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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-07 15:15:53 +00:00
stream: improve join and leave of the pipeline
simplify code Do the cleanup properly Add some docs
This commit is contained in:
Wim Taymans
parent
693dd3cfc4
commit
6f7d755894
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@ -72,7 +72,6 @@ static gboolean default_handle_message (GstRTSPMedia * media,
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GstMessage * message);
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static void finish_unprepare (GstRTSPMedia * media);
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static gboolean default_unprepare (GstRTSPMedia * media);
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static void unlock_streams (GstRTSPMedia * media);
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static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
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@ -895,22 +894,6 @@ weird_type:
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}
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}
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static void
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unlock_streams (GstRTSPMedia * media)
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{
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guint i;
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/* unlock the udp src elements */
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for (i = 0; i < media->streams->len; i++) {
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GstRTSPStream *stream;
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stream = g_ptr_array_index (media->streams, i);
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gst_element_set_locked_state (stream->udpsrc[0], FALSE);
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gst_element_set_locked_state (stream->udpsrc[1], FALSE);
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}
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}
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static void
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gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
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{
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@ -1075,24 +1058,21 @@ static void
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pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
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{
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GstRTSPStream *stream;
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gint i;
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stream = gst_rtsp_media_create_stream (media, element, pad);
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GST_INFO ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad),
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stream->idx);
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/* we will be adding elements below that will cause ASYNC_DONE to be
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* posted in the bus. We want to ignore those messages until the
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* pipeline really prerolled. */
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media->adding = TRUE;
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gst_rtsp_stream_join_bin (stream, GST_BIN (media->pipeline), media->rtpbin);
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for (i = 0; i < 2; i++) {
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gst_element_set_state (stream->udpsink[i], GST_STATE_PAUSED);
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gst_element_set_state (stream->appsink[i], GST_STATE_PAUSED);
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gst_element_set_state (stream->appqueue[i], GST_STATE_PAUSED);
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gst_element_set_state (stream->tee[i], GST_STATE_PAUSED);
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gst_element_set_state (stream->selector[i], GST_STATE_PAUSED);
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gst_element_set_state (stream->appsrc[i], GST_STATE_PAUSED);
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}
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/* join the element in the PAUSED state because this callback is
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* called from the streaming thread and it is PAUSED */
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gst_rtsp_stream_join_bin (stream, GST_BIN (media->pipeline),
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media->rtpbin, GST_STATE_PAUSED);
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media->adding = FALSE;
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}
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@ -1173,7 +1153,8 @@ gst_rtsp_media_prepare (GstRTSPMedia * media)
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stream = g_ptr_array_index (media->streams, i);
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gst_rtsp_stream_join_bin (stream, GST_BIN (media->pipeline), media->rtpbin);
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gst_rtsp_stream_join_bin (stream, GST_BIN (media->pipeline),
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media->rtpbin, GST_STATE_NULL);
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}
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for (walk = media->dynamic; walk; walk = g_list_next (walk)) {
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@ -1208,7 +1189,7 @@ gst_rtsp_media_prepare (GstRTSPMedia * media)
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/* we need to go to PLAYING */
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GST_INFO ("NO_PREROLL state change: live media %p", media);
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/* FIXME we disable seeking for live streams for now. We should perform a
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* seeking query in preroll instead and do a seeking query. */
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* seeking query in preroll instead */
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media->seekable = FALSE;
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media->is_live = TRUE;
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ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
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@ -1219,7 +1200,8 @@ gst_rtsp_media_prepare (GstRTSPMedia * media)
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goto state_failed;
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}
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/* now wait for all pads to be prerolled */
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/* now wait for all pads to be prerolled, FIXME, we should somehow be
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* able to do this async so that we don't block the server thread. */
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status = gst_rtsp_media_get_status (media);
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if (status == GST_RTSP_MEDIA_STATUS_ERROR)
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goto state_failed;
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@ -1262,7 +1244,6 @@ finish_unprepare (GstRTSPMedia * media)
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GST_DEBUG ("shutting down");
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unlock_streams (media);
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gst_element_set_state (media->pipeline, GST_STATE_NULL);
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for (i = 0; i < media->streams->len; i++) {
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@ -1278,6 +1259,7 @@ finish_unprepare (GstRTSPMedia * media)
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g_ptr_array_set_size (media->streams, 0);
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gst_bin_remove (GST_BIN (media->pipeline), media->rtpbin);
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media->rtpbin = NULL;
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gst_object_unref (media->pipeline);
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media->pipeline = NULL;
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@ -68,28 +68,12 @@ gst_rtsp_stream_finalize (GObject * obj)
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stream = GST_RTSP_STREAM (obj);
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g_assert (!stream->is_joined);
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/* we really need to be unjoined now */
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g_return_if_fail (!stream->is_joined);
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gst_object_unref (stream->payloader);
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gst_object_unref (stream->srcpad);
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if (stream->session)
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g_object_unref (stream->session);
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if (stream->caps)
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gst_caps_unref (stream->caps);
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if (stream->send_rtp_sink)
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gst_object_unref (stream->send_rtp_sink);
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if (stream->send_rtp_src)
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gst_object_unref (stream->send_rtp_src);
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if (stream->send_rtcp_src)
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gst_object_unref (stream->send_rtcp_src);
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if (stream->recv_rtcp_sink)
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gst_object_unref (stream->recv_rtcp_sink);
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if (stream->recv_rtp_sink)
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gst_object_unref (stream->recv_rtp_sink);
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g_list_free (stream->transports);
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G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
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@ -565,16 +549,18 @@ static GstAppSinkCallbacks sink_cb = {
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* @stream: a #GstRTSPStream
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* @bin: a #GstBin to join
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* @rtpbin: a rtpbin element in @bin
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* @state: the target state of the new elements
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*
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* Join the #Gstbin @bin that contains the element @rtpbin.
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*
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* @stream will link to @rtpbin, which must be inside @bin.
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* @stream will link to @rtpbin, which must be inside @bin. The elements
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* added to @bin will be set to the state given in @state.
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*
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* Returns: %TRUE on success.
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*/
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gboolean
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gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
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GstElement * rtpbin)
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GstElement * rtpbin, GstState state)
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{
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gint i, idx;
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gchar *name;
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@ -585,52 +571,42 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
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g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
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g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
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idx = stream->idx;
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if (stream->is_joined)
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return TRUE;
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GST_INFO ("stream %p joining bin", stream);
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/* create a session with the same index as the stream */
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idx = stream->idx;
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GST_INFO ("stream %p joining bin as session %d", stream, idx);
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if (!alloc_ports (stream))
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goto no_ports;
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/* add the ports to the pipeline */
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for (i = 0; i < 2; i++) {
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gst_bin_add (bin, stream->udpsink[i]);
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gst_bin_add (bin, stream->udpsrc[i]);
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}
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/* create elements for the TCP transfer */
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for (i = 0; i < 2; i++) {
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stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
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stream->appqueue[i] = gst_element_factory_make ("queue", NULL);
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stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
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g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
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g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
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gst_bin_add (bin, stream->appqueue[i]);
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gst_bin_add (bin, stream->appsink[i]);
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gst_bin_add (bin, stream->appsrc[i]);
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gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
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&sink_cb, stream, NULL);
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}
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/* hook up the stream to the RTP session elements. */
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/* get a pad for sending RTP */
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name = g_strdup_printf ("send_rtp_sink_%u", idx);
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stream->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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/* link the RTP pad to the session manager, it should not really fail unless
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* this is not really an RTP pad */
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ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
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if (ret != GST_PAD_LINK_OK)
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goto link_failed;
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/* get pads from the RTP session element for sending and receiving
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* RTP/RTCP*/
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name = g_strdup_printf ("send_rtp_src_%u", idx);
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stream->send_rtp_src = gst_element_get_static_pad (rtpbin, name);
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stream->send_src[0] = gst_element_get_static_pad (rtpbin, name);
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g_free (name);
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name = g_strdup_printf ("send_rtcp_src_%u", idx);
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stream->send_rtcp_src = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
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stream->recv_rtcp_sink = gst_element_get_request_pad (rtpbin, name);
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stream->send_src[1] = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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name = g_strdup_printf ("recv_rtp_sink_%u", idx);
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stream->recv_rtp_sink = gst_element_get_request_pad (rtpbin, name);
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stream->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
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stream->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
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g_free (name);
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/* get the session */
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g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &stream->session);
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@ -647,110 +623,119 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
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g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
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stream);
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/* link the RTP pad to the session manager */
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ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
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if (ret != GST_PAD_LINK_OK)
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goto link_failed;
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for (i = 0; i < 2; i++) {
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/* For the sender we create this bit of pipeline for both
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* RTP and RTCP. Sync and preroll are enabled on udpsink so
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* we need to add a queue before appsink to make the pipeline
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* not block. For the TCP case, we want to pump data to the
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* client as fast as possible anyway.
