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

[alsa-devel] [PATCH repost] ASoC: fsl/pcm030 add unique dai_link names
by Eric Millbrandt 22 Oct '12

22 Oct '12

Assign each dai_link a unique name to avoid this run-time error. [ 18.978043] pcm030-audio-fabric sound.2: wm9712-hifi <-> mpc5200-psc-ac97.0 mapping ok [ 19.003179] sysfs: cannot create duplicate filename '/devices/sound.2/AC97' Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- sound/soc/fsl/pcm030-audio-fabric.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 4b63ec8..4d26192 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -29,14 +29,14 @@ struct pcm030_audio_data { static struct snd_soc_dai_link pcm030_fabric_dai[] = { { - .name = "AC97", + .name = "AC97.0", .stream_name = "AC97 Analog", .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "mpc5200-psc-ac97.0", .codec_name = "wm9712-codec", }, { - .name = "AC97", + .name = "AC97.1", .stream_name = "AC97 IEC958", .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", -- 1.7.8

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Re: [alsa-devel] [PATCH] check for the kzalloc return value
by Takashi Iwai 22 Oct '12

22 Oct '12

At Mon, 22 Oct 2012 17:05:53 +0400, Denis Kirjanov wrote: > > > Signed-off-by: Denis Kirjanov <kirjanov(a)gmail.com> Thanks, applied. Takashi > --- > sound/pci/als300.c | 4 ++++ > 1 file changed, 4 insertions(+) > > diff --git a/sound/pci/als300.c b/sound/pci/als300.c > index 68c4469..e1f74d4 100644 > --- a/sound/pci/als300.c > +++ b/sound/pci/als300.c > @@ -394,6 +394,8 @@ static int snd_als300_playback_open(struct snd_pcm_substream *substream) > struct snd_als300_substream_data *data = kzalloc(sizeof(*data), > GFP_KERNEL); > > + if (!data) > + return -ENOMEM; > snd_als300_dbgcallenter(); > chip->playback_substream = substream; > runtime->hw = snd_als300_playback_hw; > @@ -425,6 +427,8 @@ static int snd_als300_capture_open(struct snd_pcm_substream *substream) > struct snd_als300_substream_data *data = kzalloc(sizeof(*data), > GFP_KERNEL); > > + if (!data) > + return -ENOMEM; > snd_als300_dbgcallenter(); > chip->capture_substream = substream; > runtime->hw = snd_als300_capture_hw; > -- > 1.7.9.5 >

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[alsa-devel] [PATCH resend 1/2] ASoC: davinci: replace private sram api with genalloc
by Daniel Mack 22 Oct '12

