March 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/3] ASoC: core: Add card mutex locking subclasses
by Liam Girdwood 08 Mar '12

08 Mar '12

This is the first part of a change that is intended to improve ASoC locking protection for DAPM and PCM operations. This part of the series adds a mutex class for the soc_card mutex. The SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the SND_SOC_CARD_CLASS_PCM class is used for the forth coming Dynamic PCM operations. The new mutex classes are required otherwise we will see a false positive mutex deadlock warning between the card initialisation and the PCM operations (something that would never deadlock in real life). Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 5 +++++ sound/soc/soc-core.c | 2 +- 2 files changed, 6 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2ebf787..70de2f8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -288,6 +288,11 @@ enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_BE = 1, }; +enum snd_soc_card_subclass { + SND_SOC_CARD_CLASS_INIT = 0, + SND_SOC_CARD_CLASS_PCM = 1, +}; + int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c90bb01..acd0d93 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1416,7 +1416,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_dai_link *dai_link; int ret, i, order; - mutex_lock(&card->mutex); + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); if (card->instantiated) { mutex_unlock(&card->mutex); -- 1.7.5.4

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Re: [alsa-devel] Random process lockup on ARM board: alsa-lib-1.0.25, FUTEX_WAIT_PRIVATE
by Clemens Ladisch 08 Mar '12

08 Mar '12

Jonathan Andrews wrote: > On Wed, 2012-02-22 at 17:23 +0100, Clemens Ladisch wrote: >> Now it doesn't use any pthread functions. > > Thanks, use or require ? Still complains if I static compile my > application but don't link with pthreads. Please try with the patch below; this should remove the last occurence of pthreads in alsa-lib. Regards, Clemens --- alsa-lib/configure.in +++ alsa-lib/configure.in @@ -512,6 +512,7 @@ if test "$HAVE_LIBDL" != "yes"; then fi if test "$HAVE_LIBPTHREAD" != "yes"; then + build_pcm_meter="no" build_pcm_share="no" fi

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[alsa-devel] [PATCH] ASoC: omap McBSP: Clear rx_irq at probe time for OMAP4
by Peter Ujfalusi 08 Mar '12

08 Mar '12

On OMAP4 we have one interrupt line per McBSP port. At proble time tx, and rx irq value will be -ENXIO, and only the tx irq will get corrected. In omap_mcbsp_request if the rx_irq is not 0 we proceed, and try to request the interrupt, which will fail on OMAP4 (rx_irq == -6). To avoid this error, clear the rx_irq at probe time on OMAP4. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/mcbsp.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 21dbb05..c3e31de 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -958,8 +958,10 @@ int __devinit omap_mcbsp_init(struct platform_device *pdev) mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx"); /* From OMAP4 there will be a single irq line */ - if (mcbsp->tx_irq == -ENXIO) + if (mcbsp->tx_irq == -ENXIO) { mcbsp->tx_irq = platform_get_irq(pdev, 0); + mcbsp->rx_irq = 0; + } res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); if (!res) { -- 1.7.8.5

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[alsa-devel] [PATCH] ASoC: dapm: Add platform stream event support.
by Liam Girdwood 07 Mar '12

