20.12.2019 20:06, Ben Dooks пишет: > On 20/12/2019 16:40, Dmitry Osipenko wrote: >> 20.12.2019 18:25, Ben Dooks пишет: >>> On 20/12/2019 15:02, Dmitry Osipenko wrote: >>>> 20.12.2019 17:56, Ben Dooks пишет: >>>>> On 20/12/2019 14:43, Dmitry Osipenko wrote: >>>>>> 20.12.2019 16:57, Jon Hunter пишет: >>>>>>> >>>>>>> On 20/12/2019 11:38, Ben Dooks wrote: >>>>>>>> On 20/12/2019 11:30, Jon Hunter wrote: >>>>>>>>> >>>>>>>>> On 25/11/2019 17:28, Dmitry Osipenko wrote: >>>>>>>>>> 25.11.2019 20:22, Dmitry Osipenko пишет: >>>>>>>>>>> 25.11.2019 13:37, Ben Dooks пишет: >>>>>>>>>>>> On 23/11/2019 21:09, Dmitry Osipenko wrote: >>>>>>>>>>>>> 18.10.2019 18:48, Ben Dooks пишет: >>>>>>>>>>>>>> From: Edward Cragg <edward.cragg@xxxxxxxxxxxxxxx> >>>>>>>>>>>>>> >>>>>>>>>>>>>> The tegra3 audio can support 24 and 32 bit sample sizes so >>>>>>>>>>>>>> add >>>>>>>>>>>>>> the >>>>>>>>>>>>>> option to the tegra30_i2s_hw_params to configure the >>>>>>>>>>>>>> S24_LE or >>>>>>>>>>>>>> S32_LE >>>>>>>>>>>>>> formats when requested. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Signed-off-by: Edward Cragg <edward.cragg@xxxxxxxxxxxxxxx> >>>>>>>>>>>>>> [ben.dooks@xxxxxxxxxxxxxxx: fixup merge of 24 and 32bit] >>>>>>>>>>>>>> [ben.dooks@xxxxxxxxxxxxxxx: add pm calls around ytdm config] >>>>>>>>>>>>>> [ben.dooks@xxxxxxxxxxxxxxx: drop debug printing to dev_dbg] >>>>>>>>>>>>>> Signed-off-by: Ben Dooks <ben.dooks@xxxxxxxxxxxxxxx> >>>>>>>>>>>>>> --- >>>>>>>>>>>>>> squash 5aeca5a055fd ASoC: tegra: i2s: >>>>>>>>>>>>>> pm_runtime_get_sync() is >>>>>>>>>>>>>> needed >>>>>>>>>>>>>> in tdm code >>>>>>>>>>>>>> >>>>>>>>>>>>>> ASoC: tegra: i2s: pm_runtime_get_sync() is needed in tdm code >>>>>>>>>>>>>> --- >>>>>>>>>>>>>> sound/soc/tegra/tegra30_i2s.c | 25 >>>>>>>>>>>>>> ++++++++++++++++++++----- >>>>>>>>>>>>>> 1 file changed, 20 insertions(+), 5 deletions(-) >>>>>>>>>>>>>> >>>>>>>>>>>>>> diff --git a/sound/soc/tegra/tegra30_i2s.c >>>>>>>>>>>>>> b/sound/soc/tegra/tegra30_i2s.c >>>>>>>>>>>>>> index 73f0dddeaef3..063f34c882af 100644 >>>>>>>>>>>>>> --- a/sound/soc/tegra/tegra30_i2s.c >>>>>>>>>>>>>> +++ b/sound/soc/tegra/tegra30_i2s.c >>>>>>>>>>>>>> @@ -127,7 +127,7 @@ static int tegra30_i2s_hw_params(struct >>>>>>>>>>>>>> snd_pcm_substream *substream, >>>>>>>>>>>>>> struct device *dev = dai->dev; >>>>>>>>>>>>>> struct tegra30_i2s *i2s = >>>>>>>>>>>>>> snd_soc_dai_get_drvdata(dai); >>>>>>>>>>>>>> unsigned int mask, val, reg; >>>>>>>>>>>>>> - int ret, sample_size, srate, i2sclock, bitcnt; >>>>>>>>>>>>>> + int ret, sample_size, srate, i2sclock, bitcnt, >>>>>>>>>>>>>> audio_bits; >>>>>>>>>>>>>> struct tegra30_ahub_cif_conf cif_conf; >>>>>>>>>>>>>> if (params_channels(params) != 2) >>>>>>>>>>>>>> @@ -137,8 +137,19 @@ static int tegra30_i2s_hw_params(struct >>>>>>>>>>>>>> snd_pcm_substream *substream, >>>>>>>>>>>>>> switch (params_format(params)) { >>>>>>>>>>>>>> case SNDRV_PCM_FORMAT_S16_LE: >>>>>>>>>>>>>> val = TEGRA30_I2S_CTRL_BIT_SIZE_16; >>>>>>>>>>>>>> + audio_bits = TEGRA30_AUDIOCIF_BITS_16; >>>>>>>>>>>>>> sample_size = 16; >>>>>>>>>>>>>> break; >>>>>>>>>>>>>> + case SNDRV_PCM_FORMAT_S24_LE: >>>>>>>>>>>>>> + val = TEGRA30_I2S_CTRL_BIT_SIZE_24; >>>>>>>>>>>>>> + audio_bits = TEGRA30_AUDIOCIF_BITS_24; >>>>>>>>>>>>>> + sample_size = 24; >>>>>>>>>>>>>> + break; >>>>>>>>>>>>>> + case SNDRV_PCM_FORMAT_S32_LE: >>>>>>>>>>>>>> + val = TEGRA30_I2S_CTRL_BIT_SIZE_32; >>>>>>>>>>>>>> + audio_bits = TEGRA30_AUDIOCIF_BITS_32; >>>>>>>>>>>>>> + sample_size = 32; >>>>>>>>>>>>>> + break; >>>>>>>>>>>>>> default: >>>>>>>>>>>>>> return -EINVAL; >>>>>>>>>>>>>> } >>>>>>>>>>>>>> @@ -170,8 +181,8 @@ static int tegra30_i2s_hw_params(struct >>>>>>>>>>>>>> snd_pcm_substream *substream, >>>>>>>>>>>>>> cif_conf.threshold = 0; >>>>>>>>>>>>>> cif_conf.audio_channels = 2; >>>>>>>>>>>>>> cif_conf.client_channels = 2; >>>>>>>>>>>>>> - cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; >>>>>>>>>>>>>> - cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; >>>>>>>>>>>>>> + cif_conf.audio_bits = audio_bits; >>>>>>>>>>>>>> + cif_conf.client_bits = audio_bits; >>>>>>>>>>>>>> cif_conf.expand = 0; >>>>>>>>>>>>>> cif_conf.stereo_conv = 0; >>>>>>>>>>>>>> cif_conf.replicate = 0; >>>>>>>>>>>>>> @@ -306,14 +317,18 @@ static const struct snd_soc_dai_driver >>>>>>>>>>>>>> tegra30_i2s_dai_template = { >>>>>>>>>>>>>> .channels_min = 2, >>>>>>>>>>>>>> .channels_max = 2, >>>>>>>>>>>>>> .rates = SNDRV_PCM_RATE_8000_96000, >>>>>>>>>>>>>> - .formats = SNDRV_PCM_FMTBIT_S16_LE, >>>>>>>>>>>>>> + .formats = SNDRV_PCM_FMTBIT_S32_LE | >>>>>>>>>>>>>> + SNDRV_PCM_FMTBIT_S24_LE | >>>>>>>>>>>>>> + SNDRV_PCM_FMTBIT_S16_LE, >>>>>>>>>>>>>> }, >>>>>>>>>>>>>> .capture = { >>>>>>>>>>>>>> .stream_name = "Capture", >>>>>>>>>>>>>> .channels_min = 2, >>>>>>>>>>>>>> .channels_max = 2, >>>>>>>>>>>>>> .rates = SNDRV_PCM_RATE_8000_96000, >>>>>>>>>>>>>> - .formats = SNDRV_PCM_FMTBIT_S16_LE, >>>>>>>>>>>>>> + .formats = SNDRV_PCM_FMTBIT_S32_LE | >>>>>>>>>>>>>> + SNDRV_PCM_FMTBIT_S24_LE | >>>>>>>>>>>>>> + SNDRV_PCM_FMTBIT_S16_LE, >>>>>>>>>>>>>> }, >>>>>>>>>>>>>> .ops = &tegra30_i2s_dai_ops, >>>>>>>>>>>>>> .symmetric_rates = 1, >>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> Hello, >>>>>>>>>>>>> >>>>>>>>>>>>> This patch breaks audio on Tegra30. I don't see errors >>>>>>>>>>>>> anywhere, but >>>>>>>>>>>>> there is no audio and reverting this patch helps. Please >>>>>>>>>>>>> fix it. >>>>>>>>>>>> >>>>>>>>>>>> What is the failure mode? I can try and take a look at this >>>>>>>>>>>> some >>>>>>>>>>>> time >>>>>>>>>>>> this week, but I am not sure if I have any boards with an >>>>>>>>>>>> actual >>>>>>>>>>>> useful >>>>>>>>>>>> audio output? >>>>>>>>>>> >>>>>>>>>>> The failure mode is that there no sound. I also noticed that >>>>>>>>>>> video >>>>>>>>>>> playback stutters a lot if movie file has audio track, seems >>>>>>>>>>> something >>>>>>>>>>> times out during of the audio playback. For now I don't have any >>>>>>>>>>> more info. >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Oh, I didn't say how to reproduce it.. for example simply playing >>>>>>>>>> big_buck_bunny_720p_h264.mov in MPV has the audio problem. >>>>>>>>>> >>>>>>>>>> https://download.blender.org/peach/bigbuckbunny_movies/big_buck_bunny_720p_h264.mov >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> Given that the audio drivers uses regmap, it could be good to >>>>>>>>> dump the >>>>>>>>> I2S/AHUB registers while the clip if playing with and without this >>>>>>>>> patch >>>>>>>>> to see the differences. I am curious if the audio is now being >>>>>>>>> played as >>>>>>>>> 24 or 32-bit instead of 16-bit now these are available. >>>>>>>>> >>>>>>>>> You could also dump the hw_params to see the format while >>>>>>>>> playing as >>>>>>>>> well ... >>>>>>>>> >>>>>>>>> $ /proc/asound/<scard-name>/pcm0p/sub0/hw_params >>>>>>>> >>>>>>>> I suppose it is also possible that the codec isn't properly >>>>>>>> doing >16 >>>>>>>> bits and the fact we now offer 24 and 32 could be an issue. I've >>>>>>>> not >>>>>>>> got anything with an audio output on it that would be easy to test. >>>>>>> >>>>>>> I thought I had tested on a Jetson TK1 (Tegra124) but it was >>>>>>> sometime >>>>>>> back. However, admittedly I may have only done simple 16-bit testing >>>>>>> with speaker-test. >>>>>>> >>>>>>> We do verify that all soundcards are detected and registered as >>>>>>> expected >>>>>>> during daily testing, but at the moment we don't have anything that >>>>>>> verifies actual playback. >>>>>> >>>>>> Please take a look at the attached logs. >>>>> >>>>> I wonder if we are falling into FIFO configuration issues with the >>>>> non-16 bit case. >>>>> >>>>> Can you try adding the following two patches? >>>> >>>> It is much better now! There is no horrible noise and no stuttering, >>>> but >>>> audio still has a "robotic" effect, like freq isn't correct. >>> >>> I wonder if there's an issue with FIFO stoking? I should also possibly >>> add the correctly stop FIFOs patch as well. >> >> I'll be happy to try more patches. >> >> Meanwhile sound on v5.5+ is broken and thus the incomplete patches >> should be reverted. > > Have you checked if just removing the 24/32 bits from .formats in > the dai template and see if the problem continues? It works. > I will try and > see if I can find the code that does the fifo level checking and > see if the problem is in the FIFO fill or something else in the > audio hub setup. Ok _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx https://mailman.alsa-project.org/mailman/listinfo/alsa-devel