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

Re: [alsa-devel] [PATCH 5/5] ASoC: mxs: correct 'direction' of device_prep_dma_cyclic
by Vinod Koul 23 Dec '11

23 Dec '11

On Wed, 2011-12-14 at 15:47 +0800, Shawn Guo wrote: > The commit 49920bc (dmaengine: add new enum dma_transfer_direction) > changes the type of parameter 'direction' of device_prep_dma_cyclic > from dma_data_direction to dma_transfer_direction. > > Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> > --- > sound/soc/mxs/mxs-pcm.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c > index 0e12f4e..105f42a 100644 > --- a/sound/soc/mxs/mxs-pcm.c > +++ b/sound/soc/mxs/mxs-pcm.c > @@ -136,7 +136,7 @@ static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, > iprtd->period_bytes * iprtd->periods, > iprtd->period_bytes, > substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? > - DMA_TO_DEVICE : DMA_FROM_DEVICE); > + DMA_MEM_TO_DEV : DMA_DEV_TO_MEM); > if (!iprtd->desc) { > dev_err(&chan->dev->device, "cannot prepare slave dma\n"); > return -EINVAL; Has this been applied in soc tree? I don't see it in linux-next or do you want this to go thru slave-dma? -- ~Vinod

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[alsa-devel] ASoC updates for 3.3
by Mark Brown 23 Dec '11

