configure.ac: Disable rtpmanager for now because it depends on CVS -base.

Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.

1 files changed, 15 insertions, 13 deletions

diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 6825e9cc..9162d76c 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -129,7 +129,7 @@ typedef struct _GstRTPBinClient GstRTPBinClient;
 static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
 
 static GstCaps *pt_map_requested (GstElement * element, guint pt,
-    GstRTPBinStream * stream);
+    GstRTPBinSession * session);
 
 /* Manages the RTP stream for one SSRC.
  *
@@ -215,9 +215,9 @@ static GstRTPBinSession *
 create_session (GstRTPBin * rtpbin, gint id)
 {
   GstRTPBinSession *sess;
-  GstElement *elem, *demux;
+  GstElement *session, *demux;
 
-  if (!(elem = gst_element_factory_make ("rtpsession", NULL)))
+  if (!(session = gst_element_factory_make ("rtpsession", NULL)))
     goto no_session;
 
   if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
@@ -227,13 +227,17 @@ create_session (GstRTPBin * rtpbin, gint id)
   sess->lock = g_mutex_new ();
   sess->id = id;
   sess->bin = rtpbin;
-  sess->session = elem;
+  sess->session = session;
   sess->demux = demux;
   sess->ptmap = g_hash_table_new (NULL, NULL);
   rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
 
-  gst_bin_add (GST_BIN_CAST (rtpbin), elem);
-  gst_element_set_state (elem, GST_STATE_PLAYING);
+  /* provide clock_rate to the session manager when needed */
+  g_signal_connect (session, "request-pt-map",
+      (GCallback) pt_map_requested, sess);
+
+  gst_bin_add (GST_BIN_CAST (rtpbin), session);
+  gst_element_set_state (session, GST_STATE_PLAYING);
   gst_bin_add (GST_BIN_CAST (rtpbin), demux);
   gst_element_set_state (demux, GST_STATE_PLAYING);
 
@@ -247,7 +251,7 @@ no_session:
   }
 no_demux:
   {
-    gst_object_unref (elem);
+    gst_object_unref (session);
     g_warning ("rtpbin: could not create rtpssrcdemux element");
     return NULL;
   }
@@ -351,7 +355,7 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
 
   /* provide clock_rate to the jitterbuffer when needed */
   g_signal_connect (buffer, "request-pt-map",
-      (GCallback) pt_map_requested, stream);
+      (GCallback) pt_map_requested, session);
 
   gst_bin_add (GST_BIN_CAST (session->bin), buffer);
   gst_element_set_state (buffer, GST_STATE_PLAYING);
@@ -590,14 +594,12 @@ new_payload_found (GstElement * element, guint pt, GstPad * pad,
 }
 
 static GstCaps *
-pt_map_requested (GstElement * element, guint pt, GstRTPBinStream * stream)
+pt_map_requested (GstElement * element, guint pt, GstRTPBinSession * session)
 {
   GstRTPBin *rtpbin;
-  GstRTPBinSession *session;
   GstCaps *caps;
 
-  rtpbin = stream->bin;
-  session = stream->session;
+  rtpbin = session->bin;
 
   GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
       session->id);
@@ -647,7 +649,7 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
    * demuxer so that it can apply a proper caps on the buffers for the
    * depayloaders. */
   stream->demux_ptreq_sig = g_signal_connect (stream->demux,
-      "request-pt-map", (GCallback) pt_map_requested, stream);
+      "request-pt-map", (GCallback) pt_map_requested, session);
 
   GST_RTP_SESSION_UNLOCK (session);