gst/rtpmanager/: Updated example pipelines in docs.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream),
(gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found),
(create_recv_rtp), (create_recv_rtcp), (create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
Updated example pipelines in docs.
Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync.
Set the default latency correctly.
Add some more points where we can get caps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Add ts-offset property to control timestamping.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp),
(gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate),
(gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Various cleanups.
Feed rtpsession manager with NTP time based on pipeline clock when
handling RTP packets and RTCP timeouts.
Perform all RTCP with the system clock.
Set caps on RTCP outgoing buffers.
* gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc),
(create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain):
* gst/rtpmanager/gstrtpssrcdemux.h:
Also demux RTCP messages.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rb), (rtp_session_process_sr),
(rtp_session_process_rr), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_send_bye),
(session_start_rtcp), (session_report_blocks), (session_cleanup),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Remove the get_time callback, the GStreamer part will feed us with
enough timing information.
Split sync timing and RTCP timing information.
Factor out common RB handling for SR and RR.
Send out SR RTCP packets for lip-sync.
Move SR and RR packet info generation to the source.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp),
(rtp_source_process_sr), (rtp_source_process_rb),
(rtp_source_get_new_sr), (rtp_source_get_new_rb),
(rtp_source_get_last_sr):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Use caps on incomming buffers to get timing information when they are
there.
Calculate clock scew of the receiver compared to the sender and adjust
the rtp timestamps.
Calculate the round trip in sources.
Do SR and RR calculations in the source.

1 files changed, 19 insertions, 19 deletions

diff --git a/gst/rtpmanager/rtpsession.h b/gst/rtpmanager/rtpsession.h
index 9380b55e..d7dbb784 100644
--- a/gst/rtpmanager/rtpsession.h
+++ b/gst/rtpmanager/rtpsession.h
@@ -82,28 +82,30 @@ typedef GstFlowReturn (*RTPSessionSendRTP) (RTPSession *sess, RTPSource *src, Gs
 typedef GstFlowReturn (*RTPSessionSendRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
 
 /**
- * RTPSessionClockRate:
+ * RTPSessionSyncRTCP:
  * @sess: an #RTPSession
- * @payload: the payload
+ * @src: the #RTPSource
+ * @buffer: the RTCP buffer ready for sending
  * @user_data: user data specified when registering
  *
- * This callback will be called when @sess needs the clock-rate of @payload.
+ * This callback will be called when @sess has and SR @buffer ready for doing
+ * synchronisation between streams.
  *
- * Returns: the clock-rate of @pt.
+ * Returns: a #GstFlowReturn.
  */
-typedef gint (*RTPSessionClockRate) (RTPSession *sess, guint8 payload, gpointer user_data);
+typedef GstFlowReturn (*RTPSessionSyncRTCP) (RTPSession *sess, RTPSource *src, GstBuffer *buffer, gpointer user_data);
 
 /**
- * RTPSessionGetTime:
+ * RTPSessionClockRate:
  * @sess: an #RTPSession
+ * @payload: the payload
  * @user_data: user data specified when registering
  *
- * This callback will be called when @sess needs the current time in
- * nanoseconds.
+ * This callback will be called when @sess needs the clock-rate of @payload.
  *
- * Returns: a #GstClockTime with the current time in nanoseconds.
+ * Returns: the clock-rate of @pt.
  */
-typedef GstClockTime (*RTPSessionGetTime) (RTPSession *sess, gpointer user_data);
+typedef gint (*RTPSessionClockRate) (RTPSession *sess, guint8 payload, gpointer user_data);
 
 /**
  * RTPSessionReconsider:
@@ -121,7 +123,7 @@ typedef void (*RTPSessionReconsider) (RTPSession *sess, gpointer user_data);
  * @RTPSessionProcessRTP: callback to process RTP packets
  * @RTPSessionSendRTP: callback for sending RTP packets
  * @RTPSessionSendRTCP: callback for sending RTCP packets
- * @RTPSessionGetTime: callback for returning the current time
+ * @RTPSessionSyncRTCP: callback for handling SR packets
  * @RTPSessionReconsider: callback for reconsidering the timeout
  *
  * These callbacks can be installed on the session manager to get notification
@@ -132,8 +134,8 @@ typedef struct {
   RTPSessionProcessRTP  process_rtp;
   RTPSessionSendRTP     send_rtp;
   RTPSessionSendRTCP    send_rtcp;
+  RTPSessionSyncRTCP    sync_rtcp;
   RTPSessionClockRate   clock_rate;
-  RTPSessionGetTime     get_time;
   RTPSessionReconsider  reconsider;
 } RTPSessionCallbacks;
 
@@ -190,8 +192,7 @@ struct _RTPSession {
 
   RTPSessionStats stats;
 
-  /* for mapping RTP time to NTP time */
-  GstClockTime  start_timestamp;
+  /* for mapping clock time to NTP time */
   GstClockTime  base_time;
 };
 
@@ -250,18 +251,17 @@ RTPSource*      rtp_session_get_source_by_cname    (RTPSession *sess, const gcha
 RTPSource*      rtp_session_create_source          (RTPSession *sess);
 
 /* processing packets from receivers */
-GstFlowReturn   rtp_session_process_rtp            (RTPSession *sess, GstBuffer *buffer);
+GstFlowReturn   rtp_session_process_rtp            (RTPSession *sess, GstBuffer *buffer, guint64 ntpnstime);
 GstFlowReturn   rtp_session_process_rtcp           (RTPSession *sess, GstBuffer *buffer);
 
 /* processing packets for sending */
-GstFlowReturn   rtp_session_send_rtp               (RTPSession *sess, GstBuffer *buffer);
-void            rtp_session_set_base_time          (RTPSession *sess, GstClockTime base_time);
-void            rtp_session_set_timestamp_sync     (RTPSession *sess, GstClockTime start_timestamp);
+GstFlowReturn   rtp_session_send_rtp               (RTPSession *sess, GstBuffer *buffer, guint64 ntptime);
+
 /* stopping the session */
 GstFlowReturn   rtp_session_send_bye               (RTPSession *sess, const gchar *reason);
 
 /* get interval for next RTCP interval */
 GstClockTime    rtp_session_next_timeout          (RTPSession *sess, GstClockTime time);
-GstFlowReturn   rtp_session_on_timeout            (RTPSession *sess, GstClockTime time);
+GstFlowReturn   rtp_session_on_timeout            (RTPSession *sess, GstClockTime time, guint64 ntpnstime);
 
 #endif /* __RTP_SESSION_H__ */