From: Mike Brady <4265913+mikebrady@users.noreply.github.com>
Date: Thu, 4 Dec 2025 11:35:00 +0000 (+0000)
Subject: Balking when connecting to HPM.
X-Git-Tag: 5.0-post-dev~46
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=0e274eb7603796dc4e9a2abf3028c292dae78005;p=thirdparty%2Fshairport-sync.git

Balking when connecting to HPM.
---

diff --git a/ap2_buffered_audio_processor.c b/ap2_buffered_audio_processor.c
index b09000c1..0ef4beef 100644
--- a/ap2_buffered_audio_processor.c
+++ b/ap2_buffered_audio_processor.c
@@ -99,10 +99,9 @@ void rtp_buffered_audio_cleanup_handler(__attribute__((unused)) void *arg) {
 }
 
 void *rtp_buffered_audio_processor(void *arg) {
-  //  #include <syscall.h>
-  //  debug(1, "rtp_buffered_audio_processor PID %d", syscall(SYS_gettid));
   rtsp_conn_info *conn = (rtsp_conn_info *)arg;
-
+#include <syscall.h>
+  debug(1, "Connection %d: rtp_buffered_audio_processor PID %d start", conn->connection_number, syscall(SYS_gettid));
   conn->incoming_ssrc = 0; // reset
   conn->resampler_ssrc = 0;
 
@@ -126,7 +125,8 @@ void *rtp_buffered_audio_processor(void *arg) {
   buffered_tcp_desc *buffered_audio = malloc(sizeof(buffered_tcp_desc));
   if (buffered_audio == NULL)
     debug(1, "cannot allocate a buffered_tcp_desc!");
-  // initialise the descriptor
+  // initialise the 
+  
   memset(buffered_audio, 0, sizeof(buffered_tcp_desc));
   pthread_cleanup_push(malloc_cleanup, &buffered_audio);
 
@@ -314,7 +314,7 @@ void *rtp_buffered_audio_processor(void *arg) {
 
       if (nread == 0) {
         // nread is 0 -- the port has been closed
-        debug(1, "buffered audio port closed!");
+        debug(1, "Connection %d: buffered audio port closed!", conn->connection_number);
         finished = 1;
       } else if (nread < 0) {
         char errorstring[1024];
@@ -516,7 +516,7 @@ void *rtp_buffered_audio_processor(void *arg) {
                   } else {
                     timestamp_difference = timestamp - expected_timestamp;
                     if (timestamp_difference != 0) {
-                      debug(2,
+                      debug(1,
                             "Connection %d: "
                             "unexpected timestamp in block %u. Actual: %u, expected: %u "
                             "difference: %d, "
@@ -541,7 +541,7 @@ void *rtp_buffered_audio_processor(void *arg) {
                     int32_t abs_timestamp_difference = -timestamp_difference;
                     if ((size_t)abs_timestamp_difference > get_ssrc_block_length(payload_ssrc)) {
                       skip_this_block = 1;
-                      debug(2,
+                      debug(1,
                             "skipping block %u because it was too far in the past. Timestamp "
                             "difference: %d, length of block: %u.",
                             seq_no, timestamp_difference, get_ssrc_block_length(payload_ssrc));
@@ -587,7 +587,7 @@ void *rtp_buffered_audio_processor(void *arg) {
       }
     }
   } while (finished == 0);
-  debug(2, "Buffered Audio Receiver RTP thread \"normal\" exit.");
+  debug(1, "Connection %d: rtp_buffered_audio_processor PID %d exiting", conn->connection_number, syscall(SYS_gettid));
   pthread_cleanup_pop(1); // buffered_tcp_reader thread creation
   pthread_cleanup_pop(1); // buffer malloc
   pthread_cleanup_pop(1); // not_full_cv
@@ -596,5 +596,6 @@ void *rtp_buffered_audio_processor(void *arg) {
   pthread_cleanup_pop(1); // descriptor malloc
   pthread_cleanup_pop(1); // pthread_t malloc
   pthread_cleanup_pop(1); // do the cleanup.
+  debug(1, "Connection %d: rtp_buffered_audio_processor PID %d finish", conn->connection_number, syscall(SYS_gettid));
   pthread_exit(NULL);
 }
diff --git a/player.c b/player.c
index 18f1e12c..1b76331b 100644
--- a/player.c
+++ b/player.c
@@ -3322,12 +3322,14 @@ void player_thread_cleanup_handler(void *arg) {
 
