| Message ID | 20190927071646.22319-1-peter.ujfalusi@ti.com |
|---|---|
| State | Accepted |
| Commit | 4378f1fbe924054a09ff0d4e39e1a581b9245252 |
| Headers | show |
| Series | [v2] ASoC: soc-pcm: Use different sequence for start/stop trigger | expand |
On 9/27/19 2:16 AM, Peter Ujfalusi wrote: > On stream stop currently we stop the DMA first followed by the CPU DAI. > This can cause underflow (playback) or overflow (capture) on the DAI side > as the DMA is no longer feeding data while the DAI is still active. > It can be observed easily if the DAI side does not have FIFO (or it is > disabled) to survive the time while the DMA is stopped, but still can > happen on relatively slow CPUs when relatively high sampling rate is used: > the FIFO is drained between the time the DMA is stopped and the DAI is > stopped. > > It can only fixed by using different sequence within trigger for 'stop' and > 'start': > case SNDRV_PCM_TRIGGER_START: > case SNDRV_PCM_TRIGGER_RESUME: > case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > Trigger order: dai_link, DMA, CPU DAI then the codec > > case SNDRV_PCM_TRIGGER_STOP: > case SNDRV_PCM_TRIGGER_SUSPEND: > case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > Trigger order: codec, CPU DAI, DMA then dai_link > > Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> This patch was tested by the Intel SOF QA folks, and we don't see any errors added by this sequence change (more precisely we see the usual random IPC timeouts we are chasing but nothing new clearly caused by this patch). And the change makes sense so Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> > --- > Hi, > > Changes since v1: > - Rebased on top 'ASoC: soc-component: fix a couple missing error assignments' > > Changes since rfc: > - Fixed missing error assignments (copy pasted the error) > - Document the trigger order in the commit message > > Regards, > Peter > > sound/soc/soc-pcm.c | 64 ++++++++++++++++++++++++++++++++++++++++++--- > 1 file changed, 61 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c > index a1b99ac57d9e..66910500e3b6 100644 > --- a/sound/soc/soc-pcm.c > +++ b/sound/soc/soc-pcm.c > @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) > return 0; > } > > -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) > { > struct snd_soc_pcm_runtime *rtd = substream->private_data; > struct snd_soc_component *component; > @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > struct snd_soc_dai *codec_dai; > int i, ret; > > - for_each_rtd_codec_dai(rtd, i, codec_dai) { > - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); > + if (rtd->dai_link->ops->trigger) { > + ret = rtd->dai_link->ops->trigger(substream, cmd); > if (ret < 0) > return ret; > } > @@ -1074,6 +1074,42 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > if (ret < 0) > return ret; > > + for_each_rtd_codec_dai(rtd, i, codec_dai) { > + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); > + if (ret < 0) > + return ret; > + } > + > + return 0; > +} > + > +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) > +{ > + struct snd_soc_pcm_runtime *rtd = substream->private_data; > + struct snd_soc_component *component; > + struct snd_soc_rtdcom_list *rtdcom; > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + struct snd_soc_dai *codec_dai; > + int i, ret; > + > + for_each_rtd_codec_dai(rtd, i, codec_dai) { > + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); > + if (ret < 0) > + return ret; > + } > + > + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); > + if (ret < 0) > + return ret; > + > + for_each_rtdcom(rtd, rtdcom) { > + component = rtdcom->component; > + > + ret = snd_soc_component_trigger(component, substream, cmd); > + if (ret < 0) > + return ret; > + } > + > if (rtd->dai_link->ops->trigger) { > ret = rtd->dai_link->ops->trigger(substream, cmd); > if (ret < 0) > @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > return 0; > } > > +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) > +{ > + int ret; > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_RESUME: > + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > + ret = soc_pcm_trigger_start(substream, cmd); > + break; > + case SNDRV_PCM_TRIGGER_STOP: > + case SNDRV_PCM_TRIGGER_SUSPEND: > + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > + ret = soc_pcm_trigger_stop(substream, cmd); > + break; > + default: > + return -EINVAL; > + } > + > + return ret; > +} > + > static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, > int cmd) > { > _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a1b99ac57d9e..66910500e3b6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1074,6 +1074,42 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) {
On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> --- Hi, Changes since v1: - Rebased on top 'ASoC: soc-component: fix a couple missing error assignments' Changes since rfc: - Fixed missing error assignments (copy pasted the error) - Document the trigger order in the commit message Regards, Peter sound/soc/soc-pcm.c | 64 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 61 insertions(+), 3 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org https://mailman.alsa-project.org/mailman/listinfo/alsa-devel