[alsa-devel] [PATCH] ASoC: core - PCM mutex per rtd
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).
The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed at runtime between the PCM device end (or Frontend - FE) and the physical DAI (Backend - BE) using regular kcontrols (just like a hardware CODEC routes audio in the analog domain). The Dynamic PCM core therefore must be able to call PCM operations for both the Frontend and Backend(s) DAIs at the same time.
Currently we have a global pcm_mutex that is used to serialise the ASoC PCM operations. This patch removes the global mutex and adds a mutex per RTD allowing the PCM operations to be reentrant and allow control of more than one DAI at at time. e.g. a frontend PCM hw_params() could configure multiple backend DAI hw_params() with similar or different hw parameters at the same time.
Signed-off-by: Liam Girdwood lrg@ti.com --- include/sound/soc.h | 8 ++++++++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 28 ++++++++++++++-------------- 3 files changed, 23 insertions(+), 14 deletions(-)
diff --git a/include/sound/soc.h b/include/sound/soc.h index d5db87e..4334ab2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -269,6 +269,11 @@ enum snd_soc_compress_type { SND_SOC_RBTREE_COMPRESSION };
+enum snd_soc_pcm_subclass { + SND_SOC_PCM_CLASS_PCM = 0, + SND_SOC_PCM_CLASS_BE = 1, +}; + int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, @@ -809,6 +814,9 @@ struct snd_soc_pcm_runtime { struct device dev; struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; + struct mutex pcm_mutex; + enum snd_soc_pcm_subclass pcm_subclass; + struct snd_pcm_ops ops;
unsigned int complete:1; unsigned int dev_registered:1; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 32d7d2f..32bc503 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1032,6 +1032,7 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd->dev.parent = card->dev; rtd->dev.release = rtd_release; rtd->dev.init_name = name; + mutex_init(&rtd->pcm_mutex); ret = device_register(&rtd->dev); if (ret < 0) { dev_err(card->dev, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9bebee8..f4864b0 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -81,7 +81,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0;
- mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
/* startup the audio subsystem */ if (cpu_dai->driver->ops->startup) { @@ -211,7 +211,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) cpu_dai->active++; codec_dai->active++; rtd->codec->active++; - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return 0;
config_err: @@ -230,7 +230,7 @@ platform_err: if (cpu_dai->driver->ops->shutdown) cpu_dai->driver->ops->shutdown(substream, cpu_dai); out: - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return ret; }
@@ -245,7 +245,7 @@ static void close_delayed_work(struct work_struct *work) container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dai;
- mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
pr_debug("pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, @@ -260,7 +260,7 @@ static void close_delayed_work(struct work_struct *work) SND_SOC_DAPM_STREAM_STOP); }
- mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); }
/* @@ -276,7 +276,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = rtd->codec;
- mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cpu_dai->playback_active--; @@ -321,7 +321,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); }
- mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return 0; }
@@ -338,7 +338,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0;
- mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) { ret = rtd->dai_link->ops->prepare(substream); @@ -391,7 +391,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(codec_dai, 0);
out: - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return ret; }
@@ -409,7 +409,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0;
- mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
if (rtd->dai_link->ops && rtd->dai_link->ops->hw_params) { ret = rtd->dai_link->ops->hw_params(substream, params); @@ -449,7 +449,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, rtd->rate = params_rate(params);
out: - mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return ret;
platform_err: @@ -464,7 +464,7 @@ codec_err: if (rtd->dai_link->ops && rtd->dai_link->ops->hw_free) rtd->dai_link->ops->hw_free(substream);
- mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return ret; }
@@ -479,7 +479,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = rtd->codec;
- mutex_lock(&pcm_mutex); + mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
/* apply codec digital mute */ if (!codec->active) @@ -500,7 +500,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai);
- mutex_unlock(&pcm_mutex); + mutex_unlock(&rtd->pcm_mutex); return 0; }
On Thu, Jun 09, 2011 at 05:04:39PM +0100, Liam Girdwood wrote:
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).
The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed at runtime between the PCM device end (or Frontend - FE) and the physical DAI (Backend - BE) using regular kcontrols (just like a hardware CODEC routes audio in the analog domain). The Dynamic PCM core therefore must be able to call PCM operations for both the Frontend and Backend(s) DAIs at the same time.
Applied, thanks.
participants (2)
-
Liam Girdwood
-
Mark Brown