From 0fe09a45c4848b5b5607b968d959fdc1821c161d Mon Sep 17 00:00:00 2001 From: Linus Torvalds Date: Fri, 19 Apr 2013 10:01:04 -0700 Subject: vm: convert snd_pcm_lib_mmap_iomem() to vm_iomap_memory() helper This is my example conversion of a few existing mmap users. The pcm mmap case is one of the more straightforward ones. Acked-by: Takashi Iwai Signed-off-by: Linus Torvalds --- sound/core/pcm_native.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 71ae86ca64ac..eb560fa32321 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3222,18 +3222,10 @@ EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap); int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - long size; - unsigned long offset; + struct snd_pcm_runtime *runtime = substream->runtime;; area->vm_page_prot = pgprot_noncached(area->vm_page_prot); - area->vm_flags |= VM_IO; - size = area->vm_end - area->vm_start; - offset = area->vm_pgoff << PAGE_SHIFT; - if (io_remap_pfn_range(area, area->vm_start, - (substream->runtime->dma_addr + offset) >> PAGE_SHIFT, - size, area->vm_page_prot)) - return -EAGAIN; - return 0; + return vm_iomap_memory(area, runtime->dma_addr, runtime->dma_bytes); } EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); -- cgit v1.2.3 From d1e1406c6ed0b92200a7de2a09fbab65661dba3c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 20 Apr 2013 19:29:00 +0200 Subject: ASoC: generic-dmaengine-pcm: Add support for half-duplex Some platforms which are half-duplex share the same DMA channel between the playback and capture stream. Add support for this to the generic dmaengine PCM driver. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 43 ++++++++++++++++++++++++----------- 1 file changed, 30 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ae0c37e66ae0..5fd5ed4c0a96 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -29,7 +29,7 @@ struct dmaengine_pcm { struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; - bool compat; + unsigned int flags; }; static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) @@ -128,6 +128,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) + return pcm->chan[0]; + if (pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); @@ -148,7 +151,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - if (!pcm->chan[i] && pcm->compat) { + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); } @@ -215,6 +218,25 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_CAPTURE] = "rx", }; +static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, + struct device_node *of_node) +{ + unsigned int i; + + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !of_node) + return; + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { + pcm->chan[0] = of_dma_request_slave_channel(of_node, "tx_rx"); + pcm->chan[1] = pcm->chan[0]; + } else { + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + pcm->chan[i] = of_dma_request_slave_channel(of_node, + dmaengine_pcm_dma_channel_names[i]); + } + } +} + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -225,23 +247,15 @@ int snd_dmaengine_pcm_register(struct device *dev, const struct snd_dmaengine_pcm_config *config, unsigned int flags) { struct dmaengine_pcm *pcm; - unsigned int i; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; pcm->config = config; + pcm->flags = flags; - if (flags & SND_DMAENGINE_PCM_FLAG_COMPAT) - pcm->compat = true; - - if (!(flags & SND_DMAENGINE_PCM_FLAG_NO_DT) && dev->of_node) { - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = of_dma_request_slave_channel(dev->of_node, - dmaengine_pcm_dma_channel_names[i]); - } - } + dmaengine_pcm_request_chan_of(pcm, dev->of_node); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_soc_add_platform(dev, &pcm->platform, @@ -272,8 +286,11 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_platform_to_pcm(platform); for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - if (pcm->chan[i]) + if (pcm->chan[i]) { dma_release_channel(pcm->chan[i]); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; + } } snd_soc_remove_platform(platform); -- cgit v1.2.3 From 57364f9ae24c3a29163c23d5219bccef7bccc96a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 20 Apr 2013 19:29:01 +0200 Subject: ASoC: mxs-pcm: Set SNDRV_PCM_INFO_HALF_DUPLEX The MXS SAIF is only half-duplex so set the SNDRV_PCM_INFO_HALF_DUPLEX flag for the PCM in order to prevent playback and capture from running at the same time. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 7bceb16d0fd9..907cdb1f989d 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -43,7 +43,8 @@ static struct snd_pcm_hardware snd_mxs_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_INTERLEAVED, + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, -- cgit v1.2.3 From 8c1bb4ecbca575ec89310a50c3d3dd475bf81fd0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 20 Apr 2013 19:29:02 +0200 Subject: ASoC: mxs: Setup dma data in DAI probe This allows us to access the DAI DMA data when we create the PCM. We'll use this when converting mxs to generic DMA engine PCM driver. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/mxs-saif.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index f13bd8730b0f..d796a393968d 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -369,7 +369,6 @@ static int mxs_saif_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); - snd_soc_dai_set_dma_data(cpu_dai, substream, &saif->dma_param); /* clear error status to 0 for each re-open */ saif->fifo_underrun = 0; @@ -605,6 +604,8 @@ static int mxs_saif_dai_probe(struct snd_soc_dai *dai) struct mxs_saif *saif = dev_get_drvdata(dai->dev); snd_soc_dai_set_drvdata(dai, saif); + dai->playback_dma_data = &saif->dma_param; + dai->capture_dma_data = &saif->dma_param; return 0; } -- cgit v1.2.3 From a8956908bf03418d7264de79e1e988628183f537 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 20 Apr 2013 19:29:03 +0200 Subject: ASoC: mxs: Use generic dmaengine PCM Use the generic dmaengine PCM driver instead of a custom implementation. Signed-off-by: Lars-Peter Clausen Tested-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/mxs/Kconfig | 2 +- sound/soc/mxs/mxs-pcm.c | 136 ++++-------------------------------------------- 2 files changed, 12 insertions(+), 126 deletions(-) (limited to 'sound') diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index b6fa77678d97..78d321cbe8b4 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -1,7 +1,7 @@ menuconfig SND_MXS_SOC tristate "SoC Audio for Freescale MXS CPUs" depends on ARCH_MXS - select SND_SOC_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the MXS SAIF interface. diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 907cdb1f989d..b41fffc056fb 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -18,27 +18,18 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#include -#include #include -#include #include -#include #include -#include -#include -#include #include -#include #include -#include #include #include #include "mxs-pcm.h" -static struct snd_pcm_hardware snd_mxs_hardware = { +static const struct snd_pcm_hardware snd_mxs_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | @@ -56,7 +47,6 @@ static struct snd_pcm_hardware snd_mxs_hardware = { .periods_max = 52, .buffer_bytes_max = 64 * 1024, .fifo_size = 32, - }; static bool filter(struct dma_chan *chan, void *param) @@ -74,129 +64,25 @@ static bool filter(struct dma_chan *chan, void *param) return true; } -static int snd_mxs_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - - return 0; -} - -static int snd_mxs_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - - snd_soc_set_runtime_hwparams(substream, &snd_mxs_hardware); - - return snd_dmaengine_pcm_open_request_chan(substream, filter, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); -} - -static int snd_mxs_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - return dma_mmap_writecombine(substream->pcm->card->dev, vma, - runtime->dma_area, - runtime->dma_addr, - runtime->dma_bytes); -} - -static struct snd_pcm_ops mxs_pcm_ops = { - .open = snd_mxs_open, - .close = snd_dmaengine_pcm_close_release_chan, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_mxs_pcm_hw_params, - .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer_no_residue, - .mmap = snd_mxs_pcm_mmap, -}; - -static int mxs_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = snd_mxs_hardware.buffer_bytes_max; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_writecombine(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) - return -ENOMEM; - buf->bytes = size; - - return 0; -} - -static u64 mxs_pcm_dmamask = DMA_BIT_MASK(32); -static int mxs_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &mxs_pcm_dmamask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = mxs_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = mxs_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - -out: - return ret; -} - -static void mxs_pcm_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; - - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - - dma_free_writecombine(pcm->card->dev, buf->bytes, - buf->area, buf->addr); - buf->area = NULL; - } -} - -static struct snd_soc_platform_driver mxs_soc_platform = { - .ops = &mxs_pcm_ops, - .pcm_new = mxs_pcm_new, - .pcm_free = mxs_pcm_free, +static const struct snd_dmaengine_pcm_config mxs_dmaengine_pcm_config = { + .pcm_hardware = &snd_mxs_hardware, + .compat_filter_fn = filter, + .prealloc_buffer_size = 64 * 1024, }; int mxs_pcm_platform_register(struct device *dev) { - return snd_soc_register_platform(dev, &mxs_soc_platform); + return snd_dmaengine_pcm_register(dev, &mxs_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT | + SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_register); void mxs_pcm_platform_unregister(struct device *dev) { - snd_soc_unregister_platform(dev); + snd_dmaengine_pcm_unregister(dev); } EXPORT_SYMBOL_GPL(mxs_pcm_platform_unregister); -- cgit v1.2.3 From 19133d2cfd9d6ad8365d94137dcd3e18f760c8e2 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 22 Apr 2013 21:48:45 +0800 Subject: ASoC: generic-dmaengine-pcm: use a more common dma name The examples in Documentation/devicetree/bindings/dma/dma.txt recommends the name for dma channel doing both RX and TX to be "rx-tx". This becomes a common pattern that has been adopted by platforms that converts to generic DMA bindings. Let's follow this common pattern in generic-dmaengine-pcm. Signed-off-by: Shawn Guo Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 5fd5ed4c0a96..98e131037fd8 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -227,7 +227,7 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = of_dma_request_slave_channel(of_node, "tx_rx"); + pcm->chan[0] = of_dma_request_slave_channel(of_node, "rx-tx"); pcm->chan[1] = pcm->chan[0]; } else { for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { -- cgit v1.2.3 From 6f1fd93e304f6f4f8b841e1b0124f3ab4c85ba2e Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Mon, 22 Apr 2013 21:57:24 +0800 Subject: ASoC: generic-dmaengine-pcm: call dma_request_slave_channel() dma_request_slave_channel() is a more appropriate API for dmaengine clients that adopt generic DMA bindings to call. Let's use it instead of of_dma_request_slave_channel() to save include. Signed-off-by: Shawn Guo Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 98e131037fd8..e29ec3cd84b1 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -21,7 +21,6 @@ #include #include #include -#include #include @@ -219,19 +218,19 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { }; static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, - struct device_node *of_node) + struct device *dev) { unsigned int i; - if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !of_node) + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { - pcm->chan[0] = of_dma_request_slave_channel(of_node, "rx-tx"); + pcm->chan[0] = dma_request_slave_channel(dev, "rx-tx"); pcm->chan[1] = pcm->chan[0]; } else { for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = of_dma_request_slave_channel(of_node, + pcm->chan[i] = dma_request_slave_channel(dev, dmaengine_pcm_dma_channel_names[i]); } } @@ -255,7 +254,7 @@ int snd_dmaengine_pcm_register(struct device *dev, pcm->config = config; pcm->flags = flags; - dmaengine_pcm_request_chan_of(pcm, dev->of_node); + dmaengine_pcm_request_chan_of(pcm, dev); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_soc_add_platform(dev, &pcm->platform, -- cgit v1.2.3 From 7c21a78104d7f2c155e2c12279282f659d773d05 Mon Sep 17 00:00:00 2001 From: Michal Bachraty Date: Fri, 19 Apr 2013 15:28:03 +0200 Subject: ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers As pointed of by Vaibhav, commit message: "ASoC: davinci-mcasp: Add support for multichannel playback" number of active serializers can be hidden into fifo_level variable, which is set in davimci-mcasp. Signed-off-by: Michal Bachraty Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 8 +++++--- sound/soc/davinci/davinci-pcm.c | 11 +++++------ sound/soc/davinci/davinci-pcm.h | 1 - 3 files changed, 10 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index aeb3a6627d6a..59143373afc6 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -834,17 +834,20 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int word_length; u8 fifo_level; u8 slots = dev->tdm_slots; + u8 active_serializers; int channels; struct snd_interval *pcm_channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); channels = pcm_channels->min; + active_serializers = (channels + slots - 1) / slots; + if (davinci_hw_common_param(dev, substream->stream, channels) == -EINVAL) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_level = dev->txnumevt; + fifo_level = dev->txnumevt * active_serializers; else - fifo_level = dev->rxnumevt; + fifo_level = dev->rxnumevt * active_serializers; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -889,7 +892,6 @@ 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; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 078031d61167..b2f27c2e5fdc 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -181,7 +181,6 @@ 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 +194,14 @@ 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 * serializers; + count /= fifo_level; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; dst_bidx = 4; - src_cidx = data_type * fifo_level * serializers; + src_cidx = data_type * fifo_level; dst_cidx = 0; } else { src = prtd->params->dma_addr; @@ -210,7 +209,7 @@ 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 * serializers; + dst_cidx = data_type * fifo_level; } acnt = prtd->params->acnt; @@ -225,8 +224,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) ASYNC); else edma_set_transfer_params(prtd->asp_link[0], acnt, - fifo_level * serializers, - count, fifo_level * serializers, + fifo_level, + count, fifo_level, ABSYNC); } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 32d7634d7b26..b6ef7039dd09 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -27,7 +27,6 @@ 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); -- cgit v1.2.3 From d486fea6babfe3ff0c382c9e4baf18f535fcee7d Mon Sep 17 00:00:00 2001 From: Michal Bachraty Date: Fri, 19 Apr 2013 15:28:44 +0200 Subject: ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes For TDM mode, BCLK-to-LCLK ratio is computed as (tdm_slots) x (word_length). I2S mode is only subset of TDM mode with specific tdm_slots = 2 channels. Also bclk_lrclk_ratio can be greater than 255, therefore u16 need to be used. Signed-off-by: Michal Bachraty Acked-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 7 ++++--- sound/soc/davinci/davinci-mcasp.h | 2 +- 2 files changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 59143373afc6..8b85049daab0 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -636,11 +636,12 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, * callback, take it into account here. That allows us to for example * send 32 bits per channel to the codec, while only 16 of them carry * audio payload. - * The clock ratio is given for a full period of data (both left and - * right channels), so it has to be divided by 2. + * The clock ratio is given for a full period of data (for I2S format + * both left and right channels), so it has to be divided by number of + * tdm-slots (for I2S - divided by 2). */ if (dev->bclk_lrclk_ratio) - word_length = dev->bclk_lrclk_ratio / 2; + word_length = dev->bclk_lrclk_ratio / dev->tdm_slots; /* mapping of the XSSZ bit-field as described in the datasheet */ fmt = (word_length >> 1) - 1; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 0edd3b5a37fd..a9ac0c11da71 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -38,7 +38,7 @@ struct davinci_audio_dev { u8 num_serializer; u8 *serial_dir; u8 version; - u8 bclk_lrclk_ratio; + u16 bclk_lrclk_ratio; /* McASP FIFO related */ u8 txnumevt; -- cgit v1.2.3 From d74bf3fa8e85f8f80738d93396d8aab3871eae1e Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Tue, 23 Apr 2013 18:30:40 +0200 Subject: ASoC: ux500: forward declare msp_i2s_platform_data We get a lot of build warnings from the msp driver like: In file included from sound/soc/ux500/ux500_msp_dai.h:21:0, from sound/soc/ux500/mop500.c:25: sound/soc/ux500/ux500_msp_i2s.h:546:11: warning: 'struct msp_i2s_platform_data' declared inside parameter list [enabled by default] struct msp_i2s_platform_data *platform_data); ^ sound/soc/ux500/ux500_msp_i2s.h:546:11: warning: its scope is only this definition or declaration, which is probably not what you want [enabled by default] The easiest solution is to add a declaration of the struct name. Signed-off-by: Arnd Bergmann Signed-off-by: Mark Brown --- sound/soc/ux500/ux500_msp_i2s.h | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 1311c0df7628..6f3e3dccbcb6 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -543,6 +543,7 @@ struct ux500_msp_dma_params { struct stedma40_chan_cfg *dma_cfg; }; +struct msp_i2s_platform_data; int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct ux500_msp **msp_p, struct msp_i2s_platform_data *platform_data); -- cgit v1.2.3