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*
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* .--------. .-----. .---------.
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* | rtpbin | | tee | | udpsink |
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* | send->sink src->sink |
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* '--------' | | '---------'
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* | | .---------. .---------.
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* | | | queue | | appsink |
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* | src->sink src->sink |
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* '-----' '---------' '---------'
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*/
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/* make tee for RTP/RTCP */
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stream->tee[i] = gst_element_factory_make ("tee", NULL);
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gst_bin_add (bin, stream->tee[i]);
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/* make tee for RTP and link to stream */
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stream->tee[0] = gst_element_factory_make ("tee", NULL);
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gst_bin_add (bin, stream->tee[0]);
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/* and link to rtpbin send pad */
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pad = gst_element_get_static_pad (stream->tee[i], "sink");
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gst_pad_link (stream->send_src[i], pad);
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gst_object_unref (pad);
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pad = gst_element_get_static_pad (stream->tee[0], "sink");
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gst_pad_link (stream->send_rtp_src, pad);
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gst_object_unref (pad);
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/* add udpsink */
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gst_bin_add (bin, stream->udpsink[i]);
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/* link RTP sink, we're pretty sure this will work. */
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teepad = gst_element_get_request_pad (stream->tee[0], "src_%u");
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pad = gst_element_get_static_pad (stream->udpsink[0], "sink");
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gst_pad_link (teepad, pad);
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gst_object_unref (pad);
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gst_object_unref (teepad);
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/* link tee to udpsink */
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teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
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pad = gst_element_get_static_pad (stream->udpsink[i], "sink");
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gst_pad_link (teepad, pad);
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gst_object_unref (pad);
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gst_object_unref (teepad);
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teepad = gst_element_get_request_pad (stream->tee[0], "src_%u");
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pad = gst_element_get_static_pad (stream->appqueue[0], "sink");
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gst_pad_link (teepad, pad);
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gst_object_unref (pad);
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gst_object_unref (teepad);
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/* make queue */
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stream->appqueue[i] = gst_element_factory_make ("queue", NULL);
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gst_bin_add (bin, stream->appqueue[i]);
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/* and link to tee */
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teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
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pad = gst_element_get_static_pad (stream->appqueue[i], "sink");
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gst_pad_link (teepad, pad);
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gst_object_unref (pad);
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gst_object_unref (teepad);
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queuepad = gst_element_get_static_pad (stream->appqueue[0], "src");
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pad = gst_element_get_static_pad (stream->appsink[0], "sink");
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gst_pad_link (queuepad, pad);
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gst_object_unref (pad);
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gst_object_unref (queuepad);
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/* make appsink */
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stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
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g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
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g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
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gst_bin_add (bin, stream->appsink[i]);
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gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
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&sink_cb, stream, NULL);
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/* and link to queue */
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queuepad = gst_element_get_static_pad (stream->appqueue[i], "src");
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pad = gst_element_get_static_pad (stream->appsink[i], "sink");
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gst_pad_link (queuepad, pad);
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gst_object_unref (pad);
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gst_object_unref (queuepad);
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/* make tee for RTCP */
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stream->tee[1] = gst_element_factory_make ("tee", NULL);
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gst_bin_add (bin, stream->tee[1]);
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/* For the receiver we create this bit of pipeline for both
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* RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
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* and it is all funneled into the rtpbin receive pad.
