Revert unusable workaround for PAUSE being sent when going NULL

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>

diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index bce6b0d..0f4f9aa 100644 (file)
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -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;