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:
|
||
|---|---|---|
| .. | ||
| qdsp6 | ||
| Kconfig | ||
| Makefile | ||
| apq8016_sbc.c | ||
| apq8096.c | ||
| common.c | ||
| common.h | ||
| lpass-apq8016.c | ||
| lpass-cpu.c | ||
| lpass-ipq806x.c | ||
| lpass-lpaif-reg.h | ||
| lpass-platform.c | ||
| lpass.h | ||
| sdm845.c | ||
| storm.c | ||