[alsa-devel] Immediate underrun with PulseAudio ALSA plugin when PA and ALSA buffer sizes differ
superquad.vortex2 at gmail.com
Tue Jul 17 09:36:52 CEST 2012
2012-7-17 上午5:39 於 "Matthew Gregan" <kinetik at flim.org> 寫道：
> I'm investigating an issue in Firefox's audio code when the PulseAudio
> plugin is in use. I posted about this on pulseaudio-discuss last week
> but I hoped I might have more success here.
> Firefox requests a particular latency (100ms, 4410 frames at 44.1kHz) via
this function does not work for all sound card drivers.
many pci sound cards have hardware constraints which you cannot select 10ms
or 100ms period time / buffer time.
> Inside the plugin (pcm_pulse.c:pulse_hw_params), that
> value is used to set up buffer_attr. When the PA stream is connected in
> pcm_pulse.c:pulse_prepare, PA may configure the stream with larger
> buffer_attr values (e.g. because the minimum sink latency has increased
> time due to underruns on the server, or because the sink hardware doesn't
> support lower latency), but this isn't reflected in pcm->buffer_attr or
> higher layers in ALSA (i.e. pcm->buffer_size is not updated).
> The problem I'm faced with is that there doesn't appear to be a way to
> detect and handle this issue at the ALSA API level, and requesting a too
> latency results in broken audio playback rather than a PCM setup failure
> a larger buffer than requested being used.
do you mean the pulse plugin advertise a low latency but the server in fact
force the application to use a large buffer with high latency?
> In the case of the PA server's minimum latency increasing over time, this
> also means that a stream that was configured and running correctly may
> while running if PA increases the minimum latency above what the PCM was
> originally configured with.
> I've attached a simple testcase that uses snd_pcm_wait,
> snd_pcm_avail_update, and snd_pcm_writei. Run it with a latency argument
> specified in milliseconds on the command line. For my local machine, 55ms
> works and 54ms fails immediately like so:
> snd_pcm_wait wakes
> snd_pcm_avail_update returns 4410
> snd_pcm_writei writes 4410
> snd_pcm_wait wakes immediately
> snd_pcm_avail_update returns -EPIPE
> (Note that when I reported this on pulseaudio-discuss, my server's minimum
> latency was 45ms, and now pacmd list-sinks | grep configured\ latency
> reports a minimum latency of 56ms)
> I'd expect to see one of the following behaviours instead:
> 1. PCM setup fails due to requesting a too small buffer.
> 2. Buffer is silently raised during setup and snd_pcm_avail_update
> the correct number of frames.
there is no negotiation between pa client and pa server about the
capability of different sound cards and you cannot change the peiod
time/buffer time when there are another pa client(s) already
> Presumably this could be achieved by having the PA plugin report valid
> values from pcm_pulse.c:pulse_hw_constraint, but I'm not sure how to query
> the necessary values from the server. This also wouldn't address the
> problem where the buffer_attr changes over time, and I'm not sure what to
> about that case.
More information about the Alsa-devel