July 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 04/10] asoc: davinci - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the DaVinci platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/davinci/davinci-evm.c | 4 ++-- sound/soc/davinci/davinci-i2s.c | 14 +++++++------- sound/soc/davinci/davinci-i2s.h | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- 4 files changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 091eae3..4249e6a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -33,8 +33,8 @@ static int evm_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_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; /* set codec DAI configuration */ diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index c3b545c..5ebf1ff 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -147,7 +147,7 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) static int davinci_i2s_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; cpu_dai->dma_data = dev->dma_params[substream->stream]; @@ -155,7 +155,7 @@ static int davinci_i2s_startup(struct snd_pcm_substream *substream) return 0; } -static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcbsp_dev *dev = cpu_dai->private_data; @@ -296,11 +296,11 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) } static int davinci_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev; struct resource *mem, *ioarea; struct evm_snd_platform_data *pdata; @@ -358,11 +358,11 @@ err_release_region: } static void davinci_i2s_remove(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_machine *machine = socdev->machine; - struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct snd_soc_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; struct davinci_mcbsp_dev *dev = cpu_dai->private_data; struct resource *mem; @@ -378,7 +378,7 @@ static void davinci_i2s_remove(struct platform_device *pdev, #define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 -struct snd_soc_cpu_dai davinci_i2s_dai = { +struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h index 9592d17..c5b0918 100644 --- a/sound/soc/davinci/davinci-i2s.h +++ b/sound/soc/davinci/davinci-i2s.h @@ -12,6 +12,6 @@ #ifndef _DAVINCI_I2S_H #define _DAVINCI_I2S_H -extern struct snd_soc_cpu_dai davinci_i2s_dai; +extern struct snd_soc_dai davinci_i2s_dai; #endif diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 6a76927..6a5e56a 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -350,7 +350,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) static u64 davinci_pcm_dmamask = 0xffffffff; static int davinci_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret; -- 1.5.4.2

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[alsa-devel] [PATCH 03/10] asoc: at91 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the AT91 platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at91/at91-pcm.c | 6 +++--- sound/soc/at91/at91-ssc.c | 12 ++++++------ sound/soc/at91/at91-ssc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 8 ++++---- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index ccac6bd..d47492b 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -318,7 +318,7 @@ static int at91_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, static u64 at91_pcm_dmamask = 0xffffffff; static int at91_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -367,7 +367,7 @@ static void at91_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at91_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; @@ -392,7 +392,7 @@ static int at91_pcm_suspend(struct platform_device *pdev, } static int at91_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at91_runtime_data *prtd; diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index bc35d00..c3625b6 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -281,7 +281,7 @@ static void at91_ssc_shutdown(struct snd_pcm_substream *substream) /* * Record the SSC system clock rate. */ -static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* @@ -303,7 +303,7 @@ static int at91_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record the DAI format for use in hw_params(). */ -static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -315,7 +315,7 @@ static int at91_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params(). */ -static int at91_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at91_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at91_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -634,7 +634,7 @@ static int at91_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at91_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -662,7 +662,7 @@ static int at91_ssc_suspend(struct platform_device *pdev, } static int at91_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at91_ssc_info *ssc_p; @@ -700,7 +700,7 @@ static int at91_ssc_resume(struct platform_device *pdev, #define AT91_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_cpu_dai at91_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at91_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at91-ssc0", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/at91/at91-ssc.h b/sound/soc/at91/at91-ssc.h index b188f97..6b7bf38 100644 --- a/sound/soc/at91/at91-ssc.h +++ b/sound/soc/at91/at91-ssc.h @@ -21,7 +21,7 @@ #define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */ #define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at91_ssc_dai[]; +extern struct snd_soc_dai at91_ssc_dai[]; #endif /* _AT91_SSC_H */ diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index ad971e7..9d42139 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -53,8 +53,8 @@ static struct clk *pllb_clk; static int eti_b1_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* cpu clock is the AT91 master clock sent to the SSC */ @@ -87,8 +87,8 @@ static int eti_b1_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_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; #ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE -- 1.5.4.2

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[alsa-devel] [PATCH 02/10] asoc: at32 - merge structs snd_soc_codec_dai and snd_soc_cpu_dai.
by Liam Girdwood 07 Jul '08

07 Jul '08

This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for AT32 platform. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --- sound/soc/at32/at32-pcm.c | 6 +++--- sound/soc/at32/at32-ssc.c | 12 ++++++------ sound/soc/at32/at32-ssc.h | 2 +- sound/soc/at32/playpaq_wm8510.c | 6 +++--- 4 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/at32/at32-pcm.c b/sound/soc/at32/at32-pcm.c index cf76e89..435f1da 100644 --- a/sound/soc/at32/at32-pcm.c +++ b/sound/soc/at32/at32-pcm.c @@ -366,7 +366,7 @@ static struct snd_pcm_ops at32_pcm_ops = { static u64 at32_pcm_dmamask = 0xffffffff; static int at32_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -422,7 +422,7 @@ static void at32_pcm_free_dma_buffers(struct snd_pcm *pcm) #ifdef CONFIG_PM static int at32_pcm_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; @@ -447,7 +447,7 @@ static int at32_pcm_suspend(struct platform_device *pdev, static int at32_pcm_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = dai->runtime; struct at32_runtime_data *prtd; diff --git a/sound/soc/at32/at32-ssc.c b/sound/soc/at32/at32-ssc.c index 0ca4410..4ef6492 100644 --- a/sound/soc/at32/at32-ssc.c +++ b/sound/soc/at32/at32-ssc.c @@ -336,7 +336,7 @@ static void at32_ssc_shutdown(struct snd_pcm_substream *substream) /* * Set the SSC system clock rate */ -static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { /* TODO: What the heck do I do here? */ @@ -348,7 +348,7 @@ static int at32_ssc_set_dai_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Record DAI format for use by hw_params() */ -static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -362,7 +362,7 @@ static int at32_ssc_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, /* * Record SSC clock dividers for use in hw_params() */ -static int at32_ssc_set_dai_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int at32_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct at32_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; @@ -670,7 +670,7 @@ static int at32_ssc_prepare(struct snd_pcm_substream *substream) #ifdef CONFIG_PM static int at32_ssc_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; @@ -699,7 +699,7 @@ static int at32_ssc_suspend(struct platform_device *pdev, static int at32_ssc_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p; u32 cr; @@ -746,7 +746,7 @@ static int at32_ssc_resume(struct platform_device *pdev, SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_S32) -struct snd_soc_cpu_dai at32_ssc_dai[NUM_SSC_DEVICES] = { +struct snd_soc_dai at32_ssc_dai[NUM_SSC_DEVICES] = { { .name = "at32-ssc0", .id = 0, diff --git a/sound/soc/at32/at32-ssc.h b/sound/soc/at32/at32-ssc.h index 3c6901a..3c052db 100644 --- a/sound/soc/at32/at32-ssc.h +++ b/sound/soc/at32/at32-ssc.h @@ -52,7 +52,7 @@ struct at32_ssc_info { #define AT32_SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */ -extern struct snd_soc_cpu_dai at32_ssc_dai[]; +extern struct snd_soc_dai at32_ssc_dai[]; diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index d6b9fd5..18ac7d7 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -82,7 +82,7 @@ static struct clk *_pll0; #if defined CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock( struct snd_pcm_hw_params *params, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; @@ -132,8 +132,8 @@ static int playpaq_wm8510_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_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct at32_ssc_info *ssc_p = cpu_dai->private_data; struct ssc_device *ssc = ssc_p->ssc; unsigned int pll_out = 0, bclk = 0, mclk_div = 0; -- 1.5.4.2

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[alsa-devel] Repeated calls to trigger and prepare in ALSA driver
by DSP Tech 07 Jul '08

07 Jul '08

Hi All, I am trying to develop ALSA PCM driver for my hardware board which has I2S-DMA interface. I ran a OSS test application. I find that trigger() and prepare() calls are being called repeatedly. Consequently, the DMA tries to reset and start playback for every buffer. Have any of you faced this problem? Any clue in this matter would be helpful. Thanks _________________________________________________________________ Tried the new MSN Messenger? It’s cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in

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[alsa-devel] ASOC: codec_reg_show and variable length registers
by Jon Smirl 07 Jul '08

