[Bug 711084] rtpmanager: add new rtprtxsend and rtprtxreceive elements for retransmission

GStreamer (bugzilla.gnome.org) bugzilla at gnome.org
Mon Nov 25 06:18:02 PST 2013


https://bugzilla.gnome.org/show_bug.cgi?id=711084
  GStreamer | gst-plugins-good | git

--- Comment #47 from George Kiagiadakis <kiagiadakis.george at gmail.com> 2013-11-25 14:17:59 UTC ---
(In reply to comment #46)
> (From update of attachment 260305 [details])
> >   if (G_UNLIKELY (priv->clock_rate == -1)) {
> >     /* no clock rate given on the caps, try to get one with the signal */
> >     if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
> >@@ -2460,6 +2468,8 @@ do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
> >   event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
> >       gst_structure_new ("GstRTPRetransmissionRequest",
> >           "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
> >+          "ssrc", G_TYPE_UINT, (guint) priv->last_ssrc,
> >+          "payload-type", G_TYPE_UINT, (guint) priv->last_pt,
> >           "running-time", G_TYPE_UINT64, timer->rtx_base,
> >           "delay", G_TYPE_UINT, GST_TIME_AS_MSECONDS (delay),
> >           "retry", G_TYPE_UINT, timer->num_rtx_retry,
> 
> The SSRC is already added by the ssrcdemux element

Right.

> I'm not sure why you would want the pt to be there, it could theoretically
> change
> for each packet and is thus not so interesting to transmit because it isn't
> related to the packet that got lost.

The pt is generally needed to be stored in rtprtxreceive so that the rtp
packets can be fully reconstructed out of rtx packets, since rtx packets do not
carry the original pt. Looking at the code though, it seems we can also store
that internally in rtprtxreceive and discard this patch completely.

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