rtpsession: Call on-new-ssrc earlier

Right now, we may call on-new-ssrc after we have processed the first
RTP packet. This prevents properly configuring the source as some
property like "probation" are copied internally for use as a
decreasing counter. For this specific property, it prevents the
application from disabling probation on auxiliary sparse stream.

Probation is harmful on sparse streams since the probation algorithm
assume frequent and contiguous RTP packets.

diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index bc952c21315ffa76771714483a9b1ab596adf55e..fefa259d4f22ac250263381682e6475ec5f7d697 100644 (file)
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -2240,6 +2240,9 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
   prevactive = RTP_SOURCE_IS_ACTIVE (source);
   oldrate = source->bitrate;
 
+  if (created)
+    on_new_ssrc (sess, source);
+
   /* let source process the packet */
   result = rtp_source_process_rtp (source, &pinfo);
 
@@ -2252,8 +2255,6 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
   if (oldrate != source->bitrate)
     sess->recalc_bandwidth = TRUE;
 
-  if (created)
-    on_new_ssrc (sess, source);
 
   if (source->validated) {
     gboolean created;

diff --git a/tests/check/elements/rtpsession.c b/tests/check/elements/rtpsession.c
index a00766fb23bde1567dd5d9835e88a84799333000..60bd1679758b27c077d74a714eda52a9f745c703 100644 (file)
--- a/tests/check/elements/rtpsession.c
+++ b/tests/check/elements/rtpsession.c
@@ -1861,6 +1861,32 @@ GST_START_TEST (test_on_sending_nacks)
 
 GST_END_TEST;
 
+static void
+disable_probation_on_new_ssrc (GObject * session, GObject * source)
+{
+  g_object_set (source, "probation", 0, NULL);
+}
+
+GST_START_TEST (test_disable_probation)
+{
+  SessionHarness *h = session_harness_new ();
+
+  g_object_set (h->internal_session, "internal-ssrc", 0xDEADBEEF, NULL);
+  g_signal_connect (h->internal_session, "on-new-ssrc",
+      G_CALLBACK (disable_probation_on_new_ssrc), NULL);
+
+  /* Receive a RTP buffer from the wire */
+  fail_unless_equals_int (GST_FLOW_OK,
+      session_harness_recv_rtp (h, generate_test_buffer (0, 0x12345678)));
+
+  /* When probation is disable, the packet should be produced immediatly */
+  fail_unless_equals_int (1, gst_harness_buffers_in_queue (h->recv_rtp_h));
+
+  session_harness_free (h);
+}
+
+GST_END_TEST;
+
 static Suite *
 rtpsession_suite (void)
 {
@@ -1890,6 +1916,7 @@ rtpsession_suite (void)
   tcase_add_test (tc_chain, test_change_sent_sdes);
   tcase_add_test (tc_chain, test_disable_sr_timestamp);
   tcase_add_test (tc_chain, test_on_sending_nacks);
+  tcase_add_test (tc_chain, test_disable_probation);
   return s;
 }