[PATCH v2 0/2] Fixes for McASP and dmaengine_pcm

[PATCH v2 0/2] Fixes for McASP and dmaengine_pcm

[Jai Luthra]

This series fixes two patches:

1. Fix the dmaengine API usage by calling dmaengine_synchronize() after
   dmaengine_terminate_async() when xrun events occur in application
2. Use the McASP AFIFO property from DT to refine the period size,
   instead of hardcoding minimum to 64 samples

Signed-off-by: Jai Luthra <j-luthra@ti.com>
---
Changes in v2:
- Fix compiler warning for prepare callback by marking it static
- Pass numevt directly to hw_rule_min_periodsize()
- Link to v1: https://lore.kernel.org/r/20240604-asoc_next-v1-0-e895c88e744d@ti.com

---
Jai Luthra (2):
      ALSA: dmaengine: Synchronize dma channel in prepare()
      ASoC: ti: davinci-mcasp: Set min period size using FIFO config

 include/sound/dmaengine_pcm.h         |  1 +
 sound/core/pcm_dmaengine.c            | 10 ++++++++++
 sound/soc/soc-generic-dmaengine-pcm.c |  8 ++++++++
 sound/soc/ti/davinci-mcasp.c          |  9 +++++++--
 4 files changed, 26 insertions(+), 2 deletions(-)
---
base-commit: d97496ca23a2d4ee80b7302849404859d9058bcd
change-id: 20240604-asoc_next-c063fcc190c6

Best regards,
-- 
Jai Luthra <j-luthra@ti.com>

[PATCH v2 1/2] ALSA: dmaengine: Synchronize dma channel in prepare()

[Jai Luthra]

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.

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[] = {

-- 
2.43.0

Re: [PATCH v2 1/2] ALSA: dmaengine: Synchronize dma channel in prepare()

[Lars-Peter Clausen]

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[] = {
>

Re: [PATCH v2 1/2] ALSA: dmaengine: Synchronize dma channel in prepare()

[Mark Brown]

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On Mon, Jun 10, 2024 at 05:45:52PM -0700, Lars-Peter Clausen wrote:
> 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.

Good point, that's a better idea.

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Re: Re: [PATCH v2 1/2] ALSA: dmaengine: Synchronize dma channel in prepare()

[Jai Luthra]

[-- Attachment #1: Type: text/plain, Size: 966 bytes --]

Hi Lars, Mark,

On Jun 11, 2024 at 11:39:12 +0100, Mark Brown wrote:
> On Mon, Jun 10, 2024 at 05:45:52PM -0700, Lars-Peter Clausen wrote:
> > 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.
> 
> Good point, that's a better idea.

Thanks for the suggestion, sending a v3 with the change.

Peter,
I've kept your R-by intact as it is a minor change.


-- 
Thanks,
Jai

GPG Fingerprint: 4DE0 D818 E5D5 75E8 D45A AFC5 43DE 91F9 249A 7145

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[PATCH v2 2/2] ASoC: ti: davinci-mcasp: Set min period size using FIFO config

[Jai Luthra]

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.

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;

-- 
2.43.0

Re: [PATCH v2 2/2] ASoC: ti: davinci-mcasp: Set min period size using FIFO config

[Péter Ujfalusi]

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;
> 

-- 
Péter

Re: [PATCH v2 0/2] Fixes for McASP and dmaengine_pcm

[Mark Brown]

On Mon, 10 Jun 2024 16:25:59 +0530, Jai Luthra wrote:
> This series fixes two patches:
> 
> 1. Fix the dmaengine API usage by calling dmaengine_synchronize() after
>    dmaengine_terminate_async() when xrun events occur in application
> 2. Use the McASP AFIFO property from DT to refine the period size,
>    instead of hardcoding minimum to 64 samples
> 
> [...]

Applied to

   https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next

Thanks!

[1/2] ALSA: dmaengine: Synchronize dma channel in prepare()
      (no commit info)
[2/2] ASoC: ti: davinci-mcasp: Set min period size using FIFO config
      commit: c5dcf8ab10606e76c1d8a0ec77f27d84a392e874

All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.

You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.

If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.

Please add any relevant lists and maintainers to the CCs when replying
to this mail.

Thanks,
Mark