[alsa-devel] [PATCH 04/16] ASoC: soc-dai: add snd_soc_dai_startup()
Kuninori Morimoto
kuninori.morimoto.gx at renesas.com
Mon Jul 22 03:33:32 CEST 2019
From: Kuninori Morimoto <kuninori.morimoto.gx at renesas.com>
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 at renesas.com>
---
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 5222b6a..0d16c5b 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 39a685e..6e19663 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 0783b05..71bfd04 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 58fc4e9..9c8713a 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)
--
2.7.4
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