23 Dec '11

The following changes since commit c45471eac2bdc271df40963ac8448d76ac434872: ASoC: Fix compile error in sound/soc/mid-x86/sst_platform.c (2011-12-16 07:50:15 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 Axel Lin (40): ASoC: Remove unused extern declarations for sh4_hac_dai and sh7760_soc_platform ASoC: Staticise sst_pcm_new and sst_soc_platform_drv ASoC: Staticise mfld_msic_dailink ASoC: pxa: Convert e740_wm9705 to use snd_soc_register_card() ASoC: pxa: Convert e750_wm9705 to use snd_soc_register_card() ASoC: pxa: Convert imote2 to use snd_soc_register_card() ASoC: pxa: Convert tosa to use snd_soc_register_card() ASoC: pxa: Convert e800_wm9712 to use snd_soc_register_card() ASoC: Convert max9850 to table based DAPM and control init ASoC: Convert uda1380 to table based DAPM and control init ASoC: Remove export of s3c_pcm_dai ASoC: Use dai_fmt in edb93xx machine driver ASoC: Use dai_fmt in snappercl15 machine driver ASoC: Rename rt562[1|2]_vol_snd_controls to alc562[1|2]_vol_snd_controls ASoC: Convert ak4104 to devm_kzalloc() ASoC: Convert ak4535 to devm_kzalloc() ASoC: Convert ak4641 to devm_kzalloc() ASoC: Convert ak4642 to devm_kzalloc() ASoC: Convert ak4671 to devm_kzalloc() ASoC: Use dai_fmt in hx4700 machine driver ASoC: Use dai_fmt in imote2 machine driver ASoC: Convert 88pm860x-codec to table based DAPM and control init ASoC: Convert cs42l51 to table based DAPM and control init ASoC: pxa: Add .owner to struct snd_soc_card ASoC: samsung: Add .owner to struct snd_soc_card ASoC: fsl: Add .owner to struct snd_soc_card ASoC: atmel: Add .owner to struct snd_soc_card ASoC: blackfin: Add .owner to struct snd_soc_card ASoC: davinci: Add .owner to struct snd_soc_card ASoC: ep93xx: Add .owner to struct snd_soc_card ASoC: tegra: Add .owner to struct snd_soc_card ASoC: au1x: Add .owner to struct snd_soc_card ASoC: imx: Add .owner to struct snd_soc_card ASoC: jz4740: Add .owner to struct snd_soc_card ASoC: nuc900: Add .owner to struct snd_soc_card ASoC: s6000: Add .owner to struct snd_soc_card ASoC: sh: Add .owner to struct snd_soc_card ASoC: txx9: Add .owner to struct snd_soc_card ASoC: kirkwood: Add .owner to struct snd_soc_card ASoC: mid-x86: Add .owner to struct snd_soc_card Leon Romanovsky (1): ASoC: Tegra machine ASoC driver for boards using ALC5332 codec Mark Brown (23): ASoC: Fix wm8995 regmap usage ASoC: Staticise asoc_idma_platform ASoC: Complete initialisation before registering Samsung PCM DAI ASoC: Remove cache default for volatile wm9081 reset register ASoC: Use standard snd_soc_cache_sync() for WM9090 ASoC: Convert wm8993 to devm_kzalloc() ASoC: Use standard register cache sync in wm8993 mfd: Test for jack detection when deciding if wm8994 should suspend ASoC: Tune the accessory detection rates for WM8996 ASoC: Use core pm_runtime callbacks for siu_dai ASoC: Use core pm_runtime callbacks for fsi ASoC: Fix partial cherry pick in wm8993 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound into for-3.3 ASoC: Convert WM8960 to devm_kzalloc() ASoC: Remove I2C ifdefs from wm8961 driver ASoC: Convert wm8961 to devm_kzalloc() ASoC: Use standard cache sync code in wm8961 Merge branch 'for-3.2' into for-3.3 ASoC: Remove ifdefs for GPIO_SYSFS Revert "ASoC: mxs: correct 'direction' of device_prep_dma_cyclic" Merge branch 'for-3.2' into for-3.3 ASoC: Declare soc_new_pcm() properly ASoC: Add trivial pm_runtime usage to Samsung DAI drivers Peter Ujfalusi (2): OMAP4: mcbsp: Enable FIFO usage ASoC: omap-mcbsp: Enable FIFO usage on OMAP4 Shawn Guo (1): ASoC: mxs: correct 'direction' of device_prep_dma_cyclic Stephen Warren (4): ASoC: Add utility to set a card's name from device tree ASoC: Add utility to parse DAPM routes from device tree ASoC: Tegra+WM8903 machine: Add device tree binding ASoC: Allow DAI links to be specified using device tree nodes .../bindings/sound/tegra-audio-wm8903.txt | 71 +++++++ arch/arm/mach-omap2/mcbsp.c | 3 + arch/arm/mach-pxa/eseries.c | 18 ++ arch/arm/mach-pxa/stargate2.c | 6 + arch/arm/mach-pxa/tosa.c | 6 + drivers/mfd/wm8994-core.c | 14 ++ include/sound/soc.h | 10 + sound/soc/atmel/sam9g20_wm8731.c | 1 + sound/soc/atmel/snd-soc-afeb9260.c | 1 + sound/soc/au1x/db1000.c | 1 + sound/soc/au1x/db1200.c | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 1 + sound/soc/blackfin/bf5xx-ad193x.c | 1 + sound/soc/blackfin/bf5xx-ad1980.c | 1 + sound/soc/blackfin/bf5xx-ad73311.c | 1 + sound/soc/blackfin/bf5xx-ssm2602.c | 1 + sound/soc/blackfin/bfin-eval-adau1373.c | 1 + sound/soc/blackfin/bfin-eval-adau1701.c | 1 + sound/soc/blackfin/bfin-eval-adav80x.c | 1 + sound/soc/codecs/88pm860x-codec.c | 15 +- sound/soc/codecs/ak4104.c | 6 +- sound/soc/codecs/ak4535.c | 6 +- sound/soc/codecs/ak4641.c | 7 +- sound/soc/codecs/ak4642.c | 6 +- sound/soc/codecs/ak4671.c | 6 +- sound/soc/codecs/alc5623.c | 12 +- sound/soc/codecs/cs42l51.c | 17 +- sound/soc/codecs/max9850.c | 17 +- sound/soc/codecs/uda1380.c | 24 +-- sound/soc/codecs/wm8960.c | 7 +- sound/soc/codecs/wm8961.c | 27 +-- sound/soc/codecs/wm8993.c | 29 +--- sound/soc/codecs/wm8995.c | 1 + sound/soc/codecs/wm8996.c | 18 +- sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9090.c | 13 +- sound/soc/davinci/davinci-evm.c | 6 + sound/soc/davinci/davinci-sffsdr.c | 1 + sound/soc/ep93xx/edb93xx.c | 15 +- sound/soc/ep93xx/simone.c | 1 + sound/soc/ep93xx/snappercl15.c | 13 +- sound/soc/fsl/efika-audio-fabric.c | 14 +- sound/soc/fsl/pcm030-audio-fabric.c | 14 +- sound/soc/imx/eukrea-tlv320.c | 1 + sound/soc/imx/mx27vis-aic32x4.c | 1 + sound/soc/imx/phycore-ac97.c | 1 + sound/soc/imx/wm1133-ev1.c | 1 + sound/soc/jz4740/qi_lb60.c | 1 + sound/soc/kirkwood/kirkwood-openrd.c | 1 + sound/soc/kirkwood/kirkwood-t5325.c | 1 + sound/soc/mid-x86/mfld_machine.c | 3 +- sound/soc/mid-x86/sst_platform.c | 4 +- sound/soc/nuc900/nuc900-audio.c | 1 + sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/pxa/corgi.c | 1 + sound/soc/pxa/e740_wm9705.c | 76 +++---- sound/soc/pxa/e750_wm9705.c | 67 +++---- sound/soc/pxa/e800_wm9712.c | 67 +++---- sound/soc/pxa/em-x270.c | 1 + sound/soc/pxa/hx4700.c | 17 +-- sound/soc/pxa/imote2.c | 58 +++--- sound/soc/pxa/magician.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/raumfeld.c | 2 + sound/soc/pxa/saarb.c | 1 + sound/soc/pxa/spitz.c | 1 + sound/soc/pxa/tavorevb3.c | 1 + sound/soc/pxa/tosa.c | 78 +++----- sound/soc/pxa/z2.c | 1 + sound/soc/pxa/zylonite.c | 1 + sound/soc/s6000/s6105-ipcam.c | 1 + sound/soc/samsung/goni_wm8994.c | 1 + sound/soc/samsung/h1940_uda1380.c | 1 + sound/soc/samsung/i2s.c | 6 +- sound/soc/samsung/idma.c | 2 +- sound/soc/samsung/jive_wm8750.c | 1 + sound/soc/samsung/littlemill.c | 1 + sound/soc/samsung/ln2440sbc_alc650.c | 1 + sound/soc/samsung/lowland.c | 1 + sound/soc/samsung/neo1973_wm8753.c | 1 + sound/soc/samsung/pcm.c | 20 +- sound/soc/samsung/rx1950_uda1380.c | 1 + sound/soc/samsung/s3c24xx_simtec_hermes.c | 1 + sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 1 + sound/soc/samsung/s3c24xx_uda134x.c | 1 + sound/soc/samsung/smartq_wm8987.c | 1 + sound/soc/samsung/smdk2443_wm9710.c | 1 + sound/soc/samsung/smdk_spdif.c | 1 + sound/soc/samsung/smdk_wm8580.c | 1 + sound/soc/samsung/smdk_wm8580pcm.c | 1 + sound/soc/samsung/smdk_wm8994.c | 1 + sound/soc/samsung/smdk_wm8994pcm.c | 1 + sound/soc/samsung/smdk_wm9713.c | 1 + sound/soc/samsung/speyside.c | 1 + sound/soc/samsung/tobermory.c | 1 + sound/soc/sh/fsi-ak4642.c | 1 + sound/soc/sh/fsi-da7210.c | 1 + sound/soc/sh/fsi-hdmi.c | 1 + sound/soc/sh/fsi.c | 4 - sound/soc/sh/migor.c | 1 + sound/soc/sh/sh7760-ac97.c | 5 +- sound/soc/sh/siu_dai.c | 6 - sound/soc/soc-core.c | 148 +++++++++++++- sound/soc/soc-jack.c | 4 - sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_alc5632.c | 214 ++++++++++++++++++++ sound/soc/tegra/tegra_wm8903.c | 129 ++++++++++--- sound/soc/tegra/trimslice.c | 1 + sound/soc/txx9/txx9aclc-generic.c | 1 + 111 files changed, 940 insertions(+), 446 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tegra-audio-wm8903.txt create mode 100644 sound/soc/tegra/tegra_alc5632.c

2 1

[alsa-devel] [RFC 0/5] compress: add compress data API to ALSA kernel
by Vinod Koul 23 Dec '11

23 Dec '11

As agreed during the ASoC workshop in May 2011, this set of patches are a proposal from Intel to add a new compressed data interface to ALSA. This proposal builds on the existing Intel solution and the experience gathered over two years with Meego and Android solutions. This proposal is no longer Intel-specific and was generalized with the addition of a basic core; codec-related definitions are based on the vendor-neutral OpenMAX AL standard. The code as is isn't functional and will need low-level hardware-specific drivers and a tighter integration with ALSA and ASoC. The main ideas will be presented during the Linux Plumbers Conference on September 9. Feedback through the mailing list would be greatly appreciated as well. Future versions will include support for capture, mmap, integration with alsa, asoc, dsp etc Support for effects is beyond the scope of this interface. Vinod Koul (5): compress API documentation compress: add compress parameter definations compress: add API header and driver header files compress: add the core file compress: add makefile and kconfig file .../sound/alsa/compress/snd_compress_data.txt | 186 ++++++ include/sound/compress_driver.h | 138 +++++ include/sound/compress_offload.h | 137 +++++ include/sound/snd_compress_params.h | 396 ++++++++++++ sound/Kconfig | 2 + sound/Makefile | 3 +- sound/compress_offload/Kconfig | 12 + sound/compress_offload/Makefile | 3 + sound/compress_offload/core.c | 632 ++++++++++++++++++++ 9 files changed, 1508 insertions(+), 1 deletions(-) create mode 100644 Documentation/sound/alsa/compress/snd_compress_data.txt create mode 100644 include/sound/compress_driver.h create mode 100644 include/sound/compress_offload.h create mode 100644 include/sound/snd_compress_params.h create mode 100644 sound/compress_offload/Kconfig create mode 100644 sound/compress_offload/Makefile create mode 100644 sound/compress_offload/core.c

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[alsa-devel] [PATCH] ASoC: Allocate PCM operations dynamically to support multiple DAIs
by Sangbeom Kim 23 Dec '11

23 Dec '11

From: Sangsu Park <sangsu4u.park(a)samsung.com> The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch allocates ASoC core PCM operations dynamically for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/core/pcm.c | 1 + sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++++-------------------- 2 files changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8928ca8..15cf27a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -769,6 +769,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream_next = substream->next; snd_pcm_timer_done(substream); snd_pcm_substream_proc_done(substream); + kfree(substream->ops); kfree(substream); substream = substream_next; } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec..3a52534 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -598,17 +598,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -616,10 +605,25 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + soc_pcm_ops = kzalloc(sizeof(*soc_pcm_ops), GFP_KERNEL); + if (soc_pcm_ops == NULL) { + snd_printk(KERN_ERR "Cannot allocate PCM OPS\n"); + return -ENOMEM; + } + + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_pcm_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->pointer = soc_pcm_pointer; + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -643,20 +647,20 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->pointer = platform->driver->ops->pointer; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); -- 1.7.1

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[alsa-devel] [PATCH 1/8] ASoC: imx: Add .owner to struct snd_soc_card
by Axel Lin 23 Dec '11

23 Dec '11

Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/imx/eukrea-tlv320.c | 1 + sound/soc/imx/mx27vis-aic32x4.c | 1 + sound/soc/imx/phycore-ac97.c | 1 + sound/soc/imx/wm1133-ev1.c | 1 + 4 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 75fb4b8..1c1fdd1 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -87,6 +87,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { static struct snd_soc_card eukrea_tlv320 = { .name = "cpuimx-audio", + .owner = THIS_MODULE, .dai_link = &eukrea_tlv320_dai, .num_links = 1, }; diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 054110b..3c2eed9 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -86,6 +86,7 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = { static struct snd_soc_card mx27vis_aic32x4 = { .name = "visstrim_m10-audio", + .owner = THIS_MODULE, .dai_link = &mx27vis_aic32x4_dai, .num_links = 1, }; diff --git a/sound/soc/imx/phycore-ac97.c b/sound/soc/imx/phycore-ac97.c index a7deb5c..6ac1211 100644 --- a/sound/soc/imx/phycore-ac97.c +++ b/sound/soc/imx/phycore-ac97.c @@ -38,6 +38,7 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { static struct snd_soc_card imx_phycore = { .name = "PhyCORE-ac97-audio", + .owner = THIS_MODULE, .dai_link = imx_phycore_dai_ac97, .num_links = ARRAY_SIZE(imx_phycore_dai_ac97), }; diff --git a/sound/soc/imx/wm1133-ev1.c b/sound/soc/imx/wm1133-ev1.c index 75b4c72..96fa761 100644 --- a/sound/soc/imx/wm1133-ev1.c +++ b/sound/soc/imx/wm1133-ev1.c @@ -254,6 +254,7 @@ static struct snd_soc_dai_link wm1133_ev1_dai = { static struct snd_soc_card wm1133_ev1 = { .name = "WM1133-EV1", + .owner = THIS_MODULE, .dai_link = &wm1133_ev1_dai, .num_links = 1, }; -- 1.7.5.4

