diff --git a/audio.h b/audio.h
index 4c86d7d..75abc08 100644
--- a/audio.h
+++ b/audio.h
@@ -22,7 +22,7 @@ typedef struct {
   void (*start)(int sample_rate);
 
   // block of samples
-  void (*play)(short buf[], int samples);
+  void (*play)(int buf[], int samples);
   void (*stop)(void);
 
   // may be null if not implemented
diff --git a/audio_alsa.c b/audio_alsa.c
index 0e9de51..c878fcb 100644
--- a/audio_alsa.c
+++ b/audio_alsa.c
@@ -39,7 +39,7 @@ static void help(void);
 static int init(int argc, char **argv);
 static void deinit(void);
 static void start(int sample_rate);
-static void play(short buf[], int samples);
+static void play(int buf[], int samples);
 static void stop(void);
 static void flush(void);
 int delay(long* the_delay);
@@ -429,7 +429,7 @@ int open_alsa_device(void) {
   }
 
   ret = snd_pcm_hw_params_set_format(alsa_handle, alsa_params,
-                                     SND_PCM_FORMAT_S16);
+                                     SND_PCM_FORMAT_S32);
   if (ret < 0) {
     die("audio_alsa: Sample format not available for device \"%s\": %s",
         alsa_out_dev, snd_strerror(ret));
@@ -693,7 +693,7 @@ int delay(long* the_delay) {
   }
 }
 
-static void play(short buf[], int samples) {
+static void play(int buf[], int samples) {
   // debug(3,"audio_alsa play called.");
   int ret = 0;
   if (alsa_handle == NULL) {
diff --git a/audio_ao.c b/audio_ao.c
index 41b823b..0e6cb0b 100644
--- a/audio_ao.c
+++ b/audio_ao.c
@@ -117,7 +117,7 @@ static int init(int argc, char **argv) {
   ao_sample_format fmt;
   memset(&fmt, 0, sizeof(fmt));
 
-  fmt.bits = 16;
+  fmt.bits = 32;
   fmt.rate = 44100;
   fmt.channels = 2;
   fmt.byte_format = AO_FMT_NATIVE;
@@ -139,7 +139,7 @@ static void start(int sample_rate) {
     die("unexpected sample rate!");
 }
 
-static void play(short buf[], int samples) { ao_play(dev, (char *)buf, samples * 4); }
+static void play(int buf[], int samples) { ao_play(dev, (char *)buf, samples * 4); }
 
 static void stop(void) {}
 
diff --git a/audio_dummy.c b/audio_dummy.c
index 659a720..bcccfcd 100644
--- a/audio_dummy.c
+++ b/audio_dummy.c
@@ -48,7 +48,7 @@ static void start(int sample_rate) {
   debug(1, "dummy audio output started at Fs=%d Hz\n", sample_rate);
 }
 
-static void play(short buf[], int samples) {}
+static void play(int buf[], int samples) {}
 
 static void stop(void) { debug(1, "dummy audio stopped\n"); }
 
diff --git a/audio_pipe.c b/audio_pipe.c
index 17bf310..314a7c5 100644
--- a/audio_pipe.c
+++ b/audio_pipe.c
@@ -48,7 +48,7 @@ static void start(int sample_rate) {
   fd = open(pipename, O_WRONLY | O_NONBLOCK);
 }
 
-static void play(short buf[], int samples) {
+static void play(int buf[], int samples) {
   // if the file is not open, try to open it.
   if (fd == -1) {
      fd = open(pipename, O_WRONLY | O_NONBLOCK); 
diff --git a/audio_pulse.c b/audio_pulse.c
index 6fa94e9..d7fce44 100644
--- a/audio_pulse.c
+++ b/audio_pulse.c
@@ -95,7 +95,7 @@ static int init(int argc, char **argv) {
 }
 
 static void pulse_connect(void) {
-  static const pa_sample_spec ss = {.format = PA_SAMPLE_S16LE, .rate = 44100, .channels = 2};
+  static const pa_sample_spec ss = {.format = PA_SAMPLE_S32LE, .rate = 44100, .channels = 2};
 
   pa_dev = pa_simple_new(pulse_options.server, pulse_options.service_name, PA_STREAM_PLAYBACK,
                          pulse_options.sink, "Shairport Stream", &ss, NULL, NULL, &pa_error);
@@ -115,8 +115,8 @@ static void start(int sample_rate) {
     die("unexpected sample rate!");
 }
 
