rtsp-stream: Always unref return value of gst_object_get_parent()

[platform/upstream/gstreamer.git] / gst / rtsp-server / rtsp-stream.c

diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
index dfbd4a3..60442ad 100644 (file)
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -1389,8 +1389,8 @@ gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
 }
 
 /**
- * gst_rtsp_media_get_retransmission_time:
- * @media: a #GstRTSPMedia
+ * gst_rtsp_stream_get_retransmission_time:
+ * @stream: a #GstRTSPStream
  *
  * Get the amount of time to store retransmission data.
  *
@@ -1410,6 +1410,13 @@ gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
   return ret;
 }
 
+/**
+ * gst_rtsp_stream_set_retransmission_pt:
+ * @stream: a #GstRTSPStream
+ * @rtx_pt: a #guint
+ *
+ * Set the payload type (pt) for retransmission of this stream.
+ */
 void
 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
 {
@@ -1431,6 +1438,14 @@ gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
   g_mutex_unlock (&stream->priv->lock);
 }
 
+/**
+ * gst_rtsp_stream_get_retransmission_pt:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the payload-type used for retransmission of this stream
+ *
+ * Returns: The retransmission PT.
+ */
 guint
 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
 {
@@ -1445,6 +1460,46 @@ gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
   return rtx_pt;
 }
 
+/**
+ * gst_rtsp_stream_set_buffer_size:
+ * @stream: a #GstRTSPStream
+ * @size: the buffer size
+ *
+ * Set the size of the UDP transmission buffer (in bytes)
+ * Needs to be set before the stream is joined to a bin.
+ *
+ * Since: 1.6
+ */
+void
+gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
+{
+  g_mutex_lock (&stream->priv->lock);
+  stream->priv->buffer_size = size;
+  g_mutex_unlock (&stream->priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_buffer_size:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the size of the UDP transmission buffer (in bytes)
+ *
+ * Returns: the size of the UDP TX buffer
+ *
+ * Since: 1.6
+ */
+guint
+gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
+{
+  guint buffer_size;
+
+  g_mutex_lock (&stream->priv->lock);
+  buffer_size = stream->priv->buffer_size;
+  g_mutex_unlock (&stream->priv->lock);
+
+  return buffer_size;
+}
+
 /* executed from streaming thread */
 static void
 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
@@ -1628,6 +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) {
@@ -1817,8 +1904,19 @@ request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
   return gst_object_ref (priv->srtpdec);
 }
 
-static GstElement *
-request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPStream * stream)
+/**
+ * gst_rtsp_stream_request_aux_sender:
+ * @stream: a #GstRTSPStream
+ * @sessid: the session id
+ *
+ * Creating a rtxsend bin
+ *
+ * Returns: (transfer full): a #GstElement.
+ *
+ * Since: 1.6
+ */
+GstElement *
+gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
 {
   GstElement *bin;
   GstPad *pad;
@@ -1827,6 +1925,8 @@ request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPStream * stream)
   guint pt, rtx_pt;
   gchar *pt_s;
 
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
   pt = gst_rtsp_stream_get_pt (stream);
   pt_s = g_strdup_printf ("%u", pt);
   rtx_pt = stream->priv->rtx_pt;
@@ -2012,11 +2112,6 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
         (GCallback) request_rtp_rtcp_decoder, stream);
   }
 
-  if (priv->rtx_time > 0 && priv->srcpad) {
-    /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
-    g_signal_connect (rtpbin, "request-aux-sender",
-        (GCallback) request_aux_sender, stream);
-  }
   if (priv->sinkpad) {
     g_signal_connect (rtpbin, "request-pt-map",
         (GCallback) request_pt_map, stream);
@@ -2072,6 +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
@@ -2505,7 +2606,8 @@ gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
         if (clock_rate) {
           GstStructure *s = gst_caps_get_structure (caps, 0);
 
-          gst_structure_get_uint (s, "clock-rate", clock_rate);
+          gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
+
           if (*clock_rate == 0 && running_time)
             *running_time = GST_CLOCK_TIME_NONE;
         }
@@ -2779,7 +2881,6 @@ update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
           gst_object_unref (pad);
           gst_object_unref (selpad);
         }
-        gst_object_unref (bin);
 
         priv->transport_sources =
             g_list_prepend (priv->transport_sources, source);
@@ -2815,6 +2916,8 @@ update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
         }
       }
 
+      gst_object_unref (bin);
+
       /* fall through for the generic case */
     }
     case GST_RTSP_LOWER_TRANS_UDP: