April 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 2/4] mfd: wm5102: Map in additional FLL control registers
by Mark Brown 11 Apr '13

11 Apr '13

On Mon, Apr 08, 2013 at 06:22:23PM +0200, Samuel Ortiz wrote: > Hi Mark, > > On Wed, Mar 06, 2013 at 02:28:14PM +0800, Mark Brown wrote: > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > --- > > drivers/mfd/wm5102-tables.c | 8 ++++++++ > > 1 file changed, 8 insertions(+) > Did you merge this patchset through the ASoC tree ? Yes, let me know if that's a problem and I can back stuff out but it seemed sensible.

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[alsa-devel] [PATCH 22/30] ASoC: samsung/idma: export idma_reg_addr_init
by Arnd Bergmann 11 Apr '13

11 Apr '13

The idma_reg_addr_init function is used by the samsung i2s driver, which can be a loadable module, so we have to export this function. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- sound/soc/samsung/idma.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index f36a541..6e5fed3 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -411,6 +411,7 @@ void idma_reg_addr_init(void __iomem *regs, dma_addr_t addr) idma.regs = regs; idma.lp_tx_addr = addr; } +EXPORT_SYMBOL_GPL(idma_reg_addr_init); static struct snd_soc_platform_driver asoc_idma_platform = { .ops = &idma_ops, -- 1.8.1.2

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[alsa-devel] [PATCH 21/30] ASoC: samsung/i2s: fix module_device_table
by Arnd Bergmann 11 Apr '13

11 Apr '13

The second argument to the module_device_table macro must be the name of the device id array. In the samsung i2s driver, there was a small typo, resulting in a build error when building it as a loadable module. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 61f2622..f7c6816 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1289,7 +1289,7 @@ static struct platform_device_id samsung_i2s_driver_ids[] = { }, {}, }; -MODULE_DEVICE_TABLE(platform, samsung-i2s-driver-ids); +MODULE_DEVICE_TABLE(platform, samsung_i2s_driver_ids); #ifdef CONFIG_OF static struct samsung_i2s_dai_data samsung_i2s_dai_data_array[] = { -- 1.8.1.2

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[alsa-devel] [PATCH 19/30] ASoC: samsung: use irq resource for idma
by Arnd Bergmann 11 Apr '13

11 Apr '13

With multiplatform kernels, we cannot use hardwired IRQ numbers in device drivers. This changes the idma driver to use a proper resource, like all other drivers do. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> --- arch/arm/plat-samsung/devs.c | 6 ++++++ sound/soc/samsung/idma.c | 10 ++++++++-- 2 files changed, 14 insertions(+), 2 deletions(-) diff --git a/arch/arm/plat-samsung/devs.c b/arch/arm/plat-samsung/devs.c index 78be9c0..b441908 100644 --- a/arch/arm/plat-samsung/devs.c +++ b/arch/arm/plat-samsung/devs.c @@ -145,14 +145,20 @@ struct platform_device s3c_device_camif = { /* ASOC DMA */ +#ifdef CONFIG_PLAT_S5P +static struct resource samsung_asoc_idma_resource = DEFINE_RES_IRQ(IRQ_I2S0); + struct platform_device samsung_asoc_idma = { .name = "samsung-idma", .id = -1, + .num_resources = 1, + .resource = &samsung_asoc_idma_resource, .dev = { .dma_mask = &samsung_device_dma_mask, .coherent_dma_mask = DMA_BIT_MASK(32), } }; +#endif /* FB */ diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index a07950b..f36a541 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -68,6 +68,8 @@ static struct idma_info { dma_addr_t lp_tx_addr; } idma; +static int idma_irq; + static void idma_getpos(dma_addr_t *src) { *src = idma.lp_tx_addr + @@ -305,7 +307,7 @@ static int idma_open(struct snd_pcm_substream *substream) if (prtd == NULL) return -ENOMEM; - ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + ret = request_irq(idma_irq, iis_irq, 0, "i2s", prtd); if (ret < 0) { pr_err("fail to claim i2s irq , ret = %d\n", ret); kfree(prtd); @@ -324,7 +326,7 @@ static int idma_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct idma_ctrl *prtd = runtime->private_data; - free_irq(IRQ_I2S0, prtd); + free_irq(idma_irq, prtd); if (!prtd) pr_err("idma_close called with prtd == NULL\n"); @@ -418,6 +420,10 @@ static struct snd_soc_platform_driver asoc_idma_platform = { static int asoc_idma_platform_probe(struct platform_device *pdev) { + idma_irq = platform_get_irq(pdev, 0); + if (idma_irq < 0) + return idma_irq; + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); } -- 1.8.1.2

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[alsa-devel] [PATCH resend] sound: at73c213: Use dev_pm_ops
by Lars-Peter Clausen 11 Apr '13

11 Apr '13

Use dev_pm_ops instead of the deprecated legacy suspend/resume callbacks. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/spi/at73c213.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 4dd60d8..a1a24b9 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1075,10 +1075,11 @@ out: return 0; } -#ifdef CONFIG_PM -static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) +#ifdef CONFIG_PM_SLEEP + +static int snd_at73c213_suspend(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); @@ -1087,9 +1088,9 @@ static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) return 0; } -static int snd_at73c213_resume(struct spi_device *spi) +static int snd_at73c213_resume(struct device *dev) { - struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; clk_enable(chip->board->dac_clk); @@ -1097,18 +1098,21 @@ static int snd_at73c213_resume(struct spi_device *spi) return 0; } + +static SIMPLE_DEV_PM_OPS(at73c213_pm_ops, snd_at73c213_suspend, + snd_at73c213_resume); +#define AT73C213_PM_OPS (&at73c213_pm_ops) + #else -#define snd_at73c213_suspend NULL -#define snd_at73c213_resume NULL +#define AT73C213_PM_OPS NULL #endif static struct spi_driver at73c213_driver = { .driver = { .name = "at73c213", + .pm = AT73C213_PM_OPS, }, .probe = snd_at73c213_probe, - .suspend = snd_at73c213_suspend, - .resume = snd_at73c213_resume, .remove = snd_at73c213_remove, }; -- 1.8.0

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[alsa-devel] [PATCH 0/3] Headphone Mic patches
by David Henningsson 11 Apr '13

11 Apr '13

I'm preparing upstreaming of some new hardware, which has headphone mic support, i e, you can use the headphone jack to plug in a TRS/TS mic instead of plugging in a headphone in that jack. For this hardware, the existing hp_mic functionality won't work as the headphone and mic are on different pins. David Henningsson (3): ALSA: hda - allow "Headphone Mic" parser flag ALSA: hda - add some small convenience functions to auto parser ALSA: hda - Handle Headphone Mic jack more generic sound/pci/hda/hda_auto_parser.c | 68 ++++++++++++++++++++++++++++++++------- sound/pci/hda/hda_auto_parser.h | 29 +++++++++++++++-- sound/pci/hda/hda_generic.c | 12 +------ sound/pci/hda/hda_jack.c | 43 +++++++++++++++++-------- 4 files changed, 113 insertions(+), 39 deletions(-) -- 1.7.9.5

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[alsa-devel] what is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE
by Zhang wei 11 Apr '13

11 Apr '13

There are two definitions In the file include/sound/asound.h SNDRV_PCM_FORMAT_S24_3LE and SNDRV_PCM_FORMAT_S24_LE. What is the difference between them? Thanks a lot. zw

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[alsa-devel] VIA VT1731
by ALEKSANDR BOIKO 11 Apr '13

11 Apr '13

Hello! I'm using DAC Audio-GD Master 7 with Voyage MPD. There is a problem with playback audio files different frequency. I've read there is a path - solution. How can I get it? Help please.

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[alsa-devel] ASOC snd_soc_dai_ops for clock mux control
by Gellner, Christoph (ADITG/SW1) 10 Apr '13

10 Apr '13

Hi all, I'm working on the driver of an audio CPU DAI IP providing - Internal clocks which can be used for master clock generation by TX as well as RX section - One oversampling clock available to RX section - One oversampling clock available to TX section. Kernel version is 3.5.7 I want to add support to the CPU DAI driver to - Specify the rate of the input clocks - Select which of the clock inputs shall be used by RX and which shall be used by TX section I'm currently wondering how to map this to existing snd_soc_dai_ops functions. Currently I see only set_sysclk, set_pll, set_clkdiv for clock control. I plan to use set_sysclk to specify the rate as well as the direction of each clock. What is currently missing for me is a way to specify the clock to be used by RX or TX section. I can't really add this in set_sysclk. In case I switch to clock X when set_sysclk is called with clock ID of X, I need a way to distinguish if RX or TX section or both need to switch to this clock. But I can't really double the internal clock X in virtual clocks X_TX, X_RX (one time for TX and one time for RX section) as the internal clock X has only one rate for both sections and this would not represent the hardware correctly. Do you have any recommendation how to implement ? What do you think about adding a new function pointer to snd_soc_dai_ops ? int (*set_clkmux)(struct snd_soc_dai *dai, int mux_id, int source_id); I think this would be the cleanest solution as it models the features of the hardware (the clock multiplexers for RX or TX sections) It might be possible to use set_pll or set_clkdiv to realize. I could use set_pll and ignore the freq_in and freq_out I could use set_clkdiv and interpret div (divider integer value) as a source_id But this is not a 100% clean solution in my opinion. Best regards Christoph

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[alsa-devel] [PATCH 1/3] ALSA: compress_core: Update calc_avail to use cumulative values
by Charles Keepax 10 Apr '13

10 Apr '13

The app_pointer is managed locally by the compress core for memory mapped DSPs but for DSPs that are not memory mapped this would have to be manually updated from within the DSP driver itself, which is hardly very idiomatic. This patch switches to using the cumulative values to calculate the available buffer space because these are already gracefully passed out of the DSP driver to the compress core and otherwise should be functionally equivalent. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 23 +++++------------------ 1 files changed, 5 insertions(+), 18 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c84abc8..27bd81a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -160,8 +160,6 @@ static int snd_compr_update_tstamp(struct snd_compr_stream *stream, static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, struct snd_compr_avail *avail) { - long avail_calc; /*this needs to be signed variable */ - memset(avail, 0, sizeof(*avail)); snd_compr_update_tstamp(stream, &avail->tstamp); /* Still need to return avail even if tstamp can't be filled in */ @@ -184,22 +182,11 @@ static size_t snd_compr_calc_avail(struct snd_compr_stream *stream, return stream->runtime->buffer_size; } - /* FIXME: this routine isn't consistent, in one test we use - * cumulative values and in the other byte offsets. Do we - * really need the byte offsets if the cumulative values have - * been updated? In the PCM interface app_ptr and hw_ptr are - * already cumulative */ - - avail_calc = stream->runtime->buffer_size - - (stream->runtime->app_pointer - stream->runtime->hw_pointer); - pr_debug("calc avail as %ld, app_ptr %lld, hw+ptr %lld\n", avail_calc, - stream->runtime->app_pointer, - stream->runtime->hw_pointer); - if (avail_calc >= stream->runtime->buffer_size) - avail_calc -= stream->runtime->buffer_size; - pr_debug("ret avail as %ld\n", avail_calc); - avail->avail = avail_calc; - return avail_calc; + avail->avail = stream->runtime->buffer_size - + (stream->runtime->total_bytes_available - + stream->runtime->total_bytes_transferred); + pr_debug("ret avail as %lld\n", avail->avail); + return avail->avail; } static inline size_t snd_compr_get_avail(struct snd_compr_stream *stream) -- 1.7.2.5

