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-rw-r--r--examples/test-mp4.c22
-rw-r--r--gst/rtsp-server/rtsp-stream.c38
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