July 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
by Liam Girdwood 20 Jul '11

20 Jul '11

In preparation for ASoC Dynamic PCM (AKA DSP) support. Provide convenience methods to retrieve the soc_card or snd_card from a DAPM context. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4205a97..aaef0ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,6 +124,36 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +/* get snd_card from DAPM context */ +static inline struct snd_card *dapm_get_snd_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card->snd_card; + else if (dapm->platform) + return dapm->platform->card->snd_card; + else + BUG(); + + /* unreachable */ + return NULL; +} + +/* get soc_card from DAPM context */ +static inline struct snd_soc_card *dapm_get_soc_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card; + else if (dapm->platform) + return dapm->platform->card; + else + BUG(); + + /* unreachable */ + return NULL; +} + static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) { if (w->codec) -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: dapm - add DAPM macro for external enum widgets
by Liam Girdwood 20 Jul '11

20 Jul '11

Add a convenience macro for external enumerated widgets. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 596387b..655acc5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -266,6 +266,12 @@ .get = snd_soc_dapm_get_enum_virt, \ .put = snd_soc_dapm_put_enum_virt, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = xget, \ + .put = xput, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_VALUE_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ -- 1.7.4.1

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[alsa-devel] [PATCH v2 2/2] ASoC: davinci: add missing break statement
by Rajashekhara, Sudhakar 20 Jul '11

20 Jul '11

In davinci_vcif_trigger() function, a break() statement was missing causing the davinci_vcif_stop() function to be called as a fallback after calling davinci_vcif_start(). Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- sound/soc/davinci/davinci-vcif.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index c957e9e..1f11525 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -159,6 +159,7 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_vcif_start(substream); + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -- 1.7.1

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[alsa-devel] [PATCH v2 1/2] ASoC: davinci: fix codec start and stop functions
by Rajashekhara, Sudhakar 20 Jul '11

20 Jul '11

According to DM365 voice codec data sheet at [1], before starting recording or playback, ADC/DAC modules should follow a reset and enable cycle. Writing a 1 to the ADC/DAC bit in the register resets the module and clearing the bit to 0 will enable the module. But the driver seems to be doing the reverse of it. [1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- sound/soc/davinci/davinci-vcif.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9259f1f..c957e9e 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -62,9 +62,9 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream) w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -80,9 +80,9 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) /* Reset transmitter/receiver and sample rate/frame sync generators */ w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } -- 1.7.1

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[alsa-devel] [PATCH v4 00/10] ARM: mxs: add audio support
by Dong Aisheng 20 Jul '11

20 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Code is written based on the branch imx-for-3.1 in sascha's tree http://git.pengutronix.de/git/imx/linux-2.6.git Tested on MX28EVK. For the ASoC-patches (1-4): Tested-by: Wolfram Sang <w.sang(a)pengutronix.de> For the mach-mxs-Patches (5-10): Reviewed-by: Wolfram Sang <w.sang(a)pengutronix.de> Changes since v3: This version only has one patch changed with another two for rebase the changed one. * [PATCH v4 06/10] ARM: mxs: add saif device Drop DIGCTRL-handling based on Wolfram's comment * [PATCH v4 07/10] ARM: mxs: add sgtl5000 i2c device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device * [PATCH v4 08/10] ARM: mxs: add mxs-sgtl5000 device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device changes since v2: * mxs-pcm - fix wrong periods_max. - For robustness replace the channels_min check with a check for the relevant substream being present * mxs-saif - Add IRQ_NONE handle - improve mxs_saif_set_clk to make it more easy to read - correct return value - request_irq sequence changed to after ioremap * mxs-sgtl5000 - clean up a few unused code * Kconfig&Makefile - SND_SOC_MXS_SGTL5000 should depend on I2C in Kconfig * mxs - Select MXS_I2C in kconfig for mxs-sgtl5000 changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 2 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 3 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 ++ arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 35 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 359 ++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 677 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++ sound/soc/mxs/mxs-sgtl5000.c | 165 ++++++ 18 files changed, 1530 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

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[alsa-devel] [PATCH 1/2] ASoC: Acknowledge WM8996 interrupts before acting on them
by Mark Brown 20 Jul '11

20 Jul '11

This closes the small race between a status being read in response to an interrupt and clearing the interrupt, meaning that if the status changes between those periods we might not get a reassertion of the interrupt. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 9 ++++----- 1 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 00f9ace..ab8e9d1 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2404,6 +2404,8 @@ static irqreturn_t wm8996_irq(int irq, void *data) } irq_val &= ~snd_soc_read(codec, WM8996_INTERRUPT_STATUS_2_MASK); + snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); + if (irq_val & (WM8996_DCS_DONE_01_EINT | WM8996_DCS_DONE_23_EINT)) { dev_dbg(codec->dev, "DC servo IRQ\n"); complete(&wm8996->dcs_done); @@ -2420,13 +2422,10 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); - if (irq_val) { - snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); - + if (irq_val) return IRQ_HANDLED; - } else { + else return IRQ_NONE; - } } static irqreturn_t wm8996_edge_irq(int irq, void *data) -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: davinci: bug fixes for DM365 voice codec driver
by Rajashekhara, Sudhakar 20 Jul '11