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*
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* .--------. .--------. .--------.
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* | udpsrc | | funnel | | rtpbin |
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* | src->sink src->sink |
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* '--------' | | '--------'
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* .--------. | |
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* | appsrc | | |
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* | src->sink |
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* '--------' '--------'
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*/
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/* make funnel for the RTP/RTCP receivers */
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stream->funnel[i] = gst_element_factory_make ("funnel", NULL);
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gst_bin_add (bin, stream->funnel[i]);
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pad = gst_element_get_static_pad (stream->tee[1], "sink");
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gst_pad_link (stream->send_rtcp_src, pad);
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gst_object_unref (pad);
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pad = gst_element_get_static_pad (stream->funnel[i], "src");
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gst_pad_link (pad, stream->recv_sink[i]);
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gst_object_unref (pad);
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/* link RTCP elements */
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teepad = gst_element_get_request_pad (stream->tee[1], "src_%u");
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pad = gst_element_get_static_pad (stream->udpsink[1], "sink");
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gst_pad_link (teepad, pad);
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gst_object_unref (pad);
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gst_object_unref (teepad);
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/* add udpsrc */
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gst_bin_add (bin, stream->udpsrc[i]);
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/* and link to the funnel */
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selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
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pad = gst_element_get_static_pad (stream->udpsrc[i], "src");
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gst_pad_link (pad, selpad);
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gst_object_unref (pad);
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gst_object_unref (selpad);
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teepad = gst_element_get_request_pad (stream->tee[1], "src_%u");
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pad = gst_element_get_static_pad (stream->appqueue[1], "sink");
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gst_pad_link (teepad, pad);
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gst_object_unref (pad);
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gst_object_unref (teepad);
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/* make and add appsrc */
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stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
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gst_bin_add (bin, stream->appsrc[i]);
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/* and link to the funnel */
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selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
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pad = gst_element_get_static_pad (stream->appsrc[i], "src");
