Message ID | 87wo8rqfi3.wl-kuninori.morimoto.gx@renesas.com (mailing list archive) |
---|---|
State | New, archived |
Headers | show |
Series | ASoC: soc-pcm cleanup step3 | expand |
On Thu, 13 Feb 2020 05:26:53 +0100, Kuninori Morimoto wrote: > > > From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> > > ALSA SoC has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere. > Having for_each_xxxx macro is useful. > This patch adds for_each_pcm_stream() for it. This macro can be put in sound/pcm.h. The similar pattern is found also generically in many non-ASoC codes, too. thanks, Takashi > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> > --- > include/sound/soc.h | 5 +++++ > sound/soc/fsl/fsl_asrc_dma.c | 4 ++-- > sound/soc/soc-core.c | 31 +++++++++++++------------------ > 3 files changed, 20 insertions(+), 20 deletions(-) > > diff --git a/include/sound/soc.h b/include/sound/soc.h > index f0e4f36f83bf..58af52efa07d 100644 > --- a/include/sound/soc.h > +++ b/include/sound/soc.h > @@ -419,6 +419,11 @@ enum snd_soc_card_subclass { > SND_SOC_CARD_CLASS_RUNTIME = 1, > }; > > +#define for_each_pcm_stream(stream) \ > + for (stream = SNDRV_PCM_STREAM_PLAYBACK; \ > + stream <= SNDRV_PCM_STREAM_LAST; \ > + stream++) > + > int snd_soc_register_card(struct snd_soc_card *card); > int snd_soc_unregister_card(struct snd_soc_card *card); > int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); > diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c > index ece130f59d15..2f5a62381f94 100644 > --- a/sound/soc/fsl/fsl_asrc_dma.c > +++ b/sound/soc/fsl/fsl_asrc_dma.c > @@ -400,7 +400,7 @@ static int fsl_asrc_dma_pcm_new(struct snd_soc_component *component, > return ret; > } > > - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { > + for_each_pcm_stream(i) { > substream = pcm->streams[i].substream; > if (!substream) > continue; > @@ -428,7 +428,7 @@ static void fsl_asrc_dma_pcm_free(struct snd_soc_component *component, > struct snd_pcm_substream *substream; > int i; > > - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { > + for_each_pcm_stream(i) { > substream = pcm->streams[i].substream; > if (!substream) > continue; > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c > index 068d809c349a..dc58ce766f3b 100644 > --- a/sound/soc/soc-core.c > +++ b/sound/soc/soc-core.c > @@ -431,6 +431,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( > struct snd_soc_component *component; > struct device *dev; > int ret; > + int stream; > > /* > * for rtd->dev > @@ -465,10 +466,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( > > rtd->dev = dev; > INIT_LIST_HEAD(&rtd->list); > - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); > - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); > - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); > - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); > + for_each_pcm_stream(stream) { > + INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients); > + INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients); > + } > dev_set_drvdata(dev, rtd); > INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); > > @@ -558,17 +559,14 @@ int snd_soc_suspend(struct device *dev) > snd_soc_flush_all_delayed_work(card); > > for_each_card_rtds(card, rtd) { > + int stream; > > if (rtd->dai_link->ignore_suspend) > continue; > > - snd_soc_dapm_stream_event(rtd, > - SNDRV_PCM_STREAM_PLAYBACK, > - SND_SOC_DAPM_STREAM_SUSPEND); > - > - snd_soc_dapm_stream_event(rtd, > - SNDRV_PCM_STREAM_CAPTURE, > - SND_SOC_DAPM_STREAM_SUSPEND); > + for_each_pcm_stream(stream) > + snd_soc_dapm_stream_event(rtd, stream, > + SND_SOC_DAPM_STREAM_SUSPEND); > } > > /* Recheck all endpoints too, their state is affected by suspend */ > @@ -664,17 +662,14 @@ static void soc_resume_deferred(struct work_struct *work) > } > > for_each_card_rtds(card, rtd) { > + int stream; > > if (rtd->dai_link->ignore_suspend) > continue; > > - snd_soc_dapm_stream_event(rtd, > - SNDRV_PCM_STREAM_PLAYBACK, > - SND_SOC_DAPM_STREAM_RESUME); > - > - snd_soc_dapm_stream_event(rtd, > - SNDRV_PCM_STREAM_CAPTURE, > - SND_SOC_DAPM_STREAM_RESUME); > + for_each_pcm_stream(stream) > + snd_soc_dapm_stream_event(rtd, stream, > + SND_SOC_DAPM_STREAM_RESUME); > } > > /* unmute any active DACs */ > -- > 2.17.1 > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel@alsa-project.org > https://mailman.alsa-project.org/mailman/listinfo/alsa-devel >
