Remove trailing whitespace from most files

22 files changed, 196 insertions, 220 deletions

diff --git a/libao2/ao_alsa.c b/libao2/ao_alsa.c
index c95aed4cf7..4c3629b8c9 100644
--- a/libao2/ao_alsa.c
+++ b/libao2/ao_alsa.c
@@ -57,7 +57,7 @@
 #include "audio_out_internal.h"
 #include "libaf/af_format.h"
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
     "ALSA-0.9.x-1.x audio output",
     "alsa",
@@ -73,7 +73,7 @@ static snd_pcm_hw_params_t *alsa_hwparams;
 static snd_pcm_sw_params_t *alsa_swparams;
 
 /* 16 sets buffersize to 16 * chunksize is as default 1024
- * which seems to be good avarge for most situations 
+ * which seems to be good avarge for most situations
  * so buffersize is 16384 frames by default */
 static int alsa_fragcount = 16;
 static snd_pcm_uframes_t chunk_size = 1024;
@@ -157,7 +157,7 @@ static int control(int cmd, void *arg)
 
       //allocate simple id
       snd_mixer_selem_id_alloca(&sid);
-	
+
       //sets simple-mixer index and name
       snd_mixer_selem_id_set_index(sid, mix_index);
       snd_mixer_selem_id_set_name(sid, mix_name);
@@ -173,7 +173,7 @@ static int control(int cmd, void *arg)
       }
 
       if ((err = snd_mixer_attach(handle, card)) < 0) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n", 
+	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Mixer attach %s error: %s\n",
 	       card, snd_strerror(err));
 	snd_mixer_close(handle);
 	return CONTROL_ERROR;
@@ -208,7 +208,7 @@ static int control(int cmd, void *arg)
 
 	//setting channels
 	if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_LEFT, set_vol)) < 0) {
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n", 
+	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting left channel, %s\n",
 		 snd_strerror(err));
 	  return CONTROL_ERROR;
 	}
@@ -217,11 +217,11 @@ static int control(int cmd, void *arg)
 	set_vol = vol->right / f_multi + pmin + 0.5;
 
 	if ((err = snd_mixer_selem_set_playback_volume(elem, SND_MIXER_SCHN_FRONT_RIGHT, set_vol)) < 0) {
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n", 
+	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Error setting right channel, %s\n",
 		 snd_strerror(err));
 	  return CONTROL_ERROR;
 	}
-	mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n", 
+	mp_msg(MSGT_AO,MSGL_DBG2,"right=%li, pmin=%li, pmax=%li, mult=%f\n",
 	       set_vol, pmin, pmax, f_multi);
 
 	if (snd_mixer_selem_has_playback_switch(elem)) {
@@ -246,7 +246,7 @@ static int control(int cmd, void *arg)
       snd_mixer_close(handle);
       return CONTROL_OK;
     }
-    
+
   } //end switch
   return CONTROL_UNKNOWN;
 }
@@ -359,7 +359,7 @@ static int init(int rate_hz, int channels, int format, int flags)
     prepause_frames = 0;
 
     snd_lib_error_set_handler(alsa_error_handler);
-    
+
     ao_data.samplerate = rate_hz;
     ao_data.format = format;
     ao_data.channels = channels;
@@ -419,7 +419,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	alsa_format = SND_PCM_FORMAT_MPEG; //? default should be -1
 	break;
       }
-    
+
     //subdevice parsing
     // set defaults
     block = 1;
@@ -547,11 +547,11 @@ static int init(int rate_hz, int channels, int format, int flags)
 		 snd_strerror(err));
 	  return 0;
 	}
-    
+
       err = snd_pcm_hw_params_set_access(alsa_handler, alsa_hwparams,
 					 SND_PCM_ACCESS_RW_INTERLEAVED);
       if (err < 0) {
-	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n", 
+	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set access type: %s\n",
 	       snd_strerror(err));
 	return 0;
       }
@@ -595,8 +595,8 @@ static int init(int rate_hz, int channels, int format, int flags)
 	}
 #endif
 
-      if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, 
-						 &ao_data.samplerate, NULL)) < 0) 
+      if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams,
+						 &ao_data.samplerate, NULL)) < 0)
         {
 	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set samplerate-2: %s\n",
 		 snd_strerror(err));
@@ -612,7 +612,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	int alsa_buffer_time = 500000; /* original 60 */
 	int alsa_period_time;
 	alsa_period_time = alsa_buffer_time/4;
-	if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams, 
+	if ((err = snd_pcm_hw_params_set_buffer_time_near(alsa_handler, alsa_hwparams,
 							  &alsa_buffer_time, NULL)) < 0)
 	  {
 	    mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set buffer time near: %s\n",
@@ -621,7 +621,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	  } else
 	    alsa_buffer_time = err;
 
-	if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams, 
+	if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams,
 							  &alsa_period_time, NULL)) < 0)
 	  /* original: alsa_buffer_time/ao_data.bps */
 	  {
@@ -631,13 +631,13 @@ static int init(int rate_hz, int channels, int format, int flags)
 	  }
 	mp_tmsg(MSGT_AO,MSGL_INFO,"[AO_ALSA] buffer_time: %d, period_time :%d\n",
 	       alsa_buffer_time, err);
-      } 
+      }
 #endif//end SET_BUFFERTIME
 