07 Mar '12

Currently stream events are only perfomed on codec stream widgets only. There is now a need to be able to perform stream events on platform widgets too. e.g. we have the ABE platform driver with several DAI links to dummy codecs. We need to be able to perform stream events on any of the dummy codec DAI links. This patch also removes the snd_soc_dai * parameter since it's already contained within the rtd * parameter. Finally makle stream event return void since no one checks it anyway. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 4 +- sound/soc/soc-core.c | 8 +--- sound/soc/soc-dapm.c | 79 +++++++++++++++++++++++++++++---------------- sound/soc/soc-pcm.c | 9 ++--- 4 files changed, 59 insertions(+), 41 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 8cd3c74..d3e45a0 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -368,8 +368,8 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); /* dapm events */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - struct snd_soc_dai *dai, int event); +void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event); void snd_soc_dapm_shutdown(struct snd_soc_card *card); /* external DAPM widget events */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f466b9d..cc9f1f1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -573,19 +573,16 @@ int snd_soc_suspend(struct device *dev) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); snd_soc_dapm_stream_event(&card->rtd[i], SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_SUSPEND); } @@ -689,17 +686,16 @@ static void soc_resume_deferred(struct work_struct *work) } for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; if (card->rtd[i].dai_link->ignore_suspend) continue; snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_PLAYBACK, codec_dai, + SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_RESUME); snd_soc_dapm_stream_event(&card->rtd[i], - SNDRV_PCM_STREAM_CAPTURE, codec_dai, + SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_RESUME); } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d15a703..8320fd7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3019,37 +3019,61 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, - int stream, struct snd_soc_dai *dai, - int event) +static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { - struct snd_soc_dapm_widget *w; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - w = dai->playback_widget; - else - w = dai->capture_widget; + struct snd_soc_dapm_widget *w_cpu, *w_codec; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; - if (!w) - return; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + w_cpu = cpu_dai->playback_widget; + w_codec = codec_dai->playback_widget; + } else { + w_cpu = cpu_dai->capture_widget; + w_codec = codec_dai->capture_widget; + } - dapm_mark_dirty(w, "stream event"); + if (w_cpu) { - switch (event) { - case SND_SOC_DAPM_STREAM_START: - w->active = 1; - break; - case SND_SOC_DAPM_STREAM_STOP: - w->active = 0; - break; - case SND_SOC_DAPM_STREAM_SUSPEND: - case SND_SOC_DAPM_STREAM_RESUME: - case SND_SOC_DAPM_STREAM_PAUSE_PUSH: - case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: - break; + dapm_mark_dirty(w_cpu, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_cpu->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_cpu->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } + } + + if (w_codec) { + + dapm_mark_dirty(w_codec, "stream event"); + + switch (event) { + case SND_SOC_DAPM_STREAM_START: + w_codec->active = 1; + break; + case SND_SOC_DAPM_STREAM_STOP: + w_codec->active = 0; + break; + case SND_SOC_DAPM_STREAM_SUSPEND: + case SND_SOC_DAPM_STREAM_RESUME: + case SND_SOC_DAPM_STREAM_PAUSE_PUSH: + case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: + break; + } } - dapm_power_widgets(dapm, event); + dapm_power_widgets(&rtd->card->dapm, event); } /** @@ -3063,15 +3087,14 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, * * Returns 0 for success else error. */ -int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, - struct snd_soc_dai *dai, int event) +void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, + int event) { struct snd_soc_card *card = rtd->card; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_PCM); - soc_dapm_stream_event(&card->dapm, stream, dai, event); + soc_dapm_stream_event(rtd, stream, event); mutex_unlock(&card->dapm_mutex); - return 0; } /** diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0ad8dca..26a60b4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -308,7 +308,7 @@ static void close_delayed_work(struct work_struct *work) if (codec_dai->pop_wait == 1) { codec_dai->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -373,7 +373,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - codec_dai, SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ @@ -384,7 +383,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) } else { /* capture streams can be powered down now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, - codec_dai, SND_SOC_DAPM_STREAM_STOP); + SND_SOC_DAPM_STREAM_STOP); } mutex_unlock(&rtd->pcm_mutex); @@ -453,8 +452,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) cancel_delayed_work(&rtd->delayed_work); } - snd_soc_dapm_stream_event(rtd, substream->stream, codec_dai, - SND_SOC_DAPM_STREAM_START); + snd_soc_dapm_stream_event(rtd, substream->stream, + SND_SOC_DAPM_STREAM_START); snd_soc_dai_digital_mute(codec_dai, 0); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: imx: imx-audmux: Fix section mismatch
by Fabio Estevam 07 Mar '12

07 Mar '12

Fix the following section mismatch warning: WARNING: vmlinux.o(.data+0x35be8): Section mismatch in reference from the variable imx_audmux_driver to the function .init.text:imx_audmux_probe() The variable imx_audmux_driver references the function __init imx_audmux_probe() If the reference is valid then annotate the variable with __init* or __refdata (see linux/init.h) or name the variable: *_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/imx/imx-audmux.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index ef425ad..601df80 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -167,7 +167,7 @@ static void __init audmux_debugfs_init(void) } } -static void __exit audmux_debugfs_remove(void) +static void __devexit audmux_debugfs_remove(void) { debugfs_remove_recursive(audmux_debugfs_root); } @@ -249,7 +249,7 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); -static int __init imx_audmux_probe(struct platform_device *pdev) +static int __devinit imx_audmux_probe(struct platform_device *pdev) { struct resource *res; const struct of_device_id *of_id = @@ -276,7 +276,7 @@ static int __init imx_audmux_probe(struct platform_device *pdev) return 0; } -static int __exit imx_audmux_remove(struct platform_device *pdev) +static int __devexit imx_audmux_remove(struct platform_device *pdev) { if (audmux_type == IMX31_AUDMUX) audmux_debugfs_remove(); @@ -287,7 +287,7 @@ static int __exit imx_audmux_remove(struct platform_device *pdev) static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, - .remove = __exit_p(imx_audmux_remove), + .remove = __devexit_p(imx_audmux_remove), .id_table = imx_audmux_ids, .driver = { .name = DRIVER_NAME, -- 1.7.1