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[alsa-devel] [PATCH v2] davinci-mcasp: Add support for multichannel playback
by Michal Bachraty 10 Apr '13

10 Apr '13

Davinci McASP has support for I2S multichannel playback. For I2S playback/receive, each serializer is capable to play 2 channels (L/R) audio data.Serializer function (Playback-receive-none) is configured in DT, depending on hardware specification. It is possible to play less channels than configured in DT. For that purpose,only specific number of active serializers are enabled. McASP FIFO need to have DMA transfer Bcnt set to number of enabled serializers, otherwise no data are transfered to McASP and Alsa generates "DMA/IRQ playback write error (DMA or IRQ trouble?)" error. Signed-off-by: Michal Bachraty <michal.bachraty(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++++++++++++----- sound/soc/davinci/davinci-pcm.c | 16 ++++++----- sound/soc/davinci/davinci-pcm.h | 1 + 3 files changed, 59 insertions(+), 14 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index afef3fb..b84bb73 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -235,6 +235,10 @@ #define DISMOD (val)(val<<2) #define TXSTATE BIT(4) #define RXSTATE BIT(5) +#define SRMOD_MASK 3 +#define SRMOD_INACTIVE 0 +#define SRMOD_TX 1 +#define SRMOD_RX 2 /* * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits @@ -657,12 +661,15 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, return 0; } -static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) +static int davinci_hw_common_param(struct davinci_audio_dev *dev, int stream, + int channels) { int i; u8 tx_ser = 0; u8 rx_ser = 0; - + u8 ser; + u8 slots = dev->tdm_slots; + u8 max_active_serializers = (channels + slots - 1) / slots; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -680,16 +687,42 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) } for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) + tx_ser++; + if (dev->serial_dir[i] == RX_MODE) + rx_ser++; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + ser = tx_ser; + else + ser = rx_ser; + + if (ser < max_active_serializers) { + dev_warn(dev->dev, "stream has more channels (%d) than are " + "enabled in mcasp (%d)\n", channels, ser * slots); + return -EINVAL; + } + + tx_ser = 0; + rx_ser = 0; + + for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) { + if (dev->serial_dir[i] == TX_MODE && + tx_ser < max_active_serializers) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); tx_ser++; - } else if (dev->serial_dir[i] == RX_MODE) { + } else if (dev->serial_dir[i] == RX_MODE && + rx_ser < max_active_serializers) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); rx_ser++; + } else { + mcasp_mod_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), + SRMOD_INACTIVE, SRMOD_MASK); } } @@ -729,6 +762,8 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); } } + + return 0; } static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) @@ -812,8 +847,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, &dev->dma_params[substream->stream]; int word_length; u8 fifo_level; + u8 slots = dev->tdm_slots; + int channels; + struct snd_interval *pcm_channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + channels = pcm_channels->min; - davinci_hw_common_param(dev, substream->stream); + if (davinci_hw_common_param(dev, substream->stream, channels)) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = dev->txnumevt; else @@ -862,6 +903,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, dma_params->acnt = dma_params->data_type; dma_params->fifo_level = fifo_level; + dma_params->active_serializers = (channels + slots - 1) / slots; davinci_config_channel_size(dev, word_length); return 0; @@ -936,13 +978,13 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, .capture = { .channels_min = 2, - .channels_max = 2, + .channels_max = 8, .rates = DAVINCI_MCASP_RATES, .formats = DAVINCI_MCASP_PCM_FMTS, }, diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index bb57552..3af8b50 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -182,6 +182,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) unsigned short acnt; unsigned int count; unsigned int fifo_level; + unsigned char serializers = prtd->params->active_serializers; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; @@ -195,14 +196,14 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) data_type = prtd->params->data_type; count = period_size / data_type; if (fifo_level) - count /= fifo_level; + count /= fifo_level * serializers; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; - dst_bidx = 0; - src_cidx = data_type * fifo_level; + dst_bidx = 4; + src_cidx = data_type * fifo_level * serializers; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +211,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) src_bidx = 0; dst_bidx = data_type; src_cidx = 0; - dst_cidx = data_type * fifo_level; + dst_cidx = data_type * fifo_level * serializers; } acnt = prtd->params->acnt; @@ -224,9 +225,10 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, ASYNC); else - edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, - count, fifo_level, - ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, + fifo_level * serializers, + count, fifo_level * serializers, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index fbb710c..0d84d32 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,6 +27,7 @@ struct davinci_pcm_dma_params { unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; unsigned int fifo_level; + unsigned char active_serializers; /* num. of active audio serializers */ }; int davinci_soc_platform_register(struct device *dev); -- 1.7.9.5

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Re: [alsa-devel] [alsa-cvslog] [ALSA GIT]ALSA utilities repository branch master updated. v1.0.26-26-ge05b903
by Takashi Iwai 10 Apr '13

10 Apr '13

At Mon, 8 Apr 2013 14:51:54 +0200 (CEST), GIT server wrote: > > This is an automated email from the git hooks/post-receive script. It was > generated because a ref change was pushed to the repository containing > the project "ALSA utilities repository". > > The branch, master has been updated > via e05b903b1fb16e967d838edac408304cd4470fee (commit) > from 46cb355d4ff41c8065b02644b0534ab42d16cb72 (commit) > > Those revisions listed above that are new to this repository have > not appeared on any other notification email; so we list those > revisions in full, below. > > - Log ----------------------------------------------------------------- > commit e05b903b1fb16e967d838edac408304cd4470fee > Author: Jaroslav Kysela <perex(a)perex.cz> > Date: Mon Apr 8 14:49:31 2013 +0200 > > alsactl: move systemd config to the daemon mode > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> I'm not thrilled by the silent default behavior change like this. This will affect all systems using systemd from now on. I understand the reason behind it, but I wonder whether it's an overkill. Yet another daemon, unconditionally no matter whether hotplug or not? Hmm... Takashi

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[alsa-devel] [PATCH] ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locations
by Daniel Mack 10 Apr '13

10 Apr '13

It turns out the devices from Playback Design need the delay quirk after usb_set_interface from clocks.c as well. Make it a proper quirks function and factor out the code to quirks.c. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/clock.c | 3 +++ sound/usb/pcm.c | 7 +------ sound/usb/quirks.c | 10 ++++++++++ sound/usb/quirks.h | 1 + 4 files changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index ae35e7d..b0ec364 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -32,6 +32,7 @@ #include "card.h" #include "helper.h" #include "clock.h" +#include "quirks.h" static struct uac_clock_source_descriptor * snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface, @@ -392,7 +393,9 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, * interface is active. */ if (rate != prev_rate) { usb_set_interface(dev, iface, 0); + snd_usb_set_interface_quirk(dev); usb_set_interface(dev, iface, fmt->altsetting); + snd_usb_set_interface_quirk(dev); } return 0; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 754cb5b..815a37d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -349,12 +349,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; - /* - * "Playback Design" products need a 50ms delay after setting the - * USB interface. - */ - if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) - mdelay(50); + snd_usb_set_interface_quirk(dev); } subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7882a82..551ae1a 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -892,6 +892,16 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) ep->skip_packets = 16; } +void snd_usb_set_interface_quirk(struct usb_device *dev) +{ + /* + * "Playback Design" products need a 50ms delay after setting the + * USB interface. + */ + if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) + mdelay(50); +} + void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index f10d00e..665e972 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -26,6 +26,7 @@ int snd_usb_is_big_endian_format(struct snd_usb_audio *chip, void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep); +void snd_usb_set_interface_quirk(struct usb_device *dev); void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); -- 1.7.10.4

2 1

Re: [alsa-devel] [PATCH] ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running
by Mark Brown 09 Apr '13

09 Apr '13

On Tue, Apr 09, 2013 at 05:13:59PM +0200, Alban Bedel wrote: > The Charge Pump needs the DSP clock to work properly, without it the > bypass to HP/LINEOUT is not working properly. This requirement is not > mentioned in the datasheet but has been confirmed by Mark Brown from > Wolfson. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH 2/2] ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running
by Mark Brown 09 Apr '13

09 Apr '13

On Wed, Mar 20, 2013 at 05:37:33PM +0100, Alban Bedel wrote: > output enable sequence requiere the DSP clock. Some input from Wolfson > will be needed to find out if this is just an undocumented requirement > or a chip bug. It seems it's an undocumented requirement. > + { "Left Output Mixer", "Left Bypass Switch", "CLK_DSP" }, > + { "Left Output Mixer", "Right Bypass Switch", "CLK_DSP" }, > + { "Right Output Mixer", "Left Bypass Switch", "CLK_DSP" }, > + { "Right Output Mixer", "Right Bypass Switch", "CLK_DSP" }, The actual change here should be to add a route: { "Charge Pump", NULL, "CLK_DSP" }, Can you submit an update for this or should I do one?

