gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.

Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
(async_jitter_queue_pop_intern_unlocked):
Fix the case where the buffer underruns and does not block.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
Rename RTCP send pad, like in the session manager.
Allow getting an RTCP pad for receiving even if we don't receive RTP.
fix handling of send_rtp_src pad.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
When no pt map could be found, fall back to the sinkpad caps.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Fix pad names.
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_create_source), (rtp_session_process_sr),
(rtp_session_send_rtp), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Unlock session when performing a callback.
Add callbacks for the internal session object.
Fix sending of RTP packets.
first attempt at adding NTP times in the SR packets.
Small debug and doc improvements.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Update stats for SR reports.

1 files changed, 13 insertions, 2 deletions

diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index e1e6aaca..7af74671 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -453,18 +453,29 @@ GstFlowReturn
 rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer)
 {
   GstFlowReturn result = GST_FLOW_OK;
+  guint len;
 
   g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
   g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
 
+  len = gst_rtp_buffer_get_payload_len (buffer);
+
   /* we are a sender now */
   src->is_sender = TRUE;
 
+  /* update stats for the SR */
+  src->stats.packets_sent++;
+  src->stats.octets_sent += len;
+
+
   /* push packet */
-  if (src->callbacks.push_rtp)
+  if (src->callbacks.push_rtp) {
+    GST_DEBUG ("pushing RTP packet %u", src->stats.packets_sent);
     result = src->callbacks.push_rtp (src, buffer, src->user_data);
-  else
+  } else {
+    GST_DEBUG ("no callback installed");
     gst_buffer_unref (buffer);
+  }
 
   return result;
 }