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-rw-r--r--sound/soc/pxa/Kconfig59
-rw-r--r--sound/soc/pxa/Makefile12
-rw-r--r--sound/soc/pxa/brownstone.c169
-rw-r--r--sound/soc/pxa/corgi.c56
-rw-r--r--sound/soc/pxa/e740_wm9705.c17
-rw-r--r--sound/soc/pxa/e750_wm9705.c17
-rw-r--r--sound/soc/pxa/e800_wm9712.c26
-rw-r--r--sound/soc/pxa/hx4700.c15
-rw-r--r--sound/soc/pxa/imote2.c7
-rw-r--r--sound/soc/pxa/magician.c60
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c56
-rw-r--r--sound/soc/pxa/mmp-pcm.c257
-rw-r--r--sound/soc/pxa/mmp-sspa.c485
-rw-r--r--sound/soc/pxa/mmp-sspa.h92
-rw-r--r--sound/soc/pxa/palm27x.c53
-rw-r--r--sound/soc/pxa/poodle.c15
-rw-r--r--sound/soc/pxa/pxa-ssp.c94
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c148
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.h3
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c44
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c40
-rw-r--r--sound/soc/pxa/saarb.c190
-rw-r--r--sound/soc/pxa/spitz.c58
-rw-r--r--sound/soc/pxa/tavorevb3.c189
-rw-r--r--sound/soc/pxa/tosa.c74
-rw-r--r--sound/soc/pxa/ttc-dkb.c171
-rw-r--r--sound/soc/pxa/zylonite.c18
27 files changed, 1606 insertions, 819 deletions
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index a0f7d3cfa47..2434b6d6167 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -8,6 +8,16 @@ config SND_PXA2XX_SOC
the PXA2xx AC97, I2S or SSP interface. You will also need
to select the audio interfaces to support below.
+config SND_MMP_SOC
+ bool "Soc Audio for Marvell MMP chips"
+ depends on ARCH_MMP
+ select MMP_SRAM
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+ select SND_ARM
+ help
+ Say Y if you want to add support for codecs attached to
+ the MMP SSPA interface.
+
config SND_PXA2XX_AC97
tristate
select SND_AC97_CODEC
@@ -26,9 +36,12 @@ config SND_PXA_SOC_SSP
tristate
select PXA_SSP
+config SND_MMP_SOC_SSPA
+ tristate
+
config SND_PXA2XX_SOC_CORGI
tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
- depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
+ depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_WM8731
help
@@ -37,7 +50,7 @@ config SND_PXA2XX_SOC_CORGI
config SND_PXA2XX_SOC_SPITZ
tristate "SoC Audio support for Sharp Zaurus SL-Cxx00"
- depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00
+ depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_WM8750
help
@@ -46,7 +59,7 @@ config SND_PXA2XX_SOC_SPITZ
config SND_PXA2XX_SOC_Z2
tristate "SoC Audio support for Zipit Z2"
- depends on SND_PXA2XX_SOC && MACH_ZIPIT2
+ depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_WM8750
help
@@ -54,7 +67,7 @@ config SND_PXA2XX_SOC_Z2
config SND_PXA2XX_SOC_POODLE
tristate "SoC Audio support for Poodle"
- depends on SND_PXA2XX_SOC && MACH_POODLE
+ depends on SND_PXA2XX_SOC && MACH_POODLE && I2C
select SND_PXA2XX_SOC_I2S
select SND_SOC_WM8731
help
@@ -118,25 +131,25 @@ config SND_PXA2XX_SOC_PALM27X
Say Y if you want to add support for SoC audio on
Palm T|X, T5, E2 or LifeDrive handheld computer.
-config SND_SOC_SAARB
- tristate "SoC Audio support for Marvell Saarb"
- depends on SND_PXA2XX_SOC && MACH_SAARB
- select MFD_88PM860X
- select SND_PXA_SOC_SSP
- select SND_SOC_88PM860X
+config SND_PXA910_SOC
+ tristate "SoC Audio for Marvell PXA910 chip"
+ depends on ARCH_MMP && SND
+ select SND_PCM
help
Say Y if you want to add support for SoC audio on the
- Marvell Saarb reference platform.
+ Marvell PXA910 reference platform.
-config SND_SOC_TAVOREVB3
- tristate "SoC Audio support for Marvell Tavor EVB3"
- depends on SND_PXA2XX_SOC && MACH_TAVOREVB3
- select MFD_88PM860X
+config SND_SOC_TTC_DKB
+ bool "SoC Audio support for TTC DKB"
+ depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y
+ select PXA_SSP
select SND_PXA_SOC_SSP
+ select SND_MMP_SOC
+ select MFD_88PM860X
select SND_SOC_88PM860X
help
- Say Y if you want to add support for SoC audio on the
- Marvell Saarb reference platform.
+ Say Y if you want to add support for SoC audio on TTC DKB
+
config SND_SOC_ZYLONITE
tristate "SoC Audio support for Marvell Zylonite"
@@ -169,7 +182,7 @@ config SND_PXA2XX_SOC_HX4700
config SND_PXA2XX_SOC_MAGICIAN
tristate "SoC Audio support for HTC Magician"
- depends on SND_PXA2XX_SOC && MACH_MAGICIAN
+ depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C
select SND_PXA2XX_SOC_I2S
select SND_PXA_SOC_SSP
select SND_SOC_UDA1380
@@ -194,3 +207,13 @@ config SND_PXA2XX_SOC_IMOTE2
help
Say Y if you want to add support for SoC audio on the
IMote 2.
+
+config SND_MMP_SOC_BROWNSTONE
+ tristate "SoC Audio support for Marvell Brownstone"
+ depends on SND_MMP_SOC && MACH_BROWNSTONE
+ select SND_MMP_SOC_SSPA
+ select MFD_WM8994
+ select SND_SOC_WM8994
+ help
+ Say Y if you want to add support for SoC audio on the
+ Marvell Brownstone reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index af357623be9..2cff67b61dc 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -3,11 +3,15 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o
snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
snd-soc-pxa-ssp-objs := pxa-ssp.o
+snd-soc-mmp-objs := mmp-pcm.o
+snd-soc-mmp-sspa-objs := mmp-sspa.o
obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
+obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o
+obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o
# PXA Machine Support
snd-soc-corgi-objs := corgi.o
@@ -19,8 +23,6 @@ snd-soc-e800-objs := e800_wm9712.o
snd-soc-spitz-objs := spitz.o
snd-soc-em-x270-objs := em-x270.o
snd-soc-palm27x-objs := palm27x.o
-snd-soc-saarb-objs := saarb.o
-snd-soc-tavorevb3-objs := tavorevb3.o
snd-soc-zylonite-objs := zylonite.o
snd-soc-hx4700-objs := hx4700.o
snd-soc-magician-objs := magician.o
@@ -28,6 +30,8 @@ snd-soc-mioa701-objs := mioa701_wm9713.o
snd-soc-z2-objs := z2.o
snd-soc-imote2-objs := imote2.o
snd-soc-raumfeld-objs := raumfeld.o
+snd-soc-brownstone-objs := brownstone.o
+snd-soc-ttc-dkb-objs := ttc-dkb.o
obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -42,8 +46,8 @@ obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o
obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o
-obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o
-obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
+obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o
+obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
new file mode 100644
index 00000000000..c8dd53f9c35
--- /dev/null
+++ b/sound/soc/pxa/brownstone.c
@@ -0,0 +1,169 @@
+/*
+ * linux/sound/soc/pxa/brownstone.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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 <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8994.h"
+#include "mmp-sspa.h"
+
+static const struct snd_kcontrol_new brownstone_dapm_control[] = {
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Main Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route brownstone_audio_map[] = {
+ {"Ext Spk", NULL, "SPKOUTLP"},
+ {"Ext Spk", NULL, "SPKOUTLN"},
+ {"Ext Spk", NULL, "SPKOUTRP"},
+ {"Ext Spk", NULL, "SPKOUTRN"},
+
+ {"Headset Stereophone", NULL, "HPOUT1L"},
+ {"Headset Stereophone", NULL, "HPOUT1R"},
+
+ {"IN1RN", NULL, "Headset Mic"},
+
+ {"DMIC1DAT", NULL, "MICBIAS1"},
+ {"MICBIAS1", NULL, "Main Mic"},
+};
+
+static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ /* set endpoints to not connected */
+ snd_soc_dapm_nc_pin(dapm, "HPOUT2P");
+ snd_soc_dapm_nc_pin(dapm, "HPOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+ snd_soc_dapm_nc_pin(dapm, "IN1LN");
+ snd_soc_dapm_nc_pin(dapm, "IN1LP");
+ snd_soc_dapm_nc_pin(dapm, "IN1RP");
+ snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RN");
+ snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+ snd_soc_dapm_nc_pin(dapm, "IN2LN");
+
+ return 0;
+}
+
+static int brownstone_wm8994_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;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ int freq_out, sspa_mclk, sysclk;
+ int sspa_div;
+
+ if (params_rate(params) > 11025) {
+ freq_out = params_rate(params) * 512;
+ sysclk = params_rate(params) * 256;
+ sspa_mclk = params_rate(params) * 64;
+ } else {
+ freq_out = params_rate(params) * 1024;
+ sysclk = params_rate(params) * 512;
+ sspa_mclk = params_rate(params) * 64;
+ }
+ sspa_div = freq_out;
+ do_div(sspa_div, sspa_mclk);
+
+ snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0);
+ snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);
+ snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk);
+
+ /* set wm8994 sysclk */
+ snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0);
+
+ return 0;
+}
+
+/* machine stream operations */
+static struct snd_soc_ops brownstone_ops = {
+ .hw_params = brownstone_wm8994_hw_params,
+};
+
+static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
+{
+ .name = "WM8994",
+ .stream_name = "WM8994 HiFi",
+ .cpu_dai_name = "mmp-sspa-dai.0",
+ .codec_dai_name = "wm8994-aif1",
+ .platform_name = "mmp-pcm-audio",
+ .codec_name = "wm8994-codec",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ops = &brownstone_ops,
+ .init = brownstone_wm8994_init,
+},
+};
+
+/* audio machine driver */
+static struct snd_soc_card brownstone = {
+ .name = "brownstone",
+ .owner = THIS_MODULE,
+ .dai_link = brownstone_wm8994_dai,
+ .num_links = ARRAY_SIZE(brownstone_wm8994_dai),
+
+ .controls = brownstone_dapm_control,
+ .num_controls = ARRAY_SIZE(brownstone_dapm_control),
+ .dapm_widgets = brownstone_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets),
+ .dapm_routes = brownstone_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map),
+};
+
+static int brownstone_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ brownstone.dev = &pdev->dev;
+ ret = snd_soc_register_card(&brownstone);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
+ return ret;
+}
+
+static int brownstone_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&brownstone);
+ return 0;
+}
+
+static struct platform_driver mmp_driver = {
+ .driver = {
+ .name = "brownstone-audio",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = brownstone_probe,
+ .remove = brownstone_remove,
+};
+
+module_platform_driver(mmp_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC Brownstone");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 863367ad89c..5a88136aa80 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -47,64 +47,63 @@ static int corgi_spk_func;
static void corgi_ext_control(struct snd_soc_dapm_context *dapm)
{
+ snd_soc_dapm_mutex_lock(dapm);
+
/* set up jack connection */
switch (corgi_jack_func) {
case CORGI_HP:
/* set = unmute headphone */
gpio_set_value(CORGI_GPIO_MUTE_L, 1);
gpio_set_value(CORGI_GPIO_MUTE_R, 1);
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case CORGI_MIC:
/* reset = mute headphone */
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 0);
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case CORGI_LINE:
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 0);
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_enable_pin(dapm, "Line Jack");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case CORGI_HEADSET:
gpio_set_value(CORGI_GPIO_MUTE_L, 0);
gpio_set_value(CORGI_GPIO_MUTE_R, 1);
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