22 Oct '12

From: Matt Porter <mporter(a)ti.com> Removes the DaVinci private SRAM API and replaces it with the genalloc API. The SRAM gen_pool is passed in pdata since DaVinci is in the early stages of DT conversion. [zonque(a)gmail.com: stub out gen_pool functions for !CONFIG_GENERIC_ALLOCATOR] Signed-off-by: Matt Porter <mporter(a)ti.com> --- include/linux/platform_data/davinci_asp.h | 3 +++ sound/soc/davinci/davinci-mcasp.c | 2 ++ sound/soc/davinci/davinci-pcm.c | 37 ++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.h | 2 ++ 4 files changed, 38 insertions(+), 6 deletions(-) diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index d0c5825..f3d6e4f 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -16,6 +16,8 @@ #ifndef __DAVINCI_ASP_H #define __DAVINCI_ASP_H +#include <linux/genalloc.h> + struct snd_platform_data { u32 tx_dma_offset; u32 rx_dma_offset; @@ -30,6 +32,7 @@ struct snd_platform_data { unsigned enable_channel_combine:1; unsigned sram_size_playback; unsigned sram_size_capture; + struct gen_pool *sram_pool; /* * If McBSP peripheral gets the clock from an external pin, diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 714e51e..52194bb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1098,6 +1098,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_playback; dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + mem->start); @@ -1115,6 +1116,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; + dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_capture; dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + mem->start); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 93ea3bf..0ae7b6d 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -16,6 +16,7 @@ #include <linux/slab.h> #include <linux/dma-mapping.h> #include <linux/kernel.h> +#include <linux/genalloc.h> #include <sound/core.h> #include <sound/pcm.h> @@ -23,7 +24,6 @@ #include <sound/soc.h> #include <asm/dma.h> -#include <mach/sram.h> #include "davinci-pcm.h" @@ -259,7 +259,9 @@ static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) } } -static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, +#ifdef CONFIG_GENERIC_ALLOCATOR +static int allocate_sram(struct snd_pcm_substream *substream, + struct gen_pool *sram_pool, unsigned size, struct snd_pcm_hardware *ppcm) { struct snd_dma_buffer *buf = &substream->dma_buffer; @@ -271,9 +273,10 @@ static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, return 0; ppcm->period_bytes_max = size; - iram_virt = sram_alloc(size, &iram_phys); + iram_virt = (void *)gen_pool_alloc(sram_pool, size); if (!iram_virt) goto exit1; + iram_phys = gen_pool_virt_to_phys(sram_pool, (unsigned)iram_virt); iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); if (!iram_dma) goto exit2; @@ -285,11 +288,33 @@ static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, return 0; exit2: if (iram_virt) - sram_free(iram_virt, size); + gen_pool_free(sram_pool, (unsigned)iram_virt, size); exit1: return -ENOMEM; } +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct gen_pool *sram_pool = prtd->params->sram_pool; + + gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); +} +#else +static int allocate_sram(struct snd_pcm_substream *substream, + struct gen_pool *sram_pool, unsigned size, + struct snd_pcm_hardware *ppcm) +{ + return 0; +} + +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ +} +#endif + /* * Only used with ping/pong. * This is called after runtime->dma_addr, period_bytes and data_type are valid @@ -676,7 +701,7 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? &pcm_hardware_playback : &pcm_hardware_capture; - allocate_sram(substream, params->sram_size, ppcm); + allocate_sram(substream, params->sram_pool, params->sram_size, ppcm); snd_soc_set_runtime_hwparams(substream, ppcm); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, @@ -819,7 +844,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) buf->area = NULL; iram_dma = buf->private_data; if (iram_dma) { - sram_free(iram_dma->area, iram_dma->bytes); + davinci_free_sram(substream, iram_dma); kfree(iram_dma); } } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fc4d01c..b6ef703 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -12,6 +12,7 @@ #ifndef _DAVINCI_PCM_H #define _DAVINCI_PCM_H +#include <linux/genalloc.h> #include <linux/platform_data/davinci_asp.h> #include <mach/edma.h> @@ -20,6 +21,7 @@ struct davinci_pcm_dma_params { unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ unsigned sram_size; + struct gen_pool *sram_pool; /* SRAM gen_pool for ping pong */ enum dma_event_q asp_chan_q; /* event queue number for ASP channel */ enum dma_event_q ram_chan_q; /* event queue number for RAM channel */ unsigned char data_type; /* xfer data type */ -- 1.7.11.7

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[alsa-devel] [PATCH] ASoC: jack: remove unused variable in snd_soc_jack_report()
by Wei Yongjun 22 Oct '12

22 Oct '12

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> The variable oldstatus is initialized but never used otherwise, so remove the unused variable. dpatch engine is used to auto generate this patch. (https://github.com/weiyj/dpatch) Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/soc-jack.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fa0fd8d..933eb25 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -66,7 +66,6 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; int enable; - int oldstatus; trace_snd_soc_jack_report(jack, mask, status); @@ -78,8 +77,6 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) mutex_lock(&jack->mutex); - oldstatus = jack->status; - jack->status &= ~mask; jack->status |= status & mask;

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[alsa-devel] [PATCH -next] ASoC: fix wm2200.c printk format warnings
by Randy Dunlap 22 Oct '12