1 0

[alsa-devel] [PATCH v3 3/3] ALSA: hda - delay resuming Haswell HDMI codec in system resume
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. In this patch, Haswell codec will claim it needs "delaying resume" by setting flag "support_delay_resume", so system resume will skip it and mark it as "delay_resumed". On later codec access, Haswell codec will be resumed and hda_call_codec_resume() will call its set_power_states ops intel_haswell_set_ power_state(). For a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ede8215..22f276c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -89,6 +89,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -975,6 +983,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(get_pin(spec, pin_idx), 1); snd_hda_jack_report_sync(codec); } @@ -1725,6 +1740,11 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); } + +#ifdef CONFIG_PM + spec->ready_to_resume = 0; +#endif + return 0; } @@ -1855,6 +1875,60 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = get_pin(spec, pin_idx); + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printk(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume do { } while (0) +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1878,6 +1952,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2 2/3] ALSA: hda - system resume may delay resuming a codec
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> On system resume, a codec may not be ready to resume for some external dependency. e.g. Intel Haswell HDMI codec depends on Gfx to initialize the display pipeline before the codec can resume. To avoid the external dependency blocking system resume, this patch allow system resume ops "snd_hda_resume" skip a not ready codec and mark it as "resume delayed". So the codec resume is delayed until later codec access, assuming the external dependency may be resolved and waiting is acceptable at that time. A codec that needs delaying resume shall explicitly set "support_delay_resume" flag on initialization in its patch ops. And it's up to the codec when it's ready and how to finish the delayed resume. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ba147d7..81e91b8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5518,7 +5518,10 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6061360..5fe7362 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -889,6 +889,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2 1/3] ALSA: hda - add new function snd_hda_bus_reset() for codec communitation error
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A new function snd_hda_bus_reset() is defined to reset all codecs on the bus. This is implemented as temporary suspend-and-resume calls to codecs as an ad hoc solution. It will be called by bus reset ops azx_bus_reset() as a rescue of codec communitation error, replacing calls to snd_hda_suspend and snd_hda_resume. It's because snd_hda_resume need change for system resume: It will delay resuming a codec that is not ready to resume for some external dependency, to avoid blocking system resume. But this change of "delay resume" is not suitable for recovering from codec communication error, which need resume the codec immediately after suspend to simulate a bus reset. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..ba147d7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5523,6 +5523,29 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + +/** + * snd_hda_bus_reset - reset the codecs on the bus. + * @bus: the HDA bus + * + * This is implemented as temporary suspend-and-resume calls + * to codecs as an ad hoc solution. + * + * Returns 0 if successful. + */ +int snd_hda_bus_reset(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + cancel_delayed_work_sync(&codec->jackpoll_work); + if (hda_codec_is_power_on(codec)) + hda_call_codec_suspend(codec, false); + hda_call_codec_resume(codec); + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_reset); #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..6061360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1076,6 +1076,7 @@ void snd_hda_unlock_devices(struct hda_bus *bus); #ifdef CONFIG_PM int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); +int snd_hda_bus_reset(struct hda_bus *bus); #endif static inline diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..6f11c7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1659,8 +1659,7 @@ static void azx_bus_reset(struct hda_bus *bus) struct azx_pcm *p; list_for_each_entry(p, &chip->pcm_list, list) snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); + snd_hda_bus_reset(chip->bus); } #endif bus->in_reset = 0; -- 1.7.10.4

1 0

[alsa-devel] [PATCH 3/3] ALSA: hda - delay resuming Haswell HDMI codec in system resume
by mengdong.lin＠intel.com 09 Apr '13

09 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. In this patch, Haswell codec will claim it needs "delaying resume" by setting flag "support_delay_resume", so system resume will skip it and mark it as "delay_resumed". On later codec access, Haswell codec will be resumed and hda_call_codec_resume() will call its set_power_states ops intel_haswell_set_ power_state(). For a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ede8215..f39e374 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -89,6 +89,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -975,6 +983,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(get_pin(spec, pin_idx), 1); snd_hda_jack_report_sync(codec); } @@ -1725,6 +1740,11 @@ static int generic_hdmi_init(struct hda_codec *codec) hdmi_init_pin(codec, pin_nid); snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); } + +#ifdef CONFIG_PM + spec->ready_to_resume = 0; +#endif + return 0; } @@ -1855,6 +1875,61 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = get_pin(spec, pin_idx); + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printd(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static inline void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = + intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume NULL +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1878,6 +1953,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ALSA: hda - Apply codec delay to wallclock.
by Dylan Reid 09 Apr '13

09 Apr '13

For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_intel.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..ec8ac71 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1889,6 +1889,27 @@ static void azx_timecounter_init(struct snd_pcm_substream *substream, tc->cycle_last = last; } +static u64 azx_add_codec_delay_nsec(struct snd_pcm_substream *substream, + u64 nsec) +{ + struct azx_pcm *apcm = snd_pcm_substream_chip(substream); + int stream = substream->stream; + struct hda_pcm_stream *hinfo = apcm->hinfo[stream]; + struct hda_codec *codec = apcm->codec; + u64 codec_nsec; + + if (hinfo->ops.get_delay) { + codec_nsec = + hinfo->ops.get_delay(hinfo, codec, substream) * 1000000; + if (stream == SNDRV_PCM_STREAM_CAPTURE) + nsec = (nsec > codec_nsec) ? nsec - codec_nsec : 0; + else if (stream == SNDRV_PCM_STREAM_PLAYBACK) + nsec += codec_nsec; + } + + return nsec; +} + static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, struct timespec *ts) { @@ -1897,6 +1918,7 @@ static int azx_get_wallclock_tstamp(struct snd_pcm_substream *substream, nsec = timecounter_read(&azx_dev->azx_tc); nsec = div_u64(nsec, 3); /* can be optimized */ + nsec = azx_add_codec_delay_nsec(substream, nsec); *ts = ns_to_timespec(nsec); -- 1.8.1.3.605.g02339dd

3 7

[alsa-devel] [PATCH 2/3] ALSA: hda - system resume may delay resuming a codec
by mengdong.lin＠intel.com 08 Apr '13

08 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> On system resume, a codec may not be ready to resume for some external dependency. e.g. Intel Haswell HDMI codec depends on Gfx to initialize the display pipeline before the codec can resume. To avoid the external dependency blocking system resume, this patch allow system resume ops "snd_hda_resume" skip a not ready codec and mark it as "resume delayed". So the codec resume is delayed until later codec access, assuming the external dependency may be resolved and waiting is acceptable at that time. A codec that needs delaying resume shall explicitly set "support_delay_resume" flag on initialization in its patch ops. And it's up to the codec when it's ready and how to finish the delayed resume. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a05af8b..53b1a1c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5518,7 +5518,10 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 6061360..5fe7362 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -889,6 +889,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/3] ALSA: hda - add new function snd_hda_bus_reset() for codec communitation error
by mengdong.lin＠intel.com 08 Apr '13

08 Apr '13

From: Mengdong Lin <mengdong.lin(a)intel.com> A new function snd_hda_bus_reset() is defined to suspend and resume all codecs on the hda bus. It will be called by bus reset ops azx_bus_reset() as a rescue of codec communitation error. snd_hda_suspend & snd_hda_resume will no longer be called by azx_bus_reset. So snd_hda_resume will only be used in system resume and may delay resuming a codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..a05af8b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5523,6 +5523,26 @@ int snd_hda_resume(struct hda_bus *bus) return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + +/** + * snd_hda_bus_reset - reset the codecs, i.e. suspend and resume the codecs + * @bus: the HDA bus + * + * Returns 0 if successful. + */ +int snd_hda_bus_reset(struct hda_bus *bus) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &bus->codec_list, list) { + cancel_delayed_work_sync(&codec->jackpoll_work); + if (hda_codec_is_power_on(codec)) + hda_call_codec_suspend(codec, false); + hda_call_codec_resume(codec); + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_bus_reset); #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c93f902..6061360 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1076,6 +1076,7 @@ void snd_hda_unlock_devices(struct hda_bus *bus); #ifdef CONFIG_PM int snd_hda_suspend(struct hda_bus *bus); int snd_hda_resume(struct hda_bus *bus); +int snd_hda_bus_reset(struct hda_bus *bus); #endif static inline diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735567e..6f11c7a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1659,8 +1659,7 @@ static void azx_bus_reset(struct hda_bus *bus) struct azx_pcm *p; list_for_each_entry(p, &chip->pcm_list, list) snd_pcm_suspend_all(p->pcm); - snd_hda_suspend(chip->bus); - snd_hda_resume(chip->bus); + snd_hda_bus_reset(chip->bus); } #endif bus->in_reset = 0; -- 1.7.10.4

2 1

Re: [alsa-devel] [RFC] revoke(2) and generic handling of things like remove_proc_entry()
by Clemens Ladisch 08 Apr '13

08 Apr '13

Al Viro wrote: > BTW, snd_card_disconnect() doesn't do anything to existing mappings; smells > like a bug, and there we do have ones with non-trivial ->mmap(). Could > ALSA folks comment? I don't know of any hotplug sound driver that maps memory from a device. All hotplug buses (PCIe, USB, FireWire) involve DMAing from/to main memory. When a sound device is unplugged, sample transfers just stop, but the buffer itself stays as long as the device node is kept opened. (In the case of a capture device, that actually makes sense.) Regards, Clemens

1 0

Re: [alsa-devel] [PATCH] ASoC: Ux500: remove test for undefined Kconfig macro
by Mark Brown 08 Apr '13

08 Apr '13

On Fri, Mar 29, 2013 at 01:28:21PM +0100, Paul Bolle wrote: > A test for CONFIG_SND_SOC_UX500_AB5500 was added in v3.5. But there > never was a corresponding Kconfig symbol so this test has always > evaluated to true. And since AB5500 support was removed in v3.5 it > appears safe to remove this test and a few lines of code. Applied, thanks.

