[alsa-devel] [PATCH v5 2/7] ASoC: tegra: Allow 24bit and 32bit samples
Dmitry Osipenko
digetx at gmail.com
Sun Dec 22 18:08:36 CET 2019
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 at 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 at codethink.co.uk>
>>>>>>>>>>>>>> [ben.dooks at codethink.co.uk: fixup merge of 24 and 32bit]
>>>>>>>>>>>>>> [ben.dooks at codethink.co.uk: add pm calls around ytdm config]
>>>>>>>>>>>>>> [ben.dooks at codethink.co.uk: drop debug printing to dev_dbg]
>>>>>>>>>>>>>> Signed-off-by: Ben Dooks <ben.dooks at 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?
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
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