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[alsa-devel] [PATCH v2] ASoC: au1x: Add .owner to struct snd_soc_card
by Axel Lin 23 Dec '11

23 Dec '11

Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: regenerate the patch against asoc tree. sound/soc/au1x/db1000.c | 1 + sound/soc/au1x/db1200.c | 2 ++ 2 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c index 127477a..8387d42 100644 --- a/sound/soc/au1x/db1000.c +++ b/sound/soc/au1x/db1000.c @@ -29,6 +29,7 @@ static struct snd_soc_dai_link db1000_ac97_dai = { static struct snd_soc_card db1000_ac97 = { .name = "DB1000_AC97", + .owner = THIS_MODULE, .dai_link = &db1000_ac97_dai, .num_links = 1, }; diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 289312c..15e5138 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -45,6 +45,7 @@ static struct snd_soc_dai_link db1200_ac97_dai = { static struct snd_soc_card db1200_ac97_machine = { .name = "DB1200_AC97", + .owner = THIS_MODULE, .dai_link = &db1200_ac97_dai, .num_links = 1, }; @@ -94,6 +95,7 @@ static struct snd_soc_dai_link db1200_i2s_dai = { static struct snd_soc_card db1200_i2s_machine = { .name = "DB1200_I2S", + .owner = THIS_MODULE, .dai_link = &db1200_i2s_dai, .num_links = 1, }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH v3 0/9] OMAP4: ASoC: Support for PandaBoard family
by Peter Ujfalusi 23 Dec '11

23 Dec '11

Hello, the following series will add ASoC support for PandaBoards. PandaBoards have different audio routings compared to SDP4430/Blaze boards, but the differences not that big to justify a new ASoC machine driver. This series is based on the first three patch from the v1 sereis (those, which has been applied by Liam). Changes since v2: - Remove direct board dependency for the ASoC machine driver (replaced by ARCH_OMAP4) - mach-omap2/board-omap4panda: Generic board revision handling function instead of dedicated for audio. - Macro twl6040_disconnect_pin replaced with a function Changes since v1: As suggested by Mark the platform data, machine driver has been converted to process feature flags instead of Board IDs. - The ASoC machine driver registers all DAPM widgets. Based on the received configuration it will disable the unused paths - PCM for dmic is only created, if the board has digital mic conencted - Jack functionality will be only used on boards supporting it - Sound card name for SDP4430 has net been changed Regards, Peter --- Peter Ujfalusi (9): include: platform_data: Platform data header for OMAP4 ASoC audio OMAP4: 4430sdp: Register platform device for OMAP4 audio ASoC: omap-abe-twl6040: Convert to platform deriver ASoC: twl6040: Convert MICBIAS to SUPPLY widget ASoC: omap-abe-twl6040: Add complete DAPM routing ASoC: omap-abe-twl6040: DAI link selection based on platform data ASoC: omap-abe-twl6040: Configure card according to platform data OMAP4: omap4panda: Enable audio support ASoC: Kconfig: OMAP4: Enable support for PandaBoards arch/arm/mach-omap2/board-4430sdp.c | 27 ++++ arch/arm/mach-omap2/board-omap4panda.c | 57 ++++++++- include/linux/platform_data/omap-abe-twl6040.h | 48 +++++++ sound/soc/codecs/twl6040.c | 16 +- sound/soc/omap/Kconfig | 4 +- sound/soc/omap/omap-abe-twl6040.c | 181 ++++++++++++++++-------- 6 files changed, 263 insertions(+), 70 deletions(-) create mode 100644 include/linux/platform_data/omap-abe-twl6040.h -- 1.7.8.1

