Add ability to set output format and output rate

audio.h

@@ -19,7 +19,7 @@ typedef struct {
   // at end of program
   void (*deinit)(void);
 
-  void (*start)(int sample_rate);
+  void (*start)(int sample_rate, int sample_format);
 
   // block of samples
   void (*play)(short buf[], int samples);

audio_alsa.c

@@ -38,7 +38,7 @@
 static void help(void);
 static int init(int argc, char **argv);
 static void deinit(void);
-static void start(int sample_rate);
+static void start(int i_sample_rate, int i_sample_format);
 static void play(short buf[], int samples);
 static void stop(void);
 static void flush(void);
@@ -68,6 +68,7 @@ audio_output audio_alsa = {
 static pthread_mutex_t alsa_mutex = PTHREAD_MUTEX_INITIALIZER;
 
 static unsigned int desired_sample_rate;
+static snd_pcm_format_t sample_format;
 
 static snd_pcm_t *alsa_handle = NULL;
 static snd_pcm_hw_params_t *alsa_params = NULL;
@@ -205,6 +206,35 @@ static int init(int argc, char **argv) {
         die("Invalid disable_synchronization option choice \"%s\". It should be \"yes\" or \"no\"");
     }
     
+    /* Get the output format, using the same names as aplay does -- S16_LE, S32_LE to begin with*/
+    if (config_lookup_string(config.cfg, "alsa.output_format", &str)) {
+      if (strcasecmp(str, "S16_LE") == 0)
+        config.output_format = SND_PCM_FORMAT_S16_LE;
+      else if (strcasecmp(str, "S24_LE") == 0)
+        config.output_format = SND_PCM_FORMAT_S24_LE;
+      else if (strcasecmp(str, "S32_LE") == 0)
+        config.output_format = SND_PCM_FORMAT_S32_LE;
+      else
+        die("Invalid output format \"%s\". It should be \"S16_LE\", \"S24_LE\" or \"S32_LE\"",str);
+    }
+
+    /* Get the output rate, which must be a multiple of 44,100*/
+    if (config_lookup_int(config.cfg, "alsa.output_rate",
+                          &value)) {
+      debug(1,"Value read for output rate is %d.",value);
+      switch(value) {
+        case 44100:
+        case 88200:
+        case 176400:
+        case 352800:
+          config.output_rate = value;
+          break;
+        default :
+          die("Invalid output rate \"%d\". It should be a multiple of 44,100 up to 352,800",value);
+      }      
+    }
+
+
     /* Get the use_mmap_if_available setting. */
     if (config_lookup_string(config.cfg, "alsa.use_mmap_if_available", &str)) {
       if (strcasecmp(str, "no") == 0)
@@ -424,11 +454,10 @@ int open_alsa_device(void) {
         alsa_out_dev, snd_strerror(ret));
   }
 
-  ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params,
-                                     SND_PCM_FORMAT_S16);
+  ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params, sample_format);
   if (ret < 0) {
-    die("audio_alsa: Sample format not available for device \"%s\": %s",
-        alsa_out_dev, snd_strerror(ret));
+    die("audio_alsa: Sample format %d not available for device \"%s\": %s",
+        sample_format, alsa_out_dev, snd_strerror(ret));
   }
 
   ret = snd_pcm_hw_params_set_channels(alsa_handle, alsa_params, 2);
@@ -640,11 +669,18 @@ int open_alsa_device(void) {
   return (0);
 }
 
