[gst-cvs] gst-plugins-base: audiortppay: use offsets for RTP timestamps

[Wim Taymans]

Module: gst-plugins-base
Branch: master
Commit: bb91a7b47ca4800e99fb1468489835549ab47c2d
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-base/commit/?id=bb91a7b47ca4800e99fb1468489835549ab47c2d

Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date:   Wed Sep  2 19:49:57 2009 +0200

audiortppay: use offsets for RTP timestamps

Have a custom sample/frame function to generate an offset that the base class
will use for generating RTP timestamps. This results in perfect RTP timestamps
on the output buffers.
Refactor setting metadata on output buffers.
Add some more functionality to _flush().
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
the next outgoing buffer.
Flush the pending data on EOS.

---

 gst-libs/gst/rtp/gstbasertpaudiopayload.c |  145 +++++++++++++++++++++++------
 1 files changed, 115 insertions(+), 30 deletions(-)

Diff:   http://cgit.freedesktop.org/gstreamer/gst-plugins-base/diff/?id=bb91a7b47ca4800e99fb1468489835549ab47c2d