From e8343410ddf08fc36a9b9cc7c51a4e53a262d4c6 Mon Sep 17 00:00:00 2001 From: Jai Luthra Date: Tue, 11 Jun 2024 18:02:55 +0530 Subject: [PATCH 1/2] ALSA: dmaengine: Synchronize dma channel after drop() 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 drop() without any synchronization. Make sure the ALSA core synchronizes the DMA channel by adding a sync_stop() hook. Reviewed-by: Peter Ujfalusi Signed-off-by: Jai Luthra Link: https://lore.kernel.org/r/20240611-asoc_next-v3-1-fcfd84b12164@ti.com Signed-off-by: Mark Brown --- 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..f6baa9a01868 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_sync_stop(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..ed07fa5693d2 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_sync_stop(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_sync_stop); + /** * 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..a63e942fdc0b 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_sync_stop(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + return snd_dmaengine_pcm_sync_stop(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, + .sync_stop = dmaengine_pcm_sync_stop, }; 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, + .sync_stop = dmaengine_pcm_sync_stop, }; static const char * const dmaengine_pcm_dma_channel_names[] = { From c5dcf8ab10606e76c1d8a0ec77f27d84a392e874 Mon Sep 17 00:00:00 2001 From: Jai Luthra Date: Tue, 11 Jun 2024 18:02:56 +0530 Subject: [PATCH 2/2] ASoC: ti: davinci-mcasp: Set min period size using FIFO config 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. Acked-by: Peter Ujfalusi Signed-off-by: Jai Luthra Link: https://lore.kernel.org/r/20240611-asoc_next-v3-2-fcfd84b12164@ti.com Signed-off-by: Mark Brown --- 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;