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, 0 insertions, 15 deletions

diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index ddbf733b..8d9d6ecf 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -170,8 +170,6 @@ rtp_source_init (RTPSource * src)
 
   src->payload = 0;
   src->clock_rate = -1;
-  src->clock_base = -1;
-  src->clock_base_time = -1;
   src->packets = g_queue_new ();
   src->seqnum_base = -1;
   src->last_rtptime = -1;
@@ -527,10 +525,6 @@ rtp_source_update_caps (RTPSource * src, GstCaps * caps)
   gst_structure_get_int (s, "clock-rate", &src->clock_rate);
   GST_DEBUG ("got clock-rate %d", src->clock_rate);
 
-  if (gst_structure_get_uint (s, "clock-base", &val))
-    src->clock_base = val;
-  GST_DEBUG ("got clock-base %" G_GINT64_FORMAT, src->clock_base);
-
   if (gst_structure_get_uint (s, "seqnum-base", &val))
     src->seqnum_base = val;
   GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
@@ -771,13 +765,6 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
 
   rtptime = gst_rtp_buffer_get_timestamp (buffer);
 
-  /* no clock-base, take first rtptime as base */
-  if (src->clock_base == -1) {
-    GST_DEBUG ("using clock-base of %" G_GUINT32_FORMAT, rtptime);
-    src->clock_base = rtptime;
-    src->clock_base_time = arrival->timestamp;
-  }
-
   /* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
    * care about the absolute value, just the difference. */
   rtparrival = gst_util_uint64_scale_int (ntpnstime, clock_rate, GST_SECOND);
@@ -923,13 +910,11 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
     } else {
       /* unacceptable jump */
       stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
-      src->clock_base = -1;
       goto bad_sequence;
     }
   } else {
     /* duplicate or reordered packet, will be filtered by jitterbuffer. */
     GST_WARNING ("duplicate or reordered packet");
-    src->clock_base = -1;
   }
 
   src->stats.octets_received += arrival->payload_len;