stream-transport: add method to handle RTP/RTCP

Call new methods instead of poking into the structures directly.

diff --git a/gst/rtsp-server/rtsp-stream-transport.c b/gst/rtsp-server/rtsp-stream-transport.c
index 3db5c39..3d97db6 100644 (file)
--- a/gst/rtsp-server/rtsp-stream-transport.c
+++ b/gst/rtsp-server/rtsp-stream-transport.c
@@ -170,3 +170,49 @@ gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
     gst_rtsp_transport_free (trans->transport);
   trans->transport = tr;
 }
+
+/**
+ * gst_rtsp_stream_transport_send_rtp:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer: a #GstBuffer
+ *
+ * Send @buffer to the installed RTP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
+    GstBuffer * buffer)
+{
+  gboolean res = FALSE;
+
+  if (trans->send_rtp)
+    res =
+        trans->send_rtp (buffer, trans->transport->interleaved.min,
+        trans->user_data);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_send_rtcp:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer: a #GstBuffer
+ *
+ * Send @buffer to the installed RTCP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
+    GstBuffer * buffer)
+{
+  gboolean res = FALSE;
+
+  if (trans->send_rtcp)
+    res =
+        trans->send_rtcp (buffer, trans->transport->interleaved.max,
+        trans->user_data);
+
+  return res;
+}

diff --git a/gst/rtsp-server/rtsp-stream-transport.h b/gst/rtsp-server/rtsp-stream-transport.h
index cbc8255..f478805 100644 (file)
--- a/gst/rtsp-server/rtsp-stream-transport.h
+++ b/gst/rtsp-server/rtsp-stream-transport.h
@@ -105,6 +105,11 @@ void                     gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamT
                                                                   gpointer user_data,
                                                                   GDestroyNotify  notify);
 
+gboolean                 gst_rtsp_stream_transport_send_rtp      (GstRTSPStreamTransport *trans,
+                                                                  GstBuffer *buffer);
+gboolean                 gst_rtsp_stream_transport_send_rtcp     (GstRTSPStreamTransport *trans,
+                                                                  GstBuffer *buffer);
+
 G_END_DECLS
 
 #endif /* __GST_RTSP_STREAM_TRANSPORT_H__ */

diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
index be262d1..f9aba19 100644 (file)
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -524,11 +524,9 @@ handle_new_sample (GstAppSink * sink, gpointer user_data)
     GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
 
     if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
-      if (tr->send_rtp)
-        tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data);
+      gst_rtsp_stream_transport_send_rtp (tr, buffer);
     } else {
-      if (tr->send_rtcp)
-        tr->send_rtcp (buffer, tr->transport->interleaved.max, tr->user_data);
+      gst_rtsp_stream_transport_send_rtcp (tr, buffer);
     }
   }
   gst_sample_unref (sample);