22 Oct '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix wm2200.c printk format warnings (seen on x86_64): sound/soc/codecs/wm2200.c:1027:4: warning: format '%d' expects type 'int', but argument 4 has type 'size_t' sound/soc/codecs/wm2200.c:1139:5: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int' sound/soc/codecs/wm2200.c:1181:2: warning: format '%d' expects type 'int', but argument 7 has type 'size_t' sound/soc/codecs/wm2200.c:1201:5: warning: format '%x' expects type 'unsigned int', but argument 3 has type 'long unsigned int' sound/soc/codecs/wm2200.c:1264:4: warning: format '%d' expects type 'int', but argument 4 has type 'size_t' sound/soc/codecs/wm2200.c:1328:5: warning: format '%d' expects type 'int', but argument 5 has type 'long unsigned int' Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org (moderated for non-subscribers) --- sound/soc/codecs/wm2200.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) --- linux-next-20121019.orig/sound/soc/codecs/wm2200.c +++ linux-next-20121019/sound/soc/codecs/wm2200.c @@ -1023,7 +1023,7 @@ static int wm2200_dsp_load(struct snd_so pos = sizeof(*header) + sizeof(*adsp1_sizes) + sizeof(*footer); if (pos >= firmware->size) { - dev_err(codec->dev, "%s: file too short, %d bytes\n", + dev_err(codec->dev, "%s: file too short, %zu bytes\n", file, firmware->size); return -EINVAL; } @@ -1135,7 +1135,7 @@ static int wm2200_dsp_load(struct snd_so } if (pos > firmware->size) - dev_warn(codec->dev, "%s.%d: %d bytes at end of file\n", + dev_warn(codec->dev, "%s.%d: %zu bytes at end of file\n", file, regions, pos - firmware->size); out: @@ -1178,7 +1178,7 @@ static int wm2200_setup_algs(struct snd_ } algs = be32_to_cpu(id.algs); - dev_info(codec->dev, "Firmware: %x v%d.%d.%d, %d algorithms\n", + dev_info(codec->dev, "Firmware: %x v%d.%d.%d, %zu algorithms\n", be32_to_cpu(id.fw.id), (be32_to_cpu(id.fw.ver) & 0xff000) >> 16, (be32_to_cpu(id.fw.ver) & 0xff00) >> 8, @@ -1196,7 +1196,7 @@ static int wm2200_setup_algs(struct snd_ } if (be32_to_cpu(val) != 0xbedead) - dev_warn(codec->dev, "Algorithm list end %x 0x%x != 0xbeadead\n", + dev_warn(codec->dev, "Algorithm list end %zx 0x%x != 0xbeadead\n", (sizeof(id) + (algs * sizeof(*alg))) / 2, be32_to_cpu(val)); @@ -1260,7 +1260,7 @@ static int wm2200_load_coeff(struct snd_ } if (sizeof(*hdr) >= firmware->size) { - dev_err(codec->dev, "%s: file too short, %d bytes\n", + dev_err(codec->dev, "%s: file too short, %zu bytes\n", file, firmware->size); return -EINVAL; } @@ -1324,7 +1324,7 @@ static int wm2200_load_coeff(struct snd_ } if (pos > firmware->size) - dev_warn(codec->dev, "%s.%d: %d bytes at end of file\n", + dev_warn(codec->dev, "%s.%d: %zu bytes at end of file\n", file, blocks, pos - firmware->size); return 0;

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Re: [alsa-devel] [PATCH 2/2] ASoC: Ux500: Control apb clock
by Mark Brown 22 Oct '12

22 Oct '12

On Mon, Oct 22, 2012 at 02:32:05PM +0200, Ulf Hansson wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > When switching to common clock driver for ux500 this clock needs to > be handled as well. Before this clock was internally managed by the > clock driver itself. Applied but why isn't this a deficiency in the common clock implementation for the platform?

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Re: [alsa-devel] [PATCH 1/2] ASoC: Ux500: Fixup use of clocks
by Mark Brown 22 Oct '12

22 Oct '12

On Mon, Oct 22, 2012 at 02:32:04PM +0200, Ulf Hansson wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > Make sure clocks are being prepared and unprepared as well > as enabled and disabled. Applied, thanks.

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Re: [alsa-devel] [PATCH 2/2] ASoC: Ux500: Control apb clock
by Linus Walleij 22 Oct '12

22 Oct '12

On Mon, Oct 22, 2012 at 2:32 PM, Ulf Hansson <ulf.hansson(a)stericsson.com> wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > When switching to common clock driver for ux500 this clock needs to > be handled as well. Before this clock was internally managed by the > clock driver itself. > > Signed-off-by: Ulf Hansson <ulf.hansson(a)linaro.org> Looks correct, Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Yours, Linus Walleij

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Re: [alsa-devel] [PATCH 1/2] ASoC: Ux500: Fixup use of clocks
by Linus Walleij 22 Oct '12

22 Oct '12

On Mon, Oct 22, 2012 at 2:32 PM, Ulf Hansson <ulf.hansson(a)stericsson.com> wrote: > From: Ulf Hansson <ulf.hansson(a)linaro.org> > > Make sure clocks are being prepared and unprepared as well > as enabled and disabled. > > Signed-off-by: Ulf Hansson <ulf.hansson(a)linaro.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Thanks, Linus Walleij

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[alsa-devel] dynamic PCM and dynamic DSP graphs
by Sebastien LEDUC 22 Oct '12

22 Oct '12

Hi Mark / Liam, We have started to use Dynamic PCM to manage our on-SoC DSP. In our case, the API exposed by the DSP is graph oriented, and allows to dynamically create and control the running graph on the DSP. What we are doing is that we use DAPM widgets and routes to manage the DSP graph dynamically: We use some dedicated widgets that have power callbacks. When a widget is powered on/off, the callback will use the DSP API to create/destroy the corresponding parts of the DSP graph. We wanted to get your feedback on this approach, so as to be sure it won't be an issue when we want to mainline our driver. We can of course share more details if needed, or even some preliminary version of the driver. Thanks in advance for your feedback Regards Sebastien

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