Re: [alsa-devel] [PATCH v5 2/7] ASoC: tegra: Allow 24bit and 32bit samples

[Ben Dooks <ben.dooks@codethink.co.uk>]

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@codethink.co.uk>
>>>>>>>>>>>>>
>>>>>>>>>>>>> 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@codethink.co.uk>
>>>>>>>>>>>>> [ben.dooks@codethink.co.uk: fixup merge of 24 and 32bit]
>>>>>>>>>>>>> [ben.dooks@codethink.co.uk: add pm calls around ytdm config]
>>>>>>>>>>>>> [ben.dooks@codethink.co.uk: drop debug printing to dev_dbg]
>>>>>>>>>>>>> Signed-off-by: Ben Dooks <ben.dooks@codethink.co.uk>
>>>>>>>>>>>>> ---
>>>>>>>>>>>>> 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? 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.


-- 
Ben Dooks				http://www.codethink.co.uk/
Senior Engineer				Codethink - Providing Genius

https://www.codethink.co.uk/privacy.html
_______________________________________________
Alsa-devel mailing list
Alsa-devel@alsa-project.org
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel