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

[GStreamer (bugzilla.gnome.org)]

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

Wim Taymans <wim.taymans> changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
 Attachment #260305|none                        |reviewed
             status|                            |

--- Comment #46 from Wim Taymans <wim.taymans at gmail.com> 2013-11-22 16:02:24 UTC ---
(From update of attachment 260305)
>   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

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.

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