-static void play(short buf[], int samples) {
-  if (pa_simple_write(pa_dev, (char *)buf, (size_t)samples * 4, &pa_error) < 0) {
+static void play(int buf[], int samples) {
+  if (pa_simple_write(pa_dev, (char *)buf, (size_t)samples * 8, &pa_error) < 0) {
     fprintf(stderr, __FILE__ ": pa_simple_write() failed: %s\n", pa_strerror(pa_error));
     if (pa_error == PA_ERR_CONNECTIONTERMINATED) {
       fprintf(stderr, __FILE__ ": reconnecting.");
diff --git a/audio_sndio.c b/audio_sndio.c
index 7093261..5248e47 100644
--- a/audio_sndio.c
+++ b/audio_sndio.c
@@ -34,7 +34,7 @@ static int init(int argc, char **argv) {
 
   sio_initpar(&par);
 
-  par.bits = 16;
+  par.bits = 32;
   par.rate = 44100;
   par.pchan = 2;
   par.le = SIO_LE_NATIVE;
@@ -59,7 +59,7 @@ static void start(int sample_rate) {
   sio_start(sio);
 }
 
-static void play(short buf[], int samples) {
+static void play(int buf[], int samples) {
   sio_write(sio, (char *)buf, samples * par.bps * par.pchan);
 }
 
diff --git a/audio_stdout.c b/audio_stdout.c
index b8b0aa5..45276b0 100644
--- a/audio_stdout.c
+++ b/audio_stdout.c
@@ -42,8 +42,8 @@ static void start(int sample_rate) {
   fd = STDOUT_FILENO;
 }
 
-static void play(short buf[], int samples) {
-  int ignore = write(fd, buf, samples * 4);
+static void play(int buf[], int samples) {
+  int ignore = write(fd, buf, samples * 8);
 }
 
 static void stop(void) {
diff --git a/player.c b/player.c
index 97eccfb..148bb9d 100644
--- a/player.c
+++ b/player.c
@@ -440,7 +440,7 @@ int32_t rand_in_range(int32_t exclusive_range_limit) {
 	return sp >> 32;  
 }
 
-static inline short dithered_vol(short sample) {
+static inline int dithered_vol(short sample) {
   long out;
 
   out = (long)sample * fix_volume;
@@ -469,7 +469,8 @@ static inline short dithered_vol(short sample) {
     		out = LONG_MIN;
     }
   }
-  return out >> 16;
+  return out;
+  //return out >> 16;
 }
 
 // get the next frame, when available. return 0 if underrun/stream reset.
@@ -697,16 +698,16 @@ static abuf_t *buffer_get_frame(void) {
                   // ab_write),ab_read,ab_write);
                   ab_buffering = 0;
                 }
-                signed short *silence;
+		signed int *silence;
                 //if (fs==0)
                 //  debug(2,"Zero length silence buffer needed with gross_frame_gap of %lld and dac_delay of %lld.",gross_frame_gap,dac_delay);
                 // the fs (number of frames of silence to play) can be zero in the DAC doesn't start ouotputting frames for a while -- it could get loaded up but not start responding for many milliseconds.
                 if (fs!=0) {
-                  silence = malloc(bytes_per_audio_frame*fs);
+                  silence = malloc(bytes_per_audio_frame*fs*2);
                   if (silence==NULL)
                     debug(1,"Failed to allocate %d byte silence buffer.",fs);
                   else {
-                    memset(silence, 0, bytes_per_audio_frame*fs);
+                    memset(silence, 0, bytes_per_audio_frame*fs*2);
                     // debug(1,"Exact frame gap is %llu; play %d frames of silence. Dac_delay is %d,
                     // with %d packets.",exact_frame_gap,fs,dac_delay,seq_diff(ab_read, ab_write));
                     config.output->play(silence, fs);
@@ -857,8 +858,15 @@ static inline short shortmean(short a, short b) {
   return r;
 }
 
