pdev->dev, "Invalid McBSP device id (%d)\n", id); - ret = -EINVAL; - goto exit; - } - mcbsp = kzalloc(sizeof(struct omap_mcbsp), GFP_KERNEL); if (!mcbsp) { ret = -ENOMEM; @@ -1207,7 +1015,7 @@ static int __devinit omap_mcbsp_probe(struct platform_device *pdev) } spin_lock_init(&mcbsp->lock); - mcbsp->id = id + 1; + mcbsp->id = pdev->id; mcbsp->free = true; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); @@ -1268,7 +1076,6 @@ static int __devinit omap_mcbsp_probe(struct platform_device *pdev) mcbsp->pdata = pdata; mcbsp->dev = &pdev->dev; - mcbsp_ptr[id] = mcbsp; platform_set_drvdata(pdev, mcbsp); pm_runtime_enable(mcbsp->dev); @@ -1323,7 +1130,7 @@ exit: return ret; } -static int __devexit omap_mcbsp_remove(struct platform_device *pdev) +int __devexit omap_mcbsp_remove(struct platform_device *pdev) { struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); @@ -1349,18 +1156,3 @@ static int __devexit omap_mcbsp_remove(struct platform_device *pdev) return 0; } - -static struct platform_driver omap_mcbsp_driver = { - .probe = omap_mcbsp_probe, - .remove = __devexit_p(omap_mcbsp_remove), - .driver = { - .name = "omap-mcbsp", - }, -}; - -module_platform_driver(omap_mcbsp_driver); - -MODULE_AUTHOR("Samuel Ortiz "); -MODULE_DESCRIPTION("OMAP McBSP core driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:omap-mcbsp"); diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 5590ab271ee..6d579938a15 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -24,6 +24,8 @@ #ifndef __ASOC_MCBSP_H #define __ASOC_MCBSP_H +#include "omap-pcm.h" + /* McBSP register numbers. Register address offset = num * reg_step */ enum { /* Common registers */ @@ -257,36 +259,92 @@ struct omap_mcbsp_reg_cfg { u16 rccr; }; -void omap_mcbsp_config(unsigned int id, +struct omap_mcbsp_st_data { + void __iomem *io_base_st; + bool running; + bool enabled; + s16 taps[128]; /* Sidetone filter coefficients */ + int nr_taps; /* Number of filter coefficients in use */ + s16 ch0gain; + s16 ch1gain; +}; + +struct omap_mcbsp_data { + struct omap_mcbsp_reg_cfg regs; + struct omap_pcm_dma_data dma_data[2]; + unsigned int fmt; + /* + * Flags indicating is the bus already activated and configured by + * another substream + */ + int active; + int configured; + unsigned int in_freq; + int clk_div; + int wlen; +}; + +struct omap_mcbsp { + struct device *dev; + unsigned long phys_base; + unsigned long phys_dma_base; + void __iomem *io_base; + u8 id; + u8 free; + + int rx_irq; + int tx_irq; + + /* DMA stuff */ + u8 dma_rx_sync; + u8 dma_tx_sync; + + /* Protect the field .free, while checking if the mcbsp is in use */ + spinlock_t lock; + struct omap_mcbsp_platform_data *pdata; + struct clk *fclk; + struct omap_mcbsp_st_data *st_data; + struct omap_mcbsp_data mcbsp_data; + int dma_op_mode; + u16 max_tx_thres; + u16 max_rx_thres; + void *reg_cache; + int reg_cache_size; +}; + +void omap_mcbsp_config(struct omap_mcbsp *mcbsp, const struct omap_mcbsp_reg_cfg *config); -void omap_mcbsp_set_tx_threshold(unsigned int id, u16 threshold); -void omap_mcbsp_set_rx_threshold(unsigned int id, u16 threshold); -u16 omap_mcbsp_get_max_tx_threshold(unsigned int id); -u16 omap_mcbsp_get_max_rx_threshold(unsigned int id); -u16 omap_mcbsp_get_fifo_size(unsigned int id); -u16 omap_mcbsp_get_tx_delay(unsigned int id); -u16 omap_mcbsp_get_rx_delay(unsigned int id); -int omap_mcbsp_get_dma_op_mode(unsigned int id); -int omap_mcbsp_request(unsigned int id); -void omap_mcbsp_free(unsigned int id); -void omap_mcbsp_start(unsigned int id, int tx, int rx); -void omap_mcbsp_stop(unsigned int id, int tx, int rx); +void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); +void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); +u16 omap_mcbsp_get_max_tx_threshold(struct omap_mcbsp *mcbsp); +u16 omap_mcbsp_get_max_rx_threshold(struct omap_mcbsp *mcbsp); +u16 omap_mcbsp_get_fifo_size(struct omap_mcbsp *mcbsp); +u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp); +u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp); +int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp); +int omap_mcbsp_request(struct omap_mcbsp *mcbsp); +void omap_mcbsp_free(struct omap_mcbsp *mcbsp); +void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx); +void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx); /* McBSP functional clock source changing function */ -int omap2_mcbsp_set_clks_src(u8 id, u8 fck_src_id); +int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); /* McBSP signal muxing API */ -void omap2_mcbsp1_mux_clkr_src(u8 mux); -void omap2_mcbsp1_mux_fsr_src(u8 mux); +void omap2_mcbsp1_mux_clkr_src(struct omap_mcbsp *mcbsp, u8 mux); +void omap2_mcbsp1_mux_fsr_src(struct omap_mcbsp *mcbsp, u8 mux); -int omap_mcbsp_dma_ch_params(unsigned int id, unsigned int stream); -int omap_mcbsp_dma_reg_params(unsigned int id, unsigned int stream); +int omap_mcbsp_dma_ch_params(struct omap_mcbsp *mcbsp, unsigned int stream); +int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, unsigned int stream); /* Sidetone specific API */ -int omap_st_set_chgain(unsigned int id, int channel, s16 chgain); -int omap_st_get_chgain(unsigned int id, int channel, s16 *chgain); -int omap_st_enable(unsigned int id); -int omap_st_disable(unsigned int id); -int omap_st_is_enabled(unsigned int id); +int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain); +int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain); +int omap_st_enable(struct omap_mcbsp *mcbsp); +int omap_st_disable(struct omap_mcbsp *mcbsp); +int omap_st_is_enabled(struct omap_mcbsp *mcbsp); + +int __devinit omap_mcbsp_probe(struct platform_device *pdev); +int __devexit omap_mcbsp_remove(struct platform_device *pdev); #endif /* __ASOC_MCBSP_H */ diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index c292bf0fd19..abac4b69075 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -275,7 +275,7 @@ static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link n810_dai = { .name = "TLV320AIC33", .stream_name = "AIC33", - .cpu_dai_name = "omap-mcbsp-dai.1", + .cpu_dai_name = "omap-mcbsp.2", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", .codec_dai_name = "tlv320aic3x-hifi", diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index f1318c1d4e1..89240025949 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -56,36 +56,18 @@ enum { OMAP_MCBSP_WORD_32, }; -struct omap_mcbsp_data { - unsigned int bus_id; - struct omap_mcbsp_reg_cfg regs; - unsigned int fmt; - /* - * Flags indicating is the bus already activated and configured by - * another substream - */ - int active; - int configured; - unsigned int in_freq; - int clk_div; - int wlen; -}; - -static struct omap_mcbsp_data mcbsp_data[NUM_LINKS]; - /* * Stream DMA parameters. DMA request line and port address are set runtime * since they are different between OMAP1 and later OMAPs */ -static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2]; - static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; struct omap_pcm_dma_data *dma_data; - int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id); + int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp); int words; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); @@ -107,9 +89,9 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) /* Configure McBSP internal buffer usage */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, words); + omap_mcbsp_set_tx_threshold(mcbsp, words); else - omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, words); + omap_mcbsp_set_rx_threshold(mcbsp, words); } static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params, @@ -119,12 +101,12 @@ static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - struct omap_mcbsp_data *mcbsp_data = rule->private; + struct omap_mcbsp *mcbsp = rule->private; struct snd_interval frames; int size; snd_interval_any(&frames); - size = omap_mcbsp_get_fifo_size(mcbsp_data->bus_id); + size = omap_mcbsp_get_fifo_size(mcbsp); frames.min = size / channels->min; frames.integer = 1; @@ -134,12 +116,11 @@ static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params, static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); - int bus_id = mcbsp_data->bus_id; + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); int err = 0; if (!cpu_dai->active) - err = omap_mcbsp_request(bus_id); + err = omap_mcbsp_request(mcbsp); /* * OMAP3 McBSP FIFO is word structured. @@ -156,7 +137,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap34xx() || cpu_is_omap44xx()) { + if (mcbsp->pdata->buffer_size) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns @@ -164,7 +145,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, omap_mcbsp_hwrule_min_buffersize, - mcbsp_data, + mcbsp, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); /* Make sure, that the period size is always even */ @@ -178,10 +159,11 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { - struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; if (!cpu_dai->active) { - omap_mcbsp_free(mcbsp_data->bus_id); + omap_mcbsp_free(mcbsp); mcbsp_data->configured = 0; } } @@ -189,7 +171,8 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { - struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { @@ -197,13 +180,13 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: mcbsp_data->active++; - omap_mcbsp_start(mcbsp_data->bus_id, play, !play); + omap_mcbsp_start(mcbsp, play, !play); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - omap_mcbsp_stop(mcbsp_data->bus_id, play, !play); + omap_mcbsp_stop(mcbsp, play, !play); mcbsp_data->active--; break; default: @@ -219,14 +202,14 @@ static snd_pcm_sframes_t omap_mcbsp_dai_delay( { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); u16 fifo_use; snd_pcm_sframes_t delay; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_use = omap_mcbsp_get_tx_delay(mcbsp_data->bus_id); + fifo_use = omap_mcbsp_get_tx_delay(mcbsp); else - fifo_use = omap_mcbsp_get_rx_delay(mcbsp_data->bus_id); + fifo_use = omap_mcbsp_get_rx_delay(mcbsp); /* * Divide the used locations with the channel count to get the @@ -242,19 +225,20 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) { - struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; struct omap_pcm_dma_data *dma_data; - int dma, bus_id = mcbsp_data->bus_id; + int dma; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; int pkt_size = 0; unsigned long port; unsigned int format, div, framesize, master; - dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream]; + dma_data = &mcbsp_data->dma_data[substream->stream]; - dma = omap_mcbsp_dma_ch_params(bus_id, substream->stream); - port = omap_mcbsp_dma_reg_params(bus_id, substream->stream); + dma = omap_mcbsp_dma_ch_params(mcbsp, substream->stream); + port = omap_mcbsp_dma_reg_params(mcbsp, substream->stream); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -268,20 +252,20 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (cpu_is_omap34xx() || cpu_is_omap44xx()) { + if (mcbsp->pdata->buffer_size) { dma_data->set_threshold = omap_mcbsp_set_threshold; /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ - if (omap_mcbsp_get_dma_op_mode(bus_id) == + if (omap_mcbsp_get_dma_op_mode(mcbsp) == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; period_words = params_period_bytes(params) / (wlen / 8); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) max_thrsh = omap_mcbsp_get_max_tx_threshold( - mcbsp_data->bus_id); + mcbsp); else max_thrsh = omap_mcbsp_get_max_rx_threshold( - mcbsp_data->bus_id); + mcbsp); /* * If the period contains less or equal number of words, * we are using the original threshold mode setup: @@ -398,7 +382,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; } - omap_mcbsp_config(bus_id, &mcbsp_data->regs); + omap_mcbsp_config(mcbsp, &mcbsp_data->regs); mcbsp_data->wlen = wlen; mcbsp_data->configured = 1; @@ -412,7 +396,8 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; bool inv_fs = false; @@ -514,7 +499,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { - struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; if (div_id != OMAP_MCBSP_CLKGDV) @@ -531,7 +517,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct omap_mcbsp_data *mcbsp_data = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; @@ -547,7 +534,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || clk_id == OMAP_MCBSP_FSR_SRC_FSR || clk_id == OMAP_MCBSP_FSR_SRC_FSX) - if (cpu_class_is_omap1() || mcbsp_data->bus_id != 0) + if (cpu_class_is_omap1() || cpu_dai->id != 1) return -EINVAL; mcbsp_data->in_freq = freq; @@ -563,7 +550,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, err = -EINVAL; break; } - err = omap2_mcbsp_set_clks_src(mcbsp_data->bus_id, + err = omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); break; case OMAP_MCBSP_SYSCLK_CLKS_EXT: @@ -571,7 +558,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, err = 0; break; } - err = omap2_mcbsp_set_clks_src(mcbsp_data->bus_id, + err = omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PAD_SRC); break; @@ -585,22 +572,22 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_CLKR_SRC_CLKR: if (cpu_class_is_omap1()) break; - omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKR); + omap2_mcbsp1_mux_clkr_src(mcbsp, CLKR_SRC_CLKR); break; case OMAP_MCBSP_CLKR_SRC_CLKX: if (cpu_class_is_omap1()) break; - omap2_mcbsp1_mux_clkr_src(CLKR_SRC_CLKX); + omap2_mcbsp1_mux_clkr_src(mcbsp, CLKR_SRC_CLKX); break; case OMAP_MCBSP_FSR_SRC_FSR: if (cpu_class_is_omap1()) break; - omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSR); + omap2_mcbsp1_mux_fsr_src(mcbsp, FSR_SRC_FSR); break; case OMAP_MCBSP_FSR_SRC_FSX: if (cpu_class_is_omap1()) break; - omap2_mcbsp1_mux_fsr_src(FSR_SRC_FSX); + omap2_mcbsp1_mux_fsr_src(mcbsp, FSR_SRC_FSX); break; default: err = -ENODEV; @@ -620,15 +607,7 @@ static const struct snd_soc_dai_ops mcbsp_dai_ops = { .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, }; -static int mcbsp_dai_probe(struct snd_soc_dai *dai) -{ - mcbsp_data[dai->id].bus_id = dai->id; - snd_soc_dai_set_drvdata(dai, &mcbsp_data[dai->id].bus_id); - return 0; -} - static struct snd_soc_dai_driver omap_mcbsp_dai = { - .probe = mcbsp_dai_probe, .playback = { .channels_min = 1, .channels_max = 16, @@ -659,11 +638,13 @@ static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, return 0; } -#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(id, channel) \ +#define OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(channel) \ static int \ -omap_mcbsp##id##_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ +omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ struct snd_ctl_elem_value *uc) \ { \ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ struct soc_mixer_control *mc = \ (struct soc_mixer_control *)kc->private_value; \ int max = mc->max; \ @@ -674,46 +655,44 @@ omap_mcbsp##id##_set_st_ch##channel##_volume(struct snd_kcontrol *kc, \ return -EINVAL; \ \ /* OMAP McBSP implementation uses index values 0..4 */ \ - return omap_st_set_chgain((id)-1, channel, val); \ + return omap_st_set_chgain(mcbsp, channel, val); \ } -#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(id, channel) \ +#define OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(channel) \ static int \ -omap_mcbsp##id##_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ +omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc, \ struct snd_ctl_elem_value *uc) \ { \ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc); \ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); \ s16 chgain; \ \ - if (omap_st_get_chgain((id)-1, channel, &chgain)) \ + if (omap_st_get_chgain(mcbsp, channel, &chgain)) \ return -EAGAIN; \ \ uc->value.integer.value[0] = chgain; \ return 0; \ } -OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 0) -OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(2, 1) -OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 0) -OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(3, 1) -OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 0) -OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(2, 1) -OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 0) -OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(3, 1) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(0) +OMAP_MCBSP_ST_SET_CHANNEL_VOLUME(1) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(0) +OMAP_MCBSP_ST_GET_CHANNEL_VOLUME(1) static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); u8 value = ucontrol->value.integer.value[0]; - if (value == omap_st_is_enabled(mc->reg)) + if (value == omap_st_is_enabled(mcbsp)) return 0; if (value) - omap_st_enable(mc->reg); + omap_st_enable(mcbsp); else - omap_st_disable(mc->reg); + omap_st_disable(mcbsp); return 1; } @@ -721,10 +700,10 @@ static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol, static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - ucontrol->value.integer.value[0] = omap_st_is_enabled(mc->reg); + ucontrol->value.integer.value[0] = omap_st_is_enabled(mcbsp); return 0; } @@ -733,12 +712,12 @@ static const struct snd_kcontrol_new omap_mcbsp2_st_controls[] = { omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 0 Volume", -32768, 32767, - omap_mcbsp2_get_st_ch0_volume, - omap_mcbsp2_set_st_ch0_volume), + omap_mcbsp_get_st_ch0_volume, + omap_mcbsp_set_st_ch0_volume), OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP2 Sidetone Channel 1 Volume", -32768, 32767, - omap_mcbsp2_get_st_ch1_volume, - omap_mcbsp2_set_st_ch1_volume), + omap_mcbsp_get_st_ch1_volume, + omap_mcbsp_set_st_ch1_volume), }; static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = { @@ -746,25 +725,30 @@ static const struct snd_kcontrol_new omap_mcbsp3_st_controls[] = { omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode), OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 0 Volume", -32768, 32767, - omap_mcbsp3_get_st_ch0_volume, - omap_mcbsp3_set_st_ch0_volume), + omap_mcbsp_get_st_ch0_volume, + omap_mcbsp_set_st_ch0_volume), OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP3 Sidetone Channel 1 Volume", -32768, 32767, - omap_mcbsp3_get_st_ch1_volume, - omap_mcbsp3_set_st_ch1_volume), + omap_mcbsp_get_st_ch1_volume, + omap_mcbsp_set_st_ch1_volume), }; -int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai) +int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd) { - if (!cpu_is_omap34xx()) + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); + + if (!mcbsp->st_data) return -ENODEV; - switch (dai->id) { - case 1: /* McBSP 2 */ - return snd_soc_add_dai_controls(dai, omap_mcbsp2_st_controls, + switch (cpu_dai->id) { + case 2: /* McBSP 2 */ + return snd_soc_add_dai_controls(cpu_dai, + omap_mcbsp2_st_controls, ARRAY_SIZE(omap_mcbsp2_st_controls)); - case 2: /* McBSP 3 */ - return snd_soc_add_dai_controls(dai, omap_mcbsp3_st_controls, + case 3: /* McBSP 3 */ + return snd_soc_add_dai_controls(cpu_dai, + omap_mcbsp3_st_controls, ARRAY_SIZE(omap_mcbsp3_st_controls)); default: break; @@ -776,18 +760,25 @@ EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); static __devinit int asoc_mcbsp_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai); + int ret; + + ret = omap_mcbsp_probe(pdev); + if (!ret) + return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai); + + return ret; } static int __devexit asoc_mcbsp_remove(struct platform_device *pdev) { + omap_mcbsp_remove(pdev); snd_soc_unregister_dai(&pdev->dev); return 0; } static struct platform_driver asoc_mcbsp_driver = { .driver = { - .name = "omap-mcbsp-dai", + .name = "omap-mcbsp", .owner = THIS_MODULE, }, diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index 476fe2add70..f877b16f19c 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -59,6 +59,6 @@ enum omap_mcbsp_div { #define NUM_LINKS 5 #endif -int omap_mcbsp_st_add_controls(struct snd_soc_dai *dai); +int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd); #endif diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h index f95fe306417..b92248cbd47 100644 --- a/sound/soc/omap/omap-pcm.h +++ b/sound/soc/omap/omap-pcm.h @@ -25,6 +25,8 @@ #ifndef __OMAP_PCM_H__ #define __OMAP_PCM_H__ +struct snd_pcm_substream; + struct omap_pcm_dma_data { char *name; /* stream identifier */ int dma_req; /* DMA request line */ diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index 3357dcc47ed..2830dfd0566 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -91,7 +91,7 @@ static struct snd_soc_ops omap3beagle_ops = { static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp-dai.1", + .cpu_dai_name = "omap-mcbsp.2", .platform_name = "omap-pcm-audio", .codec_dai_name = "twl4030-hifi", .codec_name = "twl4030-codec", diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 071fcb09b8b..3d468c9179d 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -58,7 +58,7 @@ static struct snd_soc_ops omap3evm_ops = { static struct snd_soc_dai_link omap3evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp-dai.1", + .cpu_dai_name = "omap-mcbsp.2", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 07794bd1095..4c3a0978578 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -208,7 +208,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { { .name = "PCM1773", .stream_name = "HiFi Out", - .cpu_dai_name = "omap-mcbsp-dai.1", + .cpu_dai_name = "omap-mcbsp.2", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", @@ -219,7 +219,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { }, { .name = "TWL4030", .stream_name = "Line/Mic In", - .cpu_dai_name = "omap-mcbsp-dai.3", + .cpu_dai_name = "omap-mcbsp.4", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index d859b597e7e..b1a9d64cbc5 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -96,7 +96,7 @@ static const struct snd_soc_dapm_route audio_map[] = { static struct snd_soc_dai_link osk_dai = { .name = "TLV320AIC23", .stream_name = "AIC23", - .cpu_dai_name = "omap-mcbsp-dai.0", + .cpu_dai_name = "omap-mcbsp.1", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic23-codec", diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index 2ee889c5025..6ac3e0c3c28 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -60,7 +60,7 @@ static struct snd_soc_ops overo_ops = { static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", - .cpu_dai_name = "omap-mcbsp-dai.1", + .cpu_dai_name = "omap-mcbsp.2", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 58936c730a8..2712dd232b6 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -313,7 +313,7 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); - err = omap_mcbsp_st_add_controls(rtd->cpu_dai); + err = omap_mcbsp_st_add_controls(rtd); if (err < 0) return err; @@ -353,7 +353,7 @@ static struct snd_soc_dai_link rx51_dai[] = { { .name = "TLV320AIC34", .stream_name = "AIC34", - .cpu_dai_name = "omap-mcbsp-dai.1", + .cpu_dai_name = "omap-mcbsp.2", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "omap-pcm-audio", .codec_name = "tlv320aic3x-codec.2-0018", diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 2c850662ea7..0e283226e2b 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -187,7 +187,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { { .name = "TWL4030 I2S", .stream_name = "TWL4030 Audio", - .cpu_dai_name = "omap-mcbsp-dai.1", + .cpu_dai_name = "omap-mcbsp.2", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", @@ -199,7 +199,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { { .name = "TWL4030 PCM", .stream_name = "TWL4030 Voice", - .cpu_dai_name = "omap-mcbsp-dai.2", + .cpu_dai_name = "omap-mcbsp.3", .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 981616d61f6..920e0d9e03d 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -131,7 +131,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { { .name = "TWL4030 I2S", .stream_name = "TWL4030 Audio", - .cpu_dai_name = "omap-mcbsp-dai.1", + .cpu_dai_name = "omap-mcbsp.2", .codec_dai_name = "twl4030-hifi", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", @@ -143,7 +143,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { { .name = "TWL4030 PCM", .stream_name = "TWL4030 Voice", - .cpu_dai_name = "omap-mcbsp-dai.2", + .cpu_dai_name = "omap-mcbsp.3", .codec_dai_name = "twl4030-voice", .platform_name = "omap-pcm-audio", .codec_name = "twl4030-codec", -- cgit v1.2.3-18-g5258 From 81da6a9e49b9561f325d201c54eca274f066e13b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 13 Feb 2012 15:36:49 +0200 Subject: ASoC: OMAP: mcbsp.h: Clean up bit definitions Use BIT() for bit position, correct field definition by adding mask to them, and also adding the missing spaces around '<<' Signed-off-by: Peter Ujfalusi Tested-by: Grazvydas Ignotas Tested-by: Janusz Krzysztofik Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.h | 194 ++++++++++++++++++++++++------------------------- 1 file changed, 97 insertions(+), 97 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index 6d579938a15..ec00c275ec4 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -89,130 +89,130 @@ enum { #define OMAP_ST_REG_SSELCR 0x2C /************************** McBSP SPCR1 bit definitions ***********************/ -#define RRST 0x0001 -#define RRDY 0x0002 -#define RFULL 0x0004 -#define RSYNC_ERR 0x0008 -#define RINTM(value) ((value)<<4) /* bits 4:5 */ -#define ABIS 0x0040 -#define DXENA 0x0080 -#define CLKSTP(value) ((value)<<11) /* bits 11:12 */ -#define RJUST(value) ((value)<<13) /* bits 13:14 */ -#define ALB 0x8000 -#define DLB 0x8000 +#define RRST BIT(0) +#define RRDY BIT(1) +#define RFULL BIT(2) +#define RSYNC_ERR BIT(3) +#define RINTM(value) (((value) & 0x3) << 4) /* bits 4:5 */ +#define ABIS BIT(6) +#define DXENA BIT(7) +#define CLKSTP(value) (((value) & 0x3) << 11) /* bits 11:12 */ +#define RJUST(value) (((value) & 0x3) << 13) /* bits 13:14 */ +#define ALB BIT(15) +#define DLB BIT(15) /************************** McBSP SPCR2 bit definitions ***********************/ -#define XRST 0x0001 -#define XRDY 0x0002 -#define XEMPTY 0x0004 -#define XSYNC_ERR 0x0008 -#define XINTM(value) ((value)<<4) /* bits 4:5 */ -#define GRST 0x0040 -#define FRST 0x0080 -#define SOFT 0x0100 -#define FREE 0x0200 +#define XRST BIT(0) +#define XRDY BIT(1) +#define XEMPTY BIT(2) +#define XSYNC_ERR BIT(3) +#define XINTM(value) (((value) & 0x3) << 4) /* bits 4:5 */ +#define GRST BIT(6) +#define FRST BIT(7) +#define SOFT BIT(8) +#define FREE BIT(9) /************************** McBSP PCR bit definitions *************************/ -#define CLKRP 0x0001 -#define CLKXP 0x0002 -#define FSRP 0x0004 -#define FSXP 0x0008 -#define DR_STAT 0x0010 -#define DX_STAT 0x0020 -#define CLKS_STAT 0x0040 -#define SCLKME 0x0080 -#define CLKRM 0x0100 -#define CLKXM 0x0200 -#define FSRM 0x0400 -#define FSXM 0x0800 -#define RIOEN 0x1000 -#define XIOEN 0x2000 -#define IDLE_EN 0x4000 +#define CLKRP BIT(0) +#define CLKXP BIT(1) +#define FSRP BIT(2) +#define FSXP BIT(3) +#define DR_STAT BIT(4) +#define DX_STAT BIT(5) +#define CLKS_STAT BIT(6) +#define SCLKME BIT(7) +#define CLKRM BIT(8) +#define CLKXM BIT(9) +#define FSRM BIT(10) +#define FSXM BIT(11) +#define RIOEN BIT(12) +#define XIOEN BIT(13) +#define IDLE_EN BIT(14) /************************** McBSP RCR1 bit definitions ************************/ -#define RWDLEN1(value) ((value)<<5) /* Bits 5:7 */ -#define RFRLEN1(value) ((value)<<8) /* Bits 8:14 */ +#define RWDLEN1(value) (((value) & 0x7) << 5) /* Bits 5:7 */ +#define RFRLEN1(value) (((value) & 0x7f) << 8) /* Bits 8:14 */ /************************** McBSP XCR1 bit definitions ************************/ -#define XWDLEN1(value) ((value)<<5) /* Bits 5:7 */ -#define XFRLEN1(value) ((value)<<8) /* Bits 8:14 */ +#define XWDLEN1(value) (((value) & 0x7) << 5) /* Bits 5:7 */ +#define XFRLEN1(value) (((value) & 0x7f) << 8) /* Bits 8:14 */ /*************************** McBSP RCR2 bit definitions ***********************/ -#define RDATDLY(value) (value) /* Bits 0:1 */ -#define RFIG 0x0004 -#define RCOMPAND(value) ((value)<<3) /* Bits 3:4 */ -#define RWDLEN2(value) ((value)<<5) /* Bits 5:7 */ -#define RFRLEN2(value) ((value)<<8) /* Bits 8:14 */ -#define RPHASE 0x8000 +#define RDATDLY(value) ((value) & 0x3) /* Bits 0:1 */ +#define RFIG BIT(2) +#define RCOMPAND(value) (((value) & 0x3) << 3) /* Bits 3:4 */ +#define RWDLEN2(value) (((value) & 0x7) << 5) /* Bits 5:7 */ +#define RFRLEN2(value) (((value) & 0x7f) << 8) /* Bits 8:14 */ +#define RPHASE BIT(15) /*************************** McBSP XCR2 bit definitions ***********************/ -#define XDATDLY(value) (value) /* Bits 0:1 */ -#define XFIG 0x0004 -#define XCOMPAND(value) ((value)<<3) /* Bits 3:4 */ -#define XWDLEN2(value) ((value)<<5) /* Bits 5:7 */ -#define XFRLEN2(value) ((value)<<8) /* Bits 8:14 */ -#define XPHASE 0x8000 +#define XDATDLY(value) ((value) & 0x3) /* Bits 0:1 */ +#define XFIG BIT(2) +#define XCOMPAND(value) (((value) & 0x3) << 3) /* Bits 3:4 */ +#define XWDLEN2(value) (((value) & 0x7) << 5) /* Bits 5:7 */ +#define XFRLEN2(value) (((value) & 0x7f) << 8) /* Bits 8:14 */ +#define XPHASE BIT(15) /************************* McBSP SRGR1 bit definitions ************************/ -#define CLKGDV(value) (value) /* Bits 0:7 */ -#define FWID(value) ((value)<<8) /* Bits 8:15 */ +#define CLKGDV(value) ((value) & 0x7f) /* Bits 0:7 */ +#define FWID(value) (((value) & 0xff) << 8) /* Bits 8:15 */ /************************* McBSP SRGR2 bit definitions ************************/ -#define FPER(value) (value) /* Bits 0:11 */ -#define FSGM 0x1000 -#define CLKSM 0x2000 -#define CLKSP 0x4000 -#define GSYNC 0x8000 +#define FPER(value) ((value) & 0x0fff) /* Bits 0:11 */ +#define FSGM BIT(12) +#define CLKSM BIT(13) +#define CLKSP BIT(14) +#define GSYNC BIT(15) /************************* McBSP MCR1 bit definitions *************************/ -#define RMCM 0x0001 -#define RCBLK(value) ((value)<<2) /* Bits 2:4 */ -#define RPABLK(value) ((value)<<5) /* Bits 5:6 */ -#define RPBBLK(value) ((value)<<7) /* Bits 7:8 */ +#define RMCM BIT(0) +#define RCBLK(value) (((value) & 0x7) << 2) /* Bits 2:4 */ +#define RPABLK(value) (((value) & 0x3) << 5) /* Bits 5:6 */ +#define RPBBLK(value) (((value) & 0x3) << 7) /* Bits 7:8 */ /************************* McBSP MCR2 bit definitions *************************/ -#define XMCM(value) (value) /* Bits 0:1 */ -#define XCBLK(value) ((value)<<2) /* Bits 2:4 */ -#define XPABLK(value) ((value)<<5) /* Bits 5:6 */ -#define XPBBLK(value) ((value)<<7) /* Bits 7:8 */ +#define XMCM(value) ((value) & 0x3) /* Bits 0:1 */ +#define XCBLK(value) (((value) & 0x7) << 2) /* Bits 2:4 */ +#define XPABLK(value) (((value) & 0x3) << 5) /* Bits 5:6 */ +#define XPBBLK(value) (((value) & 0x3) << 7) /* Bits 7:8 */ /*********************** McBSP XCCR bit definitions *************************/ -#define EXTCLKGATE 0x8000 -#define PPCONNECT 0x4000 -#define DXENDLY(value) ((value)<<12) /* Bits 12:13 */ -#define XFULL_CYCLE 0x0800 -#define DILB 0x0020 -#define XDMAEN 0x0008 -#define XDISABLE 0x0001 +#define XDISABLE BIT(0) +#define XDMAEN BIT(3) +#define DILB BIT(5) +#define XFULL_CYCLE BIT(11) +#define DXENDLY(value) (((value) & 0x3) << 12) /* Bits 12:13 */ +#define PPCONNECT BIT(14) +#define EXTCLKGATE BIT(15) /********************** McBSP RCCR bit definitions *************************/ -#define RFULL_CYCLE 0x0800 -#define RDMAEN 0x0008 -#define RDISABLE 0x0001 +#define RDISABLE BIT(0) +#define RDMAEN BIT(3) +#define RFULL_CYCLE BIT(11) /********************** McBSP SYSCONFIG bit definitions ********************/ -#define CLOCKACTIVITY(value) ((value)<<8) -#define SIDLEMODE(value) ((value)<<3) -#define ENAWAKEUP 0x0004 -#define SOFTRST 0x0002 +#define SOFTRST BIT(1) +#define ENAWAKEUP BIT(2) +#define SIDLEMODE(value) (((value) & 0x3) << 3) +#define CLOCKACTIVITY(value) (((value) & 0x3) << 8) /********************** McBSP SSELCR bit definitions ***********************/ -#define SIDETONEEN 0x0400 +#define SIDETONEEN BIT(10) /********************** McBSP Sidetone SYSCONFIG bit definitions ***********/ -#define ST_AUTOIDLE 0x0001 +#define ST_AUTOIDLE BIT(0) /********************** McBSP Sidetone SGAINCR bit definitions *************/ -#define ST_CH1GAIN(value) ((value<<16)) /* Bits 16:31 */ -#define ST_CH0GAIN(value) (value) /* Bits 0:15 */ +#define ST_CH0GAIN(value) ((value) & 0xffff) /* Bits 0:15 */ +#define ST_CH1GAIN(value) (((value) & 0xffff) << 16) /* Bits 16:31 */ /********************** McBSP Sidetone SFIRCR bit definitions **************/ -#define ST_FIRCOEFF(value) (value) /* Bits 0:15 */ +#define ST_FIRCOEFF(value) ((value) & 0xffff) /* Bits 0:15 */ /********************** McBSP Sidetone SSELCR bit definitions **************/ -#define ST_COEFFWRDONE 0x0004 -#define ST_COEFFWREN 0x0002 -#define ST_SIDETONEEN 0x0001 +#define ST_SIDETONEEN BIT(0) +#define ST_COEFFWREN BIT(1) +#define ST_COEFFWRDONE BIT(2) /********************** McBSP DMA operating modes **************************/ #define MCBSP_DMA_MODE_ELEMENT 0 @@ -220,15 +220,15 @@ enum { #define MCBSP_DMA_MODE_FRAME 2 /********************** McBSP WAKEUPEN bit definitions *********************/ -#define XEMPTYEOFEN 0x4000 -#define XRDYEN 0x0400 -#define XEOFEN 0x0200 -#define XFSXEN 0x0100 -#define XSYNCERREN 0x0080 -#define RRDYEN 0x0008 -#define REOFEN 0x0004 -#define RFSREN 0x0002 -#define RSYNCERREN 0x0001 +#define RSYNCERREN BIT(0) +#define RFSREN BIT(1) +#define REOFEN BIT(2) +#define RRDYEN BIT(3) +#define XSYNCERREN BIT(7) +#define XFSXEN BIT(8) +#define XEOFEN BIT(9) +#define XRDYEN BIT(10) +#define XEMPTYEOFEN BIT(14) /* we don't do multichannel for now */ struct omap_mcbsp_reg_cfg { -- cgit v1.2.3-18-g5258 From cb40b63a224b41325a8ecdbeed5290866864b9ae Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 13 Feb 2012 16:26:54 +0200 Subject: ASoC: OMAP McBSP: Remove redundant accessors We no longer need accessor functions for max_tx/rx_threshold, dma_op_mode or for the FIFO size. Signed-off-by: Peter Ujfalusi Tested-by: Grazvydas Ignotas Tested-by: Janusz Krzysztofik Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 36 ------------------------------------ sound/soc/omap/mcbsp.h | 3 --- sound/soc/omap/omap-mcbsp.c | 14 +++++--------- 3 files changed, 5 insertions(+), 48 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index be92a28e19e..95ba7e0d207 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -441,29 +441,6 @@ void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold) MCBSP_WRITE(mcbsp, THRSH1, threshold - 1); } -/* - * omap_mcbsp_get_max_tx_thres just return the current configured - * maximum threshold for transmission - */ -u16 omap_mcbsp_get_max_tx_threshold(struct omap_mcbsp *mcbsp) -{ - return mcbsp->max_tx_thres; -} - -/* - * omap_mcbsp_get_max_rx_thres just return the current configured - * maximum threshold for reception - */ -u16 omap_mcbsp_get_max_rx_threshold(struct omap_mcbsp *mcbsp) -{ - return mcbsp->max_rx_thres; -} - -u16 omap_mcbsp_get_fifo_size(struct omap_mcbsp *mcbsp) -{ - return mcbsp->pdata->buffer_size; -} - /* * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO */ @@ -504,19 +481,6 @@ u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp) return threshold - buffstat; } -/* - * omap_mcbsp_get_dma_op_mode just return the current configured - * operating mode for the mcbsp channel - */ -int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp) -{ - int dma_op_mode; - - dma_op_mode = mcbsp->dma_op_mode; - - return dma_op_mode; -} - int omap_mcbsp_request(struct omap_mcbsp *mcbsp) { void *reg_cache; diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index ec00c275ec4..a991e1dcb5e 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -316,9 +316,6 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, const struct omap_mcbsp_reg_cfg *config); void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold); -u16 omap_mcbsp_get_max_tx_threshold(struct omap_mcbsp *mcbsp); -u16 omap_mcbsp_get_max_rx_threshold(struct omap_mcbsp *mcbsp); -u16 omap_mcbsp_get_fifo_size(struct omap_mcbsp *mcbsp); u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp); u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp); int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 89240025949..9c703f18714 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -67,13 +67,12 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; struct omap_pcm_dma_data *dma_data; - int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp); int words; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ - if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) + if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) /* * Configure McBSP threshold based on either: * packet_size, when the sDMA is in packet mode, or @@ -106,7 +105,7 @@ static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params, int size; snd_interval_any(&frames); - size = omap_mcbsp_get_fifo_size(mcbsp); + size = mcbsp->pdata->buffer_size; frames.min = size / channels->min; frames.integer = 1; @@ -255,17 +254,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, if (mcbsp->pdata->buffer_size) { dma_data->set_threshold = omap_mcbsp_set_threshold; /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ - if (omap_mcbsp_get_dma_op_mode(mcbsp) == - MCBSP_DMA_MODE_THRESHOLD) { + if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) { int period_words, max_thrsh; period_words = params_period_bytes(params) / (wlen / 8); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - max_thrsh = omap_mcbsp_get_max_tx_threshold( - mcbsp); + max_thrsh = mcbsp->max_tx_thres; else - max_thrsh = omap_mcbsp_get_max_rx_threshold( - mcbsp); + max_thrsh = mcbsp->max_rx_thres; /* * If the period contains less or equal number of words, * we are using the original threshold mode setup: -- cgit v1.2.3-18-g5258 From 2ee6595069f29b918b957a013debfae83e68724d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 14 Feb 2012 14:52:42 +0200 Subject: ASoC: omap-mcbsp: Cleanup of module probe/remove code Use devm_* where it is possible to save on cleanup path. Start merging the two mcbsp file content. Move pm_runtime_enable/disable calls to ASoC probe, remove from module probe/remove time. Signed-off-by: Peter Ujfalusi Tested-by: Grazvydas Ignotas Tested-by: Janusz Krzysztofik Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 136 +++++++++++--------------------------------- sound/soc/omap/mcbsp.h | 4 +- sound/soc/omap/omap-mcbsp.c | 51 ++++++++++++++++- 3 files changed, 85 insertions(+), 106 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 95ba7e0d207..9e39c58a467 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -26,7 +26,6 @@ #include #include -#include #include "mcbsp.h" @@ -915,90 +914,56 @@ static int __devinit omap_st_add(struct omap_mcbsp *mcbsp, struct omap_mcbsp_st_data *st_data; int err; - st_data = kzalloc(sizeof(*mcbsp->st_data), GFP_KERNEL); - if (!st_data) { - err = -ENOMEM; - goto err1; - } + st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL); + if (!st_data) + return -ENOMEM; - st_data->io_base_st = ioremap(res->start, resource_size(res)); - if (!st_data->io_base_st) { - err = -ENOMEM; - goto err2; - } + st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start, + resource_size(res)); + if (!st_data->io_base_st) + return -ENOMEM; err = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); if (err) - goto err3; + return err; mcbsp->st_data = st_data; return 0; - -err3: - iounmap(st_data->io_base_st); -err2: - kfree(st_data); -err1: - return err; - -} - -static void __devexit omap_st_remove(struct omap_mcbsp *mcbsp) -{ - struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - iounmap(st_data->io_base_st); - kfree(st_data); } /* * McBSP1 and McBSP3 are directly mapped on 1610 and 1510. * 730 has only 2 McBSP, and both of them are MPU peripherals. */ -int __devinit omap_mcbsp_probe(struct platform_device *pdev) +int __devinit omap_mcbsp_init(struct platform_device *pdev) { - struct omap_mcbsp_platform_data *pdata = pdev->dev.platform_data; - struct omap_mcbsp *mcbsp; + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); struct resource *res; int ret = 0; - if (!pdata) { - dev_err(&pdev->dev, "McBSP device initialized without" - "platform data\n"); - ret = -EINVAL; - goto exit; - } - - dev_dbg(&pdev->dev, "Initializing OMAP McBSP (%d).