gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.

1 files changed, 12 insertions, 1 deletions

diff --git a/gst/rtpmanager/rtpstats.h b/gst/rtpmanager/rtpstats.h
index f82c9585d..3408300d3 100644
--- a/gst/rtpmanager/rtpstats.h
+++ b/gst/rtpmanager/rtpstats.h
@@ -150,11 +150,22 @@ typedef struct {
 #define RTP_STATS_RECEIVER_FRACTION     (1.0 - RTP_STATS_SENDER_FRACTION)
 
 /*
- * When receiving a BYE from a source, remove the source fomr the database
+ * When receiving a BYE from a source, remove the source from the database
  * after this timeout.
  */
 #define RTP_STATS_BYE_TIMEOUT           (2 * GST_SECOND)
 
+/*
+ * The maximum number of missing packets we tollerate. These are packets with a
+ * sequence number bigger than the last seen packet.
+ */
+#define RTP_MAX_DROPOUT      3000
+/*
+ * The maximum number of misordered packets we tollerate. These are packets with
+ * a sequence number smaller than the last seen packet.
+ */
+#define RTP_MAX_MISORDER     100
+
 /**
  * RTPSessionStats:
  *