[gstreamer-bugs] [Bug 590425] New: Slaved alsasrc clock not usable for RTP audio

[GStreamer (bugzilla.gnome.org)]

If you have any questions why you received this email, please see the text at
the end of this email. Replies to this email are NOT read, please see the text
at the end of this email. You can add comments to this bug at:
  http://bugzilla.gnome.org/show_bug.cgi?id=590425

  GStreamer | gst-plugins-base | Ver: git
           Summary: Slaved alsasrc clock not usable for RTP audio
           Product: GStreamer
           Version: git
          Platform: Other
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: blocker
          Priority: Normal
         Component: gst-plugins-base
        AssignedTo: gstreamer-bugs at lists.sourceforge.net
        ReportedBy: tester at tester.ca
         QAContact: gstreamer-bugs at lists.sourceforge.net
     GNOME version: Unspecified
   GNOME milestone: Unspecified


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...


-- 
See http://bugzilla.gnome.org/page.cgi?id=email.html for more info about why you received
this email, why you can't respond via email, how to stop receiving
emails (or reduce the number you receive), and how to contact someone
if you are having problems with the system.

You can add comments to this bug at http://bugzilla.gnome.org/show_bug.cgi?id=590425.