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authorWim Taymans <wim.taymans@collabora.co.uk>2009-07-27 13:17:20 +0200
committerTim-Philipp Müller <tim.muller@collabora.co.uk>2009-08-11 02:30:47 +0100
commit3747ede14a6909e11b19ae7363cd583f5ddb59ec (patch)
treebdd98731d996612f4ecc1ad847ad8f97656d3320 /gst/rtpmanager
parentd2ef095b804182f5b2f4a7a8a4d5a55088ac811d (diff)
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.
Diffstat (limited to 'gst/rtpmanager')
-rw-r--r--gst/rtpmanager/rtpsession.c15
1 files changed, 10 insertions, 5 deletions
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index fe38c36a..9d73d51d 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