     } else if (conn->airplay_stream_type == buffered_stream) {
 
-      debug(2,
+      debug(1,
             "Connection %d: Delete Buffered Audio Stream thread by player_thread_cleanup_handler",
             conn->connection_number);
       pthread_cancel(conn->rtp_buffered_audio_thread);
       pthread_join(conn->rtp_buffered_audio_thread, NULL);
-
+      debug(1,
+            "Connection %d: Deleted Buffered Audio Stream thread by player_thread_cleanup_handler",
+            conn->connection_number);
     } else {
       die("Unrecognised Stream Type");
     }
diff --git a/rtsp.c b/rtsp.c
index 742d80de..ae169ecb 100644
--- a/rtsp.c
+++ b/rtsp.c
@@ -430,10 +430,10 @@ int get_play_lock(rtsp_conn_info *conn, int allow_session_interruption) {
 
     if (principal_conn->fd > 0) {
       debug(1,
-            "Connection %d: get_play_lock forced termination. Closing RTSP connection socket %d: "
+            "Connection %d: get_play_lock forced termination in favour of connection %d. Closing RTSP connection socket %d: "
             "from %s:%u to self at "
             "%s:%u.",
-            principal_conn->connection_number, principal_conn->fd, principal_conn->client_ip_string,
+            principal_conn->connection_number, conn->connection_number, principal_conn->fd, principal_conn->client_ip_string,
             principal_conn->client_rtsp_port, principal_conn->self_ip_string,
             principal_conn->self_rtsp_port);
       close(principal_conn->fd);
@@ -2995,7 +2995,7 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
               // debug(1,"initial timing peer command: \"%s\".", timing_list_message);
               // ptp_send_control_message_string(timing_list_message);
               set_client_as_ptp_clock(conn);
-              // ptp_send_control_message_string("B"); // signify clock dependability period is
+              ptp_send_control_message_string("B"); // signify clock dependability period is
               // "B"eginning (or continuing)
               plist_dict_set_item(timingPeerInfoPlist, "Addresses", addresses);
               plist_dict_set_item(timingPeerInfoPlist, "ID",
@@ -3294,9 +3294,14 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
 
         activity_monitor_signify_activity(1);
 
+        debug(1, "Connection %d: create rtp_buffered_audio_thread", conn->connection_number);
+        
         named_pthread_create_with_priority(&conn->rtp_buffered_audio_thread, 2,
                                            &rtp_buffered_audio_processor, (void *)conn,
                                            "ap2_bat_%d", conn->connection_number);
+                                           
+                                           
+        usleep(1000000);
 
         plist_dict_set_item(stream0dict, "type", plist_new_uint(103));
         plist_dict_set_item(stream0dict, "dataPort",
diff --git a/utilities/buffered_read.c b/utilities/buffered_read.c
index 0ff7faff..faaf8b62 100644
--- a/utilities/buffered_read.c
+++ b/utilities/buffered_read.c
@@ -92,7 +92,7 @@ ssize_t buffered_read(buffered_tcp_desc *descriptor, void *buf, size_t count,
 #define STANDARD_PACKET_SIZE 4096
 
 void buffered_tcp_reader_cleanup_handler(__attribute__((unused)) void *arg) {
-  debug(2, "Buffered TCP Reader Thread Exit via Cleanup.");
+  debug(1, "Buffered TCP Reader Thread Exit via Cleanup.");
 }
 
 void *buffered_tcp_reader(void *arg) {