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, 28 insertions, 8 deletions

diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index 947fef7e..428181f2 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -40,6 +40,7 @@ enum
   SIGNAL_ON_BYE_SSRC,
   SIGNAL_ON_BYE_TIMEOUT,
   SIGNAL_ON_TIMEOUT,
+  SIGNAL_ON_SENDER_TIMEOUT,
   LAST_SIGNAL
 };
 
@@ -212,6 +213,18 @@ rtp_session_class_init (RTPSessionClass * klass)
       G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
       NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
       RTP_TYPE_SOURCE);
+  /**
+   * RTPSession::on-sender-timeout:
+   * @session: the object which received the signal
+   * @src: the RTPSource that timed out
+   *
+   * Notify of an SSRC that was a sender but timed out and became a receiver.
+   */
+  rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
+      g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
+      NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
+      RTP_TYPE_SOURCE);
 
   g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
       g_param_spec_object ("internal-source", "Internal Source",
@@ -513,6 +526,15 @@ on_timeout (RTPSession * sess, RTPSource * source)
   RTP_SESSION_LOCK (sess);
 }
 
+static void
+on_sender_timeout (RTPSession * sess, RTPSource * source)
+{
+  RTP_SESSION_UNLOCK (sess);
+  g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
+      source);
+  RTP_SESSION_LOCK (sess);
+}
+
 /**
  * rtp_session_new:
  *
@@ -908,9 +930,8 @@ check_collision (RTPSession * sess, RTPSource * source,
     RTPArrivalStats * arrival, gboolean rtp)
 {
   /* If we have not arrival address, we can't do collision checking */
-  if (!arrival->have_address) {
+  if (!arrival->have_address)
     return FALSE;
-  }
 
   if (sess->source != source) {
     /* This is not our local source, but lets check if two remote
@@ -1479,12 +1500,6 @@ rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
   if (!source)
     return;
 
-  /* we somehow need to transfer the clock_base and the base time to the next
-   * element, we use the offset and offset_end fields in the buffer for this
-   * hack */
-  GST_BUFFER_OFFSET (packet->buffer) = source->clock_base;
-  GST_BUFFER_OFFSET_END (packet->buffer) = source->clock_base_time;
-
   prevsender = RTP_SOURCE_IS_SENDER (source);
 
   /* first update the source */
@@ -2096,6 +2111,7 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
 {
   gboolean remove = FALSE;
   gboolean byetimeout = FALSE;
+  gboolean sendertimeout = FALSE;
   gboolean is_sender, is_active;
   RTPSession *sess = data->sess;
   GstClockTime interval;
@@ -2138,6 +2154,7 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
             GST_TIME_ARGS (source->last_rtp_activity));
         source->is_sender = FALSE;
         sess->stats.sender_sources--;
+        sendertimeout = TRUE;
       }
     }
   }
@@ -2153,6 +2170,9 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
       on_bye_timeout (sess, source);
     else
       on_timeout (sess, source);
+  } else {
+    if (sendertimeout)
+      on_sender_timeout (sess, source);
   }
   return remove;
 }