From: Mike Brady <4265913+mikebrady@users.noreply.github.com>
Date: Tue, 7 Dec 2021 15:09:05 +0000 (+0000)
Subject: stop removing the play lock at the end of a play session. That's the job of the rtsp... 
X-Git-Tag: 4.1-rc1~24^2~364^2~2
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=fd3d461a164bd4afcfeb76b26bdc7140f2570353;p=thirdparty%2Fshairport-sync.git

stop removing the play lock at the end of a play session. That's the job of the rtsp thread.
---

diff --git a/player.c b/player.c
index 6131ad4c..1feee08f 100644
--- a/player.c
+++ b/player.c
@@ -1717,7 +1717,7 @@ void player_thread_cleanup_handler(void *arg) {
     terminate_decoders(conn);
 
   // reset_anchor_info(conn);
-  release_play_lock(conn);
+  // release_play_lock(conn);
   conn->rtp_running = 0;
   pthread_setcancelstate(oldState, NULL);
   debug(2, "Connection %d: player terminated.", conn->connection_number);
diff --git a/rtsp.c b/rtsp.c
index 4b0827bf..959944ac 100644
--- a/rtsp.c
+++ b/rtsp.c
@@ -1879,6 +1879,7 @@ void handle_get(__attribute((unused)) rtsp_conn_info *conn, __attribute((unused)
         req->contentlength);
   resp->respcode = 500;
 }
+
 void handle_post(__attribute((unused)) rtsp_conn_info *conn,
                  __attribute((unused)) rtsp_message *req,
                  __attribute((unused)) rtsp_message *resp) {
@@ -2480,6 +2481,9 @@ void teardown_phase_two(rtsp_conn_info *conn) {
     debug(2, "Connection %d: TEARDOWN mdns_update on %s.", conn->connection_number,
           get_category_string(conn->airplay_stream_category));
     mdns_update(NULL, secondary_txt_records);
+    debug(1, "Connection %d: TEARDOWN (AP2) release play lock on %s.", conn->connection_number,
+          get_category_string(conn->airplay_stream_category));
+    release_play_lock(conn);
   }
 }
 
@@ -2533,6 +2537,9 @@ void handle_teardown(rtsp_conn_info *conn, __attribute__((unused)) rtsp_message
         "TEARDOWN: synchronously terminating the player thread of RTSP conversation thread %d (2).",
         conn->connection_number);
     player_stop(conn);
+    release_play_lock(conn);
+    debug(1, "TEARDOWN: release play lock on RTSP conversation thread %d.",
+          conn->connection_number);
     activity_monitor_signify_activity(0); // inactive, and should be after command_stop()
     debug(3, "TEARDOWN: successful termination of playing thread of RTSP conversation thread %d.",
           conn->connection_number);
@@ -2800,6 +2807,8 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
             debug(2, "Connection %d: SETUP mdns_update on %s.", conn->connection_number,
                   get_category_string(conn->airplay_stream_category));
             mdns_update(NULL, secondary_txt_records);
+            
+            
 
             resp->respcode = 200;
           } else {
@@ -2862,6 +2871,9 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
     debug(2, "Connection %d: SETUP on %s. A \"streams\" array has been found",
           conn->connection_number, get_category_string(conn->airplay_stream_category));
     if (conn->airplay_stream_category == ptp_stream) {
+    debug(1, "Connection %d: SETUP get_play_lock.",
+          conn->connection_number);    
+      get_play_lock(conn);
       // get stream[0]
       plist_t stream0 = plist_array_get_item(streams, 0);
 
@@ -2939,7 +2951,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
       case 96: {
         debug(1, "Connection %d. Realtime Audio Stream Detected.", conn->connection_number);
         debug_log_rtsp_message(2, "Realtime Audio Stream SETUP incoming message", req);
-        get_play_lock(conn);
+        // get_play_lock(conn);
         conn->airplay_stream_type = realtime_stream;
         // bind a new UDP port and get a socket
         conn->local_realtime_audio_port = 0; // any port
@@ -2999,7 +3011,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
         debug(1, "SETUP on Connection %d. Buffered Audio Stream Detected.",
               conn->connection_number);
         debug_log_rtsp_message(2, "Buffered Audio Stream SETUP incoming message", req);
-        get_play_lock(conn);
+        // get_play_lock(conn);
         conn->airplay_stream_type = buffered_stream;
         // get needed stuff
 
@@ -3229,7 +3241,7 @@ void handle_set_parameter_parameter(rtsp_conn_info *conn, rtsp_message *req,
 
     if (!strncmp(cp, "volume: ", strlen("volume: "))) {
       float volume = atof(cp + strlen("volume: "));
-      debug(2, "Connection %d: request to set AirPlay Volume to: %f.", conn->connection_number,
+      debug(1, "Connection %d: request to set AirPlay Volume to: %f.", conn->connection_number,
             volume);
       // if we are playing, go ahead and change the volume
       if (try_to_hold_play_lock(conn) == 0) {
@@ -3969,7 +3981,7 @@ static void handle_get_parameter(__attribute__((unused)) rtsp_conn_info *conn, r
 
   if ((req->content) && (req->contentlength == strlen("volume\r\n")) &&
       strstr(req->content, "volume") == req->content) {
-    debug(2, "Connection %d: Current volume (%.6f) requested", conn->connection_number,
+    debug(1, "Connection %d: Current volume (%.6f) requested", conn->connection_number,
           config.airplay_volume);
     char *p = malloc(128); // will be automatically deallocated with the response is deleted
     if (p) {
@@ -4078,6 +4090,7 @@ static void handle_set_parameter(rtsp_conn_info *conn, rtsp_message *req, rtsp_m
 
 static void handle_announce(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp) {
   debug(3, "Connection %d: ANNOUNCE", conn->connection_number);
+  debug(1, "Connection %d: ANNOUNCE get_play_lock()", conn->connection_number);
   int get_play_status = get_play_lock(conn);
   if (get_play_status != -1) {
     debug(3, "Connection %d: ANNOUNCE has acquired play lock.", conn->connection_number);