On 3/30/20 7:52 PM, Stephan Gerhold wrote:
At the moment, playing audio with PulseAudio with the qdsp6 driver results in distorted sound. It seems like its timer-based scheduling does not work properly with qdsp6 since setting tsched=0 in the PulseAudio configuration avoids the issue. Apparently this happens when the pointer() callback is not accurate enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop PulseAudio from using timer-based scheduling by default. According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html: The flag is being used in the sense explained in the previous audio meeting -- the data transfer granularity isn't fine enough but aligned to the period size (or less). q6asm-dai reports the position as multiple of prtd->pcm_count = snd_pcm_lib_period_bytes(substream) so it indeed just a multiple of the period size. Therefore adding the flag here seems appropriate and makes audio work out of the box. Fixes: 2a9e92d371db ("ASoC: qdsp6: q6asm: Add q6asm dai driver") Cc: Srinivas Kandagatla <srinivas.kandagatla@xxxxxxxxxx> Signed-off-by: Stephan Gerhold <stephan@xxxxxxxxxxx> --- I'm still quite confused about the true meaning of SNDRV_PCM_INFO_BATCH, so please correct me if I'm wrong :)
The meaning might have changed over the years, but the way it is used right now is that it means that the position pointer has limited granularity. With 'limited' being a bit fuzzy, but typically means that the granularity is worse than a few samples.
This driver definitely falls into the limited category as the granularity seems to be period size.
- Lars
