Re: [PATCH 10/11] alsaaudio: change default playback settings

[Volker Rümelin]

Am 28.12.22 um 14:52 schrieb Christian Schoenebeck:
> On Monday, December 26, 2022 4:08:37 PM CET Volker Rümelin wrote:
> > Am 21.12.22 um 12:03 schrieb Christian Schoenebeck:
> > > On Sunday, December 18, 2022 6:15:38 PM CET Volker Rümelin wrote:
> > > > The currently used default playback settings in the ALSA audio
> > > > backend are a bit unfortunate. With a few emulated audio devices,
> > > > audio playback does not work properly. Here is a short part of
> > > > the debug log while audio is playing (elapsed time in seconds).
> > > Which emulated devices are these?
> > The hda device and sb16. When I wrote this patch two months ago ac97
> > also had occasional dropouts, but at the moment ac97 works without issues.
> >
> > > > audio: Elapsed since last alsa run (running): 0.046244
> > > > audio: Elapsed since last alsa run (running): 0.023137
> > > > audio: Elapsed since last alsa run (running): 0.023170
> > > > audio: Elapsed since last alsa run (running): 0.023650
> > > > audio: Elapsed since last alsa run (running): 0.060802
> > > > audio: Elapsed since last alsa run (running): 0.031931
> > > >
> > > > For some audio devices the time of more than 23ms between updates
> > > > is too long.
> > > >
> > > > Set the period time to 5.8ms so that the maximum time between
> > > > two updates typically does not exceed 11ms. This roughly matches
> > > > the 10ms period time when doing playback with the audio timer.
> > > > After this patch the debug log looks like this.
> > > And what about dynamically adapting that value instead of reducing period time
> > > for everyone by default?
> > It seems this would be only needed for the ALSA backend. All other
> > backends with the exception of OSS are fine with a 10ms period, and the
> > ALSA audio backend also uses 10ms with -audiodev
> > alsa,out.try-poll=off,in.try-poll=off.
> OK, but all it would need was adjusting dev->timer_period appropriately either
> in audio_validate_opts() [audio/audio.c, line 2126] to handle it generalized
> or at the end of alsa_audio_init() [audio/alsaaudio.c, line 944] if
> specifically for ALSA only, no?

I think this should be the other way around. If period-length wasn't 
specified on the command line, it should be calculated from 
timer-period. With timer based playback or recording, the guest should 
be able to write or read new audio frames every timer-period 
microseconds. To have a high probability that the guest can write or 
read new frames every timer-period, the asynchronous updates of the 
audio backend have to be faster than timer-period e.g. 75-80% of 
timer-period. But that's a different patch.

> > > 23ms is usually a good trade off between low latency, CPU load and potential
> > > for audio dropouts.
> > Quite often it's longer than 23ms. For the rest of the audio backends a
> > timer period of 10ms was selected as a good trade off between CPU load
> > and audio dropouts. But you are right, this patch increases the CPU load.
> >
> > On my system the CPU load is increased by 0.9%. This was measured with a
> > Linux guest using rhythmbox for audio playback. The guest was configured
> > to use pulseaudio as sound server. The measurement was done with top -b
> > -d 10 -n 14 over a period of two minutes. The first and last measurement
> > was dropped. The average QEMU CPU load was 10.7% with and 9.8% without
> > this patch.
> >
> > I would prefer a system with a 0.9% increased CPU load where audio just
> > works over a system where you have to fine tune audio parameters.