[gstreamer-bugs] [Bug 590425] Slaved alsasrc clock with slave-method=re-timestamp 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

Olivier Crete (Tester) changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
            Summary|Slaved alsasrc clock not    |Slaved alsasrc clock with
                   |usable for RTP audio        |slave-method=re-timestamp
                   |                            |not usable for RTP audio




------- Comment #1 from Olivier Crete (Tester)  2009-08-01 00:17 UTC -------
If I set the slave-method to skew or resample it works fine.... Anyway, the
current default is really bad for us..


-- 
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.