[gst-cvs] gst-plugins-bad: rtpbin: don't do lip-sync after a BYE

[<wt...@ke...>]

Module: gst-plugins-bad
Branch: master
Commit: 9b1bdac9cc60e274ddc10b2948d9f7d70f2b6e49
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/commit/?id=9b1bdac9cc60e274ddc10b2948d9f7d70f2b6e49

Author: Wim Taymans <wim...@co...>
Date:   Mon Jul 27 13:17:20 2009 +0200

rtpbin: don't do lip-sync after a BYE

After a BYE packet from a source, stop forwarding the SR packets for lip-sync
to rtpbin. Some senders don't update their SR packets correctly after sending a
BYE and then we break lip-sync. We prefer to let the jitterbuffers drain with
the current lip-sync instead.

---

 gst/rtpmanager/rtpsession.c |   15 ++++++++++-----
 1 files changed, 10 insertions(+), 5 deletions(-)

diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index fe38c36..9d73d51 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -1577,7 +1577,7 @@ rtp_session_process_rb (RTPSession * sess, RTPSource * source,
  */
 static void
 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
-    RTPArrivalStats * arrival)
+    RTPArrivalStats * arrival, gboolean * do_sync)
 {
   guint32 senderssrc, rtptime, packet_count, octet_count;
   guint64 ntptime;
@@ -1594,6 +1594,12 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
   if (!source)
     return;
 
+  /* don't try to do lip-sync for sources that sent a BYE */
+  if (rtp_source_received_bye (source))
+    *do_sync = FALSE;
+  else
+    *do_sync = TRUE;
+
   prevsender = RTP_SOURCE_IS_SENDER (source);
 
   /* first update the source */
@@ -1816,7 +1822,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
     GstClockTime current_time)
 {
   GstRTCPPacket packet;
-  gboolean more, is_bye = FALSE, is_sr = FALSE;
+  gboolean more, is_bye = FALSE, do_sync = FALSE;
   RTPArrivalStats arrival;
   GstFlowReturn result = GST_FLOW_OK;
 
@@ -1853,8 +1859,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
 
     switch (type) {
       case GST_RTCP_TYPE_SR:
-        rtp_session_process_sr (sess, &packet, &arrival);
-        is_sr = TRUE;
+        rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
         break;
       case GST_RTCP_TYPE_RR:
         rtp_session_process_rr (sess, &packet, &arrival);
@@ -1891,7 +1896,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
   RTP_SESSION_UNLOCK (sess);
 
   /* notify caller of sr packets in the callback */
-  if (is_sr && sess->callbacks.sync_rtcp)
+  if (do_sync && sess->callbacks.sync_rtcp)
     result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
         sess->sync_rtcp_user_data);
   else