On Thu, 14 May 2009 07:39:14 -0700Grant <emailgrant@xxxxxxxxx> wrote:
> >> >> I'm playing a video in miro and I get:> >> >>> >> >> # lsof|grep speex> >> >> miro.real 9019 user mem REG 8,3 108992> >> >> 28197654 /usr/lib64/libspeex.so.1.4.0> >> >>> >> >> Does this mean dmix is using speex? If so, what else could be causing> >> >> my static problem? I basically hear static whenever dmix is involved.> >> >> If I have mpd resample with libsamplerate, I get no static.> >> >>> >> >> - Grant> >> >>> >> >> >> > Yes, you are using speex.> >>> >> I don't think my defaults.pcm.rate_converter is being obeyed. I> >> switched from "speexrate_best" to "samplerate_best" and also tried> >> removing the definition entirely, but lsof still says whichever> >> program is playing audio is opening the speex file and not the> >> libsamplerate file.> >>> >> I also tried removing speex from the system and speex disappeared from> >> lsof, but the static remained.> >>> >> > I suggest first of all to temporarily leave 'miro' aside - it's a> >> > non-trivial piece of SW which might have its own quirks.> >> >> >> > I suggest to start from very basic 'aplay' with .wav files - just to> >> > make sure ALSA works OK.> >>> >> I can definitely confirm static with aplay .wav files that doesn't> >> exist in mpd. If I don't have mpd bypass dmix I get static there too.> >> Where should I go from here?> >>> >> # lsof|grep aplay> [snip]> >>> >> - Grant> >>> >>> >> > Then, say, 'mplayer' with .flac, .mp3.> >> >> >> > You can try to increase ALSA buffers size, but I do not remember how to> >> > do this, though I remember it was easy.> >> >> >> > Regards,> >> > Sergei.> >>> >> > Then start from very basic things:> >> > 1) choose direct HW output;> > 2) choose sample rate supported by HW - if necessary, resample your> > input file by high quality stand-alone resampler;> > 3) also take care of number of bits if necessary;> > 4) start playing with ALSA buffer size.> >> > For resampling/format conversion you can use 'ecasound' or 'sox'.> >> > Disclaimer: I am not an ALSA developer, so my recommendation are from> > end user point of view.> >> > Regards,> > Sergei.> > I added this to /etc/asound.conf:> > pcm.!default {> type plug> slave.pcm {> type dmix> ipc_key 1024> slave {> pcm "hw:0"> format S24_3LE> rate 96000> }> }> }> > I can see that it works because the 96k LED lights up on the DAC, but> the static remains. I've also tried it in combination with:> > defaults.pcm.rate_converter "samplerate_best"> > I also tried various values of buffer_size and it caused some skipping> but didn't affect the static at all.> > - Grant>
(For no good reason ?) try as root, play something long, increase runtimepriority while playing.
Of course, I am not sure, I'm just suggesting to check whether priorityis the issue.
By the way, 96KHz sampling rate is a pretty high load for the system.
Regards. Sergei.
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