gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines.

Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
Add some more advanced example pipelines.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_send_rtcp):
Add some debug and FIXME.
Release LOCK when performing session cleanup.
* gst/rtpmanager/rtpsession.c: (session_report_blocks):
Add some debug.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_send_rtp):
Make sure we always send RTP packets with the session SSRC.

4 files changed, 68 insertions, 7 deletions

diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index 7f92b8c9..cb2a9f0b 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -75,9 +75,54 @@
  * </programlisting>
  * Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
  * </para>
+ * <para>
+ * <programlisting>
+ * gst-launch gstrtpbin name=rtpbin \
+ *         v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
+ *                   rtpbin.send_rtp_src_0 ! udpsink port=5000             \
+ *                   rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false \
+ *                   udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0            \
+ *         audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1    \
+ *                   rtpbin.send_rtp_src_1 ! udpsink port=5002             \
+ *                   rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false \
+ *                   udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
+ * </programlisting>
+ * Encode and payload H263 video captured from a v4l2src. Encode and payload AMR
+ * audio generated from audiotestsrc. The video is sent to session 0 in rtpbin
+ * and the audio is sent to session 1. Video packets are sent on UDP port 5000
+ * and audio packets on port 5002. The video RTCP packets for session 0 are sent
+ * on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
+ * RTCP packets for session 0 are received on port 5005 and RTCP for session 1
+ * is received on port 5007. Since RTCP packets from the sender should be sent
+ * as soon as possible, sync=false is configured on udpsink.
+ * </para>
+ * <para>
+ * <programlisting>
+ *  gst-launch -v gstrtpbin name=rtpbin                                                \
+ *     udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
+ *             port=5000 ! rtpbin.recv_rtp_sink_0                            \
+ *         rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink                \
+ *      udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0                            \
+ *      rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false             \
+ *     udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1"                                             \
+ *             port=5002 ! rtpbin.recv_rtp_sink_1                            \
+ *         rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink                       \
+ *      udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1                            \
+ *      rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false
+ * </programlisting>
+ * Receive H263 on port 5000, send it through rtpbin in session 0, depayload,
+ * decode and display the video.
+ * Receive AMR on port 5002, send it through rtpbin in session 1, depayload,
+ * decode and play the audio.
+ * Receive server RTCP packets for session 0 on port 5001 and RTCP packets for
+ * session 1 on port 5003. These packets will be used for session management and
+ * synchronisation.
+ * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
+ * on port 5007.
+ * </para>
  * </refsect2>
  *
- * Last reviewed on 2007-05-28 (0.10.5)
+ * Last reviewed on 2007-08-28 (0.10.6)
  */
 
 #ifdef HAVE_CONFIG_H
@@ -452,6 +497,7 @@ gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
   GSList *walk;
 
   GST_RTP_BIN_LOCK (bin);
+  GST_DEBUG_OBJECT (bin, "clearing pt map");
   for (walk = bin->sessions; walk; walk = g_slist_next (walk)) {
     GstRtpBinSession *session = (GstRtpBinSession *) walk->data;
 
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index 1b630243..7f5782be 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -583,8 +583,10 @@ rtcp_thread (GstRtpSession * rtpsession)
     GST_DEBUG_OBJECT (rtpsession, "unlocked %d, current %" GST_TIME_FORMAT,
         res, GST_TIME_ARGS (current_time));
 
-    /* perform actions, we ignore result. */
+    /* perform actions, we ignore result. Release lock because it might push. */
+    GST_RTP_SESSION_UNLOCK (rtpsession);
     rtp_session_on_timeout (rtpsession->priv->session, current_time);
+    GST_RTP_SESSION_LOCK (rtpsession);
   }
   GST_RTP_SESSION_UNLOCK (rtpsession);
 
@@ -637,6 +639,7 @@ stop_rtcp_thread (GstRtpSession * rtpsession)
     gst_clock_id_unschedule (rtpsession->priv->id);
   GST_RTP_SESSION_UNLOCK (rtpsession);
 
+  /* FIXME, can deadlock because the thread might be blocked in a push */
   g_thread_join (rtpsession->priv->thread);
 }
 
@@ -753,11 +756,11 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
   rtpsession = GST_RTP_SESSION (user_data);
   priv = rtpsession->priv;
 
-  GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
-
   if (rtpsession->send_rtcp_src) {
+    GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
     result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
   } else {
+    GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
     gst_buffer_unref (buffer);
     result = GST_FLOW_OK;
   }
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index 9ab3b4a0..6fa478c8 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -1659,6 +1659,7 @@ session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
         DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
       } else {
         /* No valid SR received, LSR/DLSR are set to 0 then */
+        GST_DEBUG ("no valid SR received");
         LSR = 0;
         DLSR = 0;
       }
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index 8007d54b..1a989517 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -266,8 +266,8 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
   src->stats.prev_rtptime = src->stats.last_rtptime;
   src->stats.last_rtptime = rtparrival;
 
-  GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %u",
-      rtparrival, rtptime, clock_rate, diff, src->stats.jitter);
+  GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
+      rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
 
   return;
 
@@ -446,6 +446,8 @@ rtp_source_process_bye (RTPSource * src, const gchar * reason)
  * @buffer: an RTP buffer
  *
  * Send an RTP @buffer originating from @src. This will make @src a sender.
+ * This function takes ownership of @buffer and modifies the SSRC in the RTP
+ * packet to that of @src.
  *
  * Returns: a #GstFlowReturn.
  */
@@ -467,9 +469,18 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
   src->stats.packets_sent++;
   src->stats.octets_sent += len;
 
-
   /* push packet */
   if (src->callbacks.push_rtp) {
+    guint32 ssrc;
+
+    ssrc = gst_rtp_buffer_get_ssrc (buffer);
+    if (ssrc != src->ssrc) {
+      GST_DEBUG ("updating SSRC from %u to %u", ssrc, src->ssrc);
+      buffer = gst_buffer_make_writable (buffer);
+
+      gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
+    }
+
     GST_DEBUG ("pushing RTP packet %" G_GUINT64_FORMAT,
         src->stats.packets_sent);
     result = src->callbacks.push_rtp (src, buffer, src->user_data);