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rtspsrc: Fix state changes from PAUSED to PLAYING

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This was accidentally broken in the last commit that touched this
because I missed the fall-through in the case immediately above this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/906>
This commit is contained in:
Nirbheek Chauhan 2021-03-16 00:08:43 +05:30
parent 242f3cae6d
commit f37afdafff
1 changed files with 6 additions and 3 deletions
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9
gst/rtsp/gstrtspsrc.c
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@ -9224,9 +9224,12 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
set_manager_buffer_mode (rtspsrc); set_manager_buffer_mode (rtspsrc);
/* fall-through */ /* fall-through */
case GST_STATE_CHANGE_PLAYING_TO_PAUSED: case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
if (rtspsrc->is_live && (GST_STATE_TARGET (element) == GST_STATE_PAUSED)) { if (rtspsrc->is_live &&
/* unblock the tcp tasks only if rtspsrc is going to PAUSED (not if (transition == GST_STATE_CHANGE_PAUSED_TO_PLAYING ||
* it's going to NULL) and make the loop waiting */ GST_STATE_TARGET (element) == GST_STATE_PAUSED)) {
/* Unblock the tcp tasks and make the loop waiting. But if the state
* transition is PLAY->PAUSE, then only do this if the final target
* state is PAUSED */
if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) { if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
/* make sure it is waiting before we send PAUSE or PLAY below */ /* make sure it is waiting before we send PAUSE or PLAY below */
GST_RTSP_STREAM_LOCK (rtspsrc); GST_RTSP_STREAM_LOCK (rtspsrc);