-static void start(int sample_rate) {
+static void start(int i_sample_rate, int i_sample_format) {
   // debug(2,"audio_alsa start called.");
-  if (sample_rate != 44100)
-    die("Unexpected sample rate %d -- only 44,100 supported!", sample_rate);
-  desired_sample_rate = sample_rate; // must be a variable
+  if (i_sample_rate==0)
+    desired_sample_rate = 44100; // default
+  else
+    desired_sample_rate = i_sample_rate; // must be a variable
+  
+  if (i_sample_format==0)
+    sample_format = SND_PCM_FORMAT_S16_LE; // default 
+  else
+    sample_format = i_sample_format;
+    
 }
 
 int delay(long* the_delay) {

audio_ao.c

@@ -134,9 +134,11 @@ static void deinit(void) {
   ao_shutdown();
 }
 
-static void start(int sample_rate) {
-  if (sample_rate != 44100)
+static void start(int sample_rate, int sample_format) {
+  if (sample_rate != 0)
     die("unexpected sample rate!");
+  if (sample_format != 0)
+    die("unexpected sample format!");
 }
 
 static void play(short buf[], int samples) { ao_play(dev, (char *)buf, samples * 4); }

audio_dummy.c

@@ -41,7 +41,7 @@ static int init(int argc, char **argv) { return 0; }
 
 static void deinit(void) {}
 
-static void start(int sample_rate) {
+static void start(int sample_rate, int sample_format) {
   Fs = sample_rate;
   starttime = 0;
   samples_played = 0;

audio_pipe.c

@@ -43,7 +43,7 @@ static int fd = -1;
 
 char *pipename = NULL;
 
-static void start(int sample_rate) {
+static void start(int sample_rate, int sample_format) {
   // this will leave fd as -1 if a reader hasn't been attached
   fd = open(pipename, O_WRONLY | O_NONBLOCK);
 }

audio_pulse.c

@@ -110,9 +110,11 @@ static void deinit(void) {
   pa_dev = NULL;
 }
 
-static void start(int sample_rate) {
-  if (sample_rate != 44100)
+static void start(int sample_rate, int sample_format) {
+  if (sample_rate != 0)
     die("unexpected sample rate!");
+  if (sample_format != 0)
+    die("unexpected sample format!");
 }
 
 static void play(short buf[], int samples) {

audio_sndio.c

@@ -56,6 +56,8 @@ static void deinit(void) { sio_close(sio); }
 static void start(int sample_rate) {
   if (sample_rate != par.rate)
     die("unexpected sample rate!");
+  if (sample_format != 0)
+    die("unexpected sample format!");
   sio_start(sio);
 }
 

audio_stdout.c

@@ -38,7 +38,7 @@
 
 static int fd = -1;
 
-static void start(int sample_rate) {
+static void start(int sample_rate int sample_format) {
   fd = STDOUT_FILENO;
 }
 

common.h

@@ -100,6 +100,8 @@ typedef struct {
                                             // audio backend buffer -- the DAC buffer for ALSA
   long audio_backend_latency_offset; // this will be the offset to compensate for any fixed latency there might be in the audio path
   uint32_t volume_range_db; // the range, in dB, from max dB to min dB. Zero means use the mixer's native range.
+  int output_format;
+  int output_rate;
 } shairport_cfg;
 
 // true if Shairport Sync is supposed to be sending output to the output device, false otherwise

player.c

@@ -1716,7 +1716,7 @@ int player_play(stream_cfg *stream, pthread_t *player_thread) {
 #endif
   if (rc)
     debug(1, "Error initialising condition variable.");
-  config.output->start(sampling_rate);
+  config.output->start(config.output_rate,config.output_format);
   size_t size = (PTHREAD_STACK_MIN + 256 * 1024);
   pthread_attr_t tattr;
   pthread_attr_init(&tattr);

shairport.c

@@ -1016,6 +1016,8 @@ int main(int argc, char **argv) {
   debug(1, "playback_mode is %d (0-stereo, 1-mono).", config.playback_mode);
   debug(1, "disable_synchronization is %d.", config.no_sync);
   debug(1, "use_mmap_if_available is %d.", config.no_mmap ? 0 : 1);
+  debug(1, "output_rate is %d (0 means 44,100).", config.output_rate);
+  debug(1, "output_format is %d (0 means S16_LE).", config.output_format);
   debug(1, "audio backend desired buffer length is %d.",
         config.audio_backend_buffer_desired_length);
   debug(1, "audio backend latency offset is %d.", config.audio_backend_latency_offset);