}
/**
- * 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.
*
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)
{
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)
{
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)
}
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) {
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
gst_object_unref (pad);
gst_object_unref (selpad);
}
- gst_object_unref (bin);
priv->transport_sources =
g_list_prepend (priv->transport_sources, source);
}
}
+ gst_object_unref (bin);
+
/* fall through for the generic case */
}
case GST_RTSP_LOWER_TRANS_UDP: