Enhancements

Enable the AirPlay 2 build to operate an optional classic-AirPlay-only service or to gracefully degrade to classic AirPlay, as follows:
  1. Add a new command-line option '-—service-type=<type>' and an equivalent configuration entry 'service_type = "<type>"' in the 'general' section of the configuration file,
     where <type> can be "auto", "classic" or "airplay2":
      1. "auto" (default) means that the service will be AirPlay 2 if NQPTP is running. If NQPTP is not running, classic AirPlay service will be provided instead, and in that case,
      "(Classic)" will be appended to the default AirPlay service name visible to AirPlay clients like Apple Music, for example "RaspberryPi3B (Classic)".
      2. "classic" means the service will be classic AirPlay (aka AirPlay 1).
      3. "airplay2" means the service will be the modern AirPlay 2. In this case, as distinct from "auto", if NQPTP is not running, Shairport Sync will log an error and terminate.
  2. In the systemd service file, NQPTP is now a "Want" rather than a "Require". If it's present, then it will be launched before Shairport Sync. If it's absent, Shairport Sync will launch anyway.
  3. Improve the delivery of input format changes and emit 'sdsc' metadata when changes occur.
  4. Emit new format information in the log if statistics is enabled.

Docker Changes
  1. Support for linux/arm/v6 has been dropped, as Docker is no longer supported.
  2. NQPTP is not started in the AirPlay 2 Docker image if '--service-type=classic' or '--service-type=airplay1' is in the command line options at the end of the docker run command.
     The purpose is to ensure that ports 319 and 320 are left alone when the AirPlay 2 image is set to provide Classic service only.
     Note that setting the configuration file 'service_type' to 'classic' will not prevent NQPTP from starting up -- you must use the command line option.
  3. A new "dev" target has been added. It is a large image containing the custom-built FFmpeg library, NQPTP, Avahi and D-Bus along with the Shairport Sync source and
     all necessary development tools. When started, Avahi, D-Bus and NQPTP are all installed and running. The bash shell has also been added and is entered.

Stability Improvements
  Reorganise session preemption to fully terminate the existing session before starting a new one.
  Don't delay closing the event port to wait for it to be closed at the client end.
  Add a safe_socket_close() function to ensure sockets are fully closed. Use -1 to designate closed rather than 0, to prevent attempts to reclose sockets, causing mayhem.
  Re-order FFmpeg decommissioning during teardown.

rtp.c

@@ -29,6 +29,7 @@
 #include "common.h"
 #include "player.h"
 #include "rtsp.h"
+#include "utilities/network_utilities.h"
 #include <arpa/inet.h>
 #include <errno.h>
 #include <fcntl.h>
@@ -311,11 +312,6 @@ void *rtp_audio_receiver(void *arg) {
     }
   }
 
-  /*
-  debug(3, "Audio receiver -- Server RTP thread interrupted. terminating.");
-  close(conn->audio_socket);
-  */
-
   debug(1, "Audio receiver thread \"normal\" exit -- this can't happen. Hah!");
   pthread_cleanup_pop(0); // don't execute anything here.
   debug(2, "Audio receiver thread exit.");
@@ -981,7 +977,8 @@ void *rtp_timing_receiver(void *arg) {
                   return_time);
           }
         } else {
-          debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %zd.", packet[1], nread);
+          debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %zd.", packet[1],
+                nread);
         }
       } else {
         debug(3, "Timing Receiver Thread -- dropping incoming packet to simulate a bad network.");
@@ -1387,9 +1384,11 @@ int get_ptp_anchor_local_time_info(rtsp_conn_info *conn, uint32_t *anchorRTP,
         if ((previous_offset != 0) && (previous_clock_id == actual_clock_id) &&
             ((jitter > 3000000) || (jitter < -3000000)))
           debug(1,
-                "Clock jitter: %.3f mS. Time since sample: %.3f mS. Time since start of mastership: %.3f "
+                "Clock jitter: %.3f mS. Time since sample: %.3f mS. Time since start of "
+                "mastership: %.3f "
                 "seconds.",
-                jitter * 0.000001, time_since_sample * 0.000001, time_since_start_of_mastership * 0.000000001);
+                jitter * 0.000001, time_since_sample * 0.000001,
+                time_since_start_of_mastership * 0.000000001);
 
         previous_offset = actual_offset;
         previous_clock_id = actual_clock_id;
@@ -1514,7 +1513,8 @@ int frame_to_ptp_local_time(uint32_t timestamp, uint64_t *time, rtsp_conn_info *
     int64_t time_difference = frame_difference;
     time_difference = time_difference * 1000000000;
     if (conn->input_rate == 0) {
-      debug(1,"Connection %d: in a call to frame_to_ptp_local_time, conn->input_rate is zero!", conn->connection_number);
+      debug(1, "Connection %d: in a call to frame_to_ptp_local_time, conn->input_rate is zero!",
+            conn->connection_number);
       time_difference = 0;
     } else {
       time_difference = time_difference / conn->input_rate;
@@ -1550,10 +1550,9 @@ int local_ptp_time_to_frame(uint64_t time, uint32_t *frame, rtsp_conn_info *conn
 void rtp_ap2_control_handler_cleanup_handler(void *arg) {
   rtsp_conn_info *conn = (rtsp_conn_info *)arg;
   debug(2, "Connection %d: AP2 Control Receiver Cleanup.", conn->connection_number);
-  close(conn->ap2_control_socket);
+  safe_socket_close(&conn->ap2_control_socket);
   debug(2, "Connection %d: UDP control port %u closed.", conn->connection_number,
         conn->local_ap2_control_port);
-  conn->ap2_control_socket = 0;
   conn->ap2_remote_control_socket_addr_length =
       0; // indicates to the control receiver thread that the socket address need to be
          // recreated (needed for resend requests in the realtime mode)
@@ -1806,9 +1805,9 @@ void *rtp_ap2_control_receiver(void *arg) {
 void rtp_realtime_audio_cleanup_handler(__attribute__((unused)) void *arg) {
   debug(2, "Realtime Audio Receiver Cleanup Start.");
   rtsp_conn_info *conn = (rtsp_conn_info *)arg;
-  close(conn->realtime_audio_socket);
-  debug(2, "Connection %d: closing realtime audio port %u", conn->connection_number, conn->local_realtime_audio_port);
-  conn->realtime_audio_socket = 0;
+  debug(2, "Connection %d: closing realtime audio port %u", conn->connection_number,
+        conn->local_realtime_audio_port);
+  safe_socket_close(&conn->realtime_audio_socket);
   debug(2, "Realtime Audio Receiver Cleanup Done.");
 }