rtspsrc: Do not send PAUSE command when going to GST_STATE_NULL

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>

diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 0f4f9aa..3d3cc38 100644 (file)
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -85,6 +85,19 @@
  * ]| 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
@@ -9211,8 +9224,9 @@ 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) {
-        /* unblock the tcp tasks and make the loop waiting */
+      if (rtspsrc->is_live && (GST_STATE_TARGET (element) == GST_STATE_PAUSED)) {
+        /* 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)) {
           /* make sure it is waiting before we send PAUSE or PLAY below */
           GST_RTSP_STREAM_LOCK (rtspsrc);
@@ -9247,8 +9261,9 @@ 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) {
-        /* send pause request and keep the idle task around */
+      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 */
         gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
       }
       ret = GST_STATE_CHANGE_SUCCESS;