break;
}
if (corgi_spk_func == CORGI_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
/* signal a DAPM event */
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int corgi_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- corgi_ext_control(&codec->dapm);
-
- mutex_unlock(&codec->mutex);
+ corgi_ext_control(&rtd->card->dapm);
return 0;
}
@@ -303,7 +302,7 @@ static struct snd_soc_card corgi = {
.num_dapm_routes = ARRAY_SIZE(corgi_audio_map),
};
-static int __devinit corgi_probe(struct platform_device *pdev)
+static int corgi_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &corgi;
int ret;
@@ -317,7 +316,7 @@ static int __devinit corgi_probe(struct platform_device *pdev)
return ret;
}
-static int __devexit corgi_remove(struct platform_device *pdev)
+static int corgi_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
@@ -329,9 +328,10 @@ static struct platform_driver corgi_driver = {
.driver = {
.name = "corgi-audio",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
.probe = corgi_probe,
- .remove = __devexit_p(corgi_remove),
+ .remove = corgi_remove,
};
module_platform_driver(corgi_driver);
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 7b1bc239003..c29fedab2f4 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -103,11 +103,6 @@ static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "PCBEEP");
snd_soc_dapm_nc_pin(dapm, "MIC2");
- snd_soc_dapm_new_controls(dapm, e740_dapm_widgets,
- ARRAY_SIZE(e740_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -136,6 +131,11 @@ static struct snd_soc_card e740 = {
.owner = THIS_MODULE,
.dai_link = e740_dai,
.num_links = ARRAY_SIZE(e740_dai),
+
+ .dapm_widgets = e740_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e740_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct gpio e740_audio_gpios[] = {
@@ -144,7 +144,7 @@ static struct gpio e740_audio_gpios[] = {
{ GPIO_E740_WM9705_nAVDD2, GPIOF_OUT_INIT_HIGH, "Audio power" },
};
-static int __devinit e740_probe(struct platform_device *pdev)
+static int e740_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &e740;
int ret;
@@ -165,7 +165,7 @@ static int __devinit e740_probe(struct platform_device *pdev)
return ret;
}
-static int __devexit e740_remove(struct platform_device *pdev)
+static int e740_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
@@ -178,9 +178,10 @@ static struct platform_driver e740_driver = {
.driver = {
.name = "e740-audio",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
.probe = e740_probe,
- .remove = __devexit_p(e740_remove),
+ .remove = e740_remove,
};
module_platform_driver(e740_driver);
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 47b89d71e28..ee36aba8806 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -85,11 +85,6 @@ static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "PCBEEP");
snd_soc_dapm_nc_pin(dapm, "MIC2");
- snd_soc_dapm_new_controls(dapm, e750_dapm_widgets,
- ARRAY_SIZE(e750_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -119,6 +114,11 @@ static struct snd_soc_card e750 = {
.owner = THIS_MODULE,
.dai_link = e750_dai,
.num_links = ARRAY_SIZE(e750_dai),
+
+ .dapm_widgets = e750_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e750_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct gpio e750_audio_gpios[] = {
@@ -126,7 +126,7 @@ static struct gpio e750_audio_gpios[] = {
{ GPIO_E750_SPK_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" },
};
-static int __devinit e750_probe(struct platform_device *pdev)
+static int e750_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &e750;
int ret;
@@ -147,7 +147,7 @@ static int __devinit e750_probe(struct platform_device *pdev)
return ret;
}
-static int __devexit e750_remove(struct platform_device *pdev)
+static int e750_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
@@ -160,9 +160,10 @@ static struct platform_driver e750_driver = {
.driver = {
.name = "e750-audio",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
.probe = e750_probe,
- .remove = __devexit_p(e750_remove),
+ .remove = e750_remove,
};
module_platform_driver(e750_driver);
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index ea9707ec6f2..24c2078ce70 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -71,19 +71,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
{"MIC2", NULL, "Mic (Internal2)"},
};
-static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- snd_soc_dapm_new_controls(dapm, e800_dapm_widgets,
- ARRAY_SIZE(e800_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- return 0;
-}
-
static struct snd_soc_dai_link e800_dai[] = {
{
.name = "AC97",
@@ -92,7 +79,6 @@ static struct snd_soc_dai_link e800_dai[] = {
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
- .init = e800_ac97_init,
},
{
.name = "AC97 Aux",
@@ -109,6 +95,11 @@ static struct snd_soc_card e800 = {
.owner = THIS_MODULE,
.dai_link = e800_dai,
.num_links = ARRAY_SIZE(e800_dai),
+
+ .dapm_widgets = e800_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(e800_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct gpio e800_audio_gpios[] = {
@@ -116,7 +107,7 @@ static struct gpio e800_audio_gpios[] = {
{ GPIO_E800_HP_AMP_OFF, GPIOF_OUT_INIT_HIGH, "Speaker amp" },
};
-static int __devinit e800_probe(struct platform_device *pdev)
+static int e800_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &e800;
int ret;
@@ -137,7 +128,7 @@ static int __devinit e800_probe(struct platform_device *pdev)
return ret;
}
-static int __devexit e800_remove(struct platform_device *pdev)
+static int e800_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
@@ -150,9 +141,10 @@ static struct platform_driver e800_driver = {
.driver = {
.name = "e800-audio",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
.probe = e800_probe,
- .remove = __devexit_p(e800_remove),
+ .remove = e800_remove,
};
module_platform_driver(e800_driver);
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
index 2a342c92d82..05559a725be 100644
--- a/sound/soc/pxa/hx4700.c
+++ b/sound/soc/pxa/hx4700.c
@@ -152,6 +152,13 @@ static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd)
return err;
}
+static int hx4700_card_remove(struct snd_soc_card *card)
+{
+ snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio);
+
+ return 0;
+}
+
/* hx4700 digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link hx4700_dai = {
.name = "ak4641",
@@ -170,6 +177,7 @@ static struct snd_soc_dai_link hx4700_dai = {
static struct snd_soc_card snd_soc_card_hx4700 = {
.name = "iPAQ hx4700",
.owner = THIS_MODULE,
+ .remove = hx4700_card_remove,
.dai_link = &hx4700_dai,
.num_links = 1,
.dapm_widgets = hx4700_dapm_widgets,
@@ -183,7 +191,7 @@ static struct gpio hx4700_audio_gpios[] = {
{ GPIO92_HX4700_HP_DRIVER, GPIOF_OUT_INIT_LOW, "EP_POWER" },
};
-static int __devinit hx4700_audio_probe(struct platform_device *pdev)
+static int hx4700_audio_probe(struct platform_device *pdev)
{
int ret;
@@ -204,9 +212,8 @@ static int __devinit hx4700_audio_probe(struct platform_device *pdev)
return ret;
}
-static int __devexit hx4700_audio_remove(struct platform_device *pdev)
+static int hx4700_audio_remove(struct platform_device *pdev)
{
- snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio);
snd_soc_unregister_card(&snd_soc_card_hx4700);
gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0);
@@ -223,7 +230,7 @@ static struct platform_driver hx4700_audio_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = hx4700_audio_probe,
- .remove = __devexit_p(hx4700_audio_remove),
+ .remove = hx4700_audio_remove,
};
module_platform_driver(hx4700_audio_driver);
diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c
index b93dafd32b8..fd2f4eda1fd 100644
--- a/sound/soc/pxa/imote2.c
+++ b/sound/soc/pxa/imote2.c
@@ -65,7 +65,7 @@ static struct snd_soc_card imote2 = {
.num_links = 1,
};
-static int __devinit imote2_probe(struct platform_device *pdev)
+static int imote2_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &imote2;
int ret;
@@ -79,7 +79,7 @@ static int __devinit imote2_probe(struct platform_device *pdev)
return ret;
}
-static int __devexit imote2_remove(struct platform_device *pdev)
+static int imote2_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
@@ -91,9 +91,10 @@ static struct platform_driver imote2_driver = {
.driver = {
.name = "imote2-audio",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
.probe = imote2_probe,
- .remove = __devexit_p(imote2_remove),
+ .remove = imote2_remove,
};
module_platform_driver(imote2_driver);
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index aace19e0fe2..259e048681c 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -41,44 +41,42 @@ static int magician_hp_switch;
static int magician_spk_switch = 1;
static int magician_in_sel = MAGICIAN_MIC;
-static void magician_ext_control(struct snd_soc_codec *codec)
+static void magician_ext_control(struct snd_soc_dapm_context *dapm)
{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_mutex_lock(dapm);
if (magician_spk_switch)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
if (magician_hp_switch)
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
switch (magician_in_sel) {
case MAGICIAN_MIC:
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_enable_pin(dapm, "Call Mic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic");
break;
case MAGICIAN_MIC_EXT:
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic");
break;
}
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int magician_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- magician_ext_control(codec);
-
- mutex_unlock(&codec->mutex);
+ magician_ext_control(&rtd->card->dapm);
return 0;
}
@@ -277,13 +275,13 @@ static int magician_get_hp(struct snd_kcontrol *kcontrol,
static int magician_set_hp(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 (magician_hp_switch == ucontrol->value.integer.value[0])
return 0;
magician_hp_switch = ucontrol->value.integer.value[0];
- magician_ext_control(codec);
+ magician_ext_control(&card->dapm);
return 1;
}
@@ -297,13 +295,13 @@ static int magician_get_spk(struct snd_kcontrol *kcontrol,
static int magician_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 (magician_spk_switch == ucontrol->value.integer.value[0])
return 0;
magician_spk_switch = ucontrol->value.integer.value[0];
- magician_ext_control(codec);
+ magician_ext_control(&card->dapm);
return 1;
}
@@ -400,7 +398,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
/* NC codec pins */
snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
@@ -410,19 +407,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "VINL");
snd_soc_dapm_nc_pin(dapm, "VINR");
- /* Add magician specific controls */
- err = snd_soc_add_codec_controls(codec, uda1380_magician_controls,
- ARRAY_SIZE(uda1380_magician_controls));
- if (err < 0)
- return err;
-
- /* Add magician specific widgets */
- snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
- ARRAY_SIZE(uda1380_dapm_widgets));
-
- /* Set up magician specific audio path interconnects */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -456,6 +440,12 @@ static struct snd_soc_card snd_soc_card_magician = {
.dai_link = magician_dai,
.num_links = ARRAY_SIZE(magician_dai),
+ .controls = uda1380_magician_controls,