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gst_pad_link (pad, selpad);
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gst_object_unref (pad);
|
||||
gst_object_unref (selpad);
|
||||
|
||||
queuepad = gst_element_get_static_pad (stream->appqueue[1], "src");
|
||||
pad = gst_element_get_static_pad (stream->appsink[1], "sink");
|
||||
gst_pad_link (queuepad, pad);
|
||||
gst_object_unref (pad);
|
||||
gst_object_unref (queuepad);
|
||||
|
||||
/* make selector for the RTP receivers */
|
||||
stream->selector[0] = gst_element_factory_make ("funnel", NULL);
|
||||
gst_bin_add (bin, stream->selector[0]);
|
||||
|
||||
pad = gst_element_get_static_pad (stream->selector[0], "src");
|
||||
gst_pad_link (pad, stream->recv_rtp_sink);
|
||||
gst_object_unref (pad);
|
||||
|
||||
selpad = gst_element_get_request_pad (stream->selector[0], "sink_%u");
|
||||
pad = gst_element_get_static_pad (stream->udpsrc[0], "src");
|
||||
gst_pad_link (pad, selpad);
|
||||
gst_object_unref (pad);
|
||||
gst_object_unref (selpad);
|
||||
selpad = gst_element_get_request_pad (stream->selector[0], "sink_%u");
|
||||
pad = gst_element_get_static_pad (stream->appsrc[0], "src");
|
||||
gst_pad_link (pad, selpad);
|
||||
gst_object_unref (pad);
|
||||
gst_object_unref (selpad);
|
||||
|
||||
/* make selector for the RTCP receivers */
|
||||
stream->selector[1] = gst_element_factory_make ("funnel", NULL);
|
||||
gst_bin_add (bin, stream->selector[1]);
|
||||
|
||||
pad = gst_element_get_static_pad (stream->selector[1], "src");
|
||||
gst_pad_link (pad, stream->recv_rtcp_sink);
|
||||
gst_object_unref (pad);
|
||||
|
||||
selpad = gst_element_get_request_pad (stream->selector[1], "sink_%u");
|
||||
pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
|
||||
gst_pad_link (pad, selpad);
|
||||
gst_object_unref (pad);
|
||||
gst_object_unref (selpad);
|
||||
|
||||
selpad = gst_element_get_request_pad (stream->selector[1], "sink_%u");
|
||||
pad = gst_element_get_static_pad (stream->appsrc[1], "src");
|
||||
gst_pad_link (pad, selpad);
|
||||
gst_object_unref (pad);
|
||||
gst_object_unref (selpad);
|
||||
|
||||
/* we set and keep these to playing so that they don't cause NO_PREROLL return
|
||||
* values */
|
||||
gst_element_set_state (stream->udpsrc[0], GST_STATE_PLAYING);
|
||||
gst_element_set_state (stream->udpsrc[1], GST_STATE_PLAYING);
|
||||
gst_element_set_locked_state (stream->udpsrc[0], TRUE);
|
||||
gst_element_set_locked_state (stream->udpsrc[1], TRUE);
|
||||
/* check if we need to set to a special state */
|
||||
if (state != GST_STATE_NULL) {
|
||||
gst_element_set_state (stream->udpsink[i], state);
|
||||
gst_element_set_state (stream->appsink[i], state);
|
||||
gst_element_set_state (stream->appqueue[i], state);
|
||||
gst_element_set_state (stream->tee[i], state);
|
||||
gst_element_set_state (stream->funnel[i], state);
|
||||
gst_element_set_state (stream->appsrc[i], state);
|
||||
}
|
||||
/* we set and keep these to playing so that they don't cause NO_PREROLL return
|
||||
* values */
|
||||
gst_element_set_state (stream->udpsrc[i], GST_STATE_PLAYING);
|
||||
gst_element_set_locked_state (stream->udpsrc[i], TRUE);
|
||||
}
|
||||
|
||||
/* be notified of caps changes */
|
||||
stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
|
||||
|
|
@ -769,6 +754,8 @@ no_ports:
|
|||
link_failed:
|
||||
{
|
||||
GST_WARNING ("failed to link stream %d", idx);
|
||||
gst_object_unref (stream->send_rtp_sink);
|
||||
stream->send_rtp_sink = NULL;
|
||||
return FALSE;
|
||||
}
|
||||
}
|
||||
|
|
@ -779,7 +766,8 @@ link_failed:
|
|||
* @bin: a #GstBin
|
||||
* @rtpbin: a rtpbin #GstElement
|
||||
*
|
||||
* Remove the elements of @stream from the bin
|
||||
* Remove the elements of @stream from @bin. @bin must be set
|
||||
* to the NULL state before calling this.
|
||||
*
|
||||
* Return: %TRUE on success.
