<html> <head> <base href="https://bugs.freedesktop.org/" /> </head> <body> <div> <a class="bz_bug_link bz_status_NEW " title="NEW - severe underruns with usb audio, works with pa 3.0 - buffer setup seems wrong" href="https://bugs.freedesktop.org/show_bug.cgi?id=86262#c6">Comment # 6</a> on <a class="bz_bug_link bz_status_NEW " title="NEW - severe underruns with usb audio, works with pa 3.0 - buffer setup seems wrong" href="https://bugs.freedesktop.org/show_bug.cgi?id=86262">bug 86262</a> from <a class="email" href="mailto:freedesktop-bugzilla@dm.cobite.com" title="David Mansfield <freedesktop-bugzilla@dm.cobite.com>"> David Mansfield</a> <pre>Can someone "in the know" comment on what this part of the diff means: -D: [pulseaudio] alsa-sink.c: setting avail_min=87319 +D: [pulseaudio] alsa-sink.c: setting avail_min=1 As I understand, for glitch-free audio, the playback stream should only start when at least two buffers are full. Is this saying that it will start immediately? (1 byte?) Perhaps that is the issue. In my specific case, the audio data to "play" is coming from an RTP stream, and is therefore paced by some external network client, and perhaps the stream is being kicked off too soon, and/or not rebuffering correctly on underrun. How can I diagnose this further?</pre> </div> <hr> You are receiving this mail because: <ul> <li>You are the QA Contact for the bug.</li> <li>You are the assignee for the bug.</li> </ul> </body> </html>