+ .num_controls = ARRAY_SIZE(uda1380_magician_controls),
+ .dapm_widgets = uda1380_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *magician_snd_device;
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 9c585af59b5..595eee341e9 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -56,8 +56,6 @@
#include "pxa2xx-ac97.h"
#include "../codecs/wm9713.h"
-#define ARRAY_AND_SIZE(x) (x), ARRAY_SIZE(x)
-
#define AC97_GPIO_PULL 0x58
/* Use GPIO8 for rear speaker amplifier */
@@ -129,15 +127,8 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
unsigned short reg;
- /* Add mioa701 specific widgets */
- snd_soc_dapm_new_controls(dapm, ARRAY_AND_SIZE(mioa701_dapm_widgets));
-
- /* Set up mioa701 specific audio path audio_mapnects */
- snd_soc_dapm_add_routes(dapm, ARRAY_AND_SIZE(audio_map));
-
/* Prepare GPIO8 for rear speaker amplifier */
reg = codec->driver->read(codec, AC97_GPIO_CFG);
codec->driver->write(codec, AC97_GPIO_CFG, reg | 0x0100);
@@ -146,12 +137,6 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
reg = codec->driver->read(codec, AC97_3D_CONTROL);
codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000);
- snd_soc_dapm_enable_pin(dapm, "Front Speaker");
- snd_soc_dapm_enable_pin(dapm, "Rear Speaker");
- snd_soc_dapm_enable_pin(dapm, "Front Mic");
- snd_soc_dapm_enable_pin(dapm, "GSM Line In");
- snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
-
return 0;
}
@@ -184,47 +169,44 @@ static struct snd_soc_card mioa701 = {
.owner = THIS_MODULE,
.dai_link = mioa701_dai,
.num_links = ARRAY_SIZE(mioa701_dai),
-};
-static struct platform_device *mioa701_snd_device;
+ .dapm_widgets = mioa701_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mioa701_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
static int mioa701_wm9713_probe(struct platform_device *pdev)
{
- int ret;
+ int rc;
if (!machine_is_mioa701())
return -ENODEV;
- dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
- "lead to overheating and possible destruction of your device."
- "Do not use without a good knowledge of mio's board design!\n");
-
- mioa701_snd_device = platform_device_alloc("soc-audio", -1);
- if (!mioa701_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(mioa701_snd_device, &mioa701);
-
- ret = platform_device_add(mioa701_snd_device);
- if (!ret)
- return 0;
-
- platform_device_put(mioa701_snd_device);
- return ret;
+ mioa701.dev = &pdev->dev;
+ rc = snd_soc_register_card(&mioa701);
+ if (!rc)
+ dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+ "lead to overheating and possible destruction of your device."
+ " Do not use without a good knowledge of mio's board design!\n");
+ return rc;
}
-static int __devexit mioa701_wm9713_remove(struct platform_device *pdev)
+static int mioa701_wm9713_remove(struct platform_device *pdev)
{
- platform_device_unregister(mioa701_snd_device);
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
return 0;
}
static struct platform_driver mioa701_wm9713_driver = {
.probe = mioa701_wm9713_probe,
- .remove = __devexit_p(mioa701_wm9713_remove),
+ .remove = mioa701_wm9713_remove,
.driver = {
.name = "mioa701-wm9713",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
};
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
new file mode 100644
index 00000000000..5e8d8133017
--- /dev/null
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -0,0 +1,257 @@
+/*
+ * linux/sound/soc/pxa/mmp-pcm.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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 <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/platform_data/dma-mmp_tdma.h>
+#include <linux/platform_data/mmp_audio.h>
+
+#include <sound/pxa2xx-lib.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+struct mmp_dma_data {
+ int ssp_id;
+ struct resource *dma_res;
+};
+
+#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \
+ SNDRV_PCM_INFO_MMAP_VALID | \
+ SNDRV_PCM_INFO_INTERLEAVED | \
+ SNDRV_PCM_INFO_PAUSE | \
+ SNDRV_PCM_INFO_RESUME)
+
+static struct snd_pcm_hardware mmp_pcm_hardware[] = {
+ {
+ .info = MMP_PCM_INFO,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 2048,
+ .periods_min = 2,
+ .periods_max = 32,
+ .buffer_bytes_max = 4096,
+ .fifo_size = 32,
+ },
+ {
+ .info = MMP_PCM_INFO,
+ .period_bytes_min = 1024,
+ .period_bytes_max = 2048,
+ .periods_min = 2,
+ .periods_max = 32,
+ .buffer_bytes_max = 4096,
+ .fifo_size = 32,
+ },
+};
+
+static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+ struct dma_slave_config slave_config;
+ int ret;
+
+ ret =
+ snd_dmaengine_pcm_prepare_slave_config(substream, params,
+ &slave_config);
+ if (ret)
+ return ret;
+
+ ret = dmaengine_slave_config(chan, &slave_config);
+ if (ret)
+ return ret;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+ struct mmp_dma_data *dma_data = param;
+ bool found = false;
+ char *devname;
+
+ devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name,
+ dma_data->ssp_id);
+ if ((strcmp(dev_name(chan->device->dev), devname) == 0) &&
+ (chan->chan_id == dma_data->dma_res->start)) {
+ found = true;
+ }
+
+ kfree(devname);
+ return found;
+}
+
+static int mmp_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct mmp_dma_data dma_data;
+ struct resource *r;
+
+ r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream);
+ if (!r)
+ return -EBUSY;
+
+ snd_soc_set_runtime_hwparams(substream,
+ &mmp_pcm_hardware[substream->stream]);
+
+ dma_data.dma_res = r;
+ dma_data.ssp_id = cpu_dai->id;
+
+ return snd_dmaengine_pcm_open_request_chan(substream, filter,
+ &dma_data);
+}
+
+static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned long off = vma->vm_pgoff;
+
+ vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+ return remap_pfn_range(vma, vma->vm_start,
+ __phys_to_pfn(runtime->dma_addr) + off,
+ vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+static struct snd_pcm_ops mmp_pcm_ops = {
+ .open = mmp_pcm_open,
+ .close = snd_dmaengine_pcm_close_release_chan,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = mmp_pcm_hw_params,
+ .trigger = snd_dmaengine_pcm_trigger,
+ .pointer = snd_dmaengine_pcm_pointer,
+ .mmap = mmp_pcm_mmap,
+};
+
+static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+ struct gen_pool *gpool;
+
+ gpool = sram_get_gpool("asram");
+ if (!gpool)
+ return;
+
+ for (stream = 0; stream < 2; stream++) {
+ size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+ gen_pool_free(gpool, (unsigned long)buf->area, size);
+ buf->area = NULL;
+ }
+
+ return;
+}
+
+static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
+ int stream)
+{
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+ struct gen_pool *gpool;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = substream->pcm->card->dev;
+ buf->private_data = NULL;
+
+ gpool = sram_get_gpool("asram");
+ if (!gpool)
+ return -ENOMEM;
+
+ buf->area = gen_pool_dma_alloc(gpool, size, &buf->addr);
+ if (!buf->area)
+ return -ENOMEM;
+ buf->bytes = size;
+ return 0;
+}
+
+static int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret = 0, stream;
+
+ for (stream = 0; stream < 2; stream++) {
+ substream = pcm->streams[stream].substream;
+
+ ret = mmp_pcm_preallocate_dma_buffer(substream, stream);
+ if (ret)
+ goto err;
+ }
+
+ return 0;
+
+err:
+ mmp_pcm_free_dma_buffers(pcm);
+ return ret;
+}
+
+static struct snd_soc_platform_driver mmp_soc_platform = {
+ .ops = &mmp_pcm_ops,
+ .pcm_new = mmp_pcm_new,
+ .pcm_free = mmp_pcm_free_dma_buffers,
+};
+
+static int mmp_pcm_probe(struct platform_device *pdev)
+{
+ struct mmp_audio_platdata *pdata = pdev->dev.platform_data;
+
+ if (pdata) {
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max =
+ pdata->buffer_max_playback;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max =
+ pdata->period_max_playback;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max =
+ pdata->buffer_max_capture;
+ mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max =
+ pdata->period_max_capture;
+ }
+ return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform);
+}
+
+static int mmp_pcm_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver mmp_pcm_driver = {
+ .driver = {
+ .name = "mmp-pcm-audio",
+ .owner = THIS_MODULE,
+ },
+
+ .probe = mmp_pcm_probe,
+ .remove = mmp_pcm_remove,
+};
+
+module_platform_driver(mmp_pcm_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP Soc Audio DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
new file mode 100644
index 00000000000..5bf5f1f7cac
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -0,0 +1,485 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.c
+ * Base on pxa2xx-ssp.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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
+ *
+ */
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/slab.h>
+#include <linux/pxa2xx_ssp.h>
+#include <linux/io.h>
+#include <linux/dmaengine.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
+#include "mmp-sspa.h"
+
+/*
+ * SSPA audio private data
+ */
+struct sspa_priv {
+ struct ssp_device *sspa;
+ struct snd_dmaengine_dai_dma_data *dma_params;
+ struct clk *audio_clk;
+ struct clk *sysclk;
+ int dai_fmt;
+ int running_cnt;
+};
+
+static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val)
+{
+ __raw_writel(val, sspa->mmio_base + reg);
+}
+
+static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg)
+{
+ return __raw_readl(sspa->mmio_base + reg);
+}
+
+static void mmp_sspa_tx_enable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+ sspa_sp |= SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_tx_disable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+ sspa_sp &= ~SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_enable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+ sspa_sp |= SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_disable(struct ssp_device *sspa)
+{
+ unsigned int sspa_sp;
+
+ sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+ sspa_sp &= ~SSPA_SP_S_EN;
+ sspa_sp |= SSPA_SP_WEN;
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static int mmp_sspa_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_enable(priv->sysclk);
+ clk_enable(priv->sspa->clk);
+
+ return 0;
+}
+
+static void mmp_sspa_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+ clk_disable(priv->sspa->clk);
+ clk_disable(priv->sysclk);
+
+ return;
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+
+ switch (clk_id) {
+ case MMP_SSPA_CLK_AUDIO:
+ ret = clk_set_rate(priv->audio_clk, freq);
+ if (ret)
+ return ret;
+ break;
+ case MMP_SSPA_CLK_PLL:
+ case MMP_SSPA_CLK_VCXO:
+ /* not support yet */
+ return -EINVAL;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+ int source, unsigned int freq_in,
+ unsigned int freq_out)
+{
+ struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+ int ret = 0;
+
+ switch (pll_id) {
+ case MMP_SYSCLK:
+ ret = clk_set_rate(priv->sysclk, freq_out);
+ if (ret)
+ return ret;
+ break;
+ case MMP_SSPA_CLK:
+ ret = clk_set_rate(priv->sspa->clk, freq_out);
+ if (ret)
+ return ret;
+ break;
+ default:
+ return -ENODEV;
+ }
+
+ return 0;
+}
+
+/*
+ * Set up the sspa dai format. The sspa port must be inactive
+ * before calling this function as the physical
+ * interface format is changed.
+ */
+static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+ unsigned int fmt)
+{
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ u32 sspa_sp, sspa_ctrl;
+
+ /* check if we need to change anything at all */
+ if (sspa_priv->dai_fmt == fmt)
+ return 0;
+
+ /* we can only change the settings if the port is not in use */
+ if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) ||
+ (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) {
+ dev_err(&sspa->pdev->dev,
+ "can't change hardware dai format: stream is in use\n");
+ return -EINVAL;
+ }
+
+ /* reset port settings */
+ sspa_sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH;
+ sspa_ctrl = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ sspa_sp |= SSPA_SP_MSL;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ sspa_sp |= SSPA_SP_FSP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ sspa_sp |= SSPA_TXSP_FPER(63);
+ sspa_sp |= SSPA_SP_FWID(31);
+ sspa_ctrl |= SSPA_CTL_XDATDLY(1);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+ sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH);
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+ mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+ /*
+ * FIXME: hw issue, for the tx serial port,
+ * can not config the master/slave mode;
+ * so must clean this bit.
+ * The master/slave mode has been set in the
+ * rx port.
+ */
+ sspa_sp &= ~SSPA_SP_MSL;
+ mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+
+ mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+
+ /* Since we are configuring the timings for the format by hand
+ * we have to defer some things until hw_params() where we
+ * know parameters like the sample size.
+ */
+ sspa_priv->dai_fmt = fmt;
+ return 0;
+}
+
+/*
+ * Set the SSPA audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int mmp_sspa_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ struct snd_dmaengine_dai_dma_data *dma_params;
+ u32 sspa_ctrl;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL);
+ else
+ sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL);
+
+ sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK;
+ sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1);
+ sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK;
+ sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS);
+ sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1);
+ } else {
+ mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+ mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0);
+ }
+
+ dma_params = &sspa_priv->dma_params[substream->stream];
+ dma_params->addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ (sspa->phys_base + SSPA_TXD) :
+ (sspa->phys_base + SSPA_RXD);
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params);
+ return 0;
+}
+
+static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+ struct ssp_device *sspa = sspa_priv->sspa;
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ /*
+ * whatever playback or capture, must enable rx.
+ * this is a hw issue, so need check if rx has been
+ * enabled or not; if has been enabled by another
+ * stream, do not enable again.
+ */
+ if (!sspa_priv->running_cnt)
+ mmp_sspa_rx_enable(sspa);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mmp_sspa_tx_enable(sspa);
+
+ sspa_priv->running_cnt++;
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ sspa_priv->running_cnt--;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ mmp_sspa_tx_disable(sspa);
+
+ /* have no capture stream, disable rx port */
+ if (!sspa_priv->running_cnt)
+ mmp_sspa_rx_disable(sspa);
+ break;
+
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static int mmp_sspa_probe(struct snd_soc_dai *dai)
+{
+ struct sspa_priv *priv = dev_get_drvdata(dai->dev);
+
+ snd_soc_dai_set_drvdata(dai, priv);
+ return 0;
+
+}
+
+#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000
+#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
+ .startup = mmp_sspa_startup,
+ .shutdown = mmp_sspa_shutdown,
+ .trigger = mmp_sspa_trigger,
+ .hw_params = mmp_sspa_hw_params,
+ .set_sysclk = mmp_sspa_set_dai_sysclk,
+ .set_pll = mmp_sspa_set_dai_pll,
+ .set_fmt = mmp_sspa_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver mmp_sspa_dai = {
+ .probe = mmp_sspa_probe,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 128,
+ .rates = MMP_SSPA_RATES,
+ .formats = MMP_SSPA_FORMATS,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = MMP_SSPA_RATES,
+ .formats = MMP_SSPA_FORMATS,
+ },
+ .ops = &mmp_sspa_dai_ops,
+};
+
+static const struct snd_soc_component_driver mmp_sspa_component = {
+ .name = "mmp-sspa",
+};
+
+static int asoc_mmp_sspa_probe(struct platform_device *pdev)
+{
+ struct sspa_priv *priv;
+ struct resource *res;
+
+ priv = devm_kzalloc(&pdev->dev,
+ sizeof(struct sspa_priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->sspa = devm_kzalloc(&pdev->dev,
+ sizeof(struct ssp_device), GFP_KERNEL);
+ if (priv->sspa == NULL)
+ return -ENOMEM;
+
+ priv->dma_params = devm_kzalloc(&pdev->dev,
+ 2 * sizeof(struct snd_dmaengine_dai_dma_data),
+ GFP_KERNEL);
+ if (priv->dma_params == NULL)
+ return -ENOMEM;
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ priv->sspa->mmio_base = devm_ioremap_resource(&pdev->dev, res);
+ if (IS_ERR(priv->sspa->mmio_base))
+ return PTR_ERR(priv->sspa->mmio_base);
+
+ priv->sspa->clk = devm_clk_get(&pdev->dev, NULL);
+ if (IS_ERR(priv->sspa->clk))
+ return PTR_ERR(priv->sspa->clk);
+
+ priv->audio_clk = clk_get(NULL, "mmp-audio");
+ if (IS_ERR(priv->audio_clk))
+ return PTR_ERR(priv->audio_clk);
+
+ priv->sysclk = clk_get(NULL, "mmp-sysclk");
+ if (IS_ERR(priv->sysclk)) {
+ clk_put(priv->audio_clk);
+ return PTR_ERR(priv->sysclk);
+ }
+ clk_enable(priv->audio_clk);
+ priv->dai_fmt = (unsigned int) -1;
+ platform_set_drvdata(pdev, priv);
+
+ return devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component,
+ &mmp_sspa_dai, 1);
+}
+
+static int asoc_mmp_sspa_remove(struct platform_device *pdev)
+{
+ struct sspa_priv *priv = platform_get_drvdata(pdev);
+
+ clk_disable(priv->audio_clk);
+ clk_put(priv->audio_clk);
+ clk_put(priv->sysclk);
+ return 0;
+}
+
+static struct platform_driver asoc_mmp_sspa_driver = {
+ .driver = {
+ .name = "mmp-sspa-dai",
+ .owner = THIS_MODULE,
+ },
+ .probe = asoc_mmp_sspa_probe,
+ .remove = asoc_mmp_sspa_remove,
+};
+
+module_platform_driver(asoc_mmp_sspa_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP SSPA SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h
new file mode 100644
index 00000000000..ea365cb9e78
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.h
@@ -0,0 +1,92 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.h
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * 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 _MMP_SSPA_H
+#define _MMP_SSPA_H
+
+/*
+ * SSPA Registers
+ */
+#define SSPA_RXD (0x00)
+#define SSPA_RXID (0x04)
+#define SSPA_RXCTL (0x08)
+#define SSPA_RXSP (0x0c)
+#define SSPA_RXFIFO_UL (0x10)
+#define SSPA_RXINT_MASK (0x14)
+#define SSPA_RXC (0x18)
+#define SSPA_RXFIFO_NOFS (0x1c)
+#define SSPA_RXFIFO_SIZE (0x20)
+
+#define SSPA_TXD (0x80)
+#define SSPA_TXID (0x84)
+#define SSPA_TXCTL (0x88)
+#define SSPA_TXSP (0x8c)
+#define SSPA_TXFIFO_LL (0x90)
+#define SSPA_TXINT_MASK (0x94)
+#define SSPA_TXC (0x98)
+#define SSPA_TXFIFO_NOFS (0x9c)
+#define SSPA_TXFIFO_SIZE (0xa0)
+
+/* SSPA Control Register */
+#define SSPA_CTL_XPH (1 << 31) /* Read Phase */
+#define SSPA_CTL_XFIG (1 << 15) /* Transmit Zeros when FIFO Empty */
+#define SSPA_CTL_JST (1 << 3) /* Audio Sample Justification */
+#define SSPA_CTL_XFRLEN2_MASK (7 << 24)
+#define SSPA_CTL_XFRLEN2(x) ((x) << 24) /* Transmit Frame Length in Phase 2 */
+#define SSPA_CTL_XWDLEN2_MASK (7 << 21)
+#define SSPA_CTL_XWDLEN2(x) ((x) << 21) /* Transmit Word Length in Phase 2 */
+#define SSPA_CTL_XDATDLY(x) ((x) << 19) /* Tansmit Data Delay */
+#define SSPA_CTL_XSSZ2_MASK (7 << 16)
+#define SSPA_CTL_XSSZ2(x) ((x) << 16) /* Transmit Sample Audio Size */
+#define SSPA_CTL_XFRLEN1_MASK (7 << 8)
+#define SSPA_CTL_XFRLEN1(x) ((x) << 8) /* Transmit Frame Length in Phase 1 */
+#define SSPA_CTL_XWDLEN1_MASK (7 << 5)
+#define SSPA_CTL_XWDLEN1(x) ((x) << 5) /* Transmit Word Length in Phase 1 */
+#define SSPA_CTL_XSSZ1_MASK (7 << 0)
+#define SSPA_CTL_XSSZ1(x) ((x) << 0) /* XSSZ1 */
+
+#define SSPA_CTL_8_BITS (0x0) /* Sample Size */
+#define SSPA_CTL_12_BITS (0x1)
+#define SSPA_CTL_16_BITS (0x2)
+#define SSPA_CTL_20_BITS (0x3)
+#define SSPA_CTL_24_BITS (0x4)
+#define SSPA_CTL_32_BITS (0x5)
+
+/* SSPA Serial Port Register */
+#define SSPA_SP_WEN (1 << 31) /* Write Configuration Enable */
+#define SSPA_SP_MSL (1 << 18) /* Master Slave Configuration */
+#define SSPA_SP_CLKP (1 << 17) /* CLKP Polarity Clock Edge Select */
+#define SSPA_SP_FSP (1 << 16) /* FSP Polarity Clock Edge Select */
+#define SSPA_SP_FFLUSH (1 << 2) /* FIFO Flush */
+#define SSPA_SP_S_RST (1 << 1) /* Active High Reset Signal */
+#define SSPA_SP_S_EN (1 << 0) /* Serial Clock Domain Enable */
+#define SSPA_SP_FWID(x) ((x) << 20) /* Frame-Sync Width */
+#define SSPA_TXSP_FPER(x) ((x) << 4) /* Frame-Sync Active */
