On 6/10/24 03:56, Jai Luthra wrote:

Sometimes the stream may be stopped due to XRUN events, in which case the userspace can call snd_pcm_drop() and snd_pcm_prepare() to stop and start the stream again.

In these cases, we must wait for the DMA channel to synchronize before marking the stream as prepared for playback, as the DMA channel gets stopped by snd_pcm_drop() without any synchronization.

We should really implement the sync_stop() PCM callback and let the ALSA core let care of the sync.

Reviewed-by: Peter Ujfalusi peter.ujfalusi@gmail.com Signed-off-by: Jai Luthra j-luthra@ti.com

---

include/sound/dmaengine_pcm.h | 1 + sound/core/pcm_dmaengine.c | 10 ++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 8 ++++++++ 3 files changed, 19 insertions(+)

diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index c11aaf8079fb..9c5800e5659f 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -36,6 +36,7 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, struct dma_chan *chan); int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream); +int snd_dmaengine_pcm_prepare(struct snd_pcm_substream *substream);

int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data); diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 12aa1cef11a1..dbf5c6136d68 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -349,6 +349,16 @@ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream, } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan);

+int snd_dmaengine_pcm_prepare(struct snd_pcm_substream *substream) +{

• struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
• dmaengine_synchronize(prtd->dma_chan);
• return 0;

+} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_prepare);

• /**
  • snd_dmaengine_pcm_close - Close a dmaengine based PCM substream
  • @substream: PCM substream

diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ea3bc9318412..078fcb0ba8a2 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -318,6 +318,12 @@ static int dmaengine_copy(struct snd_soc_component *component, return 0; }

+static int dmaengine_pcm_prepare(struct snd_soc_component *component,

• 		 struct snd_pcm_substream *substream)
  

+{

• return snd_dmaengine_pcm_prepare(substream);

+}

• static const struct snd_soc_component_driver dmaengine_pcm_component = { .name = SND_DMAENGINE_PCM_DRV_NAME, .probe_order = SND_SOC_COMP_ORDER_LATE,

@@ -327,6 +333,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component = { .trigger = dmaengine_pcm_trigger, .pointer = dmaengine_pcm_pointer, .pcm_construct = dmaengine_pcm_new,

• .prepare = dmaengine_pcm_prepare, };

  static const struct snd_soc_component_driver dmaengine_pcm_component_process = {

@@ -339,6 +346,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component_process = { .pointer = dmaengine_pcm_pointer, .copy = dmaengine_copy, .pcm_construct = dmaengine_pcm_new,

• .prepare = dmaengine_pcm_prepare, };

  static const char * const dmaengine_pcm_dma_channel_names[] = {

Hi,

On 6/10/24 1:56 PM, Jai Luthra wrote:

The minimum period size was enforced to 64 as older devices integrating McASP with EDMA used an internal FIFO of 64 samples.

With UDMA based platforms this internal McASP FIFO is optional, as the DMA engine internally does some buffering which is already accounted for when registering the platform. So we should read the actual FIFO configuration (txnumevt/rxnumevt) instead of hardcoding frames.min to 64.

Thank you the patch and the fix, Acked-by: Peter Ujfalusi peter.ujfalusi@gmail.com

Signed-off-by: Jai Luthra j-luthra@ti.com

sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-)

diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 1e760c315521..2b1ed91a736c 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize( { struct snd_interval *period_size = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);

• u8 numevt = *((u8 *)rule->private); struct snd_interval frames;

  snd_interval_any(&frames);

• frames.min = 64;

• frames.min = numevt; frames.integer = 1;

  return snd_interval_refine(period_size, &frames);

@@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, u32 max_channels = 0; int i, dir, ret; int tdm_slots = mcasp->tdm_slots;

• u8 *numevt;

  /* Do not allow more then one stream per direction */ if (mcasp->substreams[substream->stream])

@@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, return ret; }

• numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?

• 	 &mcasp->txnumevt :
  

• 	 &mcasp->rxnumevt;
  

  snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,

• 	    davinci_mcasp_hw_rule_min_periodsize, NULL,
  

• 	    davinci_mcasp_hw_rule_min_periodsize, numevt,
      SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
  
  

  return 0;