 #ifdef SET_CHUNKSIZE
       {
 	//set chunksize
-	if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams, 
+	if ((err = snd_pcm_hw_params_set_period_size_near(alsa_handler, alsa_hwparams,
 							  &chunk_size, NULL)) < 0)
 	  {
 	    mp_tmsg(MSGT_AO,MSGL_ERR,"[AO ALSA] Unable to set period size(%ld): %s\n",
@@ -649,7 +649,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	}
 	if ((err = snd_pcm_hw_params_set_periods_near(alsa_handler, alsa_hwparams,
 						      &alsa_fragcount, NULL)) < 0) {
-	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n", 
+	  mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Unable to set periods: %s\n",
 		 snd_strerror(err));
 	  return 0;
 	}
@@ -890,15 +890,15 @@ static int get_space(void)
 {
     snd_pcm_status_t *status;
     int ret;
-    
+
     snd_pcm_status_alloca(&status);
-    
+
     if ((ret = snd_pcm_status(alsa_handler, status)) < 0)
     {
 	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO_ALSA] Cannot get pcm status: %s\n", snd_strerror(ret));
 	return 0;
     }
-    
+
     unsigned space = snd_pcm_status_get_avail(status) * bytes_per_sample;
     if (space > ao_data.buffersize) // Buffer underrun?
         space = ao_data.buffersize;
@@ -910,10 +910,10 @@ static float get_delay(void)
 {
   if (alsa_handler) {
     snd_pcm_sframes_t delay;
-    
+
     if (snd_pcm_delay(alsa_handler, &delay) < 0)
       return 0;
-    
+
     if (delay < 0) {
       /* underrun - move the application pointer forward to catch up */
 #if SND_LIB_VERSION >= 0x000901 /* snd_pcm_forward() exists since 0.9.0rc8 */
diff --git a/libao2/ao_alsa5.c b/libao2/ao_alsa5.c
index bca4e93ed8..b2dc1efd3a 100644
--- a/libao2/ao_alsa5.c
+++ b/libao2/ao_alsa5.c
@@ -32,7 +32,7 @@
 #include "mp_msg.h"
 #include "help_mp.h"
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
     "ALSA-0.5.x audio output",
     "alsa5",
@@ -117,7 +117,7 @@ static int init(int rate_hz, int channels, int format, int flags)
 	    alsa_format.format = SND_PCM_SFMT_MPEG;
 	    break;
     }
-    
+
     switch(alsa_format.format)
     {
 	case SND_PCM_SFMT_S16_LE:
@@ -230,7 +230,7 @@ static int init(int rate_hz, int channels, int format, int flags)
     setup.format = alsa_format;
     setup.buf.stream.queue_size = ao_data.buffersize;
     setup.msbits_per_sample = ao_data.bps;
-    
+
     if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
     {
 	mp_tmsg(MSGT_AO, MSGL_ERR, "[AO ALSA5] alsa-init: error setting up channel: %s\n", snd_strerror(err));
@@ -333,10 +333,10 @@ static void audio_resume(void)
 static int play(void* data, int len, int flags)
 {
     int got_len;
-    
+
     if (!len)
 	return 0;
-    
+
     if ((got_len = snd_pcm_write(alsa_handler, data, len)) < 0)
     {
 	if (got_len == -EPIPE) /* underrun? */
@@ -365,7 +365,7 @@ static int play(void* data, int len, int flags)
 static int get_space(void)
 {
     snd_pcm_channel_status_t ch_stat;
-    
+
     ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
 
     if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
@@ -378,9 +378,9 @@ static int get_space(void)
 static float get_delay(void)
 {
     snd_pcm_channel_status_t ch_stat;
-    
+
     ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
-    
+
     if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
 	return (float)ao_data.buffersize/(float)ao_data.bps; /* error occurred */
     else
diff --git a/libao2/ao_arts.c b/libao2/ao_arts.c
index 1d1638ba2d..5e55d6f6b7 100644
--- a/libao2/ao_arts.c
+++ b/libao2/ao_arts.c
@@ -147,4 +147,3 @@ static float get_delay(void)
 	return ((float) (ao_data.buffersize - arts_stream_get(stream,
 		ARTS_P_BUFFER_SPACE))) / ((float) ao_data.bps);
 }
-
diff --git a/libao2/ao_coreaudio.c b/libao2/ao_coreaudio.c
index 3ef6d3367f..76cb9174be 100644
--- a/libao2/ao_coreaudio.c
+++ b/libao2/ao_coreaudio.c
@@ -174,7 +174,7 @@ Float32 vol;
 				ao->b_muted = 0;
 			return CONTROL_TRUE;
 		}
-		
+
 		vol=(control_vol->left+control_vol->right)*4.0/200.0;
 		err = AudioUnitSetParameter(ao->theOutputUnit, kHALOutputParam_Volume, kAudioUnitScope_Global, 0, vol, 0);
 		if(err==0) {
@@ -189,7 +189,7 @@ Float32 vol;
 	default:
 	  return CONTROL_FALSE;
 	}
-	
+
 }
 
 
@@ -237,8 +237,8 @@ static OSStatus DeviceListener( AudioDeviceID inDevice,
 static int init(int rate,int channels,int format,int flags)
 {
 AudioStreamBasicDescription inDesc;
-ComponentDescription desc; 
-Component comp; 
+ComponentDescription desc;
+Component comp;
 AURenderCallbackStruct renderCallback;