Hi Takashi Thank you for your feedback > > From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> > > > > ALSA SoC has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere. > > Having for_each_xxxx macro is useful. > > This patch adds for_each_pcm_stream() for it. > > This macro can be put in sound/pcm.h. The similar pattern is found > also generically in many non-ASoC codes, too. (snip) > > +#define for_each_pcm_stream(stream) \ > > + for (stream = SNDRV_PCM_STREAM_PLAYBACK; \ > > + stream <= SNDRV_PCM_STREAM_LAST; \ > > + stream++) I see. Will do in v2 Thank you for your help !! Best regards --- Kuninori Morimoto
On 2/12/20 10:26 PM, Kuninori Morimoto wrote: > > From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> > > ALSA SoC has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere. > Having for_each_xxxx macro is useful. > This patch adds for_each_pcm_stream() for it. Indeed, we also have ugly code in SOF for this: for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { a macro would be much nicer, good suggestion.
Hi Pierre-Louis > > From: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> > > > > ALSA SoC has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere. > > Having for_each_xxxx macro is useful. > > This patch adds for_each_pcm_stream() for it. > > Indeed, we also have ugly code in SOF for this: > > for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { > > a macro would be much nicer, good suggestion. Thank you for pointing it. I will care it in v2 :) Thank you for your help !! Best regards --- Kuninori Morimoto
diff --git a/include/sound/soc.h b/include/sound/soc.h index f0e4f36f83bf..58af52efa07d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -419,6 +419,11 @@ enum snd_soc_card_subclass { SND_SOC_CARD_CLASS_RUNTIME = 1, }; +#define for_each_pcm_stream(stream) \ + for (stream = SNDRV_PCM_STREAM_PLAYBACK; \ + stream <= SNDRV_PCM_STREAM_LAST; \ + stream++) + int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index ece130f59d15..2f5a62381f94 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -400,7 +400,7 @@ static int fsl_asrc_dma_pcm_new(struct snd_soc_component *component, return ret; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + for_each_pcm_stream(i) { substream = pcm->streams[i].substream; if (!substream) continue; @@ -428,7 +428,7 @@ static void fsl_asrc_dma_pcm_free(struct snd_soc_component *component, struct snd_pcm_substream *substream; int i; - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + for_each_pcm_stream(i) { substream = pcm->streams[i].substream; if (!substream) continue; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 068d809c349a..dc58ce766f3b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -431,6 +431,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_component *component; struct device *dev; int ret; + int stream; /* * for rtd->dev @@ -465,10 +466,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->dev = dev; INIT_LIST_HEAD(&rtd->list); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); + for_each_pcm_stream(stream) { + INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients); + } dev_set_drvdata(dev, rtd); INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); @@ -558,17 +559,14 @@ int snd_soc_suspend(struct device *dev) snd_soc_flush_all_delayed_work(card); for_each_card_rtds(card, rtd) { + int stream; if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_SUSPEND); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_SUSPEND); + for_each_pcm_stream(stream) + snd_soc_dapm_stream_event(rtd, stream, + SND_SOC_DAPM_STREAM_SUSPEND); } /* Recheck all endpoints too, their state is affected by suspend */ @@ -664,17 +662,14 @@ static void soc_resume_deferred(struct work_struct *work) } for_each_card_rtds(card, rtd) { + int stream; if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_RESUME); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_RESUME); + for_each_pcm_stream(stream) + snd_soc_dapm_stream_event(rtd, stream, + SND_SOC_DAPM_STREAM_RESUME); } /* unmute any active DACs */