1 0

[alsa-devel] [PATCH] ALSA: hda: fix mute led on HP Envy 15
by Dan Savilonis 08 Apr '13

08 Apr '13

This laptop reports the string HP_Mute_LED_P_G in DMI, but the guess made in hp_set_mute_led() is wrong. So, hard-code the GPIO id. Signed-off-by: Dan Savilonis <djs(a)n-cube.org> --- sound/pci/hda/patch_sigmatel.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1d9d642..3a6b111 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -100,6 +100,7 @@ enum { STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, + STAC_HP_ENVY_15, STAC_92HD83XXX_MODELS }; @@ -2092,6 +2093,15 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ } +static void stac92hd83xxx_fixup_hp_envy_15(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gpio_led = 0x01; /* GPIO0 */ +} + static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2101,6 +2111,7 @@ static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, spec->headset_jack = 1; } + static const struct hda_fixup stac92hd83xxx_fixups[] = { [STAC_92HD83XXX_REF] = { .type = HDA_FIXUP_PINS, @@ -2169,6 +2180,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { {} }, }, + [STAC_HP_ENVY_15] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_hp_envy_15, + .chained = true, + .chain_id = STAC_92HD83XXX_HP, + }, }; static const struct hda_model_fixup stac92hd83xxx_models[] = { @@ -2184,6 +2201,7 @@ static const struct hda_model_fixup stac92hd83xxx_models[] = { { .id = STAC_92HD83XXX_HP_MIC_LED, .name = "hp-mic-led" }, { .id = STAC_92HD83XXX_HEADSET_JACK, .name = "headset-jack" }, { .id = STAC_HP_ENVY_BASS, .name = "hp-envy-bass" }, + { .id = STAC_HP_ENVY_15, .name = "hp-envy-15" }, {} }; @@ -2269,6 +2287,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP Pavilion dv5", STAC_92HD83XXX_HP_INV_LED), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x148a, "HP Mini", STAC_92HD83XXX_HP_LED), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1688, + "HP Envy 15", STAC_HP_ENVY_15), SND_PCI_QUIRK_VENDOR(PCI_VENDOR_ID_HP, "HP", STAC_92HD83XXX_HP), {} /* terminator */ }; @@ -3968,6 +3988,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) snd_hda_add_verbs(codec, stac92hd83xxx_core_init); + snd_hda_pick_fixup(codec, stac92hd83xxx_models, stac92hd83xxx_fixup_tbl, stac92hd83xxx_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); -- 1.8.1.2

1 0

[alsa-devel] [PATCH 0/8] More dmaengine PCM cleanups
by Lars-Peter Clausen 07 Apr '13

07 Apr '13

Hi, The first few patches in this series are a few unrelated cleanups to different dmaengine PCM drivers and one fix for a small compile error I introduced in the last dmaengine PCM cleanup series. The last four patches introduce a common DAI DMA data struct, which will be used by tegra, omap and fsl/imx. Having a common DAI DMA data struct allows us to factor out some redundant code from these drivers into a common helper function in the dmaengine PCM library. This should hopefully be the last set of cleanups before the generic dmaengine PCM driver. - Lars Lars-Peter Clausen (8): ASoC: ep93xx_pcm: Fix compile error ASoC: tegra: Don't claim to support PCM pause and resume ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and capture ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_config ASoC: dmaengine-pcm: Add a common DAI DMA data struct ASoC: tegra: Use common DAI DMA data struct ASoC: fsl: Use common DAI DMA data struct ASoC: omap: Use common DAI DMA data include/sound/dmaengine_pcm.h | 24 ++++++++++++++++++ sound/soc/atmel/atmel-pcm-dma.c | 2 -- sound/soc/cirrus/ep93xx-i2s.c | 4 +-- sound/soc/fsl/fsl_ssi.c | 23 ++++++++++++------ sound/soc/fsl/imx-pcm-dma.c | 25 ++++++------------- sound/soc/fsl/imx-pcm-fiq.c | 4 +-- sound/soc/fsl/imx-pcm.h | 16 ++++-------- sound/soc/fsl/imx-ssi.c | 21 +++++++++------- sound/soc/fsl/imx-ssi.h | 7 ++++-- sound/soc/omap/am3517evm.c | 1 - sound/soc/omap/ams-delta.c | 1 - sound/soc/omap/mcbsp.c | 14 ++++++----- sound/soc/omap/mcbsp.h | 7 +++--- sound/soc/omap/n810.c | 1 - sound/soc/omap/omap-abe-twl6040.c | 1 - sound/soc/omap/omap-dmic.c | 22 ++++++++--------- sound/soc/omap/omap-hdmi.c | 24 ++++++++---------- sound/soc/omap/omap-mcbsp.c | 6 ++--- sound/soc/omap/omap-mcpdm.c | 31 +++++++++++------------- sound/soc/omap/omap-pcm.c | 42 ++++---------------------------- sound/soc/omap/omap-pcm.h | 39 ------------------------------ sound/soc/omap/omap-twl4030.c | 1 - sound/soc/omap/omap3pandora.c | 1 - sound/soc/omap/osk5912.c | 1 - sound/soc/omap/rx51.c | 1 - sound/soc/soc-dmaengine-pcm.c | 39 ++++++++++++++++++++++++++++++ sound/soc/tegra/tegra20_ac97.c | 13 +++++----- sound/soc/tegra/tegra20_ac97.h | 4 +-- sound/soc/tegra/tegra20_i2s.c | 13 +++++----- sound/soc/tegra/tegra20_i2s.h | 4 +-- sound/soc/tegra/tegra20_spdif.c | 7 +++--- sound/soc/tegra/tegra20_spdif.h | 4 +-- sound/soc/tegra/tegra30_ahub.c | 8 +++--- sound/soc/tegra/tegra30_ahub.h | 8 +++--- sound/soc/tegra/tegra30_i2s.c | 13 +++++----- sound/soc/tegra/tegra30_i2s.h | 4 +-- sound/soc/tegra/tegra_pcm.c | 40 +++--------------------------- sound/soc/tegra/tegra_pcm.h | 7 ------ sound/soc/ux500/ux500_pcm.c | 51 ++++++++++++--------------------------- sound/soc/ux500/ux500_pcm.h | 14 ----------- 40 files changed, 226 insertions(+), 322 deletions(-) delete mode 100644 sound/soc/omap/omap-pcm.h -- 1.8.0

7 25

[alsa-devel] Samsung S5PV210 I2S DMA
by Mike Dyer 07 Apr '13

07 Apr '13

Hi All, I'm attempting to play audio into a WM8960 codec attached to I2S 0 on a Samsung S5PV210. The DMA engine emits a lot of messages complaining of (I think) a read fault: dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 dma-pl330 dma-pl330.0: Reset Channel-0 CS-20000f FTC-40000 ... I've had audio playing using the I2S idma driver, so I know clocks + mixer settings are OK. I'd like to get the external DMA working because I also need to do capture. Any suggestions on how to debug? Cheers, Mike

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - Add a quirk for AC700 Chromebook.
by Dylan Reid 07 Apr '13

07 Apr '13

Correct pin configs for the Acer AC700. Most importantly indicate that SPDIF is connected, it routes to HDMI out. Similar to Aspire models, chain in the DMIC fixup and allow it to be applied to this codec (ALC269VB) as well. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_realtek.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1bd2d49..145d078 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2617,7 +2617,8 @@ static void alc271_fixup_dmic(struct hda_codec *codec, }; unsigned int cfg; - if (strcmp(codec->chip_name, "ALC271X")) + if (strcmp(codec->chip_name, "ALC271X") && + strcmp(codec->chip_name, "ALC269VB")) return; cfg = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED) @@ -2838,6 +2839,7 @@ enum { ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, + ALC269_FIXUP_ACER_AC700, }; static const struct hda_fixup alc269_fixups[] = { @@ -3010,6 +3012,19 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_AMIC_MIC2, }, + [ALC269_FIXUP_ACER_AC700] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x99a3092f }, /* int-mic */ + { 0x14, 0x99130110 }, /* speaker */ + { 0x18, 0x03a11c20 }, /* mic */ + { 0x1e, 0x0346101e }, /* SPDIF1 */ + { 0x21, 0x0321101f }, /* HP out */ + { } + }, + .chained = true, + .chain_id = ALC271_FIXUP_DMIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3034,6 +3049,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
by Eldad Zack 07 Apr '13

07 Apr '13

The usb_control_msg() function expects __u16 types and performs the endianness conversions by itself. However, in three places, a conversion is performed before it is handed over to usb_control_msg(), which leads to a double conversion (= no conversion): * snd_usb_nativeinstruments_boot_quirk() * snd_nativeinstruments_control_get() * snd_nativeinstruments_control_put() Caught by sparse: sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types) sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident> sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident> sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types) sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/mixer_quirks.c | 4 ++-- sound/usb/quirks.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 497d274..ebe9144 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -509,7 +509,7 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, - 0, cpu_to_le16(wIndex), + 0, wIndex, &tmp, sizeof(tmp), 1000); up_read(&mixer->chip->shutdown_rwsem); @@ -540,7 +540,7 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, - cpu_to_le16(wValue), cpu_to_le16(wIndex), + wValue, wIndex, NULL, 0, 1000); up_read(&mixer->chip->shutdown_rwsem); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a2ac004..68e9220 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -499,7 +499,7 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, - cpu_to_le16(1), 0, NULL, 0, 1000); + 1, 0, NULL, 0, 1000); if (ret < 0) return ret; -- 1.8.1.5

3 2

[alsa-devel] [PATCH 1/2] ALSA: add DSD formats
by Daniel Mack 07 Apr '13

07 Apr '13

This patch adds two formats for Direct Stream Digital (DSD), a pulse-density encoding format which is described here: https://en.wikipedia.org/wiki/Direct_Stream_Digital DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit stream. In order to provide a compatibility way for pushing DSD samples through ordinary PCM channels, the "DoP open Standard" was invented. See http://www.dsd-guide.com for a copy of the documentation. The two new types added by this patch describe streams that are capable of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8 or x16 data rate, respectively). DSD itself specifies samples in *bit*, while DOP and ALSA handle them as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample rare configuration, according to the following table: configured hardware 352.8kHz 705.6KHz 1411.2KHz <---- sample rate 8-bit 2.8MHz 5.6MHz 11.2MHz 16-bit 5.6MHz 11.2MHz `-----------------------------------' actual DSD sample rates Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- include/sound/pcm.h | 2 ++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm.c | 2 ++ 3 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index aa7b0a8..d957046 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -181,6 +181,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_G723_24_1B _SNDRV_PCM_FMTBIT(G723_24_1B) #define SNDRV_PCM_FMTBIT_G723_40 _SNDRV_PCM_FMTBIT(G723_40) #define SNDRV_PCM_FMTBIT_G723_40_1B _SNDRV_PCM_FMTBIT(G723_40_1B) +#define SNDRV_PCM_FMTBIT_DSD_DOP_U8 _SNDRV_PCM_FMTBIT(DSD_DOP_U8) +#define SNDRV_PCM_FMTBIT_DSD_DOP_U16 _SNDRV_PCM_FMTBIT(DSD_DOP_U16) #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_LE diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1774a5c..eb9eda8 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,7 +214,9 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_24_1B ((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_G723_40 ((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */ #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_G723_40_1B +#define SNDRV_PCM_FORMAT_DSD_DOP_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD DOP format (x8) */ +#define SNDRV_PCM_FORMAT_DSD_DOP_U16 ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD DOP format (x16) */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_DOP_U16 #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 578327e..578a761 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -209,6 +209,8 @@ static char *snd_pcm_format_names[] = { FORMAT(G723_24_1B), FORMAT(G723_40), FORMAT(G723_40_1B), + FORMAT(DSD_DOP_U8), + FORMAT(DSD_DOP_U16), }; const char *snd_pcm_format_name(snd_pcm_format_t format) -- 1.8.1.4

