Hello all,
based on last week's discussions on #farsight, I've written down some
improvement items for the gst RTP support. Here's the first one. The items
are divided into urgent items (something we must have), and more long-term
stuff (which would be nice to have already now, but without which one can
still build VoIP/conferencing apps).
For background, see:
http://gstreamer.freedesktop.org/documentation/rtp.html
http://bugzilla.gnome.org/show_bug.cgi?id=323017
Please comment either on the list, or IRC (I'll comment and summarize the
feedback).
Codec independent playout buffer element (prio:urgent)
------------------------------------------------------
A gstreamer element that takes in application/x-rtp buffers and stores
them in a playout (jitter) buffer. The buffer will contain the packets
sorted by RTP-timestamp and RTP-seqno, with duplicates removed. This
is essentially the functionality that is not provided by
gst-plugins-base-0.10/gst-libs/gst/rtp/gstbasertpdepayload.c
The element will have a separate thread that schedules out packets for
playout at the pipeline sink. The element will monitor the frequency of
incoming packets, and use this information to select a proper sleep
interval.
The element should be able to detect overrun and underrun situations
caused by clock-drift between the sender and the receiver (sink in
the pipeline the playout buffer is part)
Rationale:
- Separate element is used instead of gstbasertpdepayload.c
because the codec might change mid-stream and so the
playout buffer should be able to contain packets
for multiple codecs at the same time.
- A separate thread is needed to ensure packets are
scheduled even in the case no packets are
received from the network (pause of X msecs is fatal if X
is larger than the size of sink element buffers).
--
under work: Sofia-SIP at http://sofia-sip.sf.net
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