The patch ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c 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 467fece8fbc6774a3a3bd0981e1a342fb5022706 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx> Date: Mon, 22 Jul 2019 10:36:16 +0900 Subject: [PATCH] ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c snd_soc_dai_stream_valid() is function to check stream validity. But, some code is using it, some code are checking stream->channels_min directly. Doing samethings by different method is confusable. This patch uses same funcntion for same purpose. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx> Link: https://lore.kernel.org/r/87ftmyhmzz.wl-kuninori.morimoto.gx@xxxxxxxxxxx Signed-off-by: Mark Brown <broonie@xxxxxxxxxx> --- include/sound/soc-dai.h | 1 + sound/soc/soc-compress.c | 9 ++++----- sound/soc/soc-dai.c | 18 ++++++++++++++++++ sound/soc/soc-pcm.c | 39 ++++++++++----------------------------- 4 files changed, 33 insertions(+), 34 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0f8b09520020..dc48fe081a20 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -168,6 +168,7 @@ int snd_soc_dai_probe(struct snd_soc_dai *dai); int snd_soc_dai_remove(struct snd_soc_dai *dai); int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd, int num); +bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream); struct snd_soc_dai_ops { /* diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index ddef4ff677ce..289211069a1e 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -872,14 +872,13 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - if (codec_dai->driver->playback.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) capture = 1; - capture = capture && cpu_dai->driver->capture.channels_min; - playback = playback && cpu_dai->driver->playback.channels_min; - /* * Compress devices are unidirectional so only one of the directions * should be set, check for that (xor) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index e6f161b9f975..1c7f63871c1d 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -387,3 +387,21 @@ int snd_soc_dai_compress_new(struct snd_soc_dai *dai, return dai->driver->compress_new(rtd, num); return -ENOTSUPP; } + +/* + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * Returns true if the DAI supports the indicated stream type. + */ +bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir) +{ + struct snd_soc_pcm_stream *stream; + + if (dir == SNDRV_PCM_STREAM_PLAYBACK) + stream = &dai->driver->playback; + else + stream = &dai->driver->capture; + + /* If the codec specifies any channels at all, it supports the stream */ + return stream->channels_min; +} diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f3137723301c..fabeac164a6c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -29,24 +29,6 @@ #define DPCM_MAX_BE_USERS 8 -/* - * snd_soc_dai_stream_valid() - check if a DAI supports the given stream - * - * Returns true if the DAI supports the indicated stream type. - */ -static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) -{ - struct snd_soc_pcm_stream *codec_stream; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - codec_stream = &dai->driver->playback; - else - codec_stream = &dai->driver->capture; - - /* If the codec specifies any channels at all, it supports the stream */ - return codec_stream->channels_min; -} - /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -2688,8 +2670,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) new ? "new" : "old", fe->dai_link->name); /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min || - !fe->codec_dai->driver->playback.channels_min) + if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK) || + !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK)) goto capture; /* skip if FE isn't currently playing */ @@ -2719,8 +2701,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) capture: /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min || - !fe->codec_dai->driver->capture.channels_min) + if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE) || + !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE)) return 0; /* skip if FE isn't currently capturing */ @@ -3030,14 +3012,13 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = rtd->dai_link->dpcm_capture; } else { for_each_rtd_codec_dai(rtd, i, codec_dai) { - if (codec_dai->driver->playback.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) capture = 1; } - - capture = capture && cpu_dai->driver->capture.channels_min; - playback = playback && cpu_dai->driver->playback.channels_min; } if (rtd->dai_link->playback_only) { @@ -3375,11 +3356,11 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - if (fe->cpu_dai->driver->playback.channels_min) + if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK, buf + offset, out_count - offset); - if (fe->cpu_dai->driver->capture.channels_min) + if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE, buf + offset, out_count - offset); -- 2.20.1 _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