\n", pdev->id); - - mcbsp = kzalloc(sizeof(struct omap_mcbsp), GFP_KERNEL); - if (!mcbsp) { - ret = -ENOMEM; - goto exit; - } - spin_lock_init(&mcbsp->lock); - mcbsp->id = pdev->id; mcbsp->free = true; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!res) { res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { - dev_err(&pdev->dev, "%s:mcbsp%d has invalid memory" - "resource\n", __func__, pdev->id); - ret = -ENOMEM; - goto exit; + dev_err(mcbsp->dev, "invalid memory resource\n"); + return -ENOMEM; } } + if (!devm_request_mem_region(&pdev->dev, res->start, resource_size(res), + dev_name(&pdev->dev))) { + dev_err(mcbsp->dev, "memory region already claimed\n"); + return -ENODEV; + } + mcbsp->phys_base = res->start; mcbsp->reg_cache_size = resource_size(res); - mcbsp->io_base = ioremap(res->start, resource_size(res)); - if (!mcbsp->io_base) { - ret = -ENOMEM; - goto err_ioremap; - } + mcbsp->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!mcbsp->io_base) + return -ENOMEM; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma"); if (!res) @@ -1015,34 +980,25 @@ int __devinit omap_mcbsp_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); if (!res) { - dev_err(&pdev->dev, "%s:mcbsp%d has invalid rx DMA channel\n", - __func__, pdev->id); - ret = -ENODEV; - goto err_res; + dev_err(&pdev->dev, "invalid rx DMA channel\n"); + return -ENODEV; } mcbsp->dma_rx_sync = res->start; res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); if (!res) { - dev_err(&pdev->dev, "%s:mcbsp%d has invalid tx DMA channel\n", - __func__, pdev->id); - ret = -ENODEV; - goto err_res; + dev_err(&pdev->dev, "invalid tx DMA channel\n"); + return -ENODEV; } mcbsp->dma_tx_sync = res->start; mcbsp->fclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { ret = PTR_ERR(mcbsp->fclk); - dev_err(&pdev->dev, "unable to get fck: %d\n", ret); - goto err_res; + dev_err(mcbsp->dev, "unable to get fck: %d\n", ret); + return ret; } - mcbsp->pdata = pdata; - mcbsp->dev = &pdev->dev; - platform_set_drvdata(pdev, mcbsp); - pm_runtime_enable(mcbsp->dev); - mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT; if (mcbsp->pdata->buffer_size) { /* @@ -1082,41 +1038,17 @@ int __devinit omap_mcbsp_probe(struct platform_device *pdev) err_st: if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, - &additional_attr_group); + sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); err_thres: clk_put(mcbsp->fclk); -err_res: - iounmap(mcbsp->io_base); -err_ioremap: - kfree(mcbsp); -exit: return ret; } -int __devexit omap_mcbsp_remove(struct platform_device *pdev) +void __devexit omap_mcbsp_sysfs_remove(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - platform_set_drvdata(pdev, NULL); - if (mcbsp) { - - if (mcbsp->pdata && mcbsp->pdata->ops && - mcbsp->pdata->ops->free) - mcbsp->pdata->ops->free(mcbsp->id); - - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, - &additional_attr_group); - - if (mcbsp->st_data) - omap_st_remove(mcbsp); - - clk_put(mcbsp->fclk); - - iounmap(mcbsp->io_base); - kfree(mcbsp); - } + if (mcbsp->pdata->buffer_size) + sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return 0; + if (mcbsp->st_data) + sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); } diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index a991e1dcb5e..ac90c1a4a48 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -341,7 +341,7 @@ int omap_st_enable(struct omap_mcbsp *mcbsp); int omap_st_disable(struct omap_mcbsp *mcbsp); int omap_st_is_enabled(struct omap_mcbsp *mcbsp); -int __devinit omap_mcbsp_probe(struct platform_device *pdev); -int __devexit omap_mcbsp_remove(struct platform_device *pdev); +int __devinit omap_mcbsp_init(struct platform_device *pdev); +void __devexit omap_mcbsp_sysfs_remove(struct omap_mcbsp *mcbsp); #endif /* __ASOC_MCBSP_H */ diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 9c703f18714..69a44aa4eea 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -603,7 +604,27 @@ static const struct snd_soc_dai_ops mcbsp_dai_ops = { .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, }; +static int omap_mcbsp_probe(struct snd_soc_dai *dai) +{ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(dai); + + pm_runtime_enable(mcbsp->dev); + + return 0; +} + +static int omap_mcbsp_remove(struct snd_soc_dai *dai) +{ + struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(dai); + + pm_runtime_disable(mcbsp->dev); + + return 0; +} + static struct snd_soc_dai_driver omap_mcbsp_dai = { + .probe = omap_mcbsp_probe, + .remove = omap_mcbsp_remove, .playback = { .channels_min = 1, .channels_max = 16, @@ -756,9 +777,24 @@ EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls); static __devinit int asoc_mcbsp_probe(struct platform_device *pdev) { + struct omap_mcbsp_platform_data *pdata = dev_get_platdata(&pdev->dev); + struct omap_mcbsp *mcbsp; int ret; - ret = omap_mcbsp_probe(pdev); + if (!pdata) { + dev_err(&pdev->dev, "missing platform data.\n"); + return -EINVAL; + } + mcbsp = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcbsp), GFP_KERNEL); + if (!mcbsp) + return -ENOMEM; + + mcbsp->id = pdev->id; + mcbsp->pdata = pdata; + mcbsp->dev = &pdev->dev; + platform_set_drvdata(pdev, mcbsp); + + ret = omap_mcbsp_init(pdev); if (!ret) return snd_soc_register_dai(&pdev->dev, &omap_mcbsp_dai); @@ -767,8 +803,19 @@ static __devinit int asoc_mcbsp_probe(struct platform_device *pdev) static int __devexit asoc_mcbsp_remove(struct platform_device *pdev) { - omap_mcbsp_remove(pdev); + struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); + snd_soc_unregister_dai(&pdev->dev); + + if (mcbsp->pdata->ops && mcbsp->pdata->ops->free) + mcbsp->pdata->ops->free(mcbsp->id); + + omap_mcbsp_sysfs_remove(mcbsp); + + clk_put(mcbsp->fclk); + + platform_set_drvdata(pdev, NULL); + return 0; } -- cgit v1.2.3-18-g5258 From 256d9c251fe6800a494206b96d2572e5a98762d5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 14 Feb 2012 15:23:15 +0200 Subject: ASoC: omap-mcbsp: Merge the omap_mcbsp_data into omap_mcbsp structure Since the drivers has been merged, merge the two structures together. Signed-off-by: Peter Ujfalusi Tested-by: Grazvydas Ignotas Tested-by: Janusz Krzysztofik Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.h | 33 +++++++++++++--------------- sound/soc/omap/omap-mcbsp.c | 53 ++++++++++++++++++++------------------------- 2 files changed, 38 insertions(+), 48 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index ac90c1a4a48..d250bcc952d 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -269,27 +269,20 @@ struct omap_mcbsp_st_data { s16 ch1gain; }; -struct omap_mcbsp_data { - struct omap_mcbsp_reg_cfg regs; - struct omap_pcm_dma_data dma_data[2]; - unsigned int fmt; - /* - * Flags indicating is the bus already activated and configured by - * another substream - */ - int active; - int configured; - unsigned int in_freq; - int clk_div; - int wlen; -}; - struct omap_mcbsp { struct device *dev; + struct clk *fclk; + spinlock_t lock; unsigned long phys_base; unsigned long phys_dma_base; void __iomem *io_base; u8 id; + /* + * Flags indicating is the bus already activated and configured by + * another substream + */ + int active; + int configured; u8 free; int rx_irq; @@ -300,16 +293,20 @@ struct omap_mcbsp { u8 dma_tx_sync; /* Protect the field .free, while checking if the mcbsp is in use */ - spinlock_t lock; struct omap_mcbsp_platform_data *pdata; - struct clk *fclk; struct omap_mcbsp_st_data *st_data; - struct omap_mcbsp_data mcbsp_data; + struct omap_mcbsp_reg_cfg cfg_regs; + struct omap_pcm_dma_data dma_data[2]; int dma_op_mode; u16 max_tx_thres; u16 max_rx_thres; void *reg_cache; int reg_cache_size; + + unsigned int fmt; + unsigned int in_freq; + int clk_div; + int wlen; }; void omap_mcbsp_config(struct omap_mcbsp *mcbsp, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 69a44aa4eea..4cd7af883de 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -66,7 +66,6 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; struct omap_pcm_dma_data *dma_data; int words; @@ -83,7 +82,7 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) words = dma_data->packet_size; else words = snd_pcm_lib_period_bytes(substream) / - (mcbsp_data->wlen / 8); + (mcbsp->wlen / 8); else words = 1; @@ -160,11 +159,10 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; if (!cpu_dai->active) { omap_mcbsp_free(mcbsp); - mcbsp_data->configured = 0; + mcbsp->configured = 0; } } @@ -172,14 +170,13 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - mcbsp_data->active++; + mcbsp->active++; omap_mcbsp_start(mcbsp, play, !play); break; @@ -187,7 +184,7 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: omap_mcbsp_stop(mcbsp, play, !play); - mcbsp_data->active--; + mcbsp->active--; break; default: err = -EINVAL; @@ -226,8 +223,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; - struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; struct omap_pcm_dma_data *dma_data; int dma; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; @@ -235,7 +231,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, unsigned long port; unsigned int format, div, framesize, master; - dma_data = &mcbsp_data->dma_data[substream->stream]; + dma_data = &mcbsp->dma_data[substream->stream]; dma = omap_mcbsp_dma_ch_params(mcbsp, substream->stream); port = omap_mcbsp_dma_reg_params(mcbsp, substream->stream); @@ -303,7 +299,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - if (mcbsp_data->configured) { + if (mcbsp->configured) { /* McBSP already configured by another stream */ return 0; } @@ -312,7 +308,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->xcr2 &= ~(RPHASE | XFRLEN2(0x7f) | XWDLEN2(7)); regs->rcr1 &= ~(RFRLEN1(0x7f) | RWDLEN1(7)); regs->xcr1 &= ~(XFRLEN1(0x7f) | XWDLEN1(7)); - format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + format = mcbsp->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); if (channels == 2 && (format == SND_SOC_DAIFMT_I2S || format == SND_SOC_DAIFMT_LEFT_J)) { @@ -350,10 +346,10 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, /* In McBSP master modes, FRAME (i.e. sample rate) is generated * by _counting_ BCLKs. Calculate frame size in BCLKs */ - master = mcbsp_data->fmt & SND_SOC_DAIFMT_MASTER_MASK; + master = mcbsp->fmt & SND_SOC_DAIFMT_MASTER_MASK; if (master == SND_SOC_DAIFMT_CBS_CFS) { - div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1; - framesize = (mcbsp_data->in_freq / div) / params_rate(params); + div = mcbsp->clk_div ? mcbsp->clk_div : 1; + framesize = (mcbsp->in_freq / div) / params_rate(params); if (framesize < wlen * channels) { printk(KERN_ERR "%s: not enough bandwidth for desired rate and " @@ -379,9 +375,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; } - omap_mcbsp_config(mcbsp, &mcbsp_data->regs); - mcbsp_data->wlen = wlen; - mcbsp_data->configured = 1; + omap_mcbsp_config(mcbsp, &mcbsp->cfg_regs); + mcbsp->wlen = wlen; + mcbsp->configured = 1; return 0; } @@ -394,14 +390,13 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; - struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; bool inv_fs = false; - if (mcbsp_data->configured) + if (mcbsp->configured) return 0; - mcbsp_data->fmt = fmt; + mcbsp->fmt = fmt; memset(regs, 0, sizeof(*regs)); /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; @@ -497,13 +492,12 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; - struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; if (div_id != OMAP_MCBSP_CLKGDV) return -ENODEV; - mcbsp_data->clk_div = div; + mcbsp->clk_div = div; regs->srgr1 &= ~CLKGDV(0xff); regs->srgr1 |= CLKGDV(div - 1); @@ -515,12 +509,11 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int dir) { struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); - struct omap_mcbsp_data *mcbsp_data = &mcbsp->mcbsp_data; - struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; int err = 0; - if (mcbsp_data->active) { - if (freq == mcbsp_data->in_freq) + if (mcbsp->active) { + if (freq == mcbsp->in_freq) return 0; else return -EBUSY; @@ -534,7 +527,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, if (cpu_class_is_omap1() || cpu_dai->id != 1) return -EINVAL; - mcbsp_data->in_freq = freq; + mcbsp->in_freq = freq; regs->srgr2 &= ~CLKSM; regs->pcr0 &= ~SCLKME; -- cgit v1.2.3-18-g5258 From b8fb4907a74dbcbd0b21e02380d58e422bd4a1fe Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 14 Feb 2012 15:41:29 +0200 Subject: ASoC: omap-mcbsp: Simplify DMA configuration Configure the DMA request line, port address, and stream name at probe time instead of every time we start a stream. These settings are static in the system. Signed-off-by: Peter Ujfalusi Tested-by: Grazvydas Ignotas Tested-by: Janusz Krzysztofik Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 29 ++++++++++------------------- sound/soc/omap/mcbsp.h | 7 ------- sound/soc/omap/omap-mcbsp.c | 8 -------- 3 files changed, 10 insertions(+), 34 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 9e39c58a467..fe4734e0c18 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -175,22 +175,6 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, } } -/** - * omap_mcbsp_dma_params - returns the dma channel number - * @id - mcbsp id - * @stream - indicates the direction of data flow (rx or tx) - * - * Returns the dma channel number for the rx channel or tx channel - * based on the value of @stream for the requested mcbsp given by @id - */ -int omap_mcbsp_dma_ch_params(struct omap_mcbsp *mcbsp, unsigned int stream) -{ - if (stream) - return mcbsp->dma_rx_sync; - else - return mcbsp->dma_tx_sync; -} - /** * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register * @id - mcbsp id @@ -200,7 +184,8 @@ int omap_mcbsp_dma_ch_params(struct omap_mcbsp *mcbsp, unsigned int stream) * to be used by DMA for transferring/receiving data based on the value of * @stream for the requested mcbsp given by @id */ -int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, unsigned int stream) +static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, + unsigned int stream) { int data_reg; @@ -983,14 +968,20 @@ int __devinit omap_mcbsp_init(struct platform_device *pdev) dev_err(&pdev->dev, "invalid rx DMA channel\n"); return -ENODEV; } - mcbsp->dma_rx_sync = res->start; + /* RX DMA request number, and port address configuration */ + mcbsp->dma_data[1].name = "Audio Capture"; + mcbsp->dma_data[1].dma_req = res->start; + mcbsp->dma_data[1].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 1); res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); if (!res) { dev_err(&pdev->dev, "invalid tx DMA channel\n"); return -ENODEV; } - mcbsp->dma_tx_sync = res->start; + /* TX DMA request number, and port address configuration */ + mcbsp->dma_data[0].name = "Audio Playback"; + mcbsp->dma_data[0].dma_req = res->start; + mcbsp->dma_data[0].port_addr = omap_mcbsp_dma_reg_params(mcbsp, 0); mcbsp->fclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index d250bcc952d..a5518d71913 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -288,10 +288,6 @@ struct omap_mcbsp { int rx_irq; int tx_irq; - /* DMA stuff */ - u8 dma_rx_sync; - u8 dma_tx_sync; - /* Protect the field .free, while checking if the mcbsp is in use */ struct omap_mcbsp_platform_data *pdata; struct omap_mcbsp_st_data *st_data; @@ -328,9 +324,6 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); void omap2_mcbsp1_mux_clkr_src(struct omap_mcbsp *mcbsp, u8 mux); void omap2_mcbsp1_mux_fsr_src(struct omap_mcbsp *mcbsp, u8 mux); -int omap_mcbsp_dma_ch_params(struct omap_mcbsp *mcbsp, unsigned int stream); -int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp, unsigned int stream); - /* Sidetone specific API */ int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain); int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 4cd7af883de..10eb645ceee 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -225,17 +225,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai); struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs; struct omap_pcm_dma_data *dma_data; - int dma; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; int pkt_size = 0; - unsigned long port; unsigned int format, div, framesize, master; dma_data = &mcbsp->dma_data[substream->stream]; - dma = omap_mcbsp_dma_ch_params(mcbsp, substream->stream); - port = omap_mcbsp_dma_reg_params(mcbsp, substream->stream); - switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: dma_data->data_type = OMAP_DMA_DATA_TYPE_S16; @@ -291,9 +286,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } } - dma_data->name = substream->stream ? "Audio Capture" : "Audio Playback"; - dma_data->dma_req = dma; - dma_data->port_addr = port; dma_data->sync_mode = sync_mode; dma_data->packet_size = pkt_size; -- cgit v1.2.3-18-g5258 From e2002ab35ff7f9111081824667ce331b2c33923c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 23 Feb 2012 15:38:37 +0200 Subject: ASoC: omap: mcbsp: Use uniform st_data pointer initialization In this way we can save few lines, and have uniform way of initializing the st_data in all functions. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 21 +++++---------------- 1 file changed, 5 insertions(+), 16 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index fe4734e0c18..79f6da6381a 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -273,11 +273,9 @@ static void omap_st_chgain(struct omap_mcbsp *mcbsp) int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain) { - struct omap_mcbsp_st_data *st_data; + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; int ret = 0; - st_data = mcbsp->st_data; - if (!st_data) return -ENOENT; @@ -298,11 +296,9 @@ int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain) int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain) { - struct omap_mcbsp_st_data *st_data; + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; int ret = 0; - st_data = mcbsp->st_data; - if (!st_data) return -ENOENT; @@ -337,9 +333,7 @@ static int omap_st_start(struct omap_mcbsp *mcbsp) int omap_st_enable(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp_st_data *st_data; - - st_data = mcbsp->st_data; + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; if (!st_data) return -ENODEV; @@ -368,11 +362,9 @@ static int omap_st_stop(struct omap_mcbsp *mcbsp) int omap_st_disable(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp_st_data *st_data; + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; int ret = 0; - st_data = mcbsp->st_data; - if (!st_data) return -ENODEV; @@ -386,14 +378,11 @@ int omap_st_disable(struct omap_mcbsp *mcbsp) int omap_st_is_enabled(struct omap_mcbsp *mcbsp) { - struct omap_mcbsp_st_data *st_data; - - st_data = mcbsp->st_data; + struct omap_mcbsp_st_data *st_data = mcbsp->st_data; if (!st_data) return -ENODEV; - return st_data->enabled; } -- cgit v1.2.3-18-g5258 From 58db1dcde0dcf5143dc6a54017e4a72c25fb8db0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 23 Feb 2012 15:40:55 +0200 Subject: ASoC: omap: mcbsp: Remove redundant checks for the st_data pointer The parent functions of omap_st_start/stop also checks the validity of the st_data pointer so we do not need to do it again inside of omap_st_start/stop Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 79f6da6381a..5f6c21d4b59 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -318,7 +318,7 @@ static int omap_st_start(struct omap_mcbsp *mcbsp) { struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - if (st_data && st_data->enabled && !st_data->running) { + if (st_data->enabled && !st_data->running) { omap_st_fir_write(mcbsp, st_data->taps); omap_st_chgain(mcbsp); @@ -350,7 +350,7 @@ static int omap_st_stop(struct omap_mcbsp *mcbsp) { struct omap_mcbsp_st_data *st_data = mcbsp->st_data; - if (st_data && st_data->running) { + if (st_data->running) { if (!mcbsp->free) { omap_st_off(mcbsp); st_data->running = 0; -- cgit v1.2.3-18-g5258 From 08905d8ab4d4a264c5a700b04b9cbafe4f381037 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 5 Mar 2012 11:27:40 +0200 Subject: ASoC: omap-mcbsp: Configure wakeup in later phase Configure the WAKEUPEN register at the same time we configure the rest of the McBSP registers. In case of OMAP3+, if the sysclock has been reconfigured we are going to disable McBSP for the duration of the clock change, which will reset the McBSP registers. The WAKEUPEN register need to be configured later, so the changes will be effective during runtime. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 5f6c21d4b59..d7167932113 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -173,6 +173,9 @@ void omap_mcbsp_config(struct omap_mcbsp *mcbsp, MCBSP_WRITE(mcbsp, XCCR, config->xccr); MCBSP_WRITE(mcbsp, RCCR, config->rccr); } + /* Enable wakeup behavior */ + if (mcbsp->pdata->has_wakeup) + MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); } /** @@ -479,10 +482,6 @@ int omap_mcbsp_request(struct omap_mcbsp *mcbsp) if (mcbsp->pdata && mcbsp->pdata->ops && mcbsp->pdata->ops->request) mcbsp->pdata->ops->request(mcbsp->id - 1); - /* Enable wakeup behavior */ - if (mcbsp->pdata->has_wakeup) - MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN); - /* * Make sure that transmitter, receiver and sample-rate generator are * not running before activating IRQs. -- cgit v1.2.3-18-g5258 From e386615c01d37145aa27fd06d1f8de26f1acbb7f Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 5 Mar 2012 11:32:27 +0200 Subject: ASoC: omap-mcbsp: When closing the port select PRCM source for CLKS signal If external source for the CLKS signal selection kept after the port is no longer in use the system might refuse to go suspend. There is also a chance that the external clock is not running when next time the McBSP port is started which can result errors when we try to access McBSP registers. Reset the CLKS source back to PRCM source unconditionally. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Tested-by: Grazvydas Ignotas Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index d7167932113..21dbb0532bc 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -548,6 +548,16 @@ void omap_mcbsp_free(struct omap_mcbsp *mcbsp) reg_cache = mcbsp->reg_cache; + /* + * Select CLKS source from internal source unconditionally before + * marking the McBSP port as free. + * If the external clock source via MCBSP_CLKS pin has been selected the + * system will refuse to enter idle if the CLKS pin source is not reset + * back to internal source. + */ + if (!cpu_class_is_omap1()) + omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC); + spin_lock(&mcbsp->lock); if (mcbsp->free) dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id); -- cgit v1.2.3-18-g5258 From 73c9522e76d7147d99ff859699405a9af81fec72 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 7 Mar 2012 11:15:37 +0200 Subject: ASoC: omap McBSP: Clear rx_irq at probe time for OMAP4 On OMAP4 we have one interrupt line per McBSP port. At probe time tx, and rx irq value will be -ENXIO, and only the tx irq will get corrected. In omap_mcbsp_request if the rx_irq is not 0 we proceed, and try to request the interrupt, which will fail on OMAP4 (rx_irq == -6). To avoid this error, clear the rx_irq at probe time on OMAP4. Signed-off-by: Peter Ujfalusi Reviewed-by: Paul Menzel Acked-by: Mark Brown Acked-by: Jarkko Nikula Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 21dbb0532bc..c3e31deafa0 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -958,8 +958,10 @@ int __devinit omap_mcbsp_init(struct platform_device *pdev) mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); /* From OMAP4 there will be a single irq line */ - if (mcbsp->tx_irq == -ENXIO) + if (mcbsp->tx_irq == -ENXIO) { mcbsp->tx_irq = platform_get_irq(pdev, 0); + mcbsp->rx_irq = 0; + } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); if (!res) { -- cgit v1.2.3-18-g5258 From 33cec399048545c64d9b9a1368b968acee8acb35 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 8 Mar 2012 10:40:08 +0200 Subject: ARM/ASoC: OMAP McBSP: Move remainig defines from arch to ASoC header Clock signal muxing, and functional clock related defines are only needed in ASoC drivers. Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index a5518d71913..acc94700f5b 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -230,6 +230,18 @@ enum { #define XRDYEN BIT(10) #define XEMPTYEOFEN BIT(14) +/* CLKR signal muxing options */ +#define CLKR_SRC_CLKR 0 +#define CLKR_SRC_CLKX 1 + +/* FSR signal muxing options */ +#define FSR_SRC_FSR 0 +#define FSR_SRC_FSX 1 + +/* McBSP functional clock sources */ +#define MCBSP_CLKS_PRCM_SRC 0 +#define MCBSP_CLKS_PAD_SRC 1 + /* we don't do multichannel for now */ struct omap_mcbsp_reg_cfg { u16 spcr2; -- cgit v1.2.3-18-g5258 From cd1f08c7f64ce2093877ecafd21ee784c8ca2389 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 8 Mar 2012 11:01:37 +0200 Subject: ASoC: omap-mcbsp: Single function CLKR/FSR source mux configuration Use single function for the CLKR/FSR mux configuration. OMAP2/3 has 6 pin configuration on McBSP1 instance, while on OMAP4 McBSP4 instance have the 6 pin configuration so the omap2_mcbsp1_mux_* is not correct name for all support OMAP versions Signed-off-by: Peter Ujfalusi Acked-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 48 +++++++++++++++++++++------------------------ sound/soc/omap/mcbsp.h | 15 ++++++-------- sound/soc/omap/omap-mcbsp.c | 8 ++++---- 3 files changed, 32 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index c3e31deafa0..95413a16808 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -687,40 +687,36 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) return -EINVAL; } -void omap2_mcbsp1_mux_clkr_src(struct omap_mcbsp *mcbsp, u8 mux) +int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { - const char *src; - - if (mcbsp->id != 1) - return; + const char *signal, *src; + int ret = 0; - if (mux == CLKR_SRC_CLKR) + switch (mux) { + case CLKR_SRC_CLKR: + signal = "clkr"; src = "clkr"; - else if (mux == CLKR_SRC_CLKX) + break; + case CLKR_SRC_CLKX: + signal = "clkr"; src = "clkx"; - else - return; - - if (mcbsp->pdata->mux_signal) - mcbsp->pdata->mux_signal(mcbsp->dev, "clkr", src); -} - -void omap2_mcbsp1_mux_fsr_src(struct omap_mcbsp *mcbsp, u8 mux) -{ - const char *src; - - if (mcbsp->id != 1) - return; - - if (mux == FSR_SRC_FSR) + break; + case FSR_SRC_FSR: + signal = "fsr"; src = "fsr"; - else if (mux == FSR_SRC_FSX) + break; + case FSR_SRC_FSX: + signal = "fsr"; src = "fsx"; - else - return; + break; + default: + return -EINVAL; + } if (mcbsp->pdata->mux_signal) - mcbsp->pdata->mux_signal(mcbsp->dev, "fsr", src); + ret = mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); + + return ret; } #define max_thres(m) (mcbsp->pdata->buffer_size) diff --git a/sound/soc/omap/mcbsp.h b/sound/soc/omap/mcbsp.h index acc94700f5b..a944fcc9073 100644 --- a/sound/soc/omap/mcbsp.h +++ b/sound/soc/omap/mcbsp.h @@ -230,13 +230,11 @@ enum { #define XRDYEN BIT(10) #define XEMPTYEOFEN BIT(14) -/* CLKR signal muxing options */ -#define CLKR_SRC_CLKR 0 -#define CLKR_SRC_CLKX 1 - -/* FSR signal muxing options */ -#define FSR_SRC_FSR 0 -#define FSR_SRC_FSX 1 +/* Clock signal muxing options */ +#define CLKR_SRC_CLKR 0 /* CLKR signal is from the CLKR pin */ +#define CLKR_SRC_CLKX 1 /* CLKR signal is from the CLKX pin */ +#define FSR_SRC_FSR 2 /* FSR signal is from the FSR pin */ +#define FSR_SRC_FSX 3 /* FSR signal is from the FSX pin */ /* McBSP functional clock sources */ #define MCBSP_CLKS_PRCM_SRC 0 @@ -333,8 +331,7 @@ void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx); int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id); /* McBSP signal muxing API */ -void omap2_mcbsp1_mux_clkr_src(struct omap_mcbsp *mcbsp, u8 mux); -void omap2_mcbsp1_mux_fsr_src(struct omap_mcbsp *mcbsp, u8 mux); +int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux); /* Sidetone specific API */ int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 10eb645ceee..d8409b00843 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -554,22 +554,22 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_CLKR_SRC_CLKR: if (cpu_class_is_omap1()) break; - omap2_mcbsp1_mux_clkr_src(mcbsp, CLKR_SRC_CLKR); + err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKR); break; case OMAP_MCBSP_CLKR_SRC_CLKX: if (cpu_class_is_omap1()) break; - omap2_mcbsp1_mux_clkr_src(mcbsp, CLKR_SRC_CLKX); + err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKX); break; case OMAP_MCBSP_FSR_SRC_FSR: if (cpu_class_is_omap1()) break; - omap2_mcbsp1_mux_fsr_src(mcbsp, FSR_SRC_FSR); + err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSR); break; case OMAP_MCBSP_FSR_SRC_FSX: if (cpu_class_is_omap1()) break; - omap2_mcbsp1_mux_fsr_src(mcbsp, FSR_SRC_FSX); + err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSX); break; default: err = -ENODEV; -- cgit v1.2.3-18-g5258 From 5788c62e72b8484836ae6587c7fb65757a777a3a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 8 Mar 2012 13:34:16 +0200 Subject: ASoC: omap-mcbsp: Correct clock muxing for CLKR/FSR signals Remove the no longer valid check for McBSP1 regarding to signal mux selection (on OMAP4 McBSP4 has 6 pin setup). Only clear the srgr2, pcr0 register configuration if the requested clock configuration will actually going to touch it. In this way we can avoid issues if the CLKR/FSR mux has been configured after the clock selection. We are going to check for the valid McBSP port in the omap_mcbsp_6pin_src_mux() function based on the validity of the mux_signal callback (which is only valid for ports having 6 pin setup). Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/mcbsp.c | 9 ++++----- sound/soc/omap/omap-mcbsp.c | 34 +++++++++++++++------------------- 2 files changed, 19 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 95413a16808..e5f44440d1b 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -690,7 +690,9 @@ int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id) int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - int ret = 0; + + if (mcbsp->pdata->mux_signal) + return -EINVAL; switch (mux) { case CLKR_SRC_CLKR: @@ -713,10 +715,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) return -EINVAL; } - if (mcbsp->pdata->mux_signal) - ret = mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); - - return ret; + return mcbsp->pdata->mux_signal(mcbsp->dev, signal, src); } #define max_thres(m) (mcbsp->pdata->buffer_size) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index d8409b00843..626e2d6db20 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -511,17 +511,21 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return -EBUSY; } - /* The McBSP signal muxing functions are only available on McBSP1 */ - if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || - clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || - clk_id == OMAP_MCBSP_FSR_SRC_FSR || - clk_id == OMAP_MCBSP_FSR_SRC_FSX) - if (cpu_class_is_omap1() || cpu_dai->id != 1) - return -EINVAL; - - mcbsp->in_freq = freq; - regs->srgr2 &= ~CLKSM; - regs->pcr0 &= ~SCLKME; + if (clk_id == OMAP_MCBSP_SYSCLK_CLK || + clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK || + clk_id == OMAP_MCBSP_SYSCLK_CLKS_EXT || + clk_id == OMAP_MCBSP_SYSCLK_CLKX_EXT || + clk_id == OMAP_MCBSP_SYSCLK_CLKR_EXT) { + mcbsp->in_freq = freq; + regs->srgr2 &= ~CLKSM; + regs->pcr0 &= ~SCLKME; + } else if (cpu_class_is_omap1()) { + /* + * McBSP CLKR/FSR signal muxing functions are only available on + * OMAP2 or newer versions + */ + return -EINVAL; + } switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: @@ -552,23 +556,15 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_CLKR_SRC_CLKR: - if (cpu_class_is_omap1()) - break; err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKR); break; case OMAP_MCBSP_CLKR_SRC_CLKX: - if (cpu_class_is_omap1()) - break; err = omap_mcbsp_6pin_src_mux(mcbsp, CLKR_SRC_CLKX); break; case OMAP_MCBSP_FSR_SRC_FSR: - if (cpu_class_is_omap1()) - break; err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSR); break; case OMAP_MCBSP_FSR_SRC_FSX: - if (cpu_class_is_omap1()) - break; err = omap_mcbsp_6pin_src_mux(mcbsp, FSR_SRC_FSX); break; default: -- cgit v1.2.3-18-g5258 From 94a504c2e059fb88f05ede6d614504779275b099 Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Fri, 9 Mar 2012 01:19:15 +0200 Subject: ASoC: omap-mcbsp: fix snd_pcm_hw_rule_add arguments We are setting SNDRV_PCM_HW_PARAM_BUFFER_SIZE based on SNDRV_PCM_HW_PARAM_CHANNELS, not vice versa. This bug didn't have much impact because the rules are evaluated multiple times by the core, and intended value got set eventually. Signed-off-by: Grazvydas Ignotas Acked-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-mcbsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 626e2d6db20..6912ac7cb62 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -142,10 +142,10 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * smaller buffer than the FIFO size to avoid underruns */ snd_pcm_hw_rule_add(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, omap_mcbsp_hwrule_min_buffersize, mcbsp, - SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); + SNDRV_PCM_HW_PARAM_CHANNELS, -1); /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, -- cgit v1.2.3-18-g5258 From d0f47ff17f29740eabbd64e11705b7332241714c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 9 Mar 2012 10:26:03 +0200 Subject: ASoC: OMAP: Build config cleanup for McBSP The McBSP driver stack has been moved, and rewritten resulting a single driver - selected by CONFIG_SND_OMAP_SOC_MCBSP. There is no longer need to have CONFIG_OMAP_MCBSP anymore. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/Kconfig | 6 ------ sound/soc/omap/Makefile | 3 +-- 2 files changed, 1 insertion(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 27a3a29f7cd..e00dd0b1139 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -5,13 +5,8 @@ config SND_OMAP_SOC config SND_OMAP_SOC_DMIC tristate -config OMAP_MCBSP - tristate - depends on ARCH_OMAP - config SND_OMAP_SOC_MCBSP tristate - select OMAP_MCBSP config SND_OMAP_SOC_MCPDM tristate @@ -31,7 +26,6 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" depends on SND_OMAP_SOC && MACH_NOKIA_RX51 - select OMAP_MCBSP select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 9f8fbd554eb..1d656bce01d 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -1,13 +1,12 @@ # OMAP Platform Support snd-soc-omap-objs := omap-pcm.o snd-soc-omap-dmic-objs := omap-dmic.o -snd-soc-omap-mcbsp-objs := omap-mcbsp.o +snd-soc-omap-mcbsp-objs := omap-mcbsp.o mcbsp.o snd-soc-omap-mcpdm-objs := omap-mcpdm.o snd-soc-omap-hdmi-objs := omap-hdmi.o obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o obj-$(CONFIG_SND_OMAP_SOC_DMIC) += snd-soc-omap-dmic.o -obj-$(CONFIG_OMAP_MCBSP) += mcbsp.o obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o obj-$(CONFIG_SND_OMAP_SOC_HDMI) += snd-soc-omap-hdmi.o -- cgit v1.2.3-18-g5258 From 29e5853d618282d8277ce8a8304f7424eb60deb5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Mar 2012 12:25:03 +0100 Subject: ALSA: hda - Return the created kcontrol in __snd_hda_add_vmaster() It'll be used for adding hooks in later patches. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 9 ++++++++- sound/pci/hda/hda_local.h | 7 ++++--- sound/pci/hda/patch_analog.c | 2 +- 3 files changed, 13 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0c0ac0e1d50..b79ee344465 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2399,6 +2399,7 @@ static int init_slave_unmute(void *data, struct snd_kcontrol *slave) * @slaves: slave control names (optional) * @suffix: suffix string to each slave name (optional) * @init_slave_vol: initialize slaves to unmute/0dB + * @ctl_ret: store the vmaster kcontrol in return * * Create a virtual master control with the given name. The TLV data * must be either NULL or a valid data. @@ -2411,11 +2412,15 @@ static int init_slave_unmute(void *data, struct snd_kcontrol *slave) */ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char * const *slaves, - const char *suffix, bool init_slave_vol) + const char *suffix, bool init_slave_vol, + struct snd_kcontrol **ctl_ret) { struct snd_kcontrol *kctl; int err; + if (ctl_ret) + *ctl_ret = NULL; + err = map_slaves(codec, slaves, suffix, check_slave_present, NULL); if (err != 1) { snd_printdd("No slave found for %s\n", name); @@ -2439,6 +2444,8 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, map_slaves(codec, slaves, suffix, tlv ? init_slave_0dB : init_slave_unmute, kctl); + if (ctl_ret) + *ctl_ret = kctl; return 0; } EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index caa64686267..c3ee4ede448 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -140,10 +140,11 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, - unsigned int *tlv, const char * const *slaves, - const char *suffix, bool init_slave_vol); + unsigned int *tlv, const char * const *slaves, + const char *suffix, bool init_slave_vol, + struct snd_kcontrol **ctl_ret); #define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \ - __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true) + __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL) int snd_hda_codec_reset(struct hda_codec *codec); /* amp value bits */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index fa97a0c5ced..7143393927d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -229,7 +229,7 @@ static int ad198x_build_controls(struct hda_codec *codec) (spec->slave_vols ? spec->slave_vols : ad_slave_pfxs), "Playback Volume", - !spec->avoid_init_slave_vol); + !spec->avoid_init_slave_vol, NULL); if (err < 0) return err; } -- cgit v1.2.3-18-g5258 From 2faa3bf15ba69fa12bc53926b88982b3875abb3f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Mar 2012 12:30:22 +0100 Subject: ALSA: hda - Rewrite the mute-LED hook with vmaster hook in patch_sigmatel.c The mute-LED is controlled in patch_sigmatel.c by (ab-)using the powersave hook. This can be now rewritten with the vmaster hook instead, which is much simpler and can work even without CONFIG_SND_HDA_POWER_SAVE kconfig. A drawback is that the mute-LED corresponds _only_ to the Master mixer switch instead of checking the whole DACs. But usually this shouldn't be a big problem as PA enables the mixer elements accordingly. Also, this patch changes the code to create vmaster always even on STAC9200 and STAC925x. The former "Master" on these chips are renamed as "PCM" now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 136 +++++++++++++++-------------------------- 1 file changed, 49 insertions(+), 87 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 5988dbdedc4..6e926497b23 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -310,6 +310,8 @@ struct sigmatel_spec { unsigned long auto_capvols[MAX_ADCS_NUM]; unsigned auto_dmic_cnt; hda_nid_t auto_dmic_nids[MAX_DMICS_NUM]; + + struct snd_kcontrol *vmaster_sw_kctl; }; static const hda_nid_t stac9200_adc_nids[1] = { @@ -1007,8 +1009,8 @@ static const struct hda_verb stac9205_core_init[] = { } static const struct snd_kcontrol_new stac9200_mixer[] = { - HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xb, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xb, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0xb, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), { } /* end */ @@ -1035,8 +1037,8 @@ static const struct snd_kcontrol_new stac92hd71bxx_loopback[] = { }; static const struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xe, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x0e, 0, HDA_OUTPUT), { } /* end */ }; @@ -1074,11 +1076,19 @@ static const char * const slave_pfxs[] = { NULL }; +static void stac92xx_update_led_status(struct hda_codec *codec, int enabled); + +static void stac92xx_vmaster_hook(void *private_data, int val) +{ + stac92xx_update_led_status(private_data, val); +} + static void stac92xx_free_kctls(struct hda_codec *codec); static int stac92xx_build_controls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + unsigned int vmaster_tlv[4]; int err; int i; @@ -1135,26 +1145,29 @@ static int stac92xx_build_controls(struct hda_codec *codec) } /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { - unsigned int vmaster_tlv[4]; - snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], - HDA_OUTPUT, vmaster_tlv); - /* correct volume offset */ - vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; - /* minimum value is actually mute */ - vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, slave_pfxs, - "Playback Volume"); - if (err < 0) - return err; - } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, - "Playback Switch"); - if (err < 0) - return err; + snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], + HDA_OUTPUT, vmaster_tlv); + /* correct volume offset */ + vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; + /* minimum value is actually mute */ + vmaster_tlv[3] |= TLV_DB_SCALE_MUTE; + err = snd_hda_add_vmaster(codec, "Master Playback Volume", + vmaster_tlv, slave_pfxs, + "Playback Volume"); + if (err < 0) + return err; + + err = __snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, slave_pfxs, + "Playback Switch", true, + &spec->vmaster_sw_kctl); + if (err < 0) + return err; + + if (spec->gpio_led) { + snd_ctl_add_vmaster_hook(spec->vmaster_sw_kctl, + stac92xx_vmaster_hook, codec); + snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); } if (spec->aloopback_ctl && @@ -4419,8 +4432,7 @@ static int stac92xx_init(struct hda_codec *codec) snd_hda_jack_report_sync(codec); /* sync mute LED */ - if (spec->gpio_led) - hda_call_check_power_status(codec, 0x01); + snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); if (spec->dac_list) stac92xx_power_down(codec); return 0; @@ -5033,83 +5045,37 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, afg_power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state, true); } +#endif /* CONFIG_SND_HDA_POWER_SAVE */ +#endif /* CONFIG_PM */ -/* - * For this feature CONFIG_SND_HDA_POWER_SAVE is needed - * as mute LED state is updated in check_power_status hook - */ -static int stac92xx_update_led_status(struct hda_codec *codec) +/* update mute-LED accoring to the master switch */ +static void stac92xx_update_led_status(struct hda_codec *codec, int enabled) { struct sigmatel_spec *spec = codec->spec; - int i, num_ext_dacs, muted = 1; - unsigned int muted_lvl, notmtd_lvl; - hda_nid_t nid; + int muted = !enabled; if (!spec->gpio_led) - return 0; + return; + + /* LED state is inverted on these systems */ + if (spec->gpio_led_polarity) + muted = !muted; - for (i = 0; i < spec->multiout.num_dacs; i++) { - nid = spec->multiout.dac_nids[i]; - if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & - HDA_AMP_MUTE)) { - muted = 0; /* something heard */ - break; - } - } - if (muted && spec->multiout.hp_nid) - if (!(snd_hda_codec_amp_read(codec, - spec->multiout.hp_nid, 0, HDA_OUTPUT, 0) & - HDA_AMP_MUTE)) { - muted = 0; /* HP is not muted */ - } - num_ext_dacs = ARRAY_SIZE(spec->multiout.extra_out_nid); - for (i = 0; muted && i < num_ext_dacs; i++) { - nid = spec->multiout.extra_out_nid[i]; - if (nid == 0) - break; - if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & - HDA_AMP_MUTE)) { - muted = 0; /* extra output is not muted */ - } - } /*polarity defines *not* muted state level*/ if (!spec->vref_mute_led_nid) { if (muted) spec->gpio_data &= ~spec->gpio_led; /* orange */ else spec->gpio_data |= spec->gpio_led; /* white */ - - if (!spec->gpio_led_polarity) { - /* LED state is inverted on these systems */ - spec->gpio_data ^= spec->gpio_led; - } stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); } else { - notmtd_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; - muted_lvl = spec->gpio_led_polarity ? - AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_50; - spec->vref_led = muted ? muted_lvl : notmtd_lvl; + spec->vref_led = muted ? AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD; stac_vrefout_set(codec, spec->vref_mute_led_nid, spec->vref_led); } - return 0; } -/* - * use power check for controlling mute led of HP notebooks - */ -static int stac92xx_check_power_status(struct hda_codec *codec, - hda_nid_t nid) -{ - stac92xx_update_led_status(codec); - - return 0; -} -#endif /* CONFIG_SND_HDA_POWER_SAVE */ -#endif /* CONFIG_PM */ - static const struct hda_codec_ops stac92xx_patch_ops = { .build_controls = stac92xx_build_controls, .build_pcms = stac92xx_build_pcms, @@ -5627,8 +5593,6 @@ again: stac92xx_set_power_state; } codec->patch_ops.pre_resume = stac92xx_pre_resume; - codec->patch_ops.check_power_status = - stac92xx_check_power_status; } #endif @@ -5938,8 +5902,6 @@ again: stac92xx_set_power_state; } codec->patch_ops.pre_resume = stac92xx_pre_resume; - codec->patch_ops.check_power_status = - stac92xx_check_power_status; } #endif -- cgit v1.2.3-18-g5258 From 420b0febe54099ea9003bddad0a81e882a8472af Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Mar 2012 12:35:27 +0100 Subject: ALSA: hda - Rewrite the mute-LED control with vmaster hook for ALC269 We've had ugly static handling of the mute-LED with a powersave hook for ALC269 HP laptops just like done in patch_sigmatel.c. This is now rewritten with the new vmaster hook and a fixup code. For that, the new fixup action, ALC_FIXUP_ACT_BUILD, is introduced. It's called after build_controls is called. The reason of this new action is that vmaster hook must be added at this stage (not in init or probe). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 82 +++++++++++++++++++++++-------------------- 1 file changed, 44 insertions(+), 38 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1de0c1629ba..901547216c4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -198,6 +198,7 @@ struct alc_spec { /* for virtual master */ hda_nid_t vmaster_nid; + struct snd_kcontrol *vmaster_sw_kctl; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; int num_loopbacks; @@ -1441,6 +1442,7 @@ enum { ALC_FIXUP_ACT_PRE_PROBE, ALC_FIXUP_ACT_PROBE, ALC_FIXUP_ACT_INIT, + ALC_FIXUP_ACT_BUILD, }; static void alc_apply_fixup(struct hda_codec *codec, int action) @@ -1955,9 +1957,10 @@ static int __alc_build_controls(struct hda_codec *codec) } if (!spec->no_analog && !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, alc_slave_pfxs, - "Playback Switch"); + err = __snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, alc_slave_pfxs, + "Playback Switch", + true, &spec->vmaster_sw_kctl); if (err < 0) return err; } @@ -2042,7 +2045,11 @@ static int alc_build_controls(struct hda_codec *codec) int err = __alc_build_controls(codec); if (err < 0) return err; - return snd_hda_jack_add_kctls(codec, &spec->autocfg); + err = snd_hda_jack_add_kctls(codec, &spec->autocfg); + if (err < 0) + return err; + alc_apply_fixup(codec, ALC_FIXUP_ACT_BUILD); + return 0; } @@ -5721,35 +5728,6 @@ static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = { /* NID is set in alc_build_pcms */ }; -#ifdef CONFIG_SND_HDA_POWER_SAVE -static int alc269_mic2_for_mute_led(struct hda_codec *codec) -{ - switch (codec->subsystem_id) { - case 0x103c1586: - return 1; - } - return 0; -} - -static int alc269_mic2_mute_check_ps(struct hda_codec *codec, hda_nid_t nid) -{ - /* update mute-LED according to the speaker mute state */ - if (nid == 0x01 || nid == 0x14) { - int pinval; - if (snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0) & - HDA_AMP_MUTE) - pinval = 0x24; - else - pinval = 0x20; - /* mic2 vref pin is used for mute LED control */ - snd_hda_codec_update_cache(codec, 0x19, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinval); - } - return alc_check_power_status(codec, nid); -} -#endif /* CONFIG_SND_HDA_POWER_SAVE */ - /* different alc269-variants */ enum { ALC269_TYPE_ALC269VA, @@ -5900,6 +5878,33 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } +/* update mute-LED according to the speaker mute state via mic2 VREF pin */ +static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = enabled ? 0x20 : 0x24; + snd_hda_codec_update_cache(codec, 0x19, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pinval); +} + +static void alc269_fixup_mic2_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + if (!spec->vmaster_sw_kctl) + return; + snd_ctl_add_vmaster_hook(spec->vmaster_sw_kctl, + alc269_fixup_mic2_mute_hook, codec); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); + break; + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -5917,6 +5922,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC2_MUTE_LED, }; static const struct alc_fixup alc269_fixups[] = { @@ -6037,9 +6043,14 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC2_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic2_mute, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), @@ -6231,11 +6242,6 @@ static int patch_alc269(struct hda_codec *codec) #endif spec->shutup = alc269_shutup; -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (alc269_mic2_for_mute_led(codec)) - codec->patch_ops.check_power_status = alc269_mic2_mute_check_ps; -#endif - alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE); return 0; -- cgit v1.2.3-18-g5258 From 527c73bada6f02a35983ddb34db3a0fd4360c88c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Mar 2012 12:38:51 +0100 Subject: ALSA: hda - Add EAPD control to Conexnat auto-parser Added the vmaster hook for controlling EAPD dynamically to Conexant auto-parser. When the Master is muted, EAPDs are turned off as well. This will fix the missing mute-LED control on some machines in addition to the more power-saving in the auto-parser mode. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 36 ++++++++++++++++++++++++++++++++---- 1 file changed, 32 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5a56fda8362..f1c9aed9fa6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -70,6 +70,8 @@ struct conexant_spec { const struct snd_kcontrol_new *mixers[5]; int num_mixers; hda_nid_t vmaster_nid; + struct snd_kcontrol *vmaster_sw_kctl; + void (*vmaster_hook)(struct snd_kcontrol *, int); const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -513,9 +515,10 @@ static int conexant_build_controls(struct hda_codec *codec) } if (spec->vmaster_nid && !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, slave_pfxs, - "Playback Switch"); + err = __snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, slave_pfxs, + "Playback Switch", true, + &spec->vmaster_sw_kctl); if (err < 0) return err; } @@ -3975,6 +3978,19 @@ static void clear_unsol_on_unused_pins(struct hda_codec *codec) } } +/* turn on/off EAPD according to Master switch */ +static void cx_auto_vmaster_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct conexant_spec *spec = codec->spec; + + if (enabled && spec->pin_eapd_ctrls) { + cx_auto_update_speakers(codec); + return; + } + cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, enabled); +} + static void cx_auto_init_output(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -4079,11 +4095,13 @@ static void cx_auto_init_digital(struct hda_codec *codec) static int cx_auto_init(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; /*snd_hda_sequence_write(codec, cx_auto_init_verbs);*/ cx_auto_init_output(codec); cx_auto_init_input(codec); cx_auto_init_digital(codec); snd_hda_jack_report_sync(codec); + snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); return 0; } @@ -4329,6 +4347,11 @@ static int cx_auto_build_controls(struct hda_codec *codec) err = snd_hda_jack_add_kctls(codec, &spec->autocfg); if (err < 0) return err; + if (spec->vmaster_hook && spec->vmaster_sw_kctl) { + snd_ctl_add_vmaster_hook(spec->vmaster_sw_kctl, + spec->vmaster_hook, codec); + snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); + } return 0; } @@ -4353,7 +4376,6 @@ static int cx_auto_search_adcs(struct hda_codec *codec) return 0; } - static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = conexant_build_pcms, @@ -4455,6 +4477,12 @@ static int patch_conexant_auto(struct hda_codec *codec) apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); + /* add EAPD vmaster hook to all HP machines */ + /* NOTE: this should be applied via fixup once when the generic + * fixup code is merged to hda_codec.c + */ + spec->vmaster_hook = cx_auto_vmaster_hook; + err = cx_auto_search_adcs(codec); if (err < 0) return err; -- cgit v1.2.3-18-g5258 From c986564b3115ebd24a907515ac0b7ca2bef794f9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Mar 2012 16:31:50 +0000 Subject: ASoC: wm8994: Prevent ABBA deadlock with CODEC and accdet mutexes Currently we can the accdet mutex from within DAPM when updating the device state which means we take accdet then the CODEC mutex but we also do the locking the other way around when responding to the jackdet IRQ. Move all the jackdet use of the CODEC mutex out of the accdet lock to avoid this. Since all the DAPM interactions depend only on a single threaded IRQ this is still serialised. The locking improvements in 3.5 allow a better solution there. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 69 +++++++++++++++++++++++------------------------ 1 file changed, 34 insertions(+), 35 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index bc12d097ef0..15fcb1bb714 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3166,9 +3166,16 @@ static void wm8958_default_micdet(u16 status, void *data) /* If we have jackdet that will detect removal */ if (wm8994->jackdet) { + mutex_lock(&wm8994->accdet_lock); + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); + wm1811_jackdet_set_mode(codec, + WM1811_JACKDET_MODE_JACK); + + mutex_unlock(&wm8994->accdet_lock); + if (wm8994->pdata->jd_ext_cap) { mutex_lock(&codec->mutex); snd_soc_dapm_disable_pin(&codec->dapm, @@ -3176,9 +3183,6 @@ static void wm8958_default_micdet(u16 status, void *data) snd_soc_dapm_sync(&codec->dapm); mutex_unlock(&codec->mutex); } - - wm1811_jackdet_set_mode(codec, - WM1811_JACKDET_MODE_JACK); } } @@ -3213,6 +3217,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) struct wm8994_priv *wm8994 = data; struct snd_soc_codec *codec = wm8994->codec; int reg; + bool present; mutex_lock(&wm8994->accdet_lock); @@ -3225,11 +3230,10 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) dev_dbg(codec->dev, "JACKDET %x\n", reg); - if (reg & WM1811_JACKDET_LVL) { - dev_dbg(codec->dev, "Jack detected\n"); + present = reg & WM1811_JACKDET_LVL; - snd_soc_jack_report(wm8994->micdet[0].jack, - SND_JACK_MECHANICAL, SND_JACK_MECHANICAL); + if (present) { + dev_dbg(codec->dev, "Jack detected\n"); snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, 0); @@ -3247,32 +3251,12 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, WM8958_MICD_ENA); - - /* If required for an external cap force MICBIAS on */ - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } } else { dev_dbg(codec->dev, "Jack not detected\n"); snd_soc_update_bits(codec, WM8958_MICBIAS2, WM8958_MICB2_DISCH, WM8958_MICB2_DISCH); - if (wm8994->pdata->jd_ext_cap) { - mutex_lock(&codec->mutex); - snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); - snd_soc_dapm_sync(&codec->dapm); - mutex_unlock(&codec->mutex); - } - - snd_soc_jack_report(wm8994->micdet[0].jack, 0, - SND_JACK_MECHANICAL | SND_JACK_HEADSET | - wm8994->btn_mask); - /* Enable debounce while removed */ snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, WM1811_JACKDET_DB, WM1811_JACKDET_DB); @@ -3286,6 +3270,28 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) mutex_unlock(&wm8994->accdet_lock); + /* If required for an external cap force MICBIAS on */ + if (wm8994->pdata->jd_ext_cap) { + mutex_lock(&codec->mutex); + + if (present) + snd_soc_dapm_force_enable_pin(&codec->dapm, + "MICBIAS2"); + else + snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2"); + + snd_soc_dapm_sync(&codec->dapm); + mutex_unlock(&codec->mutex); + } + + if (present) + snd_soc_jack_report(wm8994->micdet[0].jack, + SND_JACK_MECHANICAL, SND_JACK_MECHANICAL); + else + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_MECHANICAL | SND_JACK_HEADSET | + wm8994->btn_mask); + return IRQ_HANDLED; } @@ -3389,17 +3395,13 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) struct snd_soc_codec *codec = wm8994->codec; int reg, count; - mutex_lock(&wm8994->accdet_lock); - /* * Jack detection may have detected a removal simulataneously * with an update of the MICDET status; if so it will have * stopped detection and we can ignore this interrupt. */ - if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) { - mutex_unlock(&wm8994->accdet_lock); + if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA)) return IRQ_HANDLED; - } /* We may occasionally read a detection without an impedence * range being provided - if that happens loop again. @@ -3408,7 +3410,6 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) do { reg = snd_soc_read(codec, WM8958_MIC_DETECT_3); if (reg < 0) { - mutex_unlock(&wm8994->accdet_lock); dev_err(codec->dev, "Failed to read mic detect status: %d\n", reg); @@ -3439,8 +3440,6 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) dev_warn(codec->dev, "Accessory detection with no callback\n"); out: - mutex_unlock(&wm8994->accdet_lock); - return IRQ_HANDLED; } -- cgit v1.2.3-18-g5258 From 7a08cf7022b3f6863cff004cf0531f3e44a772ea Mon Sep 17 00:00:00 2001 From: Mika Westerberg Date: Mon, 5 Mar 2012 14:02:14 +0100 Subject: ASoC: dmaengine_pcm: Reset pointer position when starting a stream Otherwise a wrong position will be reported after restarting a stream and the first few samples might be skipped. Signed-off-by: Mika Westerberg Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dmaengine-pcm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 0526cf82b54..4420b7030c8 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -142,6 +142,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) direction = snd_pcm_substream_to_dma_direction(substream); + prtd->pos = 0; desc = chan->device->device_prep_dma_cyclic(chan, substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), -- cgit v1.2.3-18-g5258 From d7a42e1033b27cea8ae137eeaa038910fe334a55 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 5 Mar 2012 14:02:15 +0100 Subject: ASoC: ep93xx-pcm: Use dmaengine PCM helper functions Signed-off-by: Lars-Peter Clausen Acked-by: Vinod Koul Tested-by: Mika Westerberg Signed-off-by: Mark Brown --- sound/soc/ep93xx/Kconfig | 1 + sound/soc/ep93xx/ep93xx-pcm.c | 148 +++++++----------------------------------- 2 files changed, 23 insertions(+), 126 deletions(-) (limited to 'sound') diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig index 91a28de9410..88143db7e75 100644 --- a/sound/soc/ep93xx/Kconfig +++ b/sound/soc/ep93xx/Kconfig @@ -1,6 +1,7 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" depends on ARCH_EP93XX && SND_SOC + select SND_SOC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the EP93xx I2S or AC97 interfaces. diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 32adca38b48..162dbb74f4c 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include @@ -52,26 +53,6 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .fifo_size = 32, }; -struct ep93xx_runtime_data -{ - int pointer_bytes; - int periods; - int period_bytes; - struct dma_chan *dma_chan; - struct ep93xx_dma_data dma_data; -}; - -static void ep93xx_pcm_dma_callback(void *data) -{ - struct snd_pcm_substream *substream = data; - struct ep93xx_runtime_data *rtd = substream->runtime->private_data; - - rtd->pointer_bytes += rtd->period_bytes; - rtd->pointer_bytes %= rtd->period_bytes * rtd->periods; - - snd_pcm_period_elapsed(substream); -} - static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) { struct ep93xx_dma_data *data = filter_param; @@ -86,98 +67,48 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) static int ep93xx_pcm_open(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = soc_rtd->cpu_dai; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct ep93xx_pcm_dma_params *dma_params; - struct ep93xx_runtime_data *rtd; - dma_cap_mask_t mask; + struct ep93xx_dma_data *dma_data; int ret; - ret = snd_pcm_hw_constraint_integer(substream->runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - return ret; - snd_soc_set_runtime_hwparams(substream, &ep93xx_pcm_hardware); - rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); - if (!rtd) + dma_data = kmalloc(sizeof(*dma_data), GFP_KERNEL); + if (!dma_data) return -ENOMEM; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - dma_cap_set(DMA_CYCLIC, mask); - dma_params = snd_soc_dai_get_dma_data(cpu_dai, substream); - rtd->dma_data.port = dma_params->dma_port; - rtd->dma_data.name = dma_params->name; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->dma_data.direction = DMA_MEM_TO_DEV; - else - rtd->dma_data.direction = DMA_DEV_TO_MEM; - - rtd->dma_chan = dma_request_channel(mask, ep93xx_pcm_dma_filter, - &rtd->dma_data); - if (!rtd->dma_chan) { - kfree(rtd); - return -EINVAL; - } - - substream->runtime->private_data = rtd; - return 0; -} + dma_data->port = dma_params->dma_port; + dma_data->name = dma_params->name; + dma_data->direction = snd_pcm_substream_to_dma_direction(substream); -static int ep93xx_pcm_close(struct snd_pcm_substream *substream) -{ - struct ep93xx_runtime_data *rtd = substream->runtime->private_data; + ret = snd_dmaengine_pcm_open(substream, ep93xx_pcm_dma_filter, dma_data); + if (ret) { + kfree(dma_data); + return ret; + } - dma_release_channel(rtd->dma_chan); - kfree(rtd); - return 0; -} + snd_dmaengine_pcm_set_data(substream, dma_data); -static int ep93xx_pcm_dma_submit(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct ep93xx_runtime_data *rtd = runtime->private_data; - struct dma_chan *chan = rtd->dma_chan; - struct dma_device *dma_dev = chan->device; - struct dma_async_tx_descriptor *desc; - - rtd->pointer_bytes = 0; - desc = dma_dev->device_prep_dma_cyclic(chan, runtime->dma_addr, - rtd->period_bytes * rtd->periods, - rtd->period_bytes, - rtd->dma_data.direction); - if (!desc) - return -EINVAL; - - desc->callback = ep93xx_pcm_dma_callback; - desc->callback_param = substream; - - dmaengine_submit(desc); return 0; } -static void ep93xx_pcm_dma_flush(struct snd_pcm_substream *substream) +static int ep93xx_pcm_close(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct ep93xx_runtime_data *rtd = runtime->private_data; + struct dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); - dmaengine_terminate_all(rtd->dma_chan); + snd_dmaengine_pcm_close(substream); + kfree(dma_data); + return 0; } static int ep93xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct ep93xx_runtime_data *rtd = runtime->private_data; - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - rtd->periods = params_periods(params); - rtd->period_bytes = params_period_bytes(params); return 0; } @@ -187,41 +118,6 @@ static int ep93xx_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int ep93xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - int ret; - - ret = 0; - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = ep93xx_pcm_dma_submit(substream); - break; - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ep93xx_pcm_dma_flush(substream); - break; - - default: - ret = -EINVAL; - break; - } - - return ret; -} - -static snd_pcm_uframes_t ep93xx_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct ep93xx_runtime_data *rtd = substream->runtime->private_data; - - /* FIXME: implement this with sub-period granularity */ - return bytes_to_frames(runtime, rtd->pointer_bytes); -} - static int ep93xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { @@ -239,8 +135,8 @@ static struct snd_pcm_ops ep93xx_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = ep93xx_pcm_hw_params, .hw_free = ep93xx_pcm_hw_free, - .trigger = ep93xx_pcm_trigger, - .pointer = ep93xx_pcm_pointer, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, .mmap = ep93xx_pcm_mmap, }; -- cgit v1.2.3-18-g5258 From d2f344b5e0a933b5b1d12f863406ee1d63e5bf8e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Mar 2012 16:59:58 +0100 Subject: ALSA: hda - Add "Mute-LED Mode" enum control Create snd_hda_add_vmaster_hook() and snd_hda_sync_vmaster_hook() helper functions to handle the mute-LED in vmaster hook more commonly. In the former function, a new enum control "Mute-LED Mode" is added. This provides user to choose whether the mute-LED should be turned on/off explicitly or to follow the master-mute status. Reviewed-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 94 ++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_local.h | 21 ++++++++++ sound/pci/hda/patch_conexant.c | 17 ++++---- sound/pci/hda/patch_realtek.c | 12 +++--- sound/pci/hda/patch_sigmatel.c | 13 +++--- 5 files changed, 135 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b79ee344465..b981ea9c644 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2450,6 +2450,100 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, } EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster); +/* + * mute-LED control using vmaster + */ +static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[] = { + "Off", "On", "Follow Master" + }; + unsigned int index; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + index = uinfo->value.enumerated.item; + if (index >= 3) + index = 2; + strcpy(uinfo->value.enumerated.name, texts[index]); + return 0; +} + +static int vmaster_mute_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol); + ucontrol->value.enumerated.item[0] = hook->mute_mode; + return 0; +} + +static int vmaster_mute_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol); + unsigned int old_mode = hook->mute_mode; + + hook->mute_mode = ucontrol->value.enumerated.item[0]; + if (hook->mute_mode > HDA_VMUTE_FOLLOW_MASTER) + hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER; + if (old_mode == hook->mute_mode) + return 0; + snd_hda_sync_vmaster_hook(hook); + return 1; +} + +static struct snd_kcontrol_new vmaster_mute_mode = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mute-LED Mode", + .info = vmaster_mute_mode_info, + .get = vmaster_mute_mode_get, + .put = vmaster_mute_mode_put, +}; + +/* + * Add a mute-LED hook with the given vmaster switch kctl + * "Mute-LED Mode" control is automatically created and associated with + * the given hook. + */ +int snd_hda_add_vmaster_hook(struct hda_codec *codec, + struct hda_vmaster_mute_hook *hook) +{ + struct snd_kcontrol *kctl; + + if (!hook->hook || !hook->sw_kctl) + return 0; + snd_ctl_add_vmaster_hook(hook->sw_kctl, hook->hook, codec); + hook->codec = codec; + hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER; + kctl = snd_ctl_new1(&vmaster_mute_mode, hook); + if (!kctl) + return -ENOMEM; + return snd_hda_ctl_add(codec, 0, kctl); +} +EXPORT_SYMBOL_HDA(snd_hda_add_vmaster_hook); + +/* + * Call the hook with the current value for synchronization + * Should be called in init callback + */ +void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook) +{ + if (!hook->hook || !hook->codec) + return; + switch (hook->mute_mode) { + case HDA_VMUTE_FOLLOW_MASTER: + snd_ctl_sync_vmaster_hook(hook->sw_kctl); + break; + default: + hook->hook(hook->codec, hook->mute_mode); + break; + } +} +EXPORT_SYMBOL_HDA(snd_hda_sync_vmaster_hook); + + /** * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch * diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index c3ee4ede448..3f82ab6a058 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -147,6 +147,27 @@ int __snd_hda_add_vmaster(struct hda_codec *codec, char *name, __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL) int snd_hda_codec_reset(struct hda_codec *codec); +enum { + HDA_VMUTE_OFF, + HDA_VMUTE_ON, + HDA_VMUTE_FOLLOW_MASTER, +}; + +struct hda_vmaster_mute_hook { + /* below two fields must be filled by the caller of + * snd_hda_add_vmaster_hook() beforehand + */ + struct snd_kcontrol *sw_kctl; + void (*hook)(void *, int); + /* below are initialized automatically */ + unsigned int mute_mode; /* HDA_VMUTE_XXX */ + struct hda_codec *codec; +}; + +int snd_hda_add_vmaster_hook(struct hda_codec *codec, + struct hda_vmaster_mute_hook *hook); +void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook); + /* amp value bits */ #define HDA_AMP_MUTE 0x80 #define HDA_AMP_UNMUTE 0x00 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f1c9aed9fa6..a21a485a413 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -70,8 +70,7 @@ struct conexant_spec { const struct snd_kcontrol_new *mixers[5]; int num_mixers; hda_nid_t vmaster_nid; - struct snd_kcontrol *vmaster_sw_kctl; - void (*vmaster_hook)(struct snd_kcontrol *, int); + struct hda_vmaster_mute_hook vmaster_mute; const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -518,7 +517,7 @@ static int conexant_build_controls(struct hda_codec *codec) err = __snd_hda_add_vmaster(codec, "Master Playback Switch", NULL, slave_pfxs, "Playback Switch", true, - &spec->vmaster_sw_kctl); + &spec->vmaster_mute.sw_kctl); if (err < 0) return err; } @@ -4101,7 +4100,7 @@ static int cx_auto_init(struct hda_codec *codec) cx_auto_init_input(codec); cx_auto_init_digital(codec); snd_hda_jack_report_sync(codec); - snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); return 0; } @@ -4347,10 +4346,10 @@ static int cx_auto_build_controls(struct hda_codec *codec) err = snd_hda_jack_add_kctls(codec, &spec->autocfg); if (err < 0) return err; - if (spec->vmaster_hook && spec->vmaster_sw_kctl) { - snd_ctl_add_vmaster_hook(spec->vmaster_sw_kctl, - spec->vmaster_hook, codec); - snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); + if (spec->vmaster_mute.hook && spec->vmaster_mute.sw_kctl) { + err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute); + if (err < 0) + return err; } return 0; } @@ -4481,7 +4480,7 @@ static int patch_conexant_auto(struct hda_codec *codec) /* NOTE: this should be applied via fixup once when the generic * fixup code is merged to hda_codec.c */ - spec->vmaster_hook = cx_auto_vmaster_hook; + spec->vmaster_mute.hook = cx_auto_vmaster_hook; err = cx_auto_search_adcs(codec); if (err < 0) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 901547216c4..b69d2fe4029 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -198,7 +198,7 @@ struct alc_spec { /* for virtual master */ hda_nid_t vmaster_nid; - struct snd_kcontrol *vmaster_sw_kctl; + struct hda_vmaster_mute_hook vmaster_mute; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; int num_loopbacks; @@ -1960,7 +1960,7 @@ static int __alc_build_controls(struct hda_codec *codec) err = __snd_hda_add_vmaster(codec, "Master Playback Switch", NULL, alc_slave_pfxs, "Playback Switch", - true, &spec->vmaster_sw_kctl); + true, &spec->vmaster_mute.sw_kctl); if (err < 0) return err; } @@ -5894,13 +5894,11 @@ static void alc269_fixup_mic2_mute(struct hda_codec *codec, struct alc_spec *spec = codec->spec; switch (action) { case ALC_FIXUP_ACT_BUILD: - if (!spec->vmaster_sw_kctl) - return; - snd_ctl_add_vmaster_hook(spec->vmaster_sw_kctl, - alc269_fixup_mic2_mute_hook, codec); + spec->vmaster_mute.hook = alc269_fixup_mic2_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute); /* fallthru */ case ALC_FIXUP_ACT_INIT: - snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); break; } } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6e926497b23..cd04e29e157 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -311,7 +311,7 @@ struct sigmatel_spec { unsigned auto_dmic_cnt; hda_nid_t auto_dmic_nids[MAX_DMICS_NUM]; - struct snd_kcontrol *vmaster_sw_kctl; + struct hda_vmaster_mute_hook vmaster_mute; }; static const hda_nid_t stac9200_adc_nids[1] = { @@ -1160,14 +1160,15 @@ static int stac92xx_build_controls(struct hda_codec *codec) err = __snd_hda_add_vmaster(codec, "Master Playback Switch", NULL, slave_pfxs, "Playback Switch", true, - &spec->vmaster_sw_kctl); + &spec->vmaster_mute.sw_kctl); if (err < 0) return err; if (spec->gpio_led) { - snd_ctl_add_vmaster_hook(spec->vmaster_sw_kctl, - stac92xx_vmaster_hook, codec); - snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); + spec->vmaster_mute.hook = stac92xx_vmaster_hook; + err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute); + if (err < 0) + return err; } if (spec->aloopback_ctl && @@ -4432,7 +4433,7 @@ static int stac92xx_init(struct hda_codec *codec) snd_hda_jack_report_sync(codec); /* sync mute LED */ - snd_ctl_sync_vmaster_hook(spec->vmaster_sw_kctl); + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); if (spec->dac_list) stac92xx_power_down(codec); return 0; -- cgit v1.2.3-18-g5258 From f29735cbef4eb6072e5ae459b556f3a061efc47e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Mar 2012 07:55:10 +0100 Subject: ALSA: hda - Add expose_enum_ctl flag to snd_hda_add_vmaster_hook() Since it's not always safe to assume that the vmaster hook is purely the mute-LED control, add the flag indicating whether to expose the mute-LED enum control or not. Currently, conexant codec sets this off for non-HP laptops where EAPD may be used really as EAPD. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 5 ++++- sound/pci/hda/hda_local.h | 3 ++- sound/pci/hda/patch_conexant.c | 21 +++++++++++++++------ sound/pci/hda/patch_realtek.c | 2 +- sound/pci/hda/patch_sigmatel.c | 2 +- 5 files changed, 23 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b981ea9c644..7a8fcc4c15f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2508,7 +2508,8 @@ static struct snd_kcontrol_new vmaster_mute_mode = { * the given hook. */ int snd_hda_add_vmaster_hook(struct hda_codec *codec, - struct hda_vmaster_mute_hook *hook) + struct hda_vmaster_mute_hook *hook, + bool expose_enum_ctl) { struct snd_kcontrol *kctl; @@ -2517,6 +2518,8 @@ int snd_hda_add_vmaster_hook(struct hda_codec *codec, snd_ctl_add_vmaster_hook(hook->sw_kctl, hook->hook, codec); hook->codec = codec; hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER; + if (!expose_enum_ctl) + return 0; kctl = snd_ctl_new1(&vmaster_mute_mode, hook); if (!kctl) return -ENOMEM; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 3f82ab6a058..0ec9248165b 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -165,7 +165,8 @@ struct hda_vmaster_mute_hook { }; int snd_hda_add_vmaster_hook(struct hda_codec *codec, - struct hda_vmaster_mute_hook *hook); + struct hda_vmaster_mute_hook *hook, + bool expose_enum_ctl); void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook); /* amp value bits */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a21a485a413..e6eafb18c8f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -71,6 +71,7 @@ struct conexant_spec { int num_mixers; hda_nid_t vmaster_nid; struct hda_vmaster_mute_hook vmaster_mute; + bool vmaster_mute_led; const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -4346,8 +4347,10 @@ static int cx_auto_build_controls(struct hda_codec *codec) err = snd_hda_jack_add_kctls(codec, &spec->autocfg); if (err < 0) return err; - if (spec->vmaster_mute.hook && spec->vmaster_mute.sw_kctl) { - err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute); + if (spec->vmaster_mute.sw_kctl) { + spec->vmaster_mute.hook = cx_auto_vmaster_hook; + err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, + spec->vmaster_mute_led); if (err < 0) return err; } @@ -4476,11 +4479,17 @@ static int patch_conexant_auto(struct hda_codec *codec) apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); - /* add EAPD vmaster hook to all HP machines */ - /* NOTE: this should be applied via fixup once when the generic - * fixup code is merged to hda_codec.c + /* Show mute-led control only on HP laptops + * This is a sort of white-list: on HP laptops, EAPD corresponds + * only to the mute-LED without actualy amp function. Meanwhile, + * others may use EAPD really as an amp switch, so it might be + * not good to expose it blindly. */ - spec->vmaster_mute.hook = cx_auto_vmaster_hook; + switch (codec->subsystem_id >> 16) { + case 0x103c: + spec->vmaster_mute_led = 1; + break; + } err = cx_auto_search_adcs(codec); if (err < 0) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b69d2fe4029..8ea2fd65432 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5895,7 +5895,7 @@ static void alc269_fixup_mic2_mute(struct hda_codec *codec, switch (action) { case ALC_FIXUP_ACT_BUILD: spec->vmaster_mute.hook = alc269_fixup_mic2_mute_hook; - snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute); + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); /* fallthru */ case ALC_FIXUP_ACT_INIT: snd_hda_sync_vmaster_hook(&spec->vmaster_mute); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index cd04e29e157..153b9ae46ba 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1166,7 +1166,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) if (spec->gpio_led) { spec->vmaster_mute.hook = stac92xx_vmaster_hook; - err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute); + err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); if (err < 0) return err; } -- cgit v1.2.3-18-g5258 From bd483d4c6c65c1c48483f2f81c603d42b39ce83b Mon Sep 17 00:00:00 2001 From: Jeffrin Jose Date: Wed, 7 Mar 2012 22:57:39 +0530 Subject: ALSA: control - Fixe a trailing white space error Fixed a trailing white space error detected in sound/core/control.c by checkpatch.pl script. Signed-off-by: Jeffrin Jose Signed-off-by: Takashi Iwai --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/control.c b/sound/core/control.c index 819a5c579a3..2487a6bb1c5 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1313,7 +1313,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, err = -EPERM; goto __kctl_end; } - err = kctl->tlv.c(kctl, op_flag, tlv.length, _tlv->tlv); + err = kctl->tlv.c(kctl, op_flag, tlv.length, _tlv->tlv); if (err > 0) { up_read(&card->controls_rwsem); snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_TLV, &kctl->id); -- cgit v1.2.3-18-g5258 From 28aa165cc52fa686a55a2a2052fdddad0fbde5eb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Mar 2012 08:07:41 +0100 Subject: ALSA: ymfpci - Fix legacy registers on S3/S4 resume We need to resume two legacy registers to recover MIDI/FM functionality on S3/S4 resume, too. Signed-off-by: Takashi Iwai --- sound/pci/ymfpci/ymfpci_main.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 03ee4e36531..a3a2eababc0 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2310,6 +2310,10 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state) for (i = 0; i < YDSXGR_NUM_SAVED_REGS; i++) chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]); chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE); + pci_read_config_word(chip->pci, PCIR_DSXG_LEGACY, + &chip->saved_dsxg_legacy); + pci_read_config_word(chip->pci, PCIR_DSXG_ELEGACY, + &chip->saved_dsxg_elegacy); snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0); snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0); snd_ymfpci_disable_dsp(chip); @@ -2344,6 +2348,11 @@ int snd_ymfpci_resume(struct pci_dev *pci) snd_ac97_resume(chip->ac97); + pci_write_config_word(chip->pci, PCIR_DSXG_LEGACY, + chip->saved_dsxg_legacy); + pci_write_config_word(chip->pci, PCIR_DSXG_ELEGACY, + chip->saved_dsxg_elegacy); + /* start hw again */ if (chip->start_count > 0) { spin_lock_irq(&chip->reg_lock); -- cgit v1.2.3-18-g5258 From 25dc16f69892182192b1234594fd3cf342ad4195 Mon Sep 17 00:00:00 2001 From: Anssi Hannula Date: Tue, 13 Mar 2012 17:43:02 +0200 Subject: ALSA: hda - fix printing of high HDMI sample rates A previous commit af65cbf296 (ALSA: hdmi: fix printout of SAD sampling rates) fixed the sample rates shown in /proc/asound/cardX/eldY and kernel log to not be entirely wrong. However, a missing rate from the array added in the patch causes HDMI rates 88.2 kHz, 96 kHz, 176.4 kHz, and 192 kHz to be shown as 96 kHz, 176.4 kHz, 192 kHz, and 384 kHz, respectively. Fix the reporting by adding the ALSA rate 64 kHz into the conversion array between 48 kHz and 88.2 kHz. Signed-off-by: Anssi Hannula Cc: Pierre-Louis Bossart Cc: stable@kernel.org Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index c1da422e085..b58b4b1687f 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -385,8 +385,8 @@ error: static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) { static unsigned int alsa_rates[] = { - 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, - 96000, 176400, 192000, 384000 + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, + 88200, 96000, 176400, 192000, 384000 }; int i, j; -- cgit v1.2.3-18-g5258 From 5ec65ee589fdaca7298b6303fd74ad6c121a8f38 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 12 Mar 2012 19:48:49 -0300 Subject: ASoC: mx27vis-aic32x4: Convert it to platform driver Convert mx27vis-aic32x4 to platform driver. Signed-off-by: Fabio Estevam Tested-by: Javier Martin Signed-off-by: Mark Brown --- sound/soc/imx/mx27vis-aic32x4.c | 41 ++++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 976f857151f..f6d04ad4bb3 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -188,22 +188,16 @@ static struct snd_soc_card mx27vis_aic32x4 = { .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; -static struct platform_device *mx27vis_aic32x4_snd_device; - -static int __init mx27vis_aic32x4_init(void) +static int __devinit mx27vis_aic32x4_probe(struct platform_device *pdev) { int ret; - mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); - if (!mx27vis_aic32x4_snd_device) - return -ENOMEM; - - platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4); - ret = platform_device_add(mx27vis_aic32x4_snd_device); - + mx27vis_aic32x4.dev = &pdev->dev; + ret = snd_soc_register_card(&mx27vis_aic32x4); if (ret) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - platform_device_put(mx27vis_aic32x4_snd_device); + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", + ret); + return ret; } /* Connect SSI0 as clock slave to SSI1 external pins */ @@ -221,22 +215,31 @@ static int __init mx27vis_aic32x4_init(void) ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); - if (ret) { + if (ret) printk(KERN_ERR "ASoC: unable to setup gpios\n"); - platform_device_put(mx27vis_aic32x4_snd_device); - } return ret; } -static void __exit mx27vis_aic32x4_exit(void) +static int __devexit mx27vis_aic32x4_remove(struct platform_device *pdev) { - platform_device_unregister(mx27vis_aic32x4_snd_device); + snd_soc_unregister_card(&mx27vis_aic32x4); + + return 0; } -module_init(mx27vis_aic32x4_init); -module_exit(mx27vis_aic32x4_exit); +static struct platform_driver mx27vis_aic32x4_audio_driver = { + .driver = { + .name = "mx27vis", + .owner = THIS_MODULE, + }, + .probe = mx27vis_aic32x4_probe, + .remove = __devexit_p(mx27vis_aic32x4_remove), +}; + +module_platform_driver(mx27vis_aic32x4_audio_driver); MODULE_AUTHOR("Javier Martin "); MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mx27vis"); -- cgit v1.2.3-18-g5258 From 7907ae3e50613ae1c6d1a10f34fcd63f4123b93d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Mar 2012 08:20:20 +0100 Subject: ALSA: hda - Fix build of patch_sigmatel.c without CONFIG_SND_HDA_POWER_SAVE Now the mute-LED is controlled without powersave hack, and the ifdefs must be removed. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 153b9ae46ba..b064e595bb6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -665,7 +665,6 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, return 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE static int stac_vrefout_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) { @@ -689,7 +688,6 @@ static int stac_vrefout_set(struct hda_codec *codec, return 1; } -#endif static unsigned int stac92xx_vref_set(struct hda_codec *codec, hda_nid_t nid, unsigned int new_vref) @@ -5011,7 +5009,6 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) return 0; } -#ifdef CONFIG_SND_HDA_POWER_SAVE static int stac92xx_pre_resume(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -5046,7 +5043,6 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, afg_power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state, true); } -#endif /* CONFIG_SND_HDA_POWER_SAVE */ #endif /* CONFIG_PM */ /* update mute-LED accoring to the master switch */ @@ -5583,7 +5579,6 @@ again: spec->gpio_led, spec->gpio_led_polarity); -#ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; @@ -5595,7 +5590,6 @@ again: } codec->patch_ops.pre_resume = stac92xx_pre_resume; } -#endif err = stac92xx_parse_auto_config(codec); if (!err) { @@ -5892,7 +5886,6 @@ again: spec->gpio_led, spec->gpio_led_polarity); -#ifdef CONFIG_SND_HDA_POWER_SAVE if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; @@ -5904,7 +5897,6 @@ again: } codec->patch_ops.pre_resume = stac92xx_pre_resume; } -#endif spec->multiout.dac_nids = spec->dac_nids; -- cgit v1.2.3-18-g5258 From 4eb98f45b77b00868dcebe4a0f00d2a36afd88c2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Mar 2012 18:38:28 +0000 Subject: ASoC: wm8996: Add 44.1kHz support The WM8996 specification has been updated to specify 44.1kHz as a supported sample rate. Update the driver to accept this configuration. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 19 ++++++++++--------- 1 file changed, 10 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 40a124c9f15..1fd63549404 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1914,7 +1914,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); - int bits, i, bclk_rate; + int bits, i, bclk_rate, best; int aifdata = 0; int lrclk = 0; int dsp = 0; @@ -1963,14 +1963,11 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream, return bits; aifdata |= (bits << WM8996_AIF1TX_WL_SHIFT) | bits; + best = 0; for (i = 0; i < ARRAY_SIZE(dsp_divs); i++) { - if (dsp_divs[i] == params_rate(params)) - break; - } - if (i == ARRAY_SIZE(dsp_divs)) { - dev_err(codec->dev, "Unsupported sample rate %dHz\n", - params_rate(params)); - return -EINVAL; + if (abs(dsp_divs[i] - params_rate(params)) < + abs(dsp_divs[best] - params_rate(params))) + best = i; } dsp |= i << dsp_shift; @@ -2030,13 +2027,16 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, } switch (wm8996->sysclk) { + case 5644800: case 6144000: snd_soc_update_bits(codec, WM8996_AIF_RATE, WM8996_SYSCLK_RATE, 0); break; + case 22579200: case 24576000: ratediv = WM8996_SYSCLK_DIV; wm8996->sysclk /= 2; + case 11289600: case 12288000: snd_soc_update_bits(codec, WM8996_AIF_RATE, WM8996_SYSCLK_RATE, WM8996_SYSCLK_RATE); @@ -3060,7 +3060,8 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8996 = { }; #define WM8996_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) #define WM8996_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -- cgit v1.2.3-18-g5258 From 4af87a939ef7092fdca267fba473cf8407d6d8e2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Mar 2012 19:48:43 +0000 Subject: ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list Allows the constraint lists to be declared const by drivers which seems reasonable; there's plenty of other constification we could do if we were being complete but this was easy and quick. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 3420bd3da5d..4d18941178e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1029,7 +1029,8 @@ static int snd_interval_ratden(struct snd_interval *i, * * Returns non-zero if the value is changed, zero if not changed. */ -int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask) +int snd_interval_list(struct snd_interval *i, unsigned int count, + const unsigned int *list, unsigned int mask) { unsigned int k; struct snd_interval list_range; -- cgit v1.2.3-18-g5258 From 1662591b2e6876b8bc041cd48837ccd297c2028f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Mar 2012 15:55:43 +0100 Subject: ALSA: pcm - Avoid GFP_ATOMIC in snd_pcm_link() GFP_ATOMIC is used in snd_pcm_link() just because the kmalloc is called inside a lock. Since this function isn't too critical for speed and is rarely called in practice, better to allocate the chunk at first before spinlock and free it in error paths, so that GFP_KERNEL can be used. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 25ed9fe41b8..3fe99e644eb 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1586,12 +1586,18 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) struct file *file; struct snd_pcm_file *pcm_file; struct snd_pcm_substream *substream1; + struct snd_pcm_group *group; file = snd_pcm_file_fd(fd); if (!file) return -EBADFD; pcm_file = file->private_data; substream1 = pcm_file->substream; + group = kmalloc(sizeof(*group), GFP_KERNEL); + if (!group) { + res = -ENOMEM; + goto _nolock; + } down_write(&snd_pcm_link_rwsem); write_lock_irq(&snd_pcm_link_rwlock); if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN || @@ -1604,11 +1610,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) goto _end; } if (!snd_pcm_stream_linked(substream)) { - substream->group = kmalloc(sizeof(struct snd_pcm_group), GFP_ATOMIC); - if (substream->group == NULL) { - res = -ENOMEM; - goto _end; - } + substream->group = group; spin_lock_init(&substream->group->lock); INIT_LIST_HEAD(&substream->group->substreams); list_add_tail(&substream->link_list, &substream->group->substreams); @@ -1620,7 +1622,10 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) _end: write_unlock_irq(&snd_pcm_link_rwlock); up_write(&snd_pcm_link_rwsem); + _nolock: fput(file); + if (res < 0) + kfree(group); return res; } -- cgit v1.2.3-18-g5258 From 181a68927b9e6ff7c0ea093c2f056eeb0552a911 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 14 Mar 2012 20:18:49 +0000 Subject: ASoC: core: Fix obscure leak of runtime array We're currently not freeing card->rtd in cases where the card is unregistered before being instantiated - convert it to devm_kzalloc() to make sure that happens. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93a0daac508..a4deebc0801 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1686,7 +1686,6 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) snd_soc_dapm_free(&card->dapm); - kfree(card->rtd); snd_card_free(card->snd_card); return 0; @@ -3112,9 +3111,10 @@ int snd_soc_register_card(struct snd_soc_card *card) soc_init_card_debugfs(card); - card->rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime) * - (card->num_links + card->num_aux_devs), - GFP_KERNEL); + card->rtd = devm_kzalloc(card->dev, + sizeof(struct snd_soc_pcm_runtime) * + (card->num_links + card->num_aux_devs), + GFP_KERNEL); if (card->rtd == NULL) return -ENOMEM; card->rtd_aux = &card->rtd[card->num_links]; -- cgit v1.2.3-18-g5258 From 0717d0f5d2737a63650a8d928360769e0d411bd5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Mar 2012 16:14:38 +0100 Subject: ALSA: usb-audio - Fix build error by consitification of rate list Signed-off-by: Takashi Iwai --- sound/usb/pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0220b0f335b..0eed6115c2d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -695,6 +695,7 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { struct audioformat *fp; + int *rate_list; int count = 0, needs_knot = 0; int err; @@ -708,7 +709,8 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, if (!needs_knot) return 0; - subs->rate_list.list = kmalloc(sizeof(int) * count, GFP_KERNEL); + subs->rate_list.list = rate_list = + kmalloc(sizeof(int) * count, GFP_KERNEL); if (!subs->rate_list.list) return -ENOMEM; subs->rate_list.count = count; @@ -717,7 +719,7 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, list_for_each_entry(fp, &subs->fmt_list, list) { int i; for (i = 0; i < fp->nr_rates; i++) - subs->rate_list.list[count++] = fp->rate_table[i]; + rate_list[count++] = fp->rate_table[i]; } err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &subs->rate_list); -- cgit v1.2.3-18-g5258 From c6b76d1f02e2ab1109d8549877a3a24c6a2b4587 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 Mar 2012 09:54:00 +0100 Subject: ALSA: au88x0 - Avoid possible Oops at unbinding The irq handler must check whether the MPU401 instance is still alive. Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 1181c5ec2d4..525f881f040 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2477,7 +2477,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) hwread(vortex->mmio, VORTEX_IRQ_STAT); handled = 1; } - if (source & IRQ_MIDI) { + if ((source & IRQ_MIDI) && vortex->rmidi) { snd_mpu401_uart_interrupt(vortex->irq, vortex->rmidi->private_data); handled = 1; -- cgit v1.2.3-18-g5258 From 350eba43fca735733a51185f26bdc30899c64a20 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 Mar 2012 16:09:03 +0100 Subject: ALSA: hda - Fix build with CONFIG_PM=n Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index b064e595bb6..33a9946b492 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5043,6 +5043,11 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg, afg_power_state); snd_hda_codec_set_power_to_all(codec, fg, power_state, true); } +#else +#define stac92xx_suspend NULL +#define stac92xx_resume NULL +#define stac92xx_pre_resume NULL +#define stac92xx_set_power_state NULL #endif /* CONFIG_PM */ /* update mute-LED accoring to the master switch */ @@ -5588,7 +5593,9 @@ again: codec->patch_ops.set_power_state = stac92xx_set_power_state; } +#ifdef CONFIG_PM codec->patch_ops.pre_resume = stac92xx_pre_resume; +#endif } err = stac92xx_parse_auto_config(codec); @@ -5895,7 +5902,9 @@ again: codec->patch_ops.set_power_state = stac92xx_set_power_state; } +#ifdef CONFIG_PM codec->patch_ops.pre_resume = stac92xx_pre_resume; +#endif } spec->multiout.dac_nids = spec->dac_nids; -- cgit v1.2.3-18-g5258 From 588fb705d560cd76d435382fb25bea7349672d80 Mon Sep 17 00:00:00 2001 From: Sangsu Park Date: Fri, 16 Mar 2012 15:40:53 +0900 Subject: ASoC: Samsung: Added to support mono recording The dma size will be changed by requested number of channel(mono/stereo) from platform. For mono recording, channels_min value should be 1. Signed-off-by: Sangsu Park Signed-off-by: Sangbeom Kim Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 6553b19c70c..6ac7b8281a0 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -559,6 +559,17 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, mod |= MOD_DC1_EN; break; case 2: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s->dma_playback.dma_size = 4; + else + i2s->dma_capture.dma_size = 4; + break; + case 1: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s->dma_playback.dma_size = 2; + else + i2s->dma_capture.dma_size = 2; + break; default: dev_err(&i2s->pdev->dev, "%d channels not supported\n", @@ -963,7 +974,7 @@ struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS; if (!sec) { - i2s->i2s_dai_drv.capture.channels_min = 2; + i2s->i2s_dai_drv.capture.channels_min = 1; i2s->i2s_dai_drv.capture.channels_max = 2; i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; -- cgit v1.2.3-18-g5258 From 70ac07bb633dee75ac554195b9a4d69adfa7803c Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 16 Mar 2012 16:32:52 -0500 Subject: ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master The WM8776 codec driver requires the machine driver to set one of the SND_SOC_DAIFMT_CBx_xxx values. The P1022DS machine driver should be setting SND_SOC_DAIFMT_CBM_CFM, but since that value was zero, no one noticed. Commit 75d9ac46 ("ASoC: Allow DAI formats to be specified in the dai_link"), however, changed the value of SND_SOC_DAIFMT_CBM_CFM from zero to a non-zero value, which means that it now needs to be specifically set by the machine driver. We also set SND_SOC_DAIFMT_NB_NF, for the same reason. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/fsl/p1022_ds.c | 24 ++++++++++++++++-------- 1 file changed, 16 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index b8898708347..46623405a2c 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -395,7 +395,8 @@ static int p1022_ds_probe(struct platform_device *pdev) } if (strcasecmp(sprop, "i2s-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_I2S; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; @@ -412,31 +413,38 @@ static int p1022_ds_probe(struct platform_device *pdev) } mdata->clk_frequency = be32_to_cpup(iprop); } else if (strcasecmp(sprop, "i2s-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_I2S; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "lj-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_LEFT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "lj-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_LEFT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "rj-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else if (strcasecmp(sprop, "ac97-slave") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_AC97; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBM_CFM; mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "ac97-master") == 0) { - mdata->dai_format = SND_SOC_DAIFMT_AC97; + mdata->dai_format = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBS_CFS; mdata->codec_clk_direction = SND_SOC_CLOCK_IN; mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; } else { -- cgit v1.2.3-18-g5258 From 273b72c8ce6b28df6b49423d775c3e59072c73c5 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 19 Mar 2012 09:12:53 +0100 Subject: ASoC: pxa-ssp: atomically set stream active masks PXA's SSP engine fails to take its current channel phase into account when enabling a stream while the engine is already running. This results in randomly swapped left/right channels on either the record or the playback side, depending on which one was enabled first. The following patch fixes this by factoring out the bit field modifications in question to a separate function that pauses the engine temporarily, modifies the bits and kicks it off again afterwards. Appearantly, a transition of SSCR0_SSE syncs both directions properly. The patch has been rolled out to quite a number of devices over the last weeks and seems to fix the issue reliably. Signed-off-by: Daniel Mack Reported-and-tested-by: Sven Neumann Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/pxa/pxa-ssp.c | 61 +++++++++++++++++++++++++++++-------------------- 1 file changed, 36 insertions(+), 25 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a16df0fa6ef..fd04ce13903 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -668,6 +668,38 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, return 0; } +static void pxa_ssp_set_running_bit(struct snd_pcm_substream *substream, + struct ssp_device *ssp, int value) +{ + uint32_t sscr0 = pxa_ssp_read_reg(ssp, SSCR0); + uint32_t sscr1 = pxa_ssp_read_reg(ssp, SSCR1); + uint32_t sspsp = pxa_ssp_read_reg(ssp, SSPSP); + uint32_t sssr = pxa_ssp_read_reg(ssp, SSSR); + + if (value && (sscr0 & SSCR0_SSE)) + pxa_ssp_write_reg(ssp, SSCR0, sscr0 & ~SSCR0_SSE); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (value) + sscr1 |= SSCR1_TSRE; + else + sscr1 &= ~SSCR1_TSRE; + } else { + if (value) + sscr1 |= SSCR1_RSRE; + else + sscr1 &= ~SSCR1_RSRE; + } + + pxa_ssp_write_reg(ssp, SSCR1, sscr1); + + if (value) { + pxa_ssp_write_reg(ssp, SSSR, sssr); + pxa_ssp_write_reg(ssp, SSPSP, sspsp); + pxa_ssp_write_reg(ssp, SSCR0, sscr0 | SSCR0_SSE); + } +} + static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { @@ -681,42 +713,21 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, pxa_ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val |= SSCR1_TSRE; - else - val |= SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_set_running_bit(substream, ssp, 1); val = pxa_ssp_read_reg(ssp, SSSR); pxa_ssp_write_reg(ssp, SSSR, val); break; case SNDRV_PCM_TRIGGER_START: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val |= SSCR1_TSRE; - else - val |= SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); - pxa_ssp_enable(ssp); + pxa_ssp_set_running_bit(substream, ssp, 1); break; case SNDRV_PCM_TRIGGER_STOP: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val &= ~SSCR1_TSRE; - else - val &= ~SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_set_running_bit(substream, ssp, 0); break; case SNDRV_PCM_TRIGGER_SUSPEND: pxa_ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - val = pxa_ssp_read_reg(ssp, SSCR1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - val &= ~SSCR1_TSRE; - else - val &= ~SSCR1_RSRE; - pxa_ssp_write_reg(ssp, SSCR1, val); + pxa_ssp_set_running_bit(substream, ssp, 0); break; default: -- cgit v1.2.3-18-g5258 From 5472bbc96f1ba666328cc2479b957ed50f5e1550 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Mar 2012 17:31:56 +0000 Subject: ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF It can just be enabled all the time with no impact. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 16 +++++----------- 1 file changed, 5 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index c08d1c2f346..f13f2886339 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -643,8 +643,6 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0), - SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0, @@ -869,11 +867,9 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout1_se_routes[] = { - { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, @@ -890,11 +886,9 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout2_se_routes[] = { - { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" }, - { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, @@ -996,6 +990,11 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, WM8993_LINEOUT2_MODE, WM8993_LINEOUT2_MODE); + if (!lineout1_diff && !lineout2_diff) + snd_soc_update_bits(codec, WM8993_ANTIPOP1, + WM8993_LINEOUT_VMID_BUF_ENA, + WM8993_LINEOUT_VMID_BUF_ENA); + if (lineout1fb) snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); @@ -1068,11 +1067,6 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, WM8993_LINEOUT2P_ENA, val); - if (!hubs->lineout1n_ena && !hubs->lineout1p_ena && - !hubs->lineout2n_ena && !hubs->lineout2p_ena) - snd_soc_update_bits(codec, WM8993_ANTIPOP1, - WM8993_LINEOUT_VMID_BUF_ENA, 0); - /* Remove the input clamps */ snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, WM8993_INPUTS_CLAMP, 0); -- cgit v1.2.3-18-g5258 From 6f8270cc9a43d767676c97df5773fdcede312a88 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 3 Mar 2012 13:06:25 +0000 Subject: ASoC: wm8994: Add missing break in resume Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 15fcb1bb714..b7f3cfc74e9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2819,6 +2819,7 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) WM1811_JACKDET_MODE_JACK); break; } + break; case WM8958: if (wm8994->jack_cb) snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, -- cgit v1.2.3-18-g5258 From 22f8d055350066b4a87de4adea8c5213cac54534 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Mar 2012 17:32:06 +0000 Subject: ASoC: wm8994: Provide VMID mode control and fix default sequence The optimal management of VMID depends on a number of factors which vary dynamically at runtime, for example the connection to a system docking station. In some circumstances it is desirable to keep VMID enabled all the time, in others it is desirable to aggressively power it up and down. Provide a callback allowing machine driver to configure either the normal power up/down mode (WM8994_VMID_MODE_NORMAL) or to maintain VMID even when idle (WM8994_VMID_MODE_FORCE). This callback, wm8994_vmid_mode(), should be called with the CODEC lock. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 186 ++++++++++++++++++++++++++++++++++++---------- sound/soc/codecs/wm8994.h | 8 ++ 2 files changed, 155 insertions(+), 39 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b7f3cfc74e9..9685dff44dd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -777,36 +777,68 @@ static void vmid_reference(struct snd_soc_codec *codec) if (wm8994->vmid_refcount == 1) { snd_soc_update_bits(codec, WM8994_ANTIPOP_1, - WM8994_LINEOUT_VMID_BUF_ENA | WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH, - WM8994_LINEOUT_VMID_BUF_ENA); + WM8994_LINEOUT2_DISCH, 0); wm_hubs_vmid_ena(codec); - /* Startup bias, VMID ramp & buffer */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | - WM8994_VMID_DISCH | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (0x2 << WM8994_VMID_RAMP_SHIFT)); + switch (wm8994->vmid_mode) { + default: + WARN_ON(0 == "Invalid VMID mode"); + case WM8994_VMID_NORMAL: + /* Startup bias, VMID ramp & buffer */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_VMID_DISCH | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (0x3 << WM8994_VMID_RAMP_SHIFT)); + + /* Main bias enable, VMID=2x40k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, + WM8994_BIAS_ENA | 0x2); + + msleep(50); - /* Main bias enable, VMID=2x40k */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | - WM8994_VMID_SEL_MASK, - WM8994_BIAS_ENA | 0x2); + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_VMID_RAMP_MASK | + WM8994_BIAS_SRC, + 0); + break; - msleep(50); + case WM8994_VMID_FORCE: + /* Startup bias, slow VMID ramp & buffer */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_VMID_DISCH | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (0x2 << WM8994_VMID_RAMP_SHIFT)); + + /* Main bias enable, VMID=2x40k */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, + WM8994_BIAS_ENA | 0x2); + + msleep(400); - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_VMID_RAMP_MASK | WM8994_BIAS_SRC, - 0); + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_VMID_RAMP_MASK | + WM8994_BIAS_SRC, + 0); + break; + } } } @@ -820,34 +852,55 @@ static void vmid_dereference(struct snd_soc_codec *codec) wm8994->vmid_refcount); if (wm8994->vmid_refcount == 0) { - /* Switch over to startup biases */ + if (wm8994->hubs.lineout1_se) + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_3, + WM8994_LINEOUT1N_ENA | + WM8994_LINEOUT1P_ENA, + WM8994_LINEOUT1N_ENA | + WM8994_LINEOUT1P_ENA); + + if (wm8994->hubs.lineout2_se) + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_3, + WM8994_LINEOUT2N_ENA | + WM8994_LINEOUT2P_ENA, + WM8994_LINEOUT2N_ENA | + WM8994_LINEOUT2P_ENA); + + /* Start discharging VMID */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, + WM8994_VMID_DISCH, WM8994_BIAS_SRC | - WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (1 << WM8994_VMID_RAMP_SHIFT)); + WM8994_VMID_DISCH); - /* Disable main biases */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | - WM8994_VMID_SEL_MASK, 0); + switch (wm8994->vmid_mode) { + case WM8994_VMID_FORCE: + msleep(350); + break; + default: + break; + } - /* Discharge VMID */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_VMID_DISCH, WM8994_VMID_DISCH); + snd_soc_update_bits(codec, WM8994_ADDITIONAL_CONTROL, + WM8994_VROI, WM8994_VROI); - /* Discharge line */ + /* Active discharge */ snd_soc_update_bits(codec, WM8994_ANTIPOP_1, WM8994_LINEOUT1_DISCH | WM8994_LINEOUT2_DISCH, WM8994_LINEOUT1_DISCH | WM8994_LINEOUT2_DISCH); - msleep(5); + msleep(150); + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_3, + WM8994_LINEOUT1N_ENA | + WM8994_LINEOUT1P_ENA | + WM8994_LINEOUT2N_ENA | + WM8994_LINEOUT2P_ENA, 0); + + snd_soc_update_bits(codec, WM8994_ADDITIONAL_CONTROL, + WM8994_VROI, 0); /* Switch off startup biases */ snd_soc_update_bits(codec, WM8994_ANTIPOP_2, @@ -855,6 +908,12 @@ static void vmid_dereference(struct snd_soc_codec *codec) WM8994_STARTUP_BIAS_ENA | WM8994_VMID_BUF_ENA | WM8994_VMID_RAMP_MASK, 0); + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0); + + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_VMID_RAMP_MASK, 0); } pm_runtime_put(codec->dev); @@ -2197,6 +2256,55 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, return 0; } +int wm8994_vmid_mode(struct snd_soc_codec *codec, enum wm8994_vmid_mode mode) +{ + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (mode) { + case WM8994_VMID_NORMAL: + if (wm8994->hubs.lineout1_se) { + snd_soc_dapm_disable_pin(&codec->dapm, + "LINEOUT1N Driver"); + snd_soc_dapm_disable_pin(&codec->dapm, + "LINEOUT1P Driver"); + } + if (wm8994->hubs.lineout2_se) { + snd_soc_dapm_disable_pin(&codec->dapm, + "LINEOUT2N Driver"); + snd_soc_dapm_disable_pin(&codec->dapm, + "LINEOUT2P Driver"); + } + + /* Do the sync with the old mode to allow it to clean up */ + snd_soc_dapm_sync(&codec->dapm); + wm8994->vmid_mode = mode; + break; + + case WM8994_VMID_FORCE: + if (wm8994->hubs.lineout1_se) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "LINEOUT1N Driver"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "LINEOUT1P Driver"); + } + if (wm8994->hubs.lineout2_se) { + snd_soc_dapm_force_enable_pin(&codec->dapm, + "LINEOUT2N Driver"); + snd_soc_dapm_force_enable_pin(&codec->dapm, + "LINEOUT2P Driver"); + } + + wm8994->vmid_mode = mode; + snd_soc_dapm_sync(&codec->dapm); + break; + + default: + return -EINVAL; + } + + return 0; +} + static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 2f4d2d12a45..c724112998d 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -32,6 +32,11 @@ #define WM8994_FLL_SRC_LRCLK 3 #define WM8994_FLL_SRC_BCLK 4 +enum wm8994_vmid_mode { + WM8994_VMID_NORMAL, + WM8994_VMID_FORCE, +}; + typedef void (*wm8958_micdet_cb)(u16 status, void *data); int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, @@ -39,6 +44,8 @@ int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8958_micdet_cb cb, void *cb_data); +int wm8994_vmid_mode(struct snd_soc_codec *codec, enum wm8994_vmid_mode mode); + int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -75,6 +82,7 @@ struct wm8994_priv { int vmid_refcount; int active_refcount; + enum wm8994_vmid_mode vmid_mode; int dac_rates[2]; int lrclk_shared[2]; -- cgit v1.2.3-18-g5258