session: improve RTP-Info

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

diff --git a/gst/rtsp-server/rtsp-stream-transport.c b/gst/rtsp-server/rtsp-stream-transport.c
index 0a03ad2..4020ee4 100644 (file)
--- a/gst/rtsp-server/rtsp-stream-transport.c
+++ b/gst/rtsp-server/rtsp-stream-transport.c
@@ -16,6 +16,27 @@
  * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, USA.
  */
+/**
+ * SECTION:rtsp-stream-transport
+ * @short_description: A media stream transport configuration
+ * @see_also: #GstRTSPStream, #GstRTSPSessionMedia
+ *
+ * The #GstRTSPStreamTransport configures the transport used by a
+ * #GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object.
+ *
+ * With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
+ * to handle the RTP and RTCP packets from the stream, for example when they
+ * need to be sent over TCP.
+ *
+ * With  gst_rtsp_stream_transport_set_active() the transports are added and
+ * removed from the stream.
+ *
+ * A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
+ * is received from the client. It will also call
+ * gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
 
 #include <string.h>
 #include <stdlib.h>
@@ -41,6 +62,7 @@ struct _GstRTSPStreamTransportPrivate
   gboolean timed_out;
 
   GstRTSPTransport *transport;
+  GstRTSPUrl *url;
 
   GObject *rtpsource;
 };
@@ -99,6 +121,9 @@ gst_rtsp_stream_transport_finalize (GObject * obj)
   if (priv->transport)
     gst_rtsp_transport_free (priv->transport);
 
+  if (priv->url)
+    gst_rtsp_url_free (priv->url);
+
   G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
 }
 
@@ -246,6 +271,102 @@ gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)
 }
 
 /**
+ * gst_rtsp_stream_transport_set_url:
+ * @trans: a #GstRTSPStreamTransport
+ * @url: (transfer none): a client #GstRTSPUrl
+ *
+ * Set @url as the client url.
+ */
+void
+gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
+    const GstRTSPUrl * url)
+{
+  GstRTSPStreamTransportPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+  priv = trans->priv;
+
+  /* keep track of the transports in the stream. */
+  if (priv->url)
+    gst_rtsp_url_free (priv->url);
+  priv->url = (url ? gst_rtsp_url_copy (url) : NULL);
+}
+
+/**
+ * gst_rtsp_stream_transport_get_url:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the url configured in @trans.
+ *
+ * Returns: (transfer none): the url configured in @trans. It remains
+ *     valid for as long as @trans is valid.
+ */
+const GstRTSPUrl *
+gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
+{
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+  return trans->priv->url;
+}
+
+ /**
+ * gst_rtsp_stream_transport_get_rtpinfo:
+ * @trans: a #GstRTSPStreamTransport
+ * @start_time: a star time
+ *
+ * Get the RTP-Info string for @trans and @start_time.
+ *
+ * Returns: the RTPInfo string for @trans and @start_time or %NULL when
+ * the RTP-Info could not be determined. g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport * trans,
+    GstClockTime start_time)
+{
+  GstRTSPStreamTransportPrivate *priv;
+  gchar *url_str;
+  GString *rtpinfo;
+  guint rtptime, seq, clock_rate;
+  GstClockTime running_time = GST_CLOCK_TIME_NONE;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+  priv = trans->priv;
+
+  if (!gst_rtsp_stream_get_rtpinfo (priv->stream, &rtptime, &seq, &clock_rate,
+          &running_time))
+    return NULL;
+
+  GST_DEBUG ("RTP time %u, seq %u, rate %u, running-time %" GST_TIME_FORMAT,
+      rtptime, seq, clock_rate, GST_TIME_ARGS (running_time));
+
+  if (GST_CLOCK_TIME_IS_VALID (running_time)
+      && GST_CLOCK_TIME_IS_VALID (start_time)) {
+    if (running_time > start_time) {
+      rtptime -=
+          gst_util_uint64_scale_int (running_time - start_time, clock_rate,
+          GST_SECOND);
+    } else {
+      rtptime +=
+          gst_util_uint64_scale_int (start_time - running_time, clock_rate,
+          GST_SECOND);
+    }
+  }
+  GST_DEBUG ("RTP time %u, for start-time %" GST_TIME_FORMAT,
+      rtptime, GST_TIME_ARGS (start_time));
+
+  rtpinfo = g_string_new ("");
+
+  url_str = gst_rtsp_url_get_request_uri (trans->priv->url);
+  g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
+      url_str, seq, rtptime);
+  g_free (url_str);
+
+  return g_string_free (rtpinfo, FALSE);
+}
+
+/**
  * gst_rtsp_stream_transport_set_active:
  * @trans: a #GstRTSPStreamTransport
  * @active: new state of @trans