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Re: [alsa-devel] [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
by Jussi Laako 06 Apr '13

06 Apr '13

On 04/05/2013 10:59 PM, Andreas Koch wrote: > However, they are interpreting the ASIO spec for DSD differently. They > express buffer size in samples, which means that for DSD it is measured > in bits. That's how pretty much everybody, including Sony who co-authored the spec, has been interpreting it so far... Otherwise all the latency calculations become inconsistent and need special exceptions for DSD. I agree the ASIO spec is not very clear, however it always talks about number of samples and then specifically mentions that DSD samples are _packed_ into bytes and thus all buffer sizes have to multiples of eight. I have a hidden configuration flag to flip that either way as necessary, but default is that sizes are in _bits_. - Jussi

2 5

[alsa-devel] [PATCH v2 1/2] alsa-lib: pcm.c: fix indentation
by Daniel Mack 06 Apr '13

06 Apr '13

Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- src/pcm/pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 05737d9..2963711 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1634,11 +1634,11 @@ static const char *const snd_pcm_type_names[] = { PCMTYPE(LADSPA), PCMTYPE(DMIX), PCMTYPE(JACK), - PCMTYPE(DSNOOP), - PCMTYPE(IEC958), + PCMTYPE(DSNOOP), + PCMTYPE(IEC958), PCMTYPE(SOFTVOL), - PCMTYPE(IOPLUG), - PCMTYPE(EXTPLUG), + PCMTYPE(IOPLUG), + PCMTYPE(EXTPLUG), PCMTYPE(MMAP_EMUL), }; -- 1.8.1.4

1 1

[alsa-devel] got a segmentation fault when use multi-channel device
by Zhang wei 06 Apr '13

06 Apr '13

hello everone. i am now debuging the multi-channel device on my embeded board with alsa-lib.The version alsa-driver and alsa-lib is 1.0.24. -----------------------------> zz@board:$ ls /dev/snd/ by-path controlC0 midiC0D0 pcmC0D0c pcmC0D0p pcmC0D1p seq timer <--------------------------- i edit the asoundrc configure file.like this: ---------------------------------------------------> pcm.!default{ type hw card 0 } ctl.!default{ type hw card 0 } ################################# ## create a virtul device which has 4 channels for playback. ################################# pcm.multi_playback{ type multi; slaves.a.pcm "hw:0,0"; slaves.a.channels 2; slaves.b.pcm "hw:0,1"; slaves.b.channels 2; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave b; bindings.2.channel 0; bindings.3.slave b; bindings.3.channel 1; } ctl.multi_playback{ type hw; card 0; } pcm.ttable{ type route; slave.pcm "multi_playback"; slave.channels 4; ttable.0.0 1; ttable.1.1 1; ttable.2.2 1; ttable.3.3 1; } ctl.ttable { type hw; card 0; } <-------------------------------------------------- and then get the Segmentation fault: --------------------------------------------------------------> [root@board /tmp]#./pb_pcm_min open...ver:[0.71] playback...capture...device=[multi_playback] #####################snd_pcm_open... zz[snd_dlsym_verify:122],vname=[_snd_config_hook_load_dlsym_config_hook_ 001] zz[snd_dlsym_verify:127] zz[snd_dlsym:167] zz[snd_dlsym:171],name=[snd_config_hook_load] zz[snd_pcm_open_conf:2163] zz[snd_dlsym_verify:122],vname=[__snd_pcm_multi_open_dlsym_pcm_001] Segmentation fault [root@board /tmp]# <--------------------------------------------------------------- To my surpris, it will be fine when run on the PC under the Ubuntu system.I used the same alsa-lib source code and the configure file asoundrc. in my codec-driver .i have not call the functions :snd_soc_add_controls() / snd_soc_dapm_new_controls() / snd_soc_dapm_add_routes(). is it the reason? But the programm have not arrived here. i have traced the source code when the Segmentation fault happened the alsa-lib was calling the function dlysm() with the param vname=__snd_pcm_multi_open_dlsym_pcm_001. But this will not happen on my PC under Ubuntu. could you help me find out what's wrong with me? thanks a lot. zw

1 0

[alsa-devel] PROBLEM: very loud sound of electric discharge on shutdown and restart with ALC882M with all kernels I tried up to 3.9 rc5
by Jenya Y 06 Apr '13

06 Apr '13

I always had this problem in linux and only today tried to search for answers. Solutions I came across involved disabling power management on Intel HDA, which is already 0 by default and also the option of muting on shutdown/ restart, which had absolutely no effect. I wish I could figure this out myself but after 4 hours of searching and trying things out and countless restarts I've given up and decided to come to you for help. Jenya On 03/20/2013 08:41 AM, Jenya Y wrote: > I still didn't receive anything from you as I already mentioned I tried > most of the 'obvious' solutions I came accross and still looking for > answers. > Maybe I'm using the wrong ML for this question. If so, maybe you could at > least suggest another HDA/ ALSA maintainer to contact in this situation. > > TIA, Jenya.

1 0

[alsa-devel] [PATCH v2] ALSA: hda/ca0132 - Update latency based on DSP state.
by Dylan Reid 05 Apr '13

05 Apr '13

The DSP in the CA0132 codec adds a variable latency to audio depending on what processing is being done. Add a new patch op to return that latency for capture and playback streams. The latency is determined by which blocks are enabled and knowing how much latency is added by each block. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v1: Rebase on tiwai's patch to add a pcm_op, this is dependent on that. Get ms to frames conversion right. Test loopback and compare measured vs reported latency. sound/pci/hda/patch_ca0132.c | 55 ++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 53 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 12eb21a..841842c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -131,6 +131,13 @@ enum { /* Effects values size*/ #define EFFECT_VALS_MAX_COUNT 12 +/* Latency introduced by DSP blocks in milliseconds. */ +#define DSP_CAPTURE_INIT_LATENCY 0 +#define DSP_CRYSTAL_VOICE_LATENCY 124 +#define DSP_PLAYBACK_INIT_LATENCY 13 +#define DSP_PLAY_ENHANCEMENT_LATENCY 30 +#define DSP_SPEAKER_OUT_LATENCY 7 + struct ct_effect { char name[44]; hda_nid_t nid; @@ -2743,6 +2750,31 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } +static unsigned int ca0132_playback_pcm_delay(struct hda_pcm_stream *info, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int latency = DSP_PLAYBACK_INIT_LATENCY; + struct snd_pcm_runtime *runtime = substream->runtime; + + if (spec->dsp_state != DSP_DOWNLOADED) + return 0; + + /* Add latency if playback enhancement and either effect is enabled. */ + if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) { + if ((spec->effects_switch[SURROUND - EFFECT_START_NID]) || + (spec->effects_switch[DIALOG_PLUS - EFFECT_START_NID])) + latency += DSP_PLAY_ENHANCEMENT_LATENCY; + } + + /* Applying Speaker EQ adds latency as well. */ + if (spec->cur_out_type == SPEAKER_OUT) + latency += DSP_SPEAKER_OUT_LATENCY; + + return (latency * runtime->rate) / 1000; +} + /* * Digital out */ @@ -2811,6 +2843,23 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, return 0; } +static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ca0132_spec *spec = codec->spec; + unsigned int latency = DSP_CAPTURE_INIT_LATENCY; + struct snd_pcm_runtime *runtime = substream->runtime; + + if (spec->dsp_state != DSP_DOWNLOADED) + return 0; + + if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]) + latency += DSP_CRYSTAL_VOICE_LATENCY; + + return (latency * runtime->rate) / 1000; +} + /* * Controls stuffs. */ @@ -4002,7 +4051,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = { .channels_max = 6, .ops = { .prepare = ca0132_playback_pcm_prepare, - .cleanup = ca0132_playback_pcm_cleanup + .cleanup = ca0132_playback_pcm_cleanup, + .get_delay = ca0132_playback_pcm_delay, }, }; @@ -4012,7 +4062,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = { .channels_max = 2, .ops = { .prepare = ca0132_capture_pcm_prepare, - .cleanup = ca0132_capture_pcm_cleanup + .cleanup = ca0132_capture_pcm_cleanup, + .get_delay = ca0132_capture_pcm_delay, }, }; -- 1.8.1.5

3 9

[alsa-devel] [PATCH 00/10] UAC2: Automatic clock switching
by Eldad Zack 05 Apr '13

05 Apr '13

Hi, This patch series is intended for-next. It provides automatic clock switching for UAC2 devices (patches #6-#8). If the current selected clock is not valid, other clocks will be checked and the first (indexwise) valid clock will be used. Patch #8 provides a module flag to turn this logic off, since in some use cases (e.g., studio work) it might not be desired. But I think most users want their audio devices to "just work", so the default is on. To make the change more obvious, I first moved the validity check around without changing any logic (#6). Patch #10 adds support for clocks with read-only sample frequency control. All other patches included are trivial clean up patches. I'd appreciate a review and/or testing this patch series. This applies to all UAC2 devices, so please let me know if anyone sees any issue. Tested with mainline. Applies against mainline (3.9-rc4), HEAD at 46a1f21a679abaaeae6db9969963dc998c9f1c1c Applies against Takashi's for-next, HEAD at 4abdbd1c2c1832e7270e546307ffb3e56b286db2 Cheers, Eldad Zack (10): ALSA: usb-audio: convert list_for_each to entry variant ALSA: usb-audio: neaten MODULE_DEVICE_TABLE placement ALSA: usb-audio: neaten EXPORT_SYMBOLS placement ALSA: usb-audio: spelling correction ALSA: usb-audio: use endianness macros ALSA: usb-audio: UAC2: do clock validity check earlier ALSA: usb-audio: UAC2: try to find and switch to valid clock ALSA: usb-audio: UAC2: auto clock selection module param ALSA: usb-audio: show err in set_sample_rate_v2 debug ALSA: usb-audio: UAC2: support read-only freq control sound/usb/card.c | 11 ++-- sound/usb/clock.c | 150 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/clock.h | 3 +- sound/usb/endpoint.c | 18 +++---- sound/usb/endpoint.h | 2 +- sound/usb/format.c | 2 +- sound/usb/midi.c | 12 ++--- sound/usb/pcm.c | 32 ++++------- sound/usb/proc.c | 7 ++- sound/usb/stream.c | 12 ++--- sound/usb/usbaudio.h | 1 + 11 files changed, 155 insertions(+), 95 deletions(-) -- 1.8.1.5

4 23

Re: [alsa-devel] Deadlock over semaphore issue with aplay while using dmix
by mateen 05 Apr '13

05 Apr '13

Hi, I checked with Takshi's patch and would like to suggest couple of changes. 1. We keep dmix_down_sem(dmix) and dmix_up_sem(dmix) as macros instead of defining them as functions. 2. We free up the semaphore in snd_pcm_dmix_close() if it is recognized that the semaphore is held up by the same process, which will be indicated by dmix->semlocked flag. Mateen. diff -Nuar dir2/pcm_direct.h dir1/pcm_direct.h --- dir2/pcm_direct.h 2009-12-16 20:48:51.000000000 +0530 +++ dir1/pcm_direct.h 2013-04-05 17:06:48.331497000 +0530 @@ -123,6 +123,7 @@ int ipc_gid; /* IPC socket gid */ int semid; /* IPC global semaphore identification */ int shmid; /* IPC global shared memory identification */ + int semlocked; snd_pcm_direct_share_t *shmptr; /* pointer to shared memory area */ snd_pcm_t *spcm; /* slave PCM handle */ snd_pcm_uframes_t appl_ptr; diff -Nuar dir2/pcm_dmix.c dir1/pcm_dmix.c --- dir2/pcm_dmix.c 2009-12-16 20:48:51.000000000 +0530 +++ dir1/pcm_dmix.c 2013-04-05 17:04:02.781109000 +0530 @@ -285,8 +285,17 @@ */ #ifndef DOC_HIDDEN #ifdef NO_CONCURRENT_ACCESS -#define dmix_down_sem(dmix) snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT) -#define dmix_up_sem(dmix) snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT) +#define dmix_down_sem(dmix) \ +{\ +if (!dmix->semlocked++)\ + snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT);\ +} +#define dmix_up_sem(dmix) \ +{\ + if (!--dmix->semlocked)\ + snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT);\ +} + #else #define dmix_down_sem(dmix) #define dmix_up_sem(dmix) @@ -764,7 +773,13 @@ if (dmix->timer) snd_timer_close(dmix->timer); - snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + if(!dmix->semlocked) + snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + else{ + snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); + snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + } + snd_pcm_close(dmix->spcm); if (dmix->server) snd_pcm_direct_server_discard(dmix); ------------------------------ ---------------------------------------- Message: 1 Date: Thu, 04 Apr 2013 18:30:01 +0200 From: Takashi Iwai <tiwai(a)suse.de> To: Jaroslav Kysela <perex(a)perex.cz> Cc: alsa-devel(a)alsa-project.org Subject: Re: [alsa-devel] Deadlock over semaphore issue with aplay while using dmix Message-ID: <s5h38v645ti.wl%tiwai(a)suse.de> Content-Type: text/plain; charset=US-ASCII At Thu, 04 Apr 2013 13:33:08 +0200, Jaroslav Kysela wrote: > > Date 4.4.2013 11:27, mateen wrote: > > Hi, > > > > I seeing sometimes deadlock issue with dmix when I press CTRL+C. > > > > Aplay's signal handler calls snd_pcm_close() if an interrupt occurs. > > snd_pcm_close() will internally call pcm->ops->close() which will fall to > > snd_pcm_dmix_close() in case you are using dmix. > > > > snd_pcm_dmix_close() will try to acquire the semaphore with > > snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT). > > The same semaphore is acquired in snd_pcm_dmix_sync_area() with > > dmix_down_sem() in case of non-concurrent access. > > > > If semaphore is acquired in snd_pcm_dmix_sync_area() which is in thread > > context and interrupt comes, which invokes the signal handler which is in > > ISR context, which calls snd_pcm_close() which in turn calls > > snd_pcm_dmix_close() then we see a deadlock since semaphore is not released > > from thread context and ISR is waiting indefinitely on the same semaphore. > > > > Please suggest a suitable solution for this. > > It seems that also other configurations (alsa-lib plugins) have trouble > with the closing from the signal handler - I hit mutex issues with the > PulseAudio plugin, too. > > The question is, how we can do a clean path in this case. Looking to the > current alsa-lib code, I would suggest to add the snd_pcm_abort() > function (may be called from the interrupt handler) to notify the > library to not ignore -EINTR return codes from poll() and other i/o ops > and pass it to the caller (application) to finish the normal close sequence. > > Opinions? Isn't it only about the direct plugins with the case where no coherent memory is available? If so, we may just avoid the deadlock by checking some internal flag like the patch below (untested)? It's no perfect but just some proof, of course. Takashi -- Regards, Shaikh Mateen S. 09423350444

2 2

[alsa-devel] Deadlock over semaphore issue with aplay while using dmix
by mateen 05 Apr '13

05 Apr '13

Hi, I seeing sometimes deadlock issue with dmix when I press CTRL+C. Aplay's signal handler calls snd_pcm_close() if an interrupt occurs. snd_pcm_close() will internally call pcm->ops->close() which will fall to snd_pcm_dmix_close() in case you are using dmix. snd_pcm_dmix_close() will try to acquire the semaphore with snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT). The same semaphore is acquired in snd_pcm_dmix_sync_area() with dmix_down_sem() in case of non-concurrent access. If semaphore is acquired in snd_pcm_dmix_sync_area() which is in thread context and interrupt comes, which invokes the signal handler which is in ISR context, which calls snd_pcm_close() which in turn calls snd_pcm_dmix_close() then we see a deadlock since semaphore is not released from thread context and ISR is waiting indefinitely on the same semaphore. Please suggest a suitable solution for this. -- Regards, Shaikh Mateen S. 09423350444

3 6

[alsa-devel] [PATCH v6 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 05 Apr '13

05 Apr '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> Tested-by: Daniel Mack <zonque(a)gmail.com> --- v6: CC Mark Brown v5: add Daniel's Tested-by v4: change subjuct to include "ALSA: ASoC: " sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

2 2

Re: [alsa-devel] [PATCH 2/2] alsa-lib: bring pcm.h and pcm.c in sync with the kernel list
by Daniel Mack 05 Apr '13

05 Apr '13

Hi Andreas, On 28.03.2013 05:58, Andreas Koch wrote: >> Am I right to assume that in your case it's the same as for DoP, >> IOW, oldest in MSB? > > Good point - thanks for bringing it up - another can of worms. > With Windows having a little endian philosophy we started early on > with the first DSD workstation by putting the oldest sample in the > LSB position. ASIO came along and adopted the same thing, but then > extended support for the opposite as well. Rigisystems (offering OEM > modules for USB audio) came along about a year ago and insisted that > there is only their way, i.e. putting the oldest bit in the MSB > position. I tried to explain to them that there are already hundreds > of products out in the market that do it differently, but that didn't > matter to them. The result is that now we have implementations with > both. Up until now that was not a huge issue, because there is no > generic Windows driver and every hardware manufacturer had to have a > custom driver. > Now of course, with your heroic effort in creating a generic driver > that would help everyone, we have a mess. Some DACs expect little > endian (mine and others as well) and others (all that are based on > Rigisystems modules) big endian positions. > > Since the time I had the discussions with Rigisystems I didn't have > any further communication with them. Maybe it is a good idea to touch > base with them again and find out what the latest is. I will try my > contact there and let you know. Did you hear back from them yet? Do you think we should allocate types for Big Endian DOP formats? Thanks, Daniel

1 0

[alsa-devel] [PATCH] pcm: fix typo, should be SND_PCM_STATE_PREPARED.
by dsvensson＠gmail.com 05 Apr '13

05 Apr '13

From: Daniel Svensson <dsvensson(a)gmail.com> Signed-off-by: Daniel Svensson <dsvensson(a)gmail.com> --- src/pcm/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 05737d9..0868dd5 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -162,7 +162,7 @@ The PCM device has accepted communication parameters and it is waiting for #snd_pcm_prepare() call to prepare the hardware for selected operation (playback or capture). -\par SND_PCM_STATE_PREPARE +\par SND_PCM_STATE_PREPARED The PCM device is prepared for operation. Application can use #snd_pcm_start() call, write or read data to start the operation. -- 1.7.12.4 (Apple Git-37)

2 1

[alsa-devel] [PATCH] ALSA: hda/cirrus - Add a quirk for Stumpy ChromeBox.
by Dylan Reid 05 Apr '13

05 Apr '13

The Stumpy ChromeBox needs its pin configs fixed up. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_cirrus.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 0d9c58f..bd8d46c 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -68,6 +68,7 @@ enum { enum { CS421X_CDB4210, CS421X_SENSE_B, + CS421X_STUMPY, }; /* Vendor-specific processing widget */ @@ -538,6 +539,7 @@ static int patch_cs420x(struct hda_codec *codec) /* CS4210 board names */ static const struct hda_model_fixup cs421x_models[] = { { .id = CS421X_CDB4210, .name = "cdb4210" }, + { .id = CS421X_STUMPY, .name = "stumpy" }, {} }; @@ -559,6 +561,17 @@ static const struct hda_pintbl cdb4210_pincfgs[] = { {} /* terminator */ }; +/* Stumpy ChromeBox */ +static const struct hda_pintbl stumpy_pincfgs[] = { + { 0x05, 0x022120f0 }, + { 0x06, 0x901700f0 }, + { 0x07, 0x02a120f0 }, + { 0x08, 0x77a70037 }, + { 0x09, 0x77a6003e }, + { 0x0a, 0x434510f0 }, + {} /* terminator */ +}; + /* Setup GPIO/SENSE for each board (if used) */ static void cs421x_fixup_sense_b(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -578,7 +591,11 @@ static const struct hda_fixup cs421x_fixups[] = { [CS421X_SENSE_B] = { .type = HDA_FIXUP_FUNC, .v.func = cs421x_fixup_sense_b, - } + }, + [CS421X_STUMPY] = { + .type = HDA_FIXUP_PINS, + .v.pins = stumpy_pincfgs, + }, }; static const struct hda_verb cs421x_coef_init_verbs[] = { -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Add fixed codec delay to runtime delay.
by Dylan Reid 05 Apr '13