|
||||
*/
|
||||
|
|
@ -796,22 +784,55 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
|
|||
if (!stream->is_joined)
|
||||
return TRUE;
|
||||
|
||||
/* all transports must be removed by now */
|
||||
g_return_val_if_fail (stream->transports == NULL, FALSE);
|
||||
|
||||
GST_INFO ("stream %p leaving bin", stream);
|
||||
|
||||
gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
|
||||
|
||||
g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
|
||||
gst_element_release_request_pad (rtpbin, stream->send_rtp_sink);
|
||||
gst_object_unref (stream->send_rtp_sink);
|
||||
stream->send_rtp_sink = NULL;
|
||||
|
||||
/* FIXME not entirely the opposite of join_bin */
|
||||
for (i = 0; i < 2; i++) {
|
||||
/* and set udpsrc to NULL now before removing */
|
||||
gst_element_set_locked_state (stream->udpsrc[i], FALSE);
|
||||
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
|
||||
|
||||
/* removing them should also nicely release the request
|
||||
* pads when they finalize */
|
||||
gst_bin_remove (bin, stream->udpsrc[i]);
|
||||
gst_bin_remove (bin, stream->udpsink[i]);
|
||||
gst_bin_remove (bin, stream->appsrc[i]);
|
||||
gst_bin_remove (bin, stream->appsink[i]);
|
||||
gst_bin_remove (bin, stream->appqueue[i]);
|
||||
gst_bin_remove (bin, stream->tee[i]);
|
||||
gst_bin_remove (bin, stream->selector[i]);
|
||||
gst_bin_remove (bin, stream->funnel[i]);
|
||||
|
||||
gst_element_release_request_pad (rtpbin, stream->recv_sink[i]);
|
||||
gst_object_unref (stream->recv_sink[i]);
|
||||
stream->recv_sink[i] = NULL;
|
||||
|
||||
stream->udpsrc[i] = NULL;
|
||||
stream->udpsink[i] = NULL;
|
||||
stream->appsrc[i] = NULL;
|
||||
stream->appsink[i] = NULL;
|
||||
stream->appqueue[i] = NULL;
|
||||
stream->tee[i] = NULL;
|
||||
stream->funnel[i] = NULL;
|
||||
}
|
||||
gst_object_unref (stream->send_src[0]);
|
||||
stream->send_src[0] = NULL;
|
||||
|
||||
gst_element_release_request_pad (rtpbin, stream->send_src[1]);
|
||||
gst_object_unref (stream->send_src[1]);
|
||||
stream->send_src[1] = NULL;
|
||||
|
||||
g_object_unref (stream->session);
|
||||
if (stream->caps)
|
||||
gst_caps_unref (stream->caps);
|
||||
|
||||
stream->is_joined = FALSE;
|
||||
|
||||
return TRUE;
|
||||
|
|
@ -862,6 +883,10 @@ gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
|
|||
{
|
||||
GstFlowReturn ret;
|
||||
|
||||
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
|
||||
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
||||
g_return_val_if_fail (stream->is_joined, FALSE);
|
||||
|
||||
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer);
|
||||
|
||||
return ret;
|
||||
|
|
@ -884,6 +909,10 @@ gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
|
|||
{
|
||||
GstFlowReturn ret;
|
||||
|
||||
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
|
||||
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
||||
g_return_val_if_fail (stream->is_joined, FALSE);
|
||||
|
||||
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer);
|
||||
|
||||
return ret;
|
||||
|
|
@ -969,6 +998,8 @@ update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
|
|||
* Add the transport in @trans to @stream. The media of @stream will
|
||||
* then also be send to the values configured in @trans.
|
||||
*
|
||||
* @stream must be joined to a bin.
|
||||
*
|
||||
* @trans must contain a valid #GstRTSPTransport.
|
||||
*
|
||||
* Returns: %TRUE if @trans was added
|
||||
|
|
@ -979,6 +1010,7 @@ gst_rtsp_stream_add_transport (GstRTSPStream * stream,
|
|||
{
|
||||
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
||||
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
|
||||
g_return_val_if_fail (stream->is_joined, FALSE);
|
||||
g_return_val_if_fail (trans->transport != NULL, FALSE);
|
||||
|
||||
return update_transport (stream, trans, TRUE);
|
||||
|
|
@ -992,6 +1024,10 @@ gst_rtsp_stream_add_transport (GstRTSPStream * stream,
|
|||
* Remove the transport in @trans from @stream. The media of @stream will
|
||||
* not be sent to the values configured in @trans.
|
||||
*
|
||||
* @stream must be joined to a bin.
|
||||
*
|
||||
* @trans must contain a valid #GstRTSPTransport.
|
||||
*
|
||||
* Returns: %TRUE if @trans was removed
|
||||
*/
|
||||
gboolean
|
||||
|
|
@ -1000,6 +1036,7 @@ gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
|
|||
{
|
||||
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
|
||||
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
|
||||
g_return_val_if_fail (stream->is_joined, FALSE);
|
||||
g_return_val_if_fail (trans->transport != NULL, FALSE);
|
||||
|
||||
return update_transport (stream, trans, FALSE);
|
||||
|
|
|
|||
|
|
@ -50,21 +50,24 @@ typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
|
|||
* @is_ipv6: should this stream be IPv6
|
||||
* @buffer_size: the UDP buffer size
|
||||
* @is_joined: if the stream is joined in a bin
|
||||
* @recv_rtp_sink: sinkpad for RTP buffers
|
||||
* @recv_rtcp_sink: sinkpad for RTCP buffers
|
||||
* @send_rtp_src: srcpad for RTP buffers
|
||||
* @send_rtcp_src: srcpad for RTCP buffers
|
||||
* @send_rtp_sink: sinkpad for sending RTP buffers
|
||||
* @recv_sink: sinkpad for receiving RTP/RTCP buffers
|
||||
* @send_src: srcpad for sending RTP/RTCP buffers
|
||||
* @session: the RTP session object
|
||||
* @udpsrc: the udp source elements for RTP/RTCP
|
||||
* @udpsink: the udp sink elements for RTP/RTCP
|
||||
* @appsrc: the app source elements for RTP/RTCP
|
||||
* @appqueue: the app queue elements for RTP/RTCP
|
||||
* @appsink: the app sink elements for RTP/RTCP
|
||||
* @tee: tee for the sending to udpsink and appsink
|
||||
* @funnel: tee for the receiving from udpsrc and appsrc
|
||||
* @server_port: the server ports for this stream
|
||||
* @caps_sig: the signal id for detecting caps
|
||||
* @caps: the caps of the stream
|
||||
* @n_active: the number of active transports in @transports
|
||||
* @transports: list of #GstStreamTransport being streamed to
|
||||
*
|
||||
* The definition of a media stream. The streams are identified by @id.
|
||||
* The definition of a media stream. The streams are identified by @idx.
|
||||
*/
|
||||
struct _GstRTSPStream {
|
||||
GObject parent;
|
||||
|
|
@ -77,11 +80,9 @@ struct _GstRTSPStream {
|
|||
gboolean is_joined;
|
||||
|
||||
/* pads on the rtpbin */
|
||||
GstPad *recv_rtcp_sink;
|
||||
GstPad *recv_rtp_sink;
|
||||
GstPad *send_rtp_sink;
|
||||
GstPad *send_rtp_src;
|
||||
GstPad *send_rtcp_src;
|
||||
GstPad *recv_sink[2];
|
||||
GstPad *send_src[2];
|
||||
|
||||
/* the RTPSession object */
|
||||
GObject *session;
|
||||
|
|
@ -96,7 +97,7 @@ struct _GstRTSPStream {
|
|||
GstElement *appsink[2];
|
||||
|
||||
GstElement *tee[2];
|
||||
GstElement *selector[2];
|
||||
GstElement *funnel[2];
|
||||
|
||||
/* server ports for sending/receiving */
|
||||
GstRTSPRange server_port;
|
||||
|
|
@ -123,7 +124,8 @@ void gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guin
|
|||
guint gst_rtsp_stream_get_mtu (GstRTSPStream * stream);
|
||||
|
||||
gboolean gst_rtsp_stream_join_bin (GstRTSPStream * stream,
|
||||
GstBin *bin, GstElement *rtpbin);
|
||||
GstBin *bin, GstElement *rtpbin,
|
||||
GstState state);
|
||||
gboolean gst_rtsp_stream_leave_bin (GstRTSPStream * stream,
|
||||
GstBin *bin, GstElement *rtpbin);
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue