Quieten some debug messages.

ap2_buffered_audio_processor.c

@@ -272,6 +272,10 @@ void *rtp_buffered_audio_processor(void *arg) {
           previous_ssrc = payload_ssrc;
           payload_ssrc = nctohl(&packet[8]);
 
+          if ((payload_ssrc != previous_ssrc) && (ssrc_is_recognised(payload_ssrc) == 0)) {
+              debug(2, "Unrecognised SSRC: %u.", payload_ssrc);
+          }
+
           if (blocks_read_since_play_began == 1) {
             debug(2, "Preparing initial decoding chain for %s.", get_ssrc_name(payload_ssrc));
             prepare_decoding_chain(conn, payload_ssrc); // needed to set the input rate...
@@ -282,19 +286,16 @@ void *rtp_buffered_audio_processor(void *arg) {
 
           if (blocks_read_since_play_began > 1) {
 
+/*
             if (payload_ssrc != previous_ssrc) {
               if (ssrc_is_recognised(payload_ssrc) == 0) {
                 debug(2, "Unrecognised SSRC: %u.", payload_ssrc);
               } else {
-                debug(1, "Connection %d: incoming audio switching to \"%s\".",
+                debug(2, "Connection %d: incoming audio switching to \"%s\".",
                       conn->connection_number, get_ssrc_name(payload_ssrc));
-                debug(2,
-                      "Reading a block: new encoding: %s, old encoding: %s. Preparing a new "
-                      "decoding chain.",
-                      get_ssrc_name(payload_ssrc), get_ssrc_name(previous_ssrc));
-                // prepare_decoding_chain(conn, payload_ssrc);
               }
             }
+*/
 
             uint32_t t_expected_seqno = (previous_seqno + 1) & 0x7fffff;
             if (t_expected_seqno != seq_no) {

ap2_event_receiver.c

@@ -188,7 +188,7 @@ void *ap2_event_receiver(void *arg) {
           debug(3, "Connection %d: Packet Received on Event Port with contents: \"%s\".",
                 conn->connection_number, packet);
         } else {
-          debug(1, "Connection %d: Event Port connection closed by client",
+          debug(2, "Connection %d: Event Port connection closed by client",
                 conn->connection_number);
           finished = 1;
         }

rtsp.c

@@ -409,7 +409,7 @@ void release_play_lock(rtsp_conn_info *conn) {
       build_bonjour_strings(principal_conn);
       mdns_update(NULL, secondary_txt_records);
 #endif
-      debug(1, "Connection %d: %s released principal_conn.", conn->connection_number,
+      debug(2, "Connection %d: %s released principal_conn.", conn->connection_number,
             get_category_string(conn->airplay_stream_category));
     }
     principal_conn = NULL; // let it go
@@ -2548,7 +2548,7 @@ void handle_teardown(rtsp_conn_info *conn, __attribute__((unused)) rtsp_message
   // receiver if and when it exits.
 
   if (conn->player_thread) {
-    debug(1, "TEARDOWN is stopping a player...");
+    debug(2, "TEARDOWN is stopping a player thread before exiting...");
     player_stop(conn);                    // this nulls the player_thread and cancels the threads...
     activity_monitor_signify_activity(0); // inactive, and should be after command_stop()
   }
@@ -2581,6 +2581,7 @@ void handle_teardown_2(rtsp_conn_info *conn, __attribute__((unused)) rtsp_messag
   debug_log_rtsp_message(2, "TEARDOWN 2: ", req);
 
   if (conn->player_thread) {
+    debug(2, "TEARDOWN 2 is stopping a player thread before exiting...");
     player_stop(conn);                    // this nulls the player_thread and cancels the threads...
     activity_monitor_signify_activity(0); // inactive, and should be after command_stop()
   }