05 Apr '13

Allow a codec to specify a delay in addition to the delay before samples are passed through DMA. Some codecs have large processing delays because of built-in DSPs that require codec-side buffers of many milliseconds. Default to zero delay if the codec doesn't specify a callback to get the processing delay. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_codec.c | 17 +++++++++++++++++ sound/pci/hda/hda_codec.h | 4 ++++ sound/pci/hda/hda_intel.c | 2 ++ 3 files changed, 23 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..1d792dd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3724,6 +3724,23 @@ void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, } EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all); +/** + * snd_hda_codec_processing_delay - get the fixed delay through the codec + * @codec: HD-audio codec + * @stream: playback or capture + * + * This is needed for codecs that contain DSPs that cause long delays. + */ +unsigned int snd_hda_codec_processing_delay(struct hda_codec *codec, + unsigned int stream) +{ + if (codec->patch_ops.get_processing_delay) + return codec->patch_ops.get_processing_delay(codec, stream); + + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_processing_delay); + /* * supported power states check */ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 23ca172..23bd1e7 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -725,6 +725,8 @@ struct hda_codec_ops { int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); #endif void (*reboot_notify)(struct hda_codec *codec); + unsigned int (*get_processing_delay)(struct hda_codec *codec, + unsigned int stream); }; /* record for amp information cache */ @@ -1063,6 +1065,8 @@ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); void snd_hda_bus_reboot_notify(struct hda_bus *bus); void snd_hda_codec_set_power_to_all(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); +unsigned int snd_hda_codec_processing_delay(struct hda_codec *codec, + unsigned int stream); int snd_hda_lock_devices(struct hda_bus *bus); void snd_hda_unlock_devices(struct hda_bus *bus); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 418bfc0..422bfaf 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2349,6 +2349,7 @@ static unsigned int azx_get_position(struct azx *chip, struct azx_dev *azx_dev, bool with_check) { + struct azx_pcm *apcm = snd_pcm_substream_chip(azx_dev->substream); unsigned int pos; int stream = azx_dev->substream->stream; int delay = 0; @@ -2400,6 +2401,7 @@ static unsigned int azx_get_position(struct azx *chip, chip->driver_caps &= ~AZX_DCAPS_COUNT_LPIB_DELAY; } azx_dev->substream->runtime->delay = + snd_hda_codec_processing_delay(apcm->codec, stream) + bytes_to_frames(azx_dev->substream->runtime, delay); } trace_azx_get_position(chip, azx_dev, pos, delay); -- 1.8.1.5

2 5

[alsa-devel] Handle large period size end-of-stream situation
by Patrick Lai 04 Apr '13

04 Apr '13

Hi, I am dealing with a use case requiring passing large buffer down to the DSP. Application in the user-space passes PCM through mmap interface. Per my understanding, ALSA is designed to work with DMA. Data is transferred in chunk equal to period size. At the end of playback, there may not be enough audio data left in the music stream to fill entire period. If remaining audio data only takes up very small chunk of period, playback takes longer to stop. Given that period size I have to deal with is quite large, this problem is observed easily. If my understanding is correct, what is the standard/recommended way of handling end of stream case? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

3 6

[alsa-devel] [PATCH v5 1/2] ALSA: ASoC: davinci-mcasp: don't configure AFSX direction in DIT mode
by yegorslists＠googlemail.com 04 Apr '13

04 Apr '13

From: Yegor Yefremov <yegorslists(a)googlemail.com> AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists(a)googlemail.com> Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> Tested-by: Daniel Mack <zonque(a)gmail.com> --- v5: add Daniel's Tested-by v4: change subjuct to include "ALSA: ASoC: " sound/soc/davinci/davinci-mcasp.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 9321e5c..90a9b99 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -772,9 +772,6 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) /* S/PDIF */ static void davinci_hw_dit_param(struct davinci_audio_dev *dev) { - /* Set the PDIR for Serialiser as output */ - mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AFSX); - /* TXMASK for 24 bits */ mcasp_set_reg(dev->base + DAVINCI_MCASP_TXMASK_REG, 0x00FFFFFF); -- 1.7.7

3 6

[alsa-devel] ASoC: BeagleBoard driver development (PCM3168)
by Wendelin Klimann 04 Apr '13

04 Apr '13

Hello Peter I am stuck again in my ASoc driver development and it would be really nice if you could give me an hint how i could solve this. Acctually i try to connect my PCM3168 Audio Codec to the McBSP on the BeagleBoard. My new driver loads fine and i got 2 soundcards: *root@beagleboard:/lib/modules/3.7.4+/kernel/sound/soc/omap# aplay -l* **** List of PLAYBACK Hardware Devices **** card 0: omap3beagle [omap3beagle], device 0: TWL4030 twl4030-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: pcm3168 [pcm3168], device 0: PCM3168 pcm3168-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Your TWL4030 driver is still working perfect with: *root@beagleboard:/lib/modules/3.7.4+/kernel/sound/soc/omap# aplay -Dhw:0,0 /home/root/fifi.wav* [ 1245.302551] omap-dma-engine omap-dma-engine: allocating channel for 33 Playing WAVE '/home/root/fifi.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo but when i play the same file with my new driver i get an error (as codec master and also as codec slave): *root@beagleboard:/lib/modules/3.7.4+/kernel/sound/soc/omap# aplay -Dhw:1,0 /home/root/fifi.wav* [ 1128.677337] omap-dma-engine omap-dma-engine: allocating channel for 17 Playing WAVE '/h[ 1128.689544] can't set codec DAI configuration - pcm3168 ome/root/fifi.wa[ 1128.695281] asoc: machine hw_params failed: -524 v' : Signed 16 b[ 1128.702514] omap-dma-engine omap-dma-engine: freeing channel for 17 it Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1145: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (124988 124989) PERIOD_SIZE: 5512 PERIOD_BYTES: 22048 PERIODS: 5 BUFFER_TIME: (624943 624944) BUFFER_SIZE: 27560 BUFFER_BYTES: 110240 TICK_TIME: 0 To check whether my machine driver (/omap-pcm3168.c/) is working i tried to implement the /twl4030.c/ codec driver into it (and deactivated the omap_twl4030_audio_init() function), which worked just fine. With this setup i changed the /.cpu_dai_name = "omap-mcbsp.2"/ to /.cpu_dai_name = "omap-mcbsp.3"/ and i adjusted the codec as slave (/SND_SOC_DAIFMT_CBS_CFS/) to test whether the McBSP3 setup is right (the PCM3168 codec was disconnected). I still got a clock signal on the McBSP2 but the McBSP2_DX had no signal anymore and on the McBSP3 i did not get any signal at all (measured with an osziloscope). I am a little bit confused as i am not shure why there is still the clk signal on the McBSP2 and furthermore i am not shure whether i test the McBSP3 properly or whether there is a proplem in the McBSP3 setup. The PinMUX for the McBSP3 is done in the kernel (/board-omap3beagle.c/) by setting: static void __init omap3_beagle_config_mcbsp3_mux(void) { omap_mux_init_signal("mcbsp3_fsx.mcbsp3_fsx", OMAP_PIN_INPUT); omap_mux_init_signal("uart2_cts.mcbsp3_dx", OMAP_PIN_OUTPUT); omap_mux_init_signal("uart2_rts.mcbsp3_dr", OMAP_PIN_INPUT); /* NOTE: Clock pins need to be in input mode */ omap_mux_init_signal("uart2_tx.mcbsp3_clkx", OMAP_PIN_INPUT); } It would be really nice if you could help me. Thanks Wendelin ****************************************************************************************************** * kernel_3.7.4+/sound/soc/omap/omap-pcm3168.c * ****************************************************************************************************** /* * PCM3168 ASoC driver for BeagleBoard. * * based on: * > omap3beagle.c -- SoC audio for OMAP3 Beagle * > Author: Steve Sakoman <steve(a)sakoman.com> * * adapted by Klimann Wendelin <wklimann(a)hotmail.com> * * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * version 2 as published by the Free Software Foundation. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA * */ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/pcm_params.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> //#include <plat/mcbsp.h> #include "omap-mcbsp.h" #include "omap-pcm.h" /* * Uncomment to test codec in slave mode or without actual codec. This makes * possible to test this driver by letting the OMAP to be DAI link master */ #define PCM3168_CODEC_SLAVE 1 static int pcm3168_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_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int fmt, div; int ret; #ifdef PCM3168_CODEC_SLAVE fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_IB_NF; #else fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF; #endif /* Set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, fmt); if (ret < 0) { printk(KERN_ERR "can't set codec DAI configuration - pcm3168\n"); return ret; } /* Set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, fmt); if (ret < 0) { printk(KERN_ERR "can't set cpu DAI configuration - pcm3168\n"); return ret; } #ifdef PCM3168_CODEC_SLAVE ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_FCLK, 48000000, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("can't set McBSP sysclk - pcm3168\n"); return ret; } /* * Calculate McBSP SRG divisor in McBSP master mode */ div = 48000000 / params_rate(params) / params_channels(params); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: div /= 16; break; case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S32_LE: div /= 32; break; }; /* * Round to maximum divisor if needed. This means that extra bit-clock * cycles are transmitted when sample rate and number of bits in frame * (channels * sample bits) are low. */ if (div >= 256) div = 256; ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, div); if (ret < 0) { pr_err("can't set SRG clock divider - pcm3168\n"); return ret; } #endif return 0; } static struct snd_soc_ops pcm3168_ops = { .hw_params = pcm3168_hw_params, }; /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link pcm3168_dai = { .name = "PCM3168", .stream_name = "PCM3168", .cpu_dai_name = "omap-mcbsp.3", .platform_name = "omap-pcm-audio", .codec_dai_name = "pcm3168-hifi", .codec_name = "pcm3168-codec.0", .ops = &pcm3168_ops, }; /* Audio machine driver */ static struct snd_soc_card snd_soc_pcm3168 = { .name = "pcm3168", .owner = THIS_MODULE, .dai_link = &pcm3168_dai, .num_links = 1, }; struct platform_device pcm3168_codec = { .name = "pcm3168-codec", .id = 0, }; struct platform_device pcm3168_soc_audio = { .name = "pcm3168-soc-audio", .id = 0, }; static int __devinit pcm3168_soc_probe(struct platform_device *pdev) { struct snd_soc_card *card = &snd_soc_pcm3168; int ret; pr_info("OMAP3 Beagle - PCM3168 ASoC init\n"); card->dev = &pdev->dev; ret = snd_soc_register_card(card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d - pcm3168\n", ret); return ret; } return 0; } static int __devexit pcm3168_soc_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); snd_soc_unregister_card(card); return 0; } static struct platform_driver pcm3168_driver = { .driver = { .name = "pcm3168-soc-audio", .owner = THIS_MODULE, }, .probe = pcm3168_soc_probe, .remove = __devexit_p(pcm3168_soc_remove), }; static int __init pcm3168_soc_init(void) { platform_device_register(&pcm3168_codec); platform_device_register(&pcm3168_soc_audio); return platform_driver_register(&pcm3168_driver); } module_init(pcm3168_soc_init); static void __exit pcm3168_soc_exit(void) { platform_driver_unregister(&pcm3168_driver); platform_device_unregister(&pcm3168_soc_audio); platform_device_unregister(&pcm3168_codec); } module_exit(pcm3168_soc_exit); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_DESCRIPTION("ALSA SoC PCM3168 add on Soundcard"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:pcm3168-soc-audio"); ****************************************************************************************************** * kernel_3.7.4+/sound/soc/codecs/pcm3168.c * ****************************************************************************************************** /* * ALSA Soc PCM3168 codec support * * Author: Klimann Wendelin <wklimann(a)hotmail.com> * * based on: * > pcm3008.c * > Author: Hugo Villeneuve * > Copyright (C) 2008 Lyrtech inc * > * > Based on AC97 Soc codec, original copyright follow: * > Copyright 2005 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * version 2 as published by the Free Software Foundation. * * This program is distributed in the hope that it will be useful, but * WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA * 02110-1301 USA */ #include <linux/init.h> #include <linux/kernel.h> #include <linux/device.h> #include <linux/gpio.h> #include <linux/slab.h> #include <linux/module.h> #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/module.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> #include <sound/pcm_params.h> //#include <sound/soc-dapm.h> #include <sound/soc.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> //#include <plat/mcbsp.h> #include "../omap/omap-mcbsp.h" #include "../omap/omap-pcm.h" #include "pcm3168.h" #define PCM3168_VERSION "0.1" #define PCM3168_RATES SNDRV_PCM_RATE_8000_96000 /*(SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000)*/ #define PCM3168_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_driver pcm3168_dai = { .name = "pcm3168-hifi", .playback = { .stream_name = "PCM3168 Playback", .channels_min = 2, .channels_max = 8, .rates = PCM3168_RATES, .formats = PCM3168_FORMATS, //SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE .sig_bits = 24, }, .capture = { .stream_name = "PCM3168 Capture", .channels_min = 2, .channels_max = 6, .rates = PCM3168_RATES, .formats = PCM3168_FORMATS, //SNDRV_PCM_FMTBIT_S16_LE, //SNDRV_PCM_FMTBIT_S32_LE .sig_bits = 24, }, }; static void pcm3168_gpio_free(struct pcm3168_setup_data *setup) { // gpio_free(setup->dem0_pin); // gpio_free(setup->dem1_pin); // gpio_free(setup->pdad_pin); // gpio_free(setup->pdda_pin); } static int pcm3168_soc_probe(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; int ret = 0; printk(KERN_ALERT "in SOC Probe codec -> kli \n"); printk(KERN_INFO "PCM3168 SoC Audio Codec %s\n", PCM3168_VERSION); return ret; gpio_err: // pcm3168_gpio_free(setup); return ret; } static int pcm3168_soc_remove(struct snd_soc_codec *codec) { struct pcm3168_setup_data *setup = codec->dev->platform_data; // pcm3168_gpio_free(setup); return 0; } #define pcm3168_soc_suspend NULL #define pcm3168_soc_resume NULL static struct snd_soc_codec_driver soc_codec_dev_pcm3168 = { .probe = pcm3168_soc_probe, .remove = pcm3168_soc_remove, .suspend = pcm3168_soc_suspend, .resume = pcm3168_soc_resume, }; static int __devinit pcm3168_codec_probe(struct platform_device *pdev) { int ret; printk(KERN_ALERT "probe pcm3168 codec -> kli \n"); ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm3168, &pcm3168_dai, 1); printk(KERN_ALERT "probe_after pcm3168 codec -> kli = %d \n", ret); return ret; } static int __devexit pcm3168_codec_remove(struct platform_device *pdev) { printk(KERN_ALERT "remove pcm3168 codec -> kli \n"); snd_soc_unregister_codec(&pdev->dev); return 0; } MODULE_ALIAS("platform:pcm3168-codec"); static struct platform_driver pcm3168_codec_driver = { .probe = pcm3168_codec_probe, .remove = __devexit_p(pcm3168_codec_remove), .driver = { .name = "pcm3168-codec", .owner = THIS_MODULE, }, }; static int __init pcm3168_modinit(void) { printk(KERN_ALERT "in init of 3168 -> kli \n"); return platform_driver_register(&pcm3168_codec_driver); } module_init(pcm3168_modinit); static void __exit pcm3168_exit(void) { printk(KERN_ALERT "in exit of 3168 -> kli \n"); platform_driver_unregister(&pcm3168_codec_driver); } module_exit(pcm3168_exit); MODULE_DESCRIPTION("Soc PCM3168 driver"); MODULE_AUTHOR("Klimann Wendelin <wklimann(a)hotmail.com>"); MODULE_LICENSE("GPL");