07 Jul '08

codec_reg_show() can't handle my TAS5504 codec. The codec has 104 (sparse out of 256) registers with variable lengths from 1 to 32 bytes. Should the register cache really be part of the core? The core could still make the sysfs entries and use a callback to print them. reg_cache_size/reg_cache_step would then become private. It's the sprintf that is the problem. I need to push that into my driver so that I can print out variable lengths. Or we need to make the core use an indirect table that encodes the lengths. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] HDA Intel ALC885 - Need new or edited model option
by Constrabus Gegami 06 Jul '08

06 Jul '08

I need a new or edited model option for this card: HDA Intel card in Alienware m15x laptop That uses codec: ALC885 Detailed specifications on the card: http://www.alienware.com/Configurator_Pages/Details_Pages/Details_Page.aspx… alsa-info.sh genterated output: http://pastebin.ca/1061774 (i switched back to no model options before i ran the script, i dont know if it leaves behind anything or if it even matters) Jack Observations: -1x auto-sensing headphone jack -1x auto-sensing microphone jack All current available model options either present me with: -working switchable internal speakers with non working headphone jack -working switchable headphone jack with non working internal speakers -non working headphone jack and internal speakers -working non switchable internal speakers with non working heaphone -working non switchable headphones with non working speakers -no working sound What is needed: - working switchable internal speakers with working headphone and mic jack (preferably switchable) Distro: Gentoo amd64 using alsa driver package (not in kernel modules) I hope that this will benefit others as well as I. If any more information is needed, please email me. -- --Constrabus

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[alsa-devel] [PATCH] ASoC: Fix warning from strict_strtoul()
by Mark Brown 06 Jul '08

06 Jul '08

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 6 +++++- 1 files changed, 5 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9e83357..456d78a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -808,7 +808,11 @@ static ssize_t dapm_pop_time_store(struct device *dev, const char *buf, size_t count) { - if (strict_strtoul(buf, 10, &pop_time) < 0) + unsigned long val; + + if (strict_strtoul(buf, 10, &val) >= 0) + pop_time = val; + else printk(KERN_ERR "Unable to parse pop_time setting\n"); return count; -- 1.5.6

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[alsa-devel] EMU10K1 Problem. Im stumped.
by Travis Place 05 Jul '08

05 Jul '08

Just had a user on #alsa with an EMU10K1 problem. To summarize, the card shows up in linux (but doesnt work), yet the card works perfectly in windows. The correct modules are loading, but ALSA doesnt see the card. It had the user unload/reload snd-emu10k1 and check dmesg output, and the relevant result is as follows. EMU10K1_Audigy: probe of 0000:00:0a.0 failed with error -16 There complete alsa-info.sh output can be found at http://pastebin.ca/1062377 I have had the user run this both with and without the onboard AC97 card enabled (just to be sure). What seems odd to me, is that the card is at 00:0a.0 and the SB Live revision is showing as (rev 0a). Just doesnt look right to me.. The card is an Creative SBLive! 'SB0220'. I have one of these exact cards here, if its needed for testing, but its never caused me any issues. Anyway, any help would be appreciated. Thanks, Travis Place (wishie)

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[alsa-devel] Sound distortions with Realtek ALC889A
by Matthias Dahl 04 Jul '08

04 Jul '08

Hello. Due to a hw defect, I recently bought a Gigabyte X48-DQ6 board which utilizes a Realtek ALC889A for sound output. At first glance everything seemed fine but during day-to-day use I figured that during playback of a 5.1 file, I've to turn up the volume quite a bit to match normal 2.0 playback or the volume levels of my old board (ADI1988B). And to make matters worse, the surround channel produces (static) noise on the center channel. If I mute every channel but the surround channel, the distortions are clearly hearable. During normal 5.1 playback, they are only hearable during scenes with a lot of talk or quieter scenes which is still very annoying unfortunately. Muting the surround channel also stops the distortions on the center channel. Also the model is not properly recognized, I've to pass model=6stack-dig to get all channels properly working with digital out. By the way, this is with a 2.6.26-rc7 x86_64 kernel running a alsa git checkout from last monday. I'd appreciate any help or hints I can get. Thanks in advance. Best regards, Matthias Dahl

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[alsa-devel] Using BD in place of DMA in soc driver
by dinesh 04 Jul '08

04 Jul '08

Hi i want to use mpc8323 ucc tdm as sound soc device but i dont know how to configure its buffer descriptor because every soc device available uses DMA. Please help.

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