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[alsa-devel] [PATCH 0/5] OMAP4: Fix CLKR/FSR mux for port with 6 pin mode
by Peter Ujfalusi 23 Dec '11

23 Dec '11

Hello, This series will fix the CLKR/FSR signal mux configuration for OMAP4 devices. McBSP4 has 6 pin configuration on McBSP4 port while OMAP2/3 has the same capability on McBSP1 port. The API for mux control will be simplified for the mcbsp users (ASoC). The series depends on earlier series sent for mcbsp: OMAP4: McBSP: Fix clock reparenting I think the ASoC patch can go via linux-omap, since we do not have other changes for McBSP coming via sound tree. Regards, Peter --- Peter Ujfalusi (5): OMAP2/3: mcbsp: Check McBSP port number in CLKR/FSR mux callback OMAP: mcbsp: Signle function for CLKR/FSR source mux configuration ASoC: omap-mcbsp: Use the new interface for configuring CLKR/FSR mux OMAP: mcbsp: Remove old CLKR/FSR mux functions OMAP4: mcbsp: Callback function for McBSP4 CLKR/FSR mux selection arch/arm/mach-omap2/mcbsp.c | 47 +++++++++++++++++++++++++++++ arch/arm/plat-omap/include/plat/mcbsp.h | 15 ++++------ arch/arm/plat-omap/mcbsp.c | 49 +++++++++++++++---------------- sound/soc/omap/omap-mcbsp.c | 12 +++++-- 4 files changed, 85 insertions(+), 38 deletions(-) -- 1.7.8.1

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[alsa-devel] [PATCH] ASoC: Allocate PCM operations dynamically to support multiple DAIs
by Sangbeom Kim 23 Dec '11

23 Dec '11

The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch allocates ASoC core PCM operations dynamically for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/core/pcm.c | 1 + sound/soc/soc-pcm.c | 44 ++++++++++++++++++++++++-------------------- 2 files changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 8928ca8..15cf27a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -769,6 +769,7 @@ static void snd_pcm_free_stream(struct snd_pcm_str * pstr) substream_next = substream->next; snd_pcm_timer_done(substream); snd_pcm_substream_proc_done(substream); + kfree(substream->ops); kfree(substream); substream = substream_next; } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec..3a52534 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -598,17 +598,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -616,10 +605,25 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + soc_pcm_ops = kzalloc(sizeof(*soc_pcm_ops), GFP_KERNEL); + if (soc_pcm_ops == NULL) { + snd_printk(KERN_ERR "Cannot allocate PCM OPS\n"); + return -ENOMEM; + } + + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_pcm_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->pointer = soc_pcm_pointer; + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -643,20 +647,20 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->pointer = platform->driver->ops->pointer; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); -- 1.7.1

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[alsa-devel] [PATCH 0/5] snd-usb endpoint rework, take 5
by Daniel Mack 22 Dec '11

22 Dec '11

I actually wanted to wait for more input, but as the flaws reported so far were rather minor, I addressed to following details in this new round as mentioned by Takashi and Clemens: - removed the faulty "(!u || u->urb)" check - guard the result of min() with the spinlock inside next_packet_size() - removed the tasklet and made the handler a function that is called in-place Thanks again for the reviews, and keep them coming :) Daniel Daniel Mack (5): ALSA: snd-usb: add snd_usb_audio-wide mutex ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: add support for implicit feedback sound/usb/card.c | 10 +- sound/usb/card.h | 62 +++ sound/usb/endpoint.c | 1454 +++++++++++++++++++++++++------------------------ sound/usb/endpoint.h | 32 +- sound/usb/pcm.c | 437 +++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 2 + 7 files changed, 1224 insertions(+), 804 deletions(-) -- 1.7.5.4

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