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[alsa-devel] [PATCH] ALSA: hda - fix typo in proc output
by David Henningsson 04 Apr '13

04 Apr '13

Rename "Digitial In" to "Digital In". This function is only used for proc output, so should not cause any problems to change. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 17286b3..e37b738 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg) "Line Out", "Speaker", "HP Out", "CD", "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" + "SPDIF In", "Digital In", "Reserved", "Other" }; return jack_types[(cfg & AC_DEFCFG_DEVICE) -- 1.7.9.5

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[alsa-devel] [PATCHv2 00/10] UAC2: Automatic clock switching
by Eldad Zack 04 Apr '13

04 Apr '13

This is version 2 of the automatic clock switching. v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-March/060734.html No patches added or removed. Rebased patches with "ALSA: usb: Work around CM6631 sample rate change bug", commit 690a863ff03d9a29ace2b752b8f802fba78a842f . Clemens, Takashi and Torstein, thank you very much for reviewing and providing comments! Changes: * Fixed formatting * (#5) Fixed type cast: u32 -> __l32 * (#6) Added rational to commit message * (#7) Fixed assignment of int to unsigned char, return error code if encountered * (#8) Fixed the parameter name, annotated the default value * (#10) Avoid second read of frequency for read-only clocks Tested with Takashi's for-next. Applies against Takashi's for-next HEAD efc33ce197e4b6aaddf1eb2ba6293f51daf3c283 . Cheers, Eldad Zack (10): ALSA: usb-audio: convert list_for_each to entry variant ALSA: usb-audio: neaten MODULE_DEVICE_TABLE placement ALSA: usb-audio: neaten EXPORT_SYMBOLS placement ALSA: usb-audio: spelling correction ALSA: usb-audio: use endianness macros ALSA: usb-audio: UAC2: do clock validity check earlier ALSA: usb-audio: UAC2: try to find and switch to valid clock ALSA: usb-audio: UAC2: auto clock selection module param ALSA: usb-audio: show err in set_sample_rate_v2 debug ALSA: usb-audio: UAC2: support read-only freq control sound/usb/card.c | 11 ++-- sound/usb/clock.c | 153 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/clock.h | 3 +- sound/usb/endpoint.c | 18 +++--- sound/usb/endpoint.h | 2 +- sound/usb/format.c | 2 +- sound/usb/midi.c | 12 ++-- sound/usb/pcm.c | 32 ++++------- sound/usb/proc.c | 7 +-- sound/usb/stream.c | 12 ++-- sound/usb/usbaudio.h | 1 + 11 files changed, 157 insertions(+), 96 deletions(-) -- 1.8.1.5

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[alsa-devel] [PATCH] Enabling Realtek ALC 671 codec
by Rainer Koenig 04 Apr '13

04 Apr '13

Changes: * Added the device ID to the modalias list and assinged ALC662 patches for it * Added 4 port support for the device ID 0671 in alc662_parse_auto_config Signed-off-by: Rainer Koenig <Rainer.Koenig(a)ts.fujitsu.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7743775..fcbf4c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6663,7 +6663,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) const hda_nid_t *ssids; if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 || + codec->vendor_id == 0x10ec0671) ssids = alc663_ssids; else ssids = alc662_ssids; @@ -7115,6 +7116,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, + { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, -- Dipl.-Inf. (FH) Rainer Koenig Project Manager Linux Clients Dept. PDG WPS R&D SW OSE Fujitsu Technology Solutions Bürgermeister-Ullrich-Str. 100 86199 Augsburg Germany Telephone: +49-821-804-3321 Telefax: +49-821-804-2131 Mail: mailto:Rainer.Koenig@ts.fujitsu.com Internet ts.fujtsu.com Company Details ts.fujitsu.com/imprint.html

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[alsa-devel] How to use implicit feedback with full duplex?
by Clemens Ladisch 04 Apr '13

04 Apr '13

Hi, I thought I'd try to use implicit feedback with my simple audio device: Interface Descriptor: bInterfaceNumber 1 bAlternateSetting 0 Interface Descriptor: bInterfaceNumber 1 bAlternateSetting 1 AudioStreaming Interface Descriptor: ... Endpoint Descriptor: bEndpointAddress 0x01 EP 1 OUT Interface Descriptor: bInterfaceNumber 2 bAlternateSetting 0 Interface Descriptor: bInterfaceNumber 2 bAlternateSetting 1 AudioStreaming Interface Descriptor: ... Endpoint Descriptor: bEndpointAddress 0x82 EP 2 IN So I wrote a patch to configure it like those M-Audio devices: --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -391,6 +391,17 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; switch (subs->stream->chip->usb_id) { + case USB_ID(0x0582, 0x0018): + if (is_playback) { + implicit_fb = 1; + ep = 0x82; + iface = usb_ifnum_to_if(dev, 2); + if (!iface || iface->num_altsetting < 2) + return -EINVAL; + alts = &iface->altsetting[1]; + goto add_sync_ep; + } + break; case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ if (is_playback) { implicit_fb = 1; This works fine when playing something: kernel: setting usb interface 1:1 kernel: Creating new playback data endpoint #1 kernel: Creating new capture data endpoint #82 kernel: Setting params for ep #1 (type 0, 8 urbs), ret=0 kernel: match_endpoint_audioformats: (fmt @ffff8801fefa8900) score 2 kernel: Setting params for ep #82 (type 0, 8 urbs), ret=0 kernel: Starting data EP @ffff8801fd4e8000 kernel: Starting sync EP @ffff8801fd2ac000 But when I then try to record at the same time, the driver refuses to configure the input endpoint (to the only format, which is already set): kernel: setting usb interface 2:1 kernel: Re-using EP 82 in iface 2,1 @ffff8801fd2ac000 kernel: Unable to change format on ep #82: already in use And despite that "alreay in use" check, the input endpoint is affected so much that playback breaks. Is full duplex supposed to work? Does it work with other devices? Regards, Clemens

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