The patch ASoC: soc-dai: add snd_soc_dai_startup() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 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 >From 5a52a04531486e2ab069b7882432c8b266db36e6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx> Date: Mon, 22 Jul 2019 10:33:32 +0900 Subject: [PATCH] ASoC: soc-dai: add snd_soc_dai_startup() Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_startup() and use it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx> Link: https://lore.kernel.org/r/87wogahn4i.wl-kuninori.morimoto.gx@xxxxxxxxxxx Signed-off-by: Mark Brown <broonie@xxxxxxxxxx> --- include/sound/soc-dai.h | 2 ++ sound/soc/soc-dai.c | 11 +++++++++++ sound/soc/soc-dapm.c | 28 ++++++++++------------------ sound/soc/soc-pcm.c | 27 +++++++++++---------------- 4 files changed, 34 insertions(+), 34 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5222b6a758f2..0d16c5bb20bb 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -150,6 +150,8 @@ int snd_soc_dai_hw_params(struct snd_soc_dai *dai, struct snd_pcm_hw_params *params); void snd_soc_dai_hw_free(struct snd_soc_dai *dai, struct snd_pcm_substream *substream); +int snd_soc_dai_startup(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream); struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 39a685e6acd5..6e196636e42f 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -289,3 +289,14 @@ void snd_soc_dai_hw_free(struct snd_soc_dai *dai, if (dai->driver->ops->hw_free) dai->driver->ops->hw_free(substream, dai); } + +int snd_soc_dai_startup(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream) +{ + int ret = 0; + + if (dai->driver->ops->startup) + ret = dai->driver->ops->startup(substream, dai); + + return ret; +} diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0783b05133ad..71bfd049480a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3828,15 +3828,11 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_source_path(w, path) { source = path->source->priv; - if (source->driver->ops->startup) { - ret = source->driver->ops->startup(&substream, - source); - if (ret < 0) { - dev_err(source->dev, - "ASoC: startup() failed: %d\n", - ret); - goto out; - } + ret = snd_soc_dai_startup(source, &substream); + if (ret < 0) { + dev_err(source->dev, + "ASoC: startup() failed: %d\n", ret); + goto out; } source->active++; ret = snd_soc_dai_hw_params(source, &substream, params); @@ -3850,15 +3846,11 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, snd_soc_dapm_widget_for_each_sink_path(w, path) { sink = path->sink->priv; - if (sink->driver->ops->startup) { - ret = sink->driver->ops->startup(&substream, - sink); - if (ret < 0) { - dev_err(sink->dev, - "ASoC: startup() failed: %d\n", - ret); - goto out; - } + ret = snd_soc_dai_startup(sink, &substream); + if (ret < 0) { + dev_err(sink->dev, + "ASoC: startup() failed: %d\n", ret); + goto out; } sink->active++; ret = snd_soc_dai_hw_params(sink, &substream, params); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 58fc4e98ab59..9c8713a3eef1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -535,13 +535,11 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { - ret = cpu_dai->driver->ops->startup(substream, cpu_dai); - if (ret < 0) { - dev_err(cpu_dai->dev, "ASoC: can't open interface" - " %s: %d\n", cpu_dai->name, ret); - goto out; - } + ret = snd_soc_dai_startup(cpu_dai, substream); + if (ret < 0) { + dev_err(cpu_dai->dev, "ASoC: can't open interface %s: %d\n", + cpu_dai->name, ret); + goto out; } ret = soc_pcm_components_open(substream, &component); @@ -549,15 +547,12 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto component_err; for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->ops->startup) { - ret = codec_dai->driver->ops->startup(substream, - codec_dai); - if (ret < 0) { - dev_err(codec_dai->dev, - "ASoC: can't open codec %s: %d\n", - codec_dai->name, ret); - goto codec_dai_err; - } + ret = snd_soc_dai_startup(codec_dai, substream); + if (ret < 0) { + dev_err(codec_dai->dev, + "ASoC: can't open codec %s: %d\n", + codec_dai->name, ret); + goto codec_dai_err; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) -- 2.20.1 _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
