gboolean timed_out;
GstRTSPTransport *transport;
+ GstRTSPUrl *url;
GObject *rtpsource;
};
return trans->priv->transport;
}
+/**
+ * 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_set_active:
* @trans: a #GstRTSPStreamTransport
GstRTSPTransport * tr);
const GstRTSPTransport * gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport *trans);
+void gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport *trans,
+ const GstRTSPUrl * url);
+const GstRTSPUrl * gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport *trans);
+
void gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,
GstRTSPSendFunc send_rtp,
GstRTSPSendFunc send_rtcp,