diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2012-06-19 23:37:19 -0700 |
---|---|---|
committer | Linus Torvalds <torvalds@linux-foundation.org> | 2012-06-19 23:37:19 -0700 |
commit | f40759e7041498577235192727322186b43aa216 (patch) | |
tree | 1e824ccc4cdd7b9d48fe617d5dcf3effac128265 | |
parent | 2fe8ac608bf1a9c947f03f5d8cbf25c91b4f1a7c (diff) | |
parent | 0b1d8e09089b69ac2e8be203fb28cd07cfe035b2 (diff) |
Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Most of changes are fairly small and driver-specific.
A remaining regression fix for USB-audio sync pipe check, a fix for
HD-audio power-up sequence, fixes for ASoC pxa-ssp compile issues, and
bunch of ASoC codec and trivial fix patches."
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: 6fire: use NULL instead of 0 for pointer assignment
ALSA: hda - Handle open while transitioning to D3.
ALSA: snd-usb: make snd_usb_substream_capture_trigger static
ALSA: snd-usb: fix sync pipe check
ASoC: tegra+wm8903: turn of mic detect when card is removed
ASoC: wm8996: Mark the CODEC as cache only when powering off on boot
ASoC: wm8996: Move reset before the initial regulator disable
ASoC: wm8996: Remove spurious regulator_bulk_free()
ASoC: wm8904: Fix cache only management
ASoC: wm8904: Fix GPIO and MICBIAS initialisation for regmap conversion
ASoC: fix pxa-ssp compiling issue under mach-mmp
ARM: MMP: add pxa910-ssp into ssp_id_table
-rw-r--r-- | arch/arm/plat-pxa/ssp.c | 1 | ||||
-rw-r--r-- | include/linux/pxa2xx_ssp.h | 2 | ||||
-rw-r--r-- | include/linux/spi/pxa2xx_spi.h | 2 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 46 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.h | 2 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8904.c | 26 | ||||
-rw-r--r-- | sound/soc/codecs/wm8996.c | 8 | ||||
-rw-r--r-- | sound/soc/pxa/pxa-ssp.c | 38 | ||||
-rw-r--r-- | sound/soc/tegra/tegra_wm8903.c | 13 | ||||
-rw-r--r-- | sound/usb/6fire/firmware.c | 2 | ||||
-rw-r--r-- | sound/usb/pcm.c | 21 |
12 files changed, 97 insertions, 66 deletions
diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index 58b79809d20..584c9bf8ed2 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -193,6 +193,7 @@ static const struct platform_device_id ssp_id_table[] = { { "pxa25x-nssp", PXA25x_NSSP }, { "pxa27x-ssp", PXA27x_SSP }, { "pxa168-ssp", PXA168_SSP }, + { "pxa910-ssp", PXA910_SSP }, { }, }; diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h index 44835fb3979..f36632061c6 100644 --- a/include/linux/pxa2xx_ssp.h +++ b/include/linux/pxa2xx_ssp.h @@ -160,7 +160,9 @@ enum pxa_ssp_type { PXA25x_SSP, /* pxa 210, 250, 255, 26x */ PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */ PXA27x_SSP, + PXA3xx_SSP, PXA168_SSP, + PXA910_SSP, CE4100_SSP, }; diff --git a/include/linux/spi/pxa2xx_spi.h b/include/linux/spi/pxa2xx_spi.h index d3e1075f7b6..c73d1445c77 100644 --- a/include/linux/spi/pxa2xx_spi.h +++ b/include/linux/spi/pxa2xx_spi.h @@ -43,7 +43,7 @@ struct pxa2xx_spi_chip { void (*cs_control)(u32 command); }; -#ifdef CONFIG_ARCH_PXA +#if defined(CONFIG_ARCH_PXA) || defined(CONFIG_ARCH_MMP) #include <linux/clk.h> #include <mach/dma.h> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 41ca803a1ff..7504e62188d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4393,20 +4393,19 @@ void snd_hda_update_power_acct(struct hda_codec *codec) codec->power_jiffies += delta; } -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -void snd_hda_power_up(struct hda_codec *codec) +/* Transition to powered up, if wait_power_down then wait for a pending + * transition to D3 to complete. A pending D3 transition is indicated + * with power_transition == -1. */ +static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) { struct hda_bus *bus = codec->bus; spin_lock(&codec->power_lock); codec->power_count++; - if (codec->power_on || codec->power_transition > 0) { + /* Return if power_on or transitioning to power_on, unless currently + * powering down. */ + if ((codec->power_on || codec->power_transition > 0) && + !(wait_power_down && codec->power_transition < 0)) { spin_unlock(&codec->power_lock); return; } @@ -4430,8 +4429,37 @@ void snd_hda_power_up(struct hda_codec *codec) codec->power_transition = 0; spin_unlock(&codec->power_lock); } + +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the power-up counter and power up the hardware really when + * not turned on yet. + */ +void snd_hda_power_up(struct hda_codec *codec) +{ + __snd_hda_power_up(codec, false); +} EXPORT_SYMBOL_HDA(snd_hda_power_up); +/** + * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending + * D3 transition to complete. This differs from snd_hda_power_up() when + * power_transition == -1. snd_hda_power_up sees this case as a nop, + * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers + * back up. + * @codec: HD-audio codec + * + * Cancel any power down operation hapenning on the work queue, then power up. + */ +void snd_hda_power_up_d3wait(struct hda_codec *codec) +{ + /* This will cancel and wait for pending power_work to complete. */ + __snd_hda_power_up(codec, true); +} +EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait); + #define power_save(codec) \ ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4fc3960c859..2fdaadbb432 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1056,10 +1056,12 @@ const char *snd_hda_get_jack_location(u32 cfg); */ #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_up(struct hda_codec *codec); +void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} +static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 02763827dde..7757536b9d5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1766,7 +1766,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); - snd_hda_power_up(apcm->codec); + snd_hda_power_up_d3wait(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { azx_release_device(azx_dev); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 65d525d74c5..812acd83fb4 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1863,6 +1863,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, return ret; } + regcache_cache_only(wm8904->regmap, false); regcache_sync(wm8904->regmap); /* Enable bias */ @@ -1899,14 +1900,8 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 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(wm8904->regmap, true); + regcache_mark_dirty(wm8904->regmap); regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); @@ -2084,10 +2079,8 @@ static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); struct wm8904_pdata *pdata = wm8904->pdata; - u16 *reg_cache = codec->reg_cache; int ret, i; - codec->cache_sync = 1; codec->control_data = wm8904->regmap; switch (wm8904->devtype) { @@ -2150,6 +2143,7 @@ static int wm8904_probe(struct snd_soc_codec *codec) goto err_enable; } + regcache_cache_only(wm8904->regmap, true); /* Change some default settings - latch VU and enable ZC */ snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, WM8904_ADC_VU, WM8904_ADC_VU); @@ -2180,14 +2174,18 @@ static int wm8904_probe(struct snd_soc_codec *codec) if (!pdata->gpio_cfg[i]) continue; - reg_cache[WM8904_GPIO_CONTROL_1 + i] - = pdata->gpio_cfg[i] & 0xffff; + regmap_update_bits(wm8904->regmap, + WM8904_GPIO_CONTROL_1 + i, + 0xffff, + pdata->gpio_cfg[i]); } /* Zero is the default value for these anyway */ for (i = 0; i < WM8904_MIC_REGS; i++) - reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i] - = pdata->mic_cfg[i]; + regmap_update_bits(wm8904->regmap, + WM8904_MIC_BIAS_CONTROL_0 + i, + 0xffff, + pdata->mic_cfg[i]); } /* Set Class W by default - this will be managed by the Class diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8af422e38fd..dc9b42b7fc4 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2837,8 +2837,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) } } - regcache_cache_only(codec->control_data, true); - /* Apply platform data settings */ snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL, WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, @@ -3051,7 +3049,6 @@ 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_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); return 0; } @@ -3206,14 +3203,15 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "revision %c\n", (reg & WM8996_CHIP_REV_MASK) + 'A'); - regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); - ret = wm8996_reset(wm8996); if (ret < 0) { dev_err(&i2c->dev, "Failed to issue reset\n"); goto err_regmap; } + regcache_cache_only(wm8996->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); + wm8996_init_gpio(wm8996); ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 1c2aa7fab3f..4da5fc55c7e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -33,7 +33,6 @@ #include <mach/hardware.h> #include <mach/dma.h> -#include <mach/audio.h> #include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" @@ -194,7 +193,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) { u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + if (ssp->type == PXA25x_SSP) { sscr0 &= ~0x0000ff00; sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ } else { @@ -212,7 +211,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp) u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); u32 div; - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + if (ssp->type == PXA25x_SSP) div = ((sscr0 >> 8) & 0xff) * 2 + 2; else div = ((sscr0 >> 8) & 0xfff) + 1; @@ -242,7 +241,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_PLL: /* Internal PLL is fixed */ - if (cpu_is_pxa25x()) + if (ssp->type == PXA25x_SSP) priv->sysclk = 1843200; else priv->sysclk = 13000000; @@ -266,11 +265,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) clk_disable(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; pxa_ssp_write_reg(ssp, SSCR0, val); - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) clk_enable(ssp->clk); return 0; @@ -294,24 +293,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, case PXA_SSP_AUDIO_DIV_SCDB: val = pxa_ssp_read_reg(ssp, SSACD); val &= ~SSACD_SCDB; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) val &= ~SSACD_SCDX8; -#endif switch (div) { case PXA_SSP_CLK_SCDB_1: val |= SSACD_SCDB; break; case PXA_SSP_CLK_SCDB_4: break; -#if defined(CONFIG_PXA3xx) case PXA_SSP_CLK_SCDB_8: - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) val |= SSACD_SCDX8; else return -EINVAL; break; -#endif default: return -EINVAL; } @@ -337,10 +332,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, struct ssp_device *ssp = priv->ssp; u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) pxa_ssp_write_reg(ssp, SSACDD, 0); -#endif switch (freq_out) { case 5622000: @@ -365,11 +358,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, break; default: -#ifdef CONFIG_PXA3xx /* PXA3xx has a clock ditherer which can be used to generate * a wider range of frequencies - calculate a value for it. */ - if (cpu_is_pxa3xx()) { + if (ssp->type == PXA3xx_SSP) { u32 val; u64 tmp = 19968; tmp *= 1000000; @@ -386,7 +378,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, val, freq_out); break; } -#endif return -EINVAL; } @@ -590,10 +581,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: -#ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) sscr0 |= SSCR0_FPCKE; -#endif sscr0 |= SSCR0_DataSize(16); break; case SNDRV_PCM_FORMAT_S24_LE: @@ -618,9 +607,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * trying and failing a lot; some of the registers * needed for that mode are only available on PXA3xx. */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) return -EINVAL; sspsp |= SSPSP_SFRMWDTH(width * 2); @@ -628,9 +615,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sspsp |= SSPSP_EDMYSTOP(3); sspsp |= SSPSP_DMYSTOP(3); sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif } else { /* The frame width is the width the LRCLK is * asserted for; the delay is expressed in diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0b0df49d9d3..3b6da91188a 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -346,6 +346,17 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_wm8903_remove(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + + wm8903_mic_detect(codec, NULL, 0, 0); + + return 0; +} + static struct snd_soc_dai_link tegra_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", @@ -363,6 +374,8 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { .dai_link = &tegra_wm8903_dai, .num_links = 1, + .remove = tegra_wm8903_remove, + .controls = tegra_wm8903_controls, .num_controls = ARRAY_SIZE(tegra_wm8903_controls), .dapm_widgets = tegra_wm8903_dapm_widgets, diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 6f9715ab32f..56ad923bf6b 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload( int ret; u8 data; struct usb_device *device = interface_to_usbdev(intf); - const struct firmware *fw = 0; + const struct firmware *fw = NULL; struct ihex_record *rec = kmalloc(sizeof(struct ihex_record), GFP_KERNEL); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index cdf8b760197..54607f8c4f6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -354,17 +354,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && get_endpoint(alts, 1)->bSynchAddress != 0 && !implicit_fb)) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + get_endpoint(alts, 1)->bmAttributes, + get_endpoint(alts, 1)->bLength, + get_endpoint(alts, 1)->bSynchAddress); return -EINVAL; } ep = get_endpoint(alts, 1)->bEndpointAddress; - if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + if (!implicit_fb && + get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) || - ( is_playback && !implicit_fb))) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); return -EINVAL; } @@ -1147,7 +1151,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea return -EINVAL; } -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, + int cmd) { int err; struct snd_usb_substream *subs = substream->runtime->private_data; |