2012-7-17 上午5:39 於 "Matthew Gregan" kinetik@flim.org 寫道：
I'm investigating an issue in Firefox's audio code when the PulseAudio
ALSA
plugin is in use. I posted about this on pulseaudio-discuss last week (
http://lists.freedesktop.org/archives/pulseaudio-discuss/2012-July/014091.ht... ),
but I hoped I might have more success here.
Firefox requests a particular latency (100ms, 4410 frames at 44.1kHz) via snd_pcm_set_params.
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
over
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
low
latency results in broken audio playback rather than a PCM setup failure
or
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
break
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:
- PCM setup fails due to requesting a too small buffer.
- Buffer is silently raised during setup and snd_pcm_avail_update
requests
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 playing/capturing
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
do
about that case.
Thanks, -mjg