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[alsa-devel] ASOC - Codecs : Renaming of spdif_tranceiver.c
by Nitin PAI 07 Mar '12

07 Mar '12

Hi, In my project I am feeling of need to have generic_codec. This will provide the required codec_dais for the usecases like -HDMI/SPDIF where there is no codec involved. -For audio over MOST, where CPLD/glue logic is used -For complex audio DSPs connected which will be mostly programmed in the user-space. -SOC manufactures who would like to ship their drivers without having to know what will be the final codec that will be present in the system. Currently there is spdif_tranceiver.c which serves the purpose for spdif, but using this for other usecases will create confusion while shipping the code. What is the best way to carry forward? 1) Use spdif_tranceiver.c for all the purpose. disadvantage: this will create more confusion while shipping. 2)Deprecate spdif_transciever.c and call it generic_codec. Change the dai names and the function names. 3)Create another generic_codec.c and keep the spdif_transceiver.c and deprecate the latter in next version. 4) Any other ideas ? Thanks, Nitin

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[alsa-devel] [PATCH] ASoC: imx: imx-audmux: Fix section mismatch
by Fabio Estevam 07 Mar '12

07 Mar '12

Fix the following section mismatch warning: WARNING: vmlinux.o(.data+0x35be8): Section mismatch in reference from the variable imx_audmux_driver to the function .init.text:imx_audmux_probe() The variable imx_audmux_driver references the function __init imx_audmux_probe() If the reference is valid then annotate the variable with __init* or __refdata (see linux/init.h) or name the variable: *_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/imx/imx-audmux.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index ef425ad..601df80 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -167,7 +167,7 @@ static void __init audmux_debugfs_init(void) } } -static void __exit audmux_debugfs_remove(void) +static void __devexit audmux_debugfs_remove(void) { debugfs_remove_recursive(audmux_debugfs_root); } @@ -249,7 +249,7 @@ int imx_audmux_v2_configure_port(unsigned int port, unsigned int ptcr, } EXPORT_SYMBOL_GPL(imx_audmux_v2_configure_port); -static int __init imx_audmux_probe(struct platform_device *pdev) +static int __devinit imx_audmux_probe(struct platform_device *pdev) { struct resource *res; const struct of_device_id *of_id = @@ -276,7 +276,7 @@ static int __init imx_audmux_probe(struct platform_device *pdev) return 0; } -static int __exit imx_audmux_remove(struct platform_device *pdev) +static int __devexit imx_audmux_remove(struct platform_device *pdev) { if (audmux_type == IMX31_AUDMUX) audmux_debugfs_remove(); @@ -287,7 +287,7 @@ static int __exit imx_audmux_remove(struct platform_device *pdev) static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, - .remove = __exit_p(imx_audmux_remove), + .remove = __devexit_p(imx_audmux_remove), .id_table = imx_audmux_ids, .driver = { .name = DRIVER_NAME, -- 1.7.1

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[alsa-devel] [PATCH v2] ASoC: DAPM - Use dev_warn for debugfs warning message.
by Liam Girdwood 07 Mar '12

07 Mar '12

Remove printk(KERN_WARNING) and use dev_warn() instead. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a4d4aa1..dcd1160 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1667,7 +1667,7 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); if (!dapm->debugfs_dapm) { - printk(KERN_WARNING + dev_warn(dapm->dev, "Failed to create DAPM debugfs directory\n"); return; } -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: dapm: Only lock CODEC for I/O if not using regmap
by Mark Brown 07 Mar '12

07 Mar '12

If we do use regmap then regmap will take care of things for us. We actually already have this check at a higher level for the current users but this makes sure we do the right thing in the future too if we need to. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6f4ea12b..25a395e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -208,7 +208,7 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) { - if (w->codec) + if (w->codec && !w->codec->using_regmap) mutex_lock(&w->codec->mutex); else if (w->platform) mutex_lock(&w->platform->mutex); @@ -216,7 +216,7 @@ static inline void soc_widget_lock(struct snd_soc_dapm_widget *w) static inline void soc_widget_unlock(struct snd_soc_dapm_widget *w) { - if (w->codec) + if (w->codec && !w->codec->using_regmap) mutex_unlock(&w->codec->mutex); else if (w->platform) mutex_unlock(&w->platform->mutex); -- 1.7.9.1

