GNOME Bugzilla – Bug 590425
Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio
Last modified: 2009-08-06 17:16:46 UTC
The clock used by alsasrc when its slaved to GstSystemClock (because its the pipeline clock) has an accuracy of ±20ms.. Which is a big problem for RTP streaming since the buffer size is around 20ms. The timestamp jump by ~1ms then ~20ms then ~1ms and it confuses the recipient.. The is also probably the cause of bug #590065. Example program: import gst, time p = gst.parse_launch("alsasrc ! identity check-perfect=1 ! fakesink name=s") p.use_clock(gst.system_clock_obtain()) p.set_state(gst.STATE_PLAYING) print p.get_clock() time.sleep(10) run with GST_DEBUG=2 ... BaseAudioSrc should probably do something to smooth the clock... This is very annoying...
If I set the slave-method to skew or resample it works fine.... Anyway, the current default is really bad for us..
Enabled in git. Overridden in pulsesrc in -good for the upcomming release.