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Revert unusable workaround for PAUSE being sent when going NULL

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Directly setting rtspsrc to the NULL state before putting the pipeline
in the NULL state usually works, but it can cause a deadlock in some
cases, so it's not a reliable mechanism to fix this.

This reverts commit f37afdafff1fd0a339966116261f5cd0de53f5d1:
"rtspsrc: Fix state changes from PAUSED to PLAYING"

and commit 76d624b2df5594a82269b94dffe8766a372d059d:
"rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/908>
This commit is contained in:
Nirbheek Chauhan 2021-03-17 11:32:08 +05:30
parent 00e73e1657
commit 55edd59f48
1 changed files with 4 additions and 22 deletions
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26
gst/rtsp/gstrtspsrc.c
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@ -85,19 +85,6 @@
* ]| Establish a connection to an RTSP server and send the raw RTP packets to a
* fakesink.
*
* NOTE: rtspsrc will send a PAUSE command to the server if you set the
* element to the PAUSED state, and will send a PLAY command if you set it to
* the PLAYING state. Sending of the PAUSE command only happens when the
* target state of the element is PAUSED. For instance, it won't happen if you
* call gst_element_set_state() on rtspsrc with %GST_STATE_NULL.
*
* BUT: due to how recursive state changes work, this doesn't apply to states
* set on parent elements, such as bins or the pipeline. Child elements will
* always see intermediate states as the target state. If you want to tear
* down rtspsrc without interrupting other clients when streaming a shared
* media, you should set rtspsrc to %GST_STATE_NULL first, and then the
* pipeline itself.
*
*/
#ifdef HAVE_CONFIG_H
@ -9224,12 +9211,8 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
set_manager_buffer_mode (rtspsrc);
/* fall-through */
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
if (rtspsrc->is_live &&
(transition == GST_STATE_CHANGE_PAUSED_TO_PLAYING ||
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 (rtspsrc->is_live) {
/* unblock the tcp tasks and make the loop waiting */
if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
/* make sure it is waiting before we send PAUSE or PLAY below */
GST_RTSP_STREAM_LOCK (rtspsrc);
@ -9264,9 +9247,8 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
ret = GST_STATE_CHANGE_SUCCESS;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
if (rtspsrc->is_live && (GST_STATE_TARGET (element) == GST_STATE_PAUSED)) {
/* send pause request only if rtspsrc is going to PAUSED (not if it's
* going to NULL) and keep the idle task around */
if (rtspsrc->is_live) {
/* send pause request and keep the idle task around */
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
}
ret = GST_STATE_CHANGE_SUCCESS;