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rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL

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This usually doesn't matter, but it is disruptive when streaming from
a shared media since it will pause all other clients when any client
exits.

This new behaviour is opt-in and should be safe because you need to
set the NULL state on rtspsrc directly, instead of just on the
pipeline. See the updated documentation for an explanation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/901>
This commit is contained in:
Nirbheek Chauhan 2021-03-15 12:57:19 +05:30
parent 78dec1e403
commit 76d624b2df
1 changed files with 19 additions and 4 deletions
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23
gst/rtsp/gstrtspsrc.c
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@ -85,6 +85,19 @@
* ]| Establish a connection to an RTSP server and send the raw RTP packets to a * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
* fakesink. * 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 #ifdef HAVE_CONFIG_H
@ -9211,8 +9224,9 @@ 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) { if (rtspsrc->is_live && (GST_STATE_TARGET (element) == GST_STATE_PAUSED)) {
/* unblock the tcp tasks and make the loop waiting */ /* unblock the tcp tasks only if rtspsrc is going to PAUSED (not if
* it's going to NULL) and make the loop waiting */
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);
@ -9247,8 +9261,9 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
ret = GST_STATE_CHANGE_SUCCESS; ret = GST_STATE_CHANGE_SUCCESS;
break; break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED: case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
if (rtspsrc->is_live) { if (rtspsrc->is_live && (GST_STATE_TARGET (element) == GST_STATE_PAUSED)) {
/* send pause request and keep the idle task around */ /* send pause request only if rtspsrc is going to PAUSED (not if it's
* going to NULL) and keep the idle task around */
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP); gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
} }
ret = GST_STATE_CHANGE_SUCCESS; ret = GST_STATE_CHANGE_SUCCESS;