UDP port numbers are now chosen from port 7000 up. Two options are introduced to control this. joerg-krause's stack erro fix installed.

common.h

@@ -48,6 +48,8 @@ typedef struct {
 #endif
   uint8_t hw_addr[6];
   int port;
+  int udp_port_base;
+  int udp_port_range;
   int ignore_volume_control;
   int resyncthreshold; // if it get's out of whack my more than this, resync. Zero means never
                        // resync.

configure.ac

@@ -2,7 +2,7 @@
 # Process this file with autoconf to produce a configure script.
 
 AC_PREREQ([2.50])
-AC_INIT([shairport-sync], [2.3.2], [mikebrady@eircom.net])
+AC_INIT([shairport-sync], [2.3.3], [mikebrady@eircom.net])
 AM_INIT_AUTOMAKE
 AC_CONFIG_SRCDIR([shairport.c])
 AC_CONFIG_HEADERS([config.h])

player.c

@@ -36,12 +36,13 @@
 #include <pthread.h>
 #include <math.h>
 #include <sys/stat.h>
-#include <sys/signal.h>
+#include <signal.h>
 #include <sys/syslog.h>
 #include <assert.h>
 #include <fcntl.h>
 #include <stdlib.h>
 #include <errno.h>
+#include <limits.h>
 
 #include "config.h"
 
@@ -1271,8 +1272,14 @@ int player_play(stream_cfg *stream) {
   if (rc)
     debug(1, "Error initialising condition variable.");
   config.output->start(sampling_rate);
-  pthread_create(&player_thread, NULL, player_thread_func, NULL);
-
+  size_t size = (PTHREAD_STACK_MIN + 128 * 1024);
+  pthread_attr_t tattr;
+  pthread_attr_init(&tattr);
+  rc = pthread_attr_setstacksize(&tattr, size);
+  if (rc)
+    debug(1, "Error setting stack size for player_thread: %s", strerror(errno));
+  pthread_create(&player_thread, &tattr, player_thread_func, NULL);
+  pthread_attr_destroy(&tattr);
   return 0;
 }
 

rtp.c

@@ -396,7 +396,7 @@ static void *rtp_timing_receiver(void *arg) {
   return NULL;
 }
 
-static int bind_port(SOCKADDR *remote, int *sock, int desired_port) {
+static int bind_port(SOCKADDR *remote, int *sock) {
   struct addrinfo hints, *info;
 
   memset(&hints, 0, sizeof(hints));
@@ -405,21 +405,32 @@ static int bind_port(SOCKADDR *remote, int *sock, int desired_port) {
   hints.ai_flags = AI_PASSIVE;
 
   char buffer[10];
-  snprintf(buffer, 10, "%d", desired_port);
+  
+  // look for a port in the range, if any was specified.
+  int desired_port = config.udp_port_base;
+  int ret;
+  do {
+    snprintf(buffer, 10, "%d", desired_port);
 
-  int ret = getaddrinfo(NULL, buffer, &hints, &info);
+    ret = getaddrinfo(NULL, buffer, &hints, &info);
 
-  if (ret < 0)
-    die("failed to get usable addrinfo?! %s.", gai_strerror(ret));
+    if (ret < 0)
+      die("failed to get usable addrinfo?! %s.", gai_strerror(ret));
 
-  *sock = socket(remote->SAFAMILY, SOCK_DGRAM, IPPROTO_UDP);
-  ret = bind(*sock, info->ai_addr, info->ai_addrlen);
+    *sock = socket(remote->SAFAMILY, SOCK_DGRAM, IPPROTO_UDP);
+    ret = bind(*sock, info->ai_addr, info->ai_addrlen);
 
-  freeaddrinfo(info);
-
-  if (ret < 0)
-    die("could not bind a UDP port!");
+    freeaddrinfo(info);
 
+  } while ((ret<0) && (errno==EADDRINUSE) && (desired_port!=0) && (desired_port++ < config.udp_port_base+config.udp_port_range));
+  
+  debug(1,"UDP port chosen: %d.",desired_port);
+  
+  if (ret < 0) {
+     die("error: could not bind a UDP port!");
+  }
+ 
+  
   int sport;
   SOCKADDR local;
   socklen_t local_len = sizeof(local);
@@ -510,9 +521,9 @@ void rtp_setup(SOCKADDR *remote, int cport, int tport, uint32_t active_remote, i
   // now, we open three sockets -- one for the audio stream, one for the timing and one for the
   // control
 
-  *lsport = bind_port(remote, &audio_socket, 0);
-  *lcport = bind_port(remote, &control_socket, 0);
-  *ltport = bind_port(remote, &timing_socket, 0);
+  *lsport = bind_port(remote, &audio_socket);
+  *lcport = bind_port(remote, &control_socket);
+  *ltport = bind_port(remote, &timing_socket);
 
   debug(2, "listening for audio, control and timing on ports %d, %d, %d.", *lsport, *lcport,
         *ltport);

rtsp.c

@@ -39,7 +39,7 @@
 #include <errno.h>
 #include <fcntl.h>
 #include <pthread.h>
-#include <sys/poll.h>
+#include <poll.h>
 
 #include "config.h"
 

scripts/shairport-sync.conf

@@ -8,6 +8,8 @@ general =
 //	mdns_backend = "avahi"; // not used, not tested
 //	output_backend = "alsa"; // alsa is default, other possibilities are "stdout", "pulse", "dummy". Only alsa supports synchronisation.
 //	port = 5000;
+//  udp_port_base = 7000; // start allocation UDP ports from this port number
+//  udp_port_range = 100; // look for free ports in this number of places, starting at the UDP port base (only three are needed).
 //	password = "secret"; // default is no password
 //	interpolation = "basic"; // aka "stuffing". Default is "basic", alternative is "soxr"
 //	statistics = "no"; // print statistics in the log

scripts/shairport-sync.conf~

@@ -0,0 +1,75 @@
+// Sample Configuration File for Shairport Sync
+// Commented out settings are generally the defaults, except where noted.
+
+// General Settings
+general =
+{
+//	name = "Shairport Sync Player"; // This is the name the service will advertise to iTunes. The default is "Shairport Sync on <hostname>"
+//	mdns_backend = "avahi"; // not used, not tested
+//	output_backend = "alsa"; // alsa is default, other possibilities are "stdout", "pulse", "dummy". Only alsa supports synchronisation.
+//	port = 5000;
+//	password = "secret"; // default is no password
+//	interpolation = "basic"; // aka "stuffing". Default is "basic", alternative is "soxr"
+//	statistics = "no"; // print statistics in the log
+//	drift = 88; // allow this number of frames of drift before correcting it
+//	resync_threshold = 2205; // a synchronisation error greater than this will cause resynchronisation; 0 disables it
+//	log_verbosity = 0; // "0" means no verbosity, "3" is most verbose.
+//  ignore_volume_control = "no"; // set this to "yes" if you want the volume to be at 100% no matter what the source's volume control is set to.
+};
+
+// Latencies for different sources.
+latencies =
+{
+//	default = 88200;
+//	itunes = 99400;
+//	airplay = 88200;
+//	forkedDaapd = 99400;
+};
+
+
+// How to deal with metadata, including artwork
+metadata =
+{
+//	enabled = "no";
+//	include_cover_art = "no";
+//	pipe_name = "/tmp/shairport-sync-metadata";
+};
+
+// Advanced parameters for controlling how a Shairport Sync runs
+sessioncontrol = 
+{
+//	run_this_before_play_begins = "/path/to/application and args";
+//	run_this_after_play_ends = "/path/to/application and args";
+//	wait_for_completion = "no";
+//	allow_session_interruption = "no";
+//	session_timeout = 120;
+};
+
+//
+// Back End Settings
+//
+
+// These are parameters for the alsa back end, the only back end that supports synchronisation
+{
+//  output_device = "default";
+//  mixer_type = "software"; // "software" or "hardware"
+//  mixer_device = "default"; //actually, the mixer default is whatever the output_device is. Normally you wouldn't have to use this.
+//  mixer_control_name = "PCM"; // the name of the mixer to use -- there is no default.
+//  audio_backend_latency_offset = 0; // Set this offset to compensate for a fixed delay in the audio back end. E.g. if the output device delays by 100 ms, set this to -4410.
+//  audio_backend_buffer_desired_length = 6615; // If set too small, buffer underflow occurs on low-powered machines. Too long and the response times with software mixer become annoying.
+};
+
+// These are parameters for the pipe back end, an experimental back end that directs output to a pipe.
+pipe =
+{
+//  audio_backend_latency_offset = 0; // Set this offset to compensate for a fixed delay in the audio back end. E.g. if the output device delays by 100 ms, set this to -4410.
+//  audio_backend_buffer_desired_length = 44100;
+//  name = "/path/to/pipe"; // there is no default pipe name for the output
+};
+
+// These are parameters for the stdout back end, an experimental back end that directs output to stdout.
+stdout =
+{
+//  audio_backend_latency_offset = 0; // Set this offset to compensate for a fixed delay in the audio back end. E.g. if the output device delays by 100 ms, set this to -4410.
+//  audio_backend_buffer_desired_length = 44100;
+};

shairport.c

@@ -369,6 +369,24 @@ int parse_options(int argc, char **argv) {
         config.port = value;
     }
 
+    /* Get the udp port base setting. */
+    if (config_lookup_int(config.cfg, "general.udp_port_base", &value)) {
+      if ((value < 0) || (value > 65535))
+        die("Invalid port number  \"%sd\". It should be between 0 and 65535, default is 7000",
+            value);
+      else
+        config.udp_port_base = value;
+    }
+
+    /* Get the udp port range setting. This is number of ports that will be tried for free ports , starting at the port base. Only three ports are needed. */
+    if (config_lookup_int(config.cfg, "general.udp_port_range", &value)) {
+      if ((value < 0) || (value > 65535))
+        die("Invalid port range  \"%sd\". It should be between 0 and 65535, default is 100",
+            value);
+      else
+        config.udp_port_range = value;
+    }
+
     /* Get the password setting. */
     if (config_lookup_string(config.cfg, "general.password", &str))
       config.password = (char *)str;
@@ -606,6 +624,8 @@ int main(int argc, char **argv) {
   set_requested_connection_state_to_output(
       1); // we expect to be able to connect to the output device
   config.audio_backend_buffer_desired_length = 6615; // 0.15 seconds.
+  config.udp_port_base = 7000;
+  config.udp_port_range = 100;
 
   // this is a bit weird, but apparently necessary
   char *basec = strdup(argv[0]);
@@ -780,6 +800,8 @@ int main(int argc, char **argv) {
   debug(2, "statistics_requester status is %d.", config.statistics_requested);
   debug(2, "daemon status is %d.", config.daemonise);
   debug(2, "rtsp listening port is %d.", config.port);
+  debug(2, "udp base port is %d.", config.udp_port_base);
+  debug(2, "udp port range is %d.", config.udp_port_range);
   debug(2, "Shairport Sync player name is \"%s\".", config.apname);
   debug(2, "Audio Output name is \"%s\".", config.output_name);
   debug(2, "on-start action is \"%s\".", config.cmd_start);