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[alsa-devel] [RFC 0/7] ASoC: Introduce dmaengine pcm helper functions
by Lars-Peter Clausen 07 Mar '12

07 Mar '12

Same series as send out yesterday (sans the cleanup patches which already have been applied), since I forgot to Cc alsa-devel. Sorry for the noice if you got the series twice. This patch series introduces a new set of helper functions for dmaengine based PCM drivers. Currently we have 6 different dmaengine based PCM drivers which are all more or less similar in structure, but all have also individual requirements which can't easily be generalized. So that's why this series does not introduce a generic dmaengine PCM driver but rather a set of helper functions which can be used to implement the common bits between such dmaengine based PCM drivers. The helper functions only provide infrastructure code for dmaengines which provide thed evice_prep_dma_cyclic. While it is possible to add support for dmaengine, which only provide device_prep_slave_sg, it is in my opinion a better idea to either implement the missing functionality in the dmaengine driver or provide a emulation layer for cyclic transfer within the dmaengine framework itself. This has the advantage that it is also available outside of ALSA. This series converts the three PCM drivers which already use the circular dmaengine prepare function. Since I don't have access to hardware using the converted drivers the patches have only been compile tested. But the helper functions introduced in this series have been runtime tested with a platform which is not upstream yet. The first two patches restructure the iMX and MXS drivers to request their DMA channel in the PCM open callback instead of the hwparams callback. This is done to make their structure more similar to what it will be when using the helper functions and make the actual patch converting them to the new dmaengine PCM helper functions less intrusive. The next patch introduces the helper functions and the last three patches respectively convert the imx, mxs and ep39xx dmaengine based PCM drivers to use the new set of helper functions. The set of helper functions consists of a open, a close, a trigger and a pointer callback and also a function to convert hwparams to a dma_slave_config. The trigger and pointer callbacks can usually be used directly for the pcm_ops callbacks. The open callback wants to be called from a driver specific open function which may do some driver specific setup tasks and provide the helper open callback with a dma filter function. If the driver specific open functions allocates resources they need to be freed in a driver specific close function which has to call the helper close callback. If not the helper close callback can be assigned directly to the pcm_ops struct. If a driver needs to keep additional driver specific data around it can be done by using snd_dmaengine_pcm_{get,set}_data. Normally a driver should not require to keep additional data around, but all of the converted drivers need this, because of this horrible abusive pattern where the dma channels private field gets assigned configuration data in the filter callback. We should hopefully be able to get rid of this pattern with Guennadi Liakhovetski's patches[1], which add a slave configuration parameter to dma_request_channel, and with it the need for storing extra private data. While the DMA helper functions only use generic ALSA functions and nothing ASoC specific I've placed the, under ASoC for now since the only users are ASoC driver. - Lars [1] https://lkml.org/lkml/2012/2/1/227 Lars-Peter Clausen (7): ASoC: imx-ssi: Set dma data early ASoC: imx-pcm: Request DMA channel early ASoC: mxs-pcm: Request DMA channel early ASoC: Add dmaengine PCM helper functions ASoC: imx-pcm-dma: Use dmaengine PCM helper functions ASoC: mxs-pcm: Use dmaengine PCM helper functions ASoC: ep93xx-pcm: Use dmaengine PCM helper functions include/sound/dmaengine_pcm.h | 49 +++++++ sound/soc/Kconfig | 3 + sound/soc/Makefile | 3 + sound/soc/ep93xx/Kconfig | 1 + sound/soc/ep93xx/ep93xx-pcm.c | 148 +++----------------- sound/soc/imx/Kconfig | 1 + sound/soc/imx/imx-pcm-dma-mx2.c | 196 ++++---------------------- sound/soc/imx/imx-ssi.c | 28 +++- sound/soc/mxs/Kconfig | 1 + sound/soc/mxs/mxs-pcm.c | 152 ++++----------------- sound/soc/mxs/mxs-pcm.h | 12 -- sound/soc/soc-dmaengine-pcm.c | 287 +++++++++++++++++++++++++++++++++++++++ 12 files changed, 443 insertions(+), 438 deletions(-) create mode 100644 include/sound/dmaengine_pcm.h create mode 100644 sound/soc/soc-dmaengine-pcm.c -- 1.7.9

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