stream: listen to sender ssrc signals

https://bugzilla.gnome.org/show_bug.cgi?id=746747

2 files changed, 60 insertions, 0 deletions

diff --git a/examples/test-mp4.c b/examples/test-mp4.c
index 34b8906..2ebe98c 100644
--- a/examples/test-mp4.c
+++ b/examples/test-mp4.c
@@ -51,6 +51,26 @@ on_ssrc_active (GObject * session, GObject * source, GstRTSPMedia * media)
   }
 }
 
+static void
+on_sender_ssrc_active (GObject * session, GObject * source,
+    GstRTSPMedia * media)
+{
+  GstStructure *stats;
+
+  GST_INFO ("source %p in session %p is active", source, session);
+
+  g_object_get (source, "stats", &stats, NULL);
+  if (stats) {
+    gchar *sstr;
+
+    sstr = gst_structure_to_string (stats);
+    g_print ("Sender stats:\nstructure: %s\n", sstr);
+    g_free (sstr);
+
+    gst_structure_free (stats);
+  }
+}
+
 /* signal callback when the media is prepared for streaming. We can get the
  * session manager for each of the streams and connect to some signals. */
 static void
@@ -75,6 +95,8 @@ media_prepared_cb (GstRTSPMedia * media)
 
     g_signal_connect (session, "on-ssrc-active",
         (GCallback) on_ssrc_active, media);
+    g_signal_connect (session, "on-sender-ssrc-active",
+        (GCallback) on_sender_ssrc_active, media);
   }
 }
 
diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
index 4417c43..4e5f644 100644
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -1683,6 +1683,38 @@ on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
 }
 
 static void
+on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GST_INFO ("%p: new sender source %p", stream, source);
+#ifndef DUMP_STATS
+  {
+    GstStructure *stats;
+    g_object_get (source, "stats", &stats, NULL);
+    if (stats) {
+      dump_structure (stats);
+      gst_structure_free (stats);
+    }
+  }
+#endif
+}
+
+static void
+on_sender_ssrc_active (GObject * session, GObject * source,
+    GstRTSPStream * stream)
+{
+#ifndef DUMP_STATS
+  {
+    GstStructure *stats;
+    g_object_get (source, "stats", &stats, NULL);
+    if (stats) {
+      dump_structure (stats);
+      gst_structure_free (stats);
+    }
+  }
+#endif
+}
+
+static void
 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
 {
   if (is_rtp) {
@@ -2135,6 +2167,12 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
   g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
       stream);
 
+  /* signal for sender ssrc */
+  g_signal_connect (priv->session, "on-new-sender-ssrc",
+      (GCallback) on_new_sender_ssrc, stream);
+  g_signal_connect (priv->session, "on-sender-ssrc-active",
+      (GCallback) on_sender_ssrc_active, stream);
+
   for (i = 0; i < 2; i++) {
     GstPad *teepad, *queuepad;
     /* For the sender we create this bit of pipeline for both