+static void stuff_buffer_normal(short *inptr, int length, int *outptr){
+  int i;
+  for (i = 0; i < length; i++)
+    *outptr++ = *inptr++;
+}
+
+
 // stuff: 1 means add 1; 0 means do nothing; -1 means remove 1
-static int stuff_buffer_basic(short *inptr, int length, short *outptr, int stuff) {
+static int stuff_buffer_basic(short *inptr, int length, int *outptr, int stuff) {
   if ((stuff > 1) || (stuff < -1) || (length <100)) {
     // debug(1, "Stuff argument to stuff_buffer must be from -1 to +1 and length >100.");
     return length;
@@ -1033,14 +1041,15 @@ static void *player_thread_func(void *arg) {
   char rnstate[256];
   initstate(time(NULL), rnstate, 256);
 
-  signed short *inbuf, *outbuf, *silence;
+  signed short *inbuf;
+  signed int *outbuf;
   outbuf = malloc(bytes_per_audio_frame*(frame_size+max_frame_size_change));
   if (outbuf==NULL)
     debug(1,"Failed to allocate memory for an output buffer.");
-  silence = malloc(bytes_per_audio_frame*frame_size);
-  if (silence==NULL)
+//  silence = malloc(bytes_per_audio_frame*frame_size);
+//  if (silence==NULL)
     debug(1,"Failed to allocate memory for a silence buffer.");
-  memset(silence, 0, bytes_per_audio_frame*frame_size);
+//  memset(silence, 0, bytes_per_audio_frame*frame_size);
   late_packet_message_sent = 0;
   first_packet_timestamp = 0;
   missing_packets = late_packets = too_late_packets = resend_requests = 0;
@@ -1100,8 +1109,10 @@ static void *player_thread_func(void *arg) {
           else {
             if (inframe->length==0)
               debug(1,"empty frame to play -- skipping it.");
-            else
-              config.output->play(inframe->data, inframe->length);
+	    else {///
+	      stuff_buffer_normal(inframe->data, inframe->length, outbuf);
+              config.output->play(outbuf, inframe->length);
+	    }
           }
         } else {
           // We have a frame of data. We need to see if we want to add or remove a frame from it to
@@ -1251,8 +1262,10 @@ static void *player_thread_func(void *arg) {
               else {
                 if (inframe->length==0)
                   debug(1,"empty frame to play -- skipping it (2).");
-                else
-                  config.output->play(inframe->data, inframe->length);
+		else {///
+		  stuff_buffer_normal(inframe->data, inframe->length, outbuf);
+                  config.output->play(outbuf, inframe->length);
+		}
               }
             } else {
 #ifdef HAVE_LIBSOXR
@@ -1270,23 +1283,6 @@ static void *player_thread_func(void *arg) {
               //          if (amount_to_stuff) debug(1,"Standard stuff...");
               play_samples = stuff_buffer_basic(inframe->data, inframe->length, outbuf, amount_to_stuff);
 #endif
-
-              /*
-              {
-                int co;
-                int is_silent=1;
-                short *p = outbuf;
-                for (co=0;co<play_samples;co++) {
-                  if (*p!=0)
-                    is_silent=0;
-                  p++;
-                }
-                if (is_silent)
-                  debug(1,"Silence!");
-              }
-              */
-              
-              
               if (outbuf==NULL)
                 debug(1,"NULL outbuf to play -- skipping it.");
               else {
@@ -1331,8 +1327,10 @@ static void *player_thread_func(void *arg) {
               else {
                 if (inframe->length==0)
                   debug(1,"empty frame to play -- skipping it (3).");
-                else
-                  config.output->play(inframe->data, inframe->length);
+		else {///
+		  stuff_buffer_normal(inframe->data, inframe->length, outbuf);
+                  config.output->play(outbuf, inframe->length);
+		}
               }
             else {
               play_samples = stuff_buffer_basic(inframe->data, inframe->length, outbuf, 0); // no stuffing, but volume adjustment
@@ -1498,7 +1496,7 @@ static void *player_thread_func(void *arg) {
   	config.output->stop();
   usleep(100000); // allow this time to (?) allow the alsa subsystem to finish cleaning up after itself. 50 ms seems too short
   free(outbuf);
-  free(silence);
+//  free(silence);
   debug(1,"Shut down audio, control and timing threads");
   itr.please_stop = 1;
   pthread_kill(rtp_audio_thread, SIGUSR1);