20 Jul '11

This patch fixes the following bugs found in DaVinci voice codec driver. a. According to DM365 voice codec data sheet at [1], before starting recording or playback, ADC/DAC modules should follow a reset and enable cycle. Writing a 1 to the ADC/DAC bit in the register resets the module and clearing the bit to 0 will enable the module. But the driver seems to be doing the reverse of it. b. In davinci_vcif_trigger() function, a break() statement was missing causing the davinci_vcif_stop() function to be called as a fallback after calling davinci_vcif_start(). [1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- sound/soc/davinci/davinci-vcif.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9259f1f..1f11525 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -62,9 +62,9 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream) w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -80,9 +80,9 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) /* Reset transmitter/receiver and sample rate/frame sync generators */ w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -159,6 +159,7 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_vcif_start(substream); + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -- 1.7.1

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[alsa-devel] [PATCH] ASoC: dapm - Add DAPM stream completion event.
by Liam Girdwood 20 Jul '11

20 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. This adds a callback function to be called at the completion of a DAPM stream event. This can be used by DSP components to perform calculations based on DAPM graphs after completion of stream events. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 2 ++ sound/soc/soc-dapm.c | 4 ++++ 4 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 198997d..596387b 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -524,6 +524,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int (*stream_event)(struct snd_soc_dapm_context *dapm); + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a..4c42f02 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -634,6 +634,9 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + /* codec stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; @@ -669,6 +672,9 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + /* platform stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f..159f144 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3140,6 +3140,7 @@ int snd_soc_register_platform(struct device *dev, platform->driver = platform_drv; platform->dapm.dev = dev; platform->dapm.platform = platform; + platform->dapm.stream_event = platform_drv->stream_event; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); @@ -3252,6 +3253,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.dev = dev; codec->dapm.codec = codec; codec->dapm.seq_notifier = codec_drv->seq_notifier; + codec->dapm.stream_event = codec_drv->stream_event; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aaef0ef..931dd7a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2764,6 +2764,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, } dapm_power_widgets(dapm, event); + + /* do we need to notify any clients that DAPM stream is complete */ + if (dapm->stream_event) + dapm->stream_event(dapm); } /** -- 1.7.4.1

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Re: [alsa-devel] [patch][saa7134] do not change mute state for capturing audio
by Stas Sergeev 20 Jul '11

20 Jul '11

15.07.2011 05:38, Mauro Carvalho Chehab wrote: > If you want, feel free to propose a patch fixing that logic at saa7134, instead > of just removing it. Hi, I've just verified that pulseaudio indeed does the sound capturing on startup: --- saa7134[0]/alsa: saa7134[0] at 0xfe8fb800 irq 22 registered as card 2 saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: irq: field oops [even] --- So your proposal is not going to fix anything at all. Can we get back to discussing/applying mine then? And if the other drivers has that autounmute logic, then I suggest removing it there as well. You have not named any use-case for it, so I think there is none. I also think that the whole auto-unmute logic in your drivers is entirely flawed: for instance, I don't think recording from the sound card will automatically unmute its line-in or something else, so you are probably not following the generic alsa style here. I am adding alsa-devel to CC to find out what they think about that whole auto-unmute question.

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[alsa-devel] [PATCH v3 00/10] ARM: mxs: add audio support
by Dong Aisheng 20 Jul '11

20 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Code is written based on the branch imx-for-3.1 in sascha's tree http://git.pengutronix.de/git/imx/linux-2.6.git Tested on MX28EVK. changes since v2: * mxs-pcm - fix wrong periods_max. - For robustness replace the channels_min check with a check for the relevant substream being present * mxs-saif - Add IRQ_NONE handle - improve mxs_saif_set_clk to make it more easy to read - correct return value - request_irq sequence changed to after ioremap * mxs-sgtl5000 - clean up a few unused code * Kconfig&Makefile - SND_SOC_MXS_SGTL5000 should depend on I2C in Kconfig * mxs - Select MXS_I2C in kconfig for mxs-sgtl5000 changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 2 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 3 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 ++ arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 40 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 359 ++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 677 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++ sound/soc/mxs/mxs-sgtl5000.c | 165 ++++++ 18 files changed, 1535 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

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