The patch ASoC:hdac_hda:use correct format to setup hda codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 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 03d0aa4d4fddce4a5d865d819a4d98bfc3d451e6 Mon Sep 17 00:00:00 2001 From: Rander Wang <rander.wang@xxxxxxxxxxxxxxx> Date: Fri, 8 Mar 2019 16:38:58 +0800 Subject: [PATCH] ASoC:hdac_hda:use correct format to setup hda codec The current implementation of the hdac_hda codec results in zero-valued samples on capture and noise with headset playback when SOF is used on platforms with an on-board HDaudio codec. This is root-caused to SOF using be_hw_params_fixup, and the prepare() call using invalid runtime fields to determine the format. This patch moves the format handling to the hw_params() callback, as done already for hdac_hdmi, to make sure the fixed-up information is taken into account but keeps the codec initialization in prepare() as the stream_tag is only available at that time. Moving everything in the prepare() callback is possible but the code is less elegant so this two-step solution was chosen. The solution was tested with the SST driver with no regressions, and all the issues with SOF playback and capture are solved. Signed-off-by: Rander Wang <rander.wang@xxxxxxxxxxxxxxx> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx> Signed-off-by: Mark Brown <broonie@xxxxxxxxxx> --- sound/soc/codecs/hdac_hda.c | 53 +++++++++++++++++++++++++++---------- sound/soc/codecs/hdac_hda.h | 1 + 2 files changed, 40 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index ffecdaaa8cf2..f889d94c8e3c 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -38,6 +38,9 @@ static void hdac_hda_dai_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); +static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, @@ -50,6 +53,7 @@ static const struct snd_soc_dai_ops hdac_hda_dai_ops = { .startup = hdac_hda_dai_open, .shutdown = hdac_hda_dai_close, .prepare = hdac_hda_dai_prepare, + .hw_params = hdac_hda_dai_hw_params, .hw_free = hdac_hda_dai_hw_free, .set_tdm_slot = hdac_hda_dai_set_tdm_slot, }; @@ -139,6 +143,39 @@ static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + unsigned int format_val; + unsigned int maxbps; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + maxbps = dai->driver->playback.sig_bits; + else + maxbps = dai->driver->capture.sig_bits; + + hda_pvt = snd_soc_component_get_drvdata(component); + format_val = snd_hdac_calc_stream_format(params_rate(params), + params_channels(params), + params_format(params), + maxbps, + 0); + if (!format_val) { + dev_err(dai->dev, + "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + params_rate(params), params_channels(params), + params_format(params), maxbps); + + return -EINVAL; + } + + hda_pvt->pcm[dai->id].format_val[substream->stream] = format_val; + return 0; +} + static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -162,10 +199,9 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct snd_pcm_runtime *runtime = substream->runtime; struct hdac_device *hdev; - struct hda_pcm_stream *hda_stream; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; @@ -179,19 +215,8 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - format_val = snd_hdac_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hda_stream->maxbps, - 0); - if (!format_val) { - dev_err(&hdev->dev, - "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); - return -EINVAL; - } - stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream]; + format_val = hda_pvt->pcm[dai->id].format_val[substream->stream]; ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream, stream, format_val, substream); diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h index e444ef593360..6b1bd4f428e7 100644 --- a/sound/soc/codecs/hdac_hda.h +++ b/sound/soc/codecs/hdac_hda.h @@ -8,6 +8,7 @@ struct hdac_hda_pcm { int stream_tag[2]; + unsigned int format_val[2]; }; struct hdac_hda_priv { -- 2.20.1 _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
