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Bug 590425 - Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio
Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio
Status: RESOLVED FIXED
Product: GStreamer
Classification: Platform
Component: gst-plugins-base
git master
Other Linux
: Normal blocker
: 0.10.25
Assigned To: GStreamer Maintainers
GStreamer Maintainers
Depends on:
Blocks:
 
 
Reported: 2009-07-31 23:39 UTC by Olivier Crête
Modified: 2009-08-06 17:16 UTC
See Also:
GNOME target: ---
GNOME version: ---


Description Olivier Crête 2009-07-31 23:39:04 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...
Comment 1 Olivier Crête 2009-08-01 00:17:16 UTC 
If I set the slave-method to skew or resample it works fine.... Anyway, the current default is really bad for us..
Comment 2 Wim Taymans 2009-08-06 17:16:46 UTC 
Enabled in git. Overridden in pulsesrc in -good for the upcomming release.