mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-27 12:11:13 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state

Browse source 
If we consider the RTSP state, what can happen is that it is PLAYING but the
element already asynchronously tried to PAUSE and it just did not happen yet.

We would then override this setting to PAUSED (while the element actually is
in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
to produce packets while the sinks are all PAUSED, piling up thousands of
packets in the rtpjitterbuffer and other elements and finally failing.
This commit is contained in:
Sebastian Dröge 2016-06-28 10:57:27 +03:00
parent d6f597db20
commit c18b609c06
1 changed files with 7 additions and 3 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

10
gst/rtsp/gstrtspsrc.c
View file

@ -2170,9 +2170,8 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
if ((stop = seeksegment.stop) == -1) if ((stop = seeksegment.stop) == -1)
stop = seeksegment.duration; stop = seeksegment.duration;
playing = (src->state == GST_RTSP_STATE_PLAYING);
/* if we were playing, pause first */ /* if we were playing, pause first */
playing = (src->state == GST_RTSP_STATE_PLAYING);
if (playing) { if (playing) {
/* obtain current position in case seek fails */ /* obtain current position in case seek fails */
gst_rtspsrc_get_position (src); gst_rtspsrc_get_position (src);
@ -2185,7 +2184,12 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
/* PLAY will add the range header now. */ /* PLAY will add the range header now. */
src->need_range = TRUE; src->need_range = TRUE;
/* and continue playing */ /* and continue playing if needed */
GST_OBJECT_LOCK (src);
playing = (GST_STATE_PENDING (src) == GST_STATE_VOID_PENDING
&& GST_STATE (src) == GST_STATE_PLAYING)
|| (GST_STATE_PENDING (src) == GST_STATE_PLAYING);
GST_OBJECT_UNLOCK (src);
if (playing) if (playing)
gst_rtspsrc_play (src, &seeksegment, FALSE); gst_rtspsrc_play (src, &seeksegment, FALSE);