+
+/* sspa clock sources */
+#define MMP_SSPA_CLK_PLL 0
+#define MMP_SSPA_CLK_VCXO 1
+#define MMP_SSPA_CLK_AUDIO 3
+
+/* sspa pll id */
+#define MMP_SYSCLK 0
+#define MMP_SSPA_CLK 1
+
+#endif /* _MMP_SSPA_H */
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index db24bc685bd..17f9521ff6e 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -25,7 +25,7 @@
#include <asm/mach-types.h>
#include <mach/audio.h>
-#include <mach/palmasoc.h>
+#include <linux/platform_data/asoc-palm27x.h>
#include "../codecs/wm9712.h"
#include "pxa2xx-ac97.h"
@@ -79,25 +79,6 @@ static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
- /* add palm27x specific widgets */
- err = snd_soc_dapm_new_controls(dapm, palm27x_dapm_widgets,
- ARRAY_SIZE(palm27x_dapm_widgets));
- if (err)
- return err;
-
- /* set up palm27x specific audio path audio_map */
- err = snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
- if (err)
- return err;
-
- /* connected pins */
- if (machine_is_palmld())
- snd_soc_dapm_enable_pin(dapm, "MIC1");
- snd_soc_dapm_enable_pin(dapm, "HPOUTL");
- snd_soc_dapm_enable_pin(dapm, "HPOUTR");
- snd_soc_dapm_enable_pin(dapm, "LOUT2");
- snd_soc_dapm_enable_pin(dapm, "ROUT2");
-
/* not connected pins */
snd_soc_dapm_nc_pin(dapm, "OUT3");
snd_soc_dapm_nc_pin(dapm, "MONOOUT");
@@ -149,10 +130,12 @@ static struct snd_soc_card palm27x_asoc = {
.owner = THIS_MODULE,
.dai_link = palm27x_dai,
.num_links = ARRAY_SIZE(palm27x_dai),
+ .dapm_widgets = palm27x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(palm27x_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map)
};
-static struct platform_device *palm27x_snd_device;
-
static int palm27x_asoc_probe(struct platform_device *pdev)
{
int ret;
@@ -169,36 +152,28 @@ static int palm27x_asoc_probe(struct platform_device *pdev)
hs_jack_gpios[0].gpio = ((struct palm27x_asoc_info *)
(pdev->dev.platform_data))->jack_gpio;
- palm27x_snd_device = platform_device_alloc("soc-audio", -1);
- if (!palm27x_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(palm27x_snd_device, &palm27x_asoc);
- ret = platform_device_add(palm27x_snd_device);
-
- if (ret != 0)
- goto put_device;
-
- return 0;
-
-put_device:
- platform_device_put(palm27x_snd_device);
+ palm27x_asoc.dev = &pdev->dev;
+ ret = snd_soc_register_card(&palm27x_asoc);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+ ret);
return ret;
}
-static int __devexit palm27x_asoc_remove(struct platform_device *pdev)
+static int palm27x_asoc_remove(struct platform_device *pdev)
{
- platform_device_unregister(palm27x_snd_device);
+ snd_soc_unregister_card(&palm27x_asoc);
return 0;
}
static struct platform_driver palm27x_wm9712_driver = {
.probe = palm27x_asoc_probe,
- .remove = __devexit_p(palm27x_asoc_remove),
+ .remove = palm27x_asoc_remove,
.driver = {
.name = "palm27x-asoc",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
};
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index d2cc8173503..21f34006531 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -74,14 +74,9 @@ static void poodle_ext_control(struct snd_soc_dapm_context *dapm)
static int poodle_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- poodle_ext_control(&codec->dapm);
-
- mutex_unlock(&codec->mutex);
+ poodle_ext_control(&rtd->card->dapm);
return 0;
}
@@ -235,7 +230,6 @@ static int poodle_wm8731_init(struct snd_soc_pcm_runtime *rtd)
snd_soc_dapm_nc_pin(dapm, "LLINEIN");
snd_soc_dapm_nc_pin(dapm, "RLINEIN");
- snd_soc_dapm_enable_pin(dapm, "MICIN");
return 0;
}
@@ -269,7 +263,7 @@ static struct snd_soc_card poodle = {
.num_dapm_routes = ARRAY_SIZE(poodle_audio_map),
};
-static int __devinit poodle_probe(struct platform_device *pdev)
+static int poodle_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &poodle;
int ret;
@@ -291,7 +285,7 @@ static int __devinit poodle_probe(struct platform_device *pdev)
return ret;
}
-static int __devexit poodle_remove(struct platform_device *pdev)
+static int poodle_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
@@ -303,9 +297,10 @@ static struct platform_driver poodle_driver = {
.driver = {
.name = "poodle-audio",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
.probe = poodle_probe,
- .remove = __devexit_p(poodle_remove),
+ .remove = poodle_remove,
};
module_platform_driver(poodle_driver);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 4da5fc55c7e..199a8b37755 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -21,6 +21,8 @@
#include <linux/clk.h>
#include <linux/io.h>
#include <linux/pxa2xx_ssp.h>
+#include <linux/of.h>
+#include <linux/dmaengine.h>
#include <asm/irq.h>
@@ -30,9 +32,7 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
-
-#include <mach/hardware.h>
-#include <mach/dma.h>
+#include <sound/dmaengine_pcm.h>
#include "../../arm/pxa2xx-pcm.h"
#include "pxa-ssp.h"
@@ -79,27 +79,13 @@ static void pxa_ssp_disable(struct ssp_device *ssp)
__raw_writel(sscr0, ssp->mmio_base + SSCR0);
}
-struct pxa2xx_pcm_dma_data {
- struct pxa2xx_pcm_dma_params params;
- char name[20];
-};
-
static void pxa_ssp_set_dma_params(struct ssp_device *ssp, int width4,
- int out, struct pxa2xx_pcm_dma_params *dma_data)
+ int out, struct snd_dmaengine_dai_dma_data *dma)
{
- struct pxa2xx_pcm_dma_data *dma;
-
- dma = container_of(dma_data, struct pxa2xx_pcm_dma_data, params);
-
- snprintf(dma->name, 20, "SSP%d PCM %s %s", ssp->port_id,
- width4 ? "32-bit" : "16-bit", out ? "out" : "in");
-
- dma->params.name = dma->name;
- dma->params.drcmr = &DRCMR(out ? ssp->drcmr_tx : ssp->drcmr_rx);
- dma->params.dcmd = (out ? (DCMD_INCSRCADDR | DCMD_FLOWTRG) :
- (DCMD_INCTRGADDR | DCMD_FLOWSRC)) |
- (width4 ? DCMD_WIDTH4 : DCMD_WIDTH2) | DCMD_BURST16;
- dma->params.dev_addr = ssp->phys_base + SSDR;
+ dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES :
+ DMA_SLAVE_BUSWIDTH_2_BYTES;
+ dma->maxburst = 16;
+ dma->addr = ssp->phys_base + SSDR;
}
static int pxa_ssp_startup(struct snd_pcm_substream *substream,
@@ -107,7 +93,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
{
struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
struct ssp_device *ssp = priv->ssp;
- struct pxa2xx_pcm_dma_data *dma;
+ struct snd_dmaengine_dai_dma_data *dma;
int ret = 0;
if (!cpu_dai->active) {
@@ -115,10 +101,14 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
pxa_ssp_disable(ssp);
}
- dma = kzalloc(sizeof(struct pxa2xx_pcm_dma_data), GFP_KERNEL);
+ dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL);
if (!dma)
return -ENOMEM;
- snd_soc_dai_set_dma_data(cpu_dai, substream, &dma->params);
+
+ dma->filter_data = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+ &ssp->drcmr_tx : &ssp->drcmr_rx;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma);
return ret;
}
@@ -559,7 +549,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
u32 sspsp;
int width = snd_pcm_format_physical_width(params_format(params));
int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf;
- struct pxa2xx_pcm_dma_params *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
@@ -719,6 +709,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
static int pxa_ssp_probe(struct snd_soc_dai *dai)
{
+ struct device *dev = dai->dev;
struct ssp_priv *priv;
int ret;
@@ -726,10 +717,26 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
if (!priv)
return -ENOMEM;
- priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
- if (priv->ssp == NULL) {
- ret = -ENODEV;
- goto err_priv;
+ if (dev->of_node) {
+ struct device_node *ssp_handle;
+
+ ssp_handle = of_parse_phandle(dev->of_node, "port", 0);
+ if (!ssp_handle) {
+ dev_err(dev, "unable to get 'port' phandle\n");
+ return -ENODEV;
+ }
+
+ priv->ssp = pxa_ssp_request_of(ssp_handle, "SoC audio");
+ if (priv->ssp == NULL) {
+ ret = -ENODEV;
+ goto err_priv;
+ }
+ } else {
+ priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
+ if (priv->ssp == NULL) {
+ ret = -ENODEV;
+ goto err_priv;
+ }
}
priv->dai_fmt = (unsigned int) -1;
@@ -794,25 +801,38 @@ static struct snd_soc_dai_driver pxa_ssp_dai = {
.ops = &pxa_ssp_dai_ops,
};
-static __devinit int asoc_ssp_probe(struct platform_device *pdev)
+static const struct snd_soc_component_driver pxa_ssp_component = {
+ .name = "pxa-ssp",
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id pxa_ssp_of_ids[] = {
+ { .compatible = "mrvl,pxa-ssp-dai" },
+ {}
+};
+#endif
+
+static int asoc_ssp_probe(struct platform_device *pdev)
{
- return snd_soc_register_dai(&pdev->dev, &pxa_ssp_dai);
+ return snd_soc_register_component(&pdev->dev, &pxa_ssp_component,
+ &pxa_ssp_dai, 1);
}
-static int __devexit asoc_ssp_remove(struct platform_device *pdev)
+static int asoc_ssp_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
static struct platform_driver asoc_ssp_driver = {
.driver = {
- .name = "pxa-ssp-dai",
- .owner = THIS_MODULE,
+ .name = "pxa-ssp-dai",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(pxa_ssp_of_ids),
},
.probe = asoc_ssp_probe,
- .remove = __devexit_p(asoc_ssp_remove),
+ .remove = asoc_ssp_remove,
};
module_platform_driver(asoc_ssp_driver);
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 06ea2744cc8..ae956e3f4b9 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -14,15 +14,16 @@
#include <linux/io.h>
#include <linux/module.h>
#include <linux/platform_device.h>
+#include <linux/dmaengine.h>
#include <sound/core.h>
#include <sound/ac97_codec.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include <mach/hardware.h>
#include <mach/regs-ac97.h>
-#include <mach/dma.h>
#include <mach/audio.h>
#include "pxa2xx-ac97.h"
@@ -41,85 +42,58 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97)
pxa2xx_ac97_finish_reset(ac97);
}
-struct snd_ac97_bus_ops soc_ac97_ops = {
+static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
.read = pxa2xx_ac97_read,
.write = pxa2xx_ac97_write,
.warm_reset = pxa2xx_ac97_warm_reset,
.reset = pxa2xx_ac97_cold_reset,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = {
- .name = "AC97 PCM Stereo out",
- .dev_addr = __PREG(PCDR),
- .drcmr = &DRCMR(12),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
+ .addr = __PREG(PCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_ac97_pcm_stereo_in_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = {
- .name = "AC97 PCM Stereo in",
- .dev_addr = __PREG(PCDR),
- .drcmr = &DRCMR(11),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
+ .addr = __PREG(PCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_ac97_pcm_stereo_out_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = {
- .name = "AC97 Aux PCM (Slot 5) Mono out",
- .dev_addr = __PREG(MODR),
- .drcmr = &DRCMR(10),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST16 | DCMD_WIDTH2,
+static unsigned long pxa2xx_ac97_pcm_aux_mono_out_req = 10;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = {
+ .addr = __PREG(MODR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .maxburst = 16,
+ .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = {
- .name = "AC97 Aux PCM (Slot 5) Mono in",
- .dev_addr = __PREG(MODR),
- .drcmr = &DRCMR(9),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH2,
+static unsigned long pxa2xx_ac97_pcm_aux_mono_in_req = 9;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = {
+ .addr = __PREG(MODR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .maxburst = 16,
+ .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = {
- .name = "AC97 Mic PCM (Slot 6) Mono in",
- .dev_addr = __PREG(MCDR),
- .drcmr = &DRCMR(8),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST16 | DCMD_WIDTH2,
+static unsigned long pxa2xx_ac97_pcm_aux_mic_mono_req = 8;
+static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = {
+ .addr = __PREG(MCDR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES,
+ .maxburst = 16,
+ .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req,
};
-#ifdef CONFIG_PM
-static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai)
-{
- return pxa2xx_ac97_hw_suspend();
-}
-
-static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
-{
- return pxa2xx_ac97_hw_resume();
-}
-
-#else
-#define pxa2xx_ac97_suspend NULL
-#define pxa2xx_ac97_resume NULL
-#endif
-
-static int __devinit pxa2xx_ac97_probe(struct snd_soc_dai *dai)
-{
- return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev));
-}
-
-static int pxa2xx_ac97_remove(struct snd_soc_dai *dai)
-{
- pxa2xx_ac97_hw_remove(to_platform_device(dai->dev));
- return 0;
-}
-
static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai)
{
- struct pxa2xx_pcm_dma_params *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dma_data = &pxa2xx_ac97_pcm_stereo_out;
@@ -135,7 +109,7 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *cpu_dai)
{
- struct pxa2xx_pcm_dma_params *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dma_data = &pxa2xx_ac97_pcm_aux_mono_out;
@@ -184,10 +158,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = {
{
.name = "pxa2xx-ac97",
.ac97_control = 1,
- .probe = pxa2xx_ac97_probe,
- .remove = pxa2xx_ac97_remove,
- .suspend = pxa2xx_ac97_suspend,
- .resume = pxa2xx_ac97_resume,
.playback = {
.stream_name = "AC97 Playback",
.channels_min = 2,
@@ -232,35 +202,69 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = {
},
};
-EXPORT_SYMBOL_GPL(soc_ac97_ops);
+static const struct snd_soc_component_driver pxa_ac97_component = {
+ .name = "pxa-ac97",
+};
-static __devinit int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
+static int pxa2xx_ac97_dev_probe(struct platform_device *pdev)
{
+ int ret;
+
if (pdev->id != -1) {
dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n");
return -ENXIO;
}
+ ret = pxa2xx_ac97_hw_probe(pdev);
+ if (ret) {
+ dev_err(&pdev->dev, "PXA2xx AC97 hw probe error (%d)\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops);
+ if (ret != 0)
+ return ret;
+
/* Punt most of the init to the SoC probe; we may need the machine
* driver to do interesting things with the clocking to get us up
* and running.
*/
- return snd_soc_register_dais(&pdev->dev, pxa_ac97_dai_driver,
- ARRAY_SIZE(pxa_ac97_dai_driver));
+ return snd_soc_register_component(&pdev->dev, &pxa_ac97_component,
+ pxa_ac97_dai_driver, ARRAY_SIZE(pxa_ac97_dai_driver));
}
-static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev)
+static int pxa2xx_ac97_dev_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(pxa_ac97_dai_driver));
+ snd_soc_unregister_component(&pdev->dev);
+ snd_soc_set_ac97_ops(NULL);
+ pxa2xx_ac97_hw_remove(pdev);
return 0;
}
+#ifdef CONFIG_PM_SLEEP
+static int pxa2xx_ac97_dev_suspend(struct device *dev)
+{
+ return pxa2xx_ac97_hw_suspend();
+}
+
+static int pxa2xx_ac97_dev_resume(struct device *dev)
+{
+ return pxa2xx_ac97_hw_resume();
+}
+
+static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops,
+ pxa2xx_ac97_dev_suspend, pxa2xx_ac97_dev_resume);
+#endif
+
static struct platform_driver pxa2xx_ac97_driver = {
.probe = pxa2xx_ac97_dev_probe,
- .remove = __devexit_p(pxa2xx_ac97_dev_remove),
+ .remove = pxa2xx_ac97_dev_remove,
.driver = {
.name = "pxa2xx-ac97",
.owner = THIS_MODULE,
+#ifdef CONFIG_PM_SLEEP
+ .pm = &pxa2xx_ac97_pm_ops,
+#endif
},
};
diff --git a/sound/soc/pxa/pxa2xx-ac97.h b/sound/soc/pxa/pxa2xx-ac97.h
index eda891e6f31..a49c21ba384 100644
--- a/sound/soc/pxa/pxa2xx-ac97.h
+++ b/sound/soc/pxa/pxa2xx-ac97.h
@@ -14,7 +14,4 @@
#define PXA2XX_DAI_AC97_AUX 1
#define PXA2XX_DAI_AC97_MIC 2
-/* platform data */
-extern struct snd_ac97_bus_ops pxa2xx_ac97_ops;
-
#endif
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 3075a426124..c0d648d3339 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -23,9 +23,9 @@
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include <mach/hardware.h>
-#include <mach/dma.h>
#include <mach/audio.h>
#include "pxa2xx-i2s.h"
@@ -82,20 +82,20 @@ static struct pxa_i2s_port pxa_i2s;
static struct clk *clk_i2s;
static int clk_ena = 0;
-static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = {
- .name = "I2S PCM Stereo out",
- .dev_addr = __PREG(SADR),
- .drcmr = &DRCMR(3),
- .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = {
+ .addr = __PREG(SADR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_i2s_pcm_stereo_out_req,
};
-static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
- .name = "I2S PCM Stereo in",
- .dev_addr = __PREG(SADR),
- .drcmr = &DRCMR(2),
- .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC |
- DCMD_BURST32 | DCMD_WIDTH4,
+static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2;
+static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = {
+ .addr = __PREG(SADR),
+ .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
+ .maxburst = 32,
+ .filter_data = &pxa2xx_i2s_pcm_stereo_in_req,
};
static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
@@ -163,9 +163,10 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
- struct pxa2xx_pcm_dma_params *dma_data;
+ struct snd_dmaengine_dai_dma_data *dma_data;
- BUG_ON(IS_ERR(clk_i2s));
+ if (WARN_ON(IS_ERR(clk_i2s)))
+ return -EINVAL;
clk_prepare_enable(clk_i2s);
clk_ena = 1;
pxa_i2s_wait();
@@ -360,20 +361,25 @@ static struct snd_soc_dai_driver pxa_i2s_dai = {
.symmetric_rates = 1,
};
+static const struct snd_soc_component_driver pxa_i2s_component = {
+ .name = "pxa-i2s",
+};
+
static int pxa2xx_i2s_drv_probe(struct platform_device *pdev)
{
- return snd_soc_register_dai(&pdev->dev, &pxa_i2s_dai);
+ return snd_soc_register_component(&pdev->dev, &pxa_i2s_component,
+ &pxa_i2s_dai, 1);
}
-static int __devexit pxa2xx_i2s_drv_remove(struct platform_device *pdev)
+static int pxa2xx_i2s_drv_remove(struct platform_device *pdev)
{
- snd_soc_unregister_dai(&pdev->dev);
+ snd_soc_unregister_component(&pdev->dev);
return 0;
}
static struct platform_driver pxa2xx_i2s_driver = {
.probe = pxa2xx_i2s_drv_probe,
- .remove = __devexit_p(pxa2xx_i2s_drv_remove),
+ .remove = pxa2xx_i2s_drv_remove,
.driver = {
.name = "pxa2xx-i2s",
diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c
index fdd6bedef9b..42f2f017598 100644
--- a/sound/soc/pxa/pxa2xx-pcm.c
+++ b/sound/soc/pxa/pxa2xx-pcm.c
@@ -12,10 +12,15 @@
#include <linux/dma-mapping.h>
#include <linux/module.h>
+#include <linux/dmaengine.h>
+#include <linux/of.h>
+
+#include <mach/dma.h>
#include <sound/core.h>
#include <sound/soc.h>
#include <sound/pxa2xx-lib.h>
+#include <sound/dmaengine_pcm.h>
#include "../../arm/pxa2xx-pcm.h"
@@ -25,7 +30,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_runtime *runtime = substream->runtime;
struct pxa2xx_runtime_data *prtd = runtime->private_data;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct pxa2xx_pcm_dma_params *dma;
+ struct snd_dmaengine_dai_dma_data *dma;
int ret;
dma = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
@@ -39,7 +44,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
* with different params */
if (prtd->params == NULL) {
prtd->params = dma;
- ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+ ret = pxa_request_dma("name", DMA_PRIO_LOW,
pxa2xx_pcm_dma_irq, substream);
if (ret < 0)
return ret;
@@ -47,7 +52,7 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
} else if (prtd->params != dma) {
pxa_free_dma(prtd->dma_ch);
prtd->params = dma;
- ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW,
+ ret = pxa_request_dma("name", DMA_PRIO_LOW,
pxa2xx_pcm_dma_irq, substream);
if (ret < 0)
return ret;
@@ -84,18 +89,15 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = {
.mmap = pxa2xx_pcm_mmap,
};
-static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32);
-
static int pxa2xx_soc_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;
+ int ret;
- if (!card->dev->dma_mask)
- card->dev->dma_mask = &pxa2xx_pcm_dmamask;
- if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
ret = pxa2xx_pcm_preallocate_dma_buffer(pcm,
@@ -120,25 +122,33 @@ static struct snd_soc_platform_driver pxa2xx_soc_platform = {
.pcm_free = pxa2xx_pcm_free_dma_buffers,
};
-static int __devinit pxa2xx_soc_platform_probe(struct platform_device *pdev)
+static int pxa2xx_soc_platform_probe(struct platform_device *pdev)
{
return snd_soc_register_platform(&pdev->dev, &pxa2xx_soc_platform);
}
-static int __devexit pxa2xx_soc_platform_remove(struct platform_device *pdev)
+static int pxa2xx_soc_platform_remove(struct platform_device *pdev)
{
snd_soc_unregister_platform(&pdev->dev);
return 0;
}
+#ifdef CONFIG_OF
+static const struct of_device_id snd_soc_pxa_audio_match[] = {
+ { .compatible = "mrvl,pxa-pcm-audio" },
+ { }
+};
+#endif
+
static struct platform_driver pxa_pcm_driver = {
.driver = {
- .name = "pxa-pcm-audio",
- .owner = THIS_MODULE,
+ .name = "pxa-pcm-audio",
+ .owner = THIS_MODULE,
+ .of_match_table = of_match_ptr(snd_soc_pxa_audio_match),
},
.probe = pxa2xx_soc_platform_probe,
- .remove = __devexit_p(pxa2xx_soc_platform_remove),
+ .remove = pxa2xx_soc_platform_remove,
};
module_platform_driver(pxa_pcm_driver);
diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c
deleted file mode 100644
index c34146b776b..00000000000
--- a/sound/soc/pxa/saarb.c
+++ /dev/null
@@ -1,190 +0,0 @@
-/*
- * saarb.c -- SoC audio for saarb
- *
- * Copyright (C) 2010 Marvell International Ltd.
- * Haojian Zhuang <haojian.zhuang@marvell.com>
- *
- * 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 <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-#include <linux/clk.h>
-#include <linux/i2c.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-
-#include "../codecs/88pm860x-codec.h"
-#include "pxa-ssp.h"
-
-static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd);
-
-static struct platform_device *saarb_snd_device;
-
-static struct snd_soc_jack hs_jack, mic_jack;
-
-static struct snd_soc_jack_pin hs_jack_pins[] = {
- { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
-};
-
-static struct snd_soc_jack_pin mic_jack_pins[] = {
- { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
-};
-
-/* saarb machine dapm widgets */
-static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Stereophone", NULL),
- SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
- SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
- SND_SOC_DAPM_SPK("Ext Speaker", NULL),
- SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
-};
-
-/* saarb machine audio map */
-static const struct snd_soc_dapm_route saarb_audio_map[] = {
- {"Headset Stereophone", NULL, "HS1"},
- {"Headset Stereophone", NULL, "HS2"},
-
- {"Ext Speaker", NULL, "LSP"},
- {"Ext Speaker", NULL, "LSN"},
-
- {"Lineout Out 1", NULL, "LINEOUT1"},
- {"Lineout Out 2", NULL, "LINEOUT2"},
-
- {"MIC1P", NULL, "Mic1 Bias"},
- {"MIC1N", NULL, "Mic1 Bias"},
- {"Mic1 Bias", NULL, "Ext Mic 1"},
-
- {"MIC2P", NULL, "Mic1 Bias"},
- {"MIC2N", NULL, "Mic1 Bias"},
- {"Mic1 Bias", NULL, "Headset Mic 2"},
-
- {"MIC3P", NULL, "Mic3 Bias"},
- {"MIC3N", NULL, "Mic3 Bias"},
- {"Mic3 Bias", NULL, "Ext Mic 3"},
-};
-
-static int saarb_i2s_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;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int width = snd_pcm_format_physical_width(params_format(params));
- int ret;
-
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
- PM860X_CLK_DIR_OUT);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
-
- return ret;
-}
-
-static struct snd_soc_ops saarb_i2s_ops = {
- .hw_params = saarb_i2s_hw_params,
-};
-
-static struct snd_soc_dai_link saarb_dai[] = {
- {
- .name = "88PM860x I2S",
- .stream_name = "I2S Audio",
- .cpu_dai_name = "pxa-ssp-dai.1",
- .codec_dai_name = "88pm860x-i2s",
- .platform_name = "pxa-pcm-audio",
- .codec_name = "88pm860x-codec",
- .init = saarb_pm860x_init,
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &saarb_i2s_ops,
- },
-};
-
-static struct snd_soc_card snd_soc_card_saarb = {
- .name = "Saarb",
- .owner = THIS_MODULE,
- .dai_link = saarb_dai,
- .num_links = ARRAY_SIZE(saarb_dai),
-
- .dapm_widgets = saarb_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets),
- .dapm_routes = saarb_audio_map,
- .num_dapm_routes = ARRAY_SIZE(saarb_audio_map),
-};
-
-static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- /* connected pins */
- snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
- snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
- snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
- snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
-
- /* Headset jack detection */
- snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
- | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
- &hs_jack);
- snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
- hs_jack_pins);
- snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
- &mic_jack);
- snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
- mic_jack_pins);
-
- /* headphone, microphone detection & headset short detection */
- pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
- SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
- pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
- return 0;
-}
-
-static int __init saarb_init(void)
-{
- int ret;
-
- if (!machine_is_saarb())
- return -ENODEV;
- saarb_snd_device = platform_device_alloc("soc-audio", -1);
- if (!saarb_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb);
-
- ret = platform_device_add(saarb_snd_device);
- if (ret)
- platform_device_put(saarb_snd_device);
-
- return ret;
-}
-
-static void __exit saarb_exit(void)
-{
- platform_device_unregister(saarb_snd_device);
-}
-
-module_init(saarb_init);
-module_exit(saarb_exit);
-
-MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
-MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index fc052d8247f..1373b017a95 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -46,74 +46,74 @@ static int spitz_mic_gpio;
static void spitz_ext_control(struct snd_soc_dapm_context *dapm)
{
+ snd_soc_dapm_mutex_lock(dapm);
+
if (spitz_spk_func == SPITZ_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
/* set up jack connection */
switch (spitz_jack_func) {
case SPITZ_HP:
/* enable and unmute hp jack, disable mic bias */
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 1);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_MIC:
/* enable mic jack and bias, mute hp */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_LINE:
/* enable line jack, disable mic bias and mute hp */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_enable_pin(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
case SPITZ_HEADSET:
/* enable and unmute headset jack enable mic bias, mute L hp */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
- snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 1);
break;
case SPITZ_HP_OFF:
/* jack removed, everything off */
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
- snd_soc_dapm_disable_pin(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin(dapm, "Line Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
gpio_set_value(SPITZ_GPIO_MUTE_L, 0);
gpio_set_value(SPITZ_GPIO_MUTE_R, 0);
break;
}
- snd_soc_dapm_sync(dapm);
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int spitz_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- spitz_ext_control(&codec->dapm);
-
- mutex_unlock(&codec->mutex);
+ spitz_ext_control(&rtd->card->dapm);
return 0;
}
diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c
deleted file mode 100644
index 8b5ab8f7272..00000000000
--- a/sound/soc/pxa/tavorevb3.c
+++ /dev/null
@@ -1,189 +0,0 @@
-/*
- * tavorevb3.c -- SoC audio for Tavor EVB3
- *
- * Copyright (C) 2010 Marvell International Ltd.
- * Haojian Zhuang <haojian.zhuang@marvell.com>
- *
- * 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 <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-#include <linux/clk.h>
-#include <linux/i2c.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-
-#include "../codecs/88pm860x-codec.h"
-#include "pxa-ssp.h"
-
-static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd);
-
-static struct platform_device *evb3_snd_device;
-
-static struct snd_soc_jack hs_jack, mic_jack;
-
-static struct snd_soc_jack_pin hs_jack_pins[] = {
- { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
-};
-
-static struct snd_soc_jack_pin mic_jack_pins[] = {
- { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
-};
-
-/* tavorevb3 machine dapm widgets */
-static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headset Stereophone", NULL),
- SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
- SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
- SND_SOC_DAPM_SPK("Ext Speaker", NULL),
- SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
- SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
- SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
-};
-
-/* tavorevb3 machine audio map */
-static const struct snd_soc_dapm_route evb3_audio_map[] = {
- {"Headset Stereophone", NULL, "HS1"},
- {"Headset Stereophone", NULL, "HS2"},
-
- {"Ext Speaker", NULL, "LSP"},
- {"Ext Speaker", NULL, "LSN"},
-
- {"Lineout Out 1", NULL, "LINEOUT1"},
- {"Lineout Out 2", NULL, "LINEOUT2"},
-
- {"MIC1P", NULL, "Mic1 Bias"},
- {"MIC1N", NULL, "Mic1 Bias"},
- {"Mic1 Bias", NULL, "Ext Mic 1"},
-
- {"MIC2P", NULL, "Mic1 Bias"},
- {"MIC2N", NULL, "Mic1 Bias"},
- {"Mic1 Bias", NULL, "Headset Mic 2"},
-
- {"MIC3P", NULL, "Mic3 Bias"},
- {"MIC3N", NULL, "Mic3 Bias"},
- {"Mic3 Bias", NULL, "Ext Mic 3"},
-};
-
-static int evb3_i2s_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;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int width = snd_pcm_format_physical_width(params_format(params));
- int ret;
-
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
- PM860X_CLK_DIR_OUT);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
- if (ret < 0)
- return ret;
-
- ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
- return ret;
-}
-
-static struct snd_soc_ops evb3_i2s_ops = {
- .hw_params = evb3_i2s_hw_params,
-};
-
-static struct snd_soc_dai_link evb3_dai[] = {
- {
- .name = "88PM860x I2S",
- .stream_name = "I2S Audio",
- .cpu_dai_name = "pxa-ssp-dai.1",
- .codec_dai_name = "88pm860x-i2s",
- .platform_name = "pxa-pcm-audio",
- .codec_name = "88pm860x-codec",
- .init = evb3_pm860x_init,
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .ops = &evb3_i2s_ops,
- },
-};
-
-static struct snd_soc_card snd_soc_card_evb3 = {
- .name = "Tavor EVB3",
- .owner = THIS_MODULE,
- .dai_link = evb3_dai,
- .num_links = ARRAY_SIZE(evb3_dai),
-
- .dapm_widgets = evb3_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(evb3_dapm_widgets),
- .dapm_routes = evb3_audio_map,
- .num_dapm_routes = ARRAY_SIZE(evb3_audio_map),
-};
-
-static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- /* connected pins */
- snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
- snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
- snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
- snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
-
- /* Headset jack detection */
- snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
- | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
- &hs_jack);
- snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
- hs_jack_pins);
- snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
- &mic_jack);
- snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
- mic_jack_pins);
-
- /* headphone, microphone detection & headset short detection */
- pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
- SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
- pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
- return 0;
-}
-
-static int __init tavorevb3_init(void)
-{
- int ret;
-
- if (!machine_is_tavorevb3())
- return -ENODEV;
- evb3_snd_device = platform_device_alloc("soc-audio", -1);
- if (!evb3_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3);
-
- ret = platform_device_add(evb3_snd_device);
- if (ret)
- platform_device_put(evb3_snd_device);
-
- return ret;
-}
-
-static void __exit tavorevb3_exit(void)
-{
- platform_device_unregister(evb3_snd_device);
-}
-
-module_init(tavorevb3_init);
-module_exit(tavorevb3_exit);
-
-MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
-MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 2aec63f3706..4a956d1cb26 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -44,48 +44,46 @@
static int tosa_jack_func;
static int tosa_spk_func;
-static void tosa_ext_control(struct snd_soc_codec *codec)
+static void tosa_ext_control(struct snd_soc_dapm_context *dapm)
{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_mutex_lock(dapm);
/* set up jack connection */
switch (tosa_jack_func) {
case TOSA_HP:
- snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case TOSA_MIC_INT:
- snd_soc_dapm_enable_pin(dapm, "Mic (Internal)");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
break;
case TOSA_HEADSET:
- snd_soc_dapm_disable_pin(dapm, "Mic (Internal)");
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Headset Jack");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
break;
}
if (tosa_spk_func == TOSA_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Speaker");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
else
- snd_soc_dapm_disable_pin(dapm, "Speaker");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+
+ snd_soc_dapm_sync_unlocked(dapm);
- snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_mutex_unlock(dapm);
}
static int tosa_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_codec *codec = rtd->codec;
-
- mutex_lock(&codec->mutex);
/* check the jack status at stream startup */
- tosa_ext_control(codec);
-
- mutex_unlock(&codec->mutex);
+ tosa_ext_control(&rtd->card->dapm);
return 0;
}
@@ -104,13 +102,13 @@ static int tosa_get_jack(struct snd_kcontrol *kcontrol,
static int tosa_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 (tosa_jack_func == ucontrol->value.integer.value[0])
return 0;
tosa_jack_func = ucontrol->value.integer.value[0];
- tosa_ext_control(codec);
+ tosa_ext_control(&card->dapm);
return 1;
}
@@ -124,13 +122,13 @@ static int tosa_get_spk(struct snd_kcontrol *kcontrol,
static int tosa_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 (tosa_spk_func == ucontrol->value.integer.value[0])
return 0;
tosa_spk_func = ucontrol->value.integer.value[0];
- tosa_ext_control(codec);
+ tosa_ext_control(&card->dapm);
return 1;
}
@@ -191,24 +189,10 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
snd_soc_dapm_nc_pin(dapm, "OUT3");
snd_soc_dapm_nc_pin(dapm, "MONOOUT");
- /* add tosa specific controls */
- err = snd_soc_add_codec_controls(codec, tosa_controls,
- ARRAY_SIZE(tosa_controls));
- if (err < 0)
- return err;
-
- /* add tosa specific widgets */
- snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets,
- ARRAY_SIZE(tosa_dapm_widgets));
-
- /* set up tosa specific audio path audio_map */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
return 0;
}
@@ -239,9 +223,16 @@ static struct snd_soc_card tosa = {
.owner = THIS_MODULE,
.dai_link = tosa_dai,
.num_links = ARRAY_SIZE(tosa_dai),
+
+ .controls = tosa_controls,
+ .num_controls = ARRAY_SIZE(tosa_controls),
+ .dapm_widgets = tosa_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
-static int __devinit tosa_probe(struct platform_device *pdev)
+static int tosa_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &tosa;
int ret;
@@ -262,7 +253,7 @@ static int __devinit tosa_probe(struct platform_device *pdev)
return ret;
}
-static int __devexit tosa_remove(struct platform_device *pdev)
+static int tosa_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
@@ -275,9 +266,10 @@ static struct platform_driver tosa_driver = {
.driver = {
.name = "tosa-audio",
.owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
},
.probe = tosa_probe,
- .remove = __devexit_p(tosa_remove),
+ .remove = tosa_remove,
};
module_platform_driver(tosa_driver);
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
new file mode 100644
index 00000000000..9d7c5b7e953
--- /dev/null
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -0,0 +1,171 @@
+/*
+ * linux/sound/soc/pxa/ttc_dkb.c
+ *
+ * Copyright (C) 2012 Marvell International Ltd.
+ *
+ * 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
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <sound/pcm_params.h>
+#include "../codecs/88pm860x-codec.h"
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+ { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
+};
+
+/* ttc machine dapm widgets */
+static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+ SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+ SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+ SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+ SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+ SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* ttc machine audio map */
+static const struct snd_soc_dapm_route ttc_audio_map[] = {
+ {"Headset Stereophone", NULL, "HS1"},
+ {"Headset Stereophone", NULL, "HS2"},
+
+ {"Ext Speaker", NULL, "LSP"},
+ {"Ext Speaker", NULL, "LSN"},
+
+ {"Lineout Out 1", NULL, "LINEOUT1"},
+ {"Lineout Out 2", NULL, "LINEOUT2"},
+
+ {"MIC1P", NULL, "Mic1 Bias"},
+ {"MIC1N", NULL, "Mic1 Bias"},
+ {"Mic1 Bias", NULL, "Ext Mic 1"},
+
+ {"MIC2P", NULL, "Mic1 Bias"},
+ {"MIC2N", NULL, "Mic1 Bias"},
+ {"Mic1 Bias", NULL, "Headset Mic 2"},
+
+ {"MIC3P", NULL, "Mic3 Bias"},
+ {"MIC3N", NULL, "Mic3 Bias"},
+ {"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+ snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
+
+ /* Headset jack detection */
+ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+ | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+ &hs_jack);
+ snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+ hs_jack_pins);
+ snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+ &mic_jack);
+ snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+ mic_jack_pins);
+
+ /* headphone, microphone detection & headset short detection */
+ pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+ SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+ pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+
+ return 0;
+}
+
+/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
+{
+ .name = "88pm860x i2s",
+ .stream_name = "audio playback",
+ .codec_name = "88pm860x-codec",
+ .platform_name = "mmp-pcm-audio",
+ .cpu_dai_name = "pxa-ssp-dai.1",
+ .codec_dai_name = "88pm860x-i2s",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .init = ttc_pm860x_init,
+},
+};
+
+/* ttc/td audio machine driver */
+static struct snd_soc_card ttc_dkb_card = {
+ .name = "ttc-dkb-hifi",
+ .owner = THIS_MODULE,
+ .dai_link = ttc_pm860x_hifi_dai,
+ .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
+
+ .dapm_widgets = ttc_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets),
+ .dapm_routes = ttc_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(ttc_audio_map),
+};
+
+static int ttc_dkb_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &ttc_dkb_card;
+ int ret;
+
+ 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 int ttc_dkb_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+ snd_soc_unregister_card(card);
+
+ return 0;
+}
+
+static struct platform_driver ttc_dkb_driver = {
+ .driver = {
+ .name = "ttc-dkb-audio",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = ttc_dkb_probe,
+ .remove = ttc_dkb_remove,
+};
+
+module_platform_driver(ttc_dkb_driver);
+
+/* Module information */
+MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC TTC DKB");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:ttc-dkb-audio");
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index ceb656695b0..23bf991e95d 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -71,22 +71,10 @@ static const struct snd_soc_dapm_route audio_map[] = {
static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
if (clk_pout)
snd_soc_dai_set_pll(rtd->codec_dai, 0, 0,
clk_get_rate(pout), 0);
- snd_soc_dapm_new_controls(dapm, zylonite_dapm_widgets,
- ARRAY_SIZE(zylonite_dapm_widgets));
-
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- /* Static setup for now */
- snd_soc_dapm_enable_pin(dapm, "Headphone");
- snd_soc_dapm_enable_pin(dapm, "Headset Earpiece");
-
return 0;
}
@@ -256,7 +244,11 @@ static struct snd_soc_card zylonite = {
.resume_pre = &zylonite_resume_pre,
.dai_link = zylonite_dai,
.num_links = ARRAY_SIZE(zylonite_dai),
- .owner = THIS_MODULE,
+
+ .dapm_widgets = zylonite_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(zylonite_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *zylonite_snd_ac97_device;