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, 23 insertions, 0 deletions

diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index cc794b62..e78e972d 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -193,6 +193,7 @@ enum
   SIGNAL_ON_BYE_SSRC,
   SIGNAL_ON_BYE_TIMEOUT,
   SIGNAL_ON_TIMEOUT,
+  SIGNAL_ON_SENDER_TIMEOUT,
   LAST_SIGNAL
 };
 
@@ -416,6 +417,13 @@ on_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
       src->ssrc);
 }
 
+static void
+on_sender_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
+{
+  g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
+      src->ssrc);
+}
+
 GST_BOILERPLATE (GstRtpSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
 
 static void
@@ -574,6 +582,18 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass)
       g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
       G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_timeout),
       NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
+  /**
+   * GstRtpSession::on-sender-timeout:
+   * @sess: the object which received the signal
+   * @ssrc: the SSRC 
+   *
+   * Notify of a sender SSRC that has timed out and became a receiver
+   */
+  gst_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 (GstRtpSessionClass,
+          on_sender_timeout), NULL, NULL, g_cclosure_marshal_VOID__UINT,
+      G_TYPE_NONE, 1, G_TYPE_UINT);
 
   g_object_class_install_property (gobject_class, PROP_NTP_NS_BASE,
       g_param_spec_uint64 ("ntp-ns-base", "NTP base time",
@@ -655,6 +675,7 @@ gst_rtp_session_init (GstRtpSession * rtpsession, GstRtpSessionClass * klass)
   rtpsession->priv->lock = g_mutex_new ();
   rtpsession->priv->sysclock = gst_system_clock_obtain ();
   rtpsession->priv->session = rtp_session_new ();
+
   /* configure callbacks */
   rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession);
   /* configure signals */
@@ -674,6 +695,8 @@ gst_rtp_session_init (GstRtpSession * rtpsession, GstRtpSessionClass * klass)
       (GCallback) on_bye_timeout, rtpsession);
   g_signal_connect (rtpsession->priv->session, "on-timeout",
       (GCallback) on_timeout, rtpsession);
+  g_signal_connect (rtpsession->priv->session, "on-sender-timeout",
+      (GCallback) on_sender_timeout, rtpsession);
   rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
       (GDestroyNotify) gst_caps_unref);