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

[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

2 1

[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

2 1

[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

2 1

[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

2 1

[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

2 4

[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

2 1

[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.

2 1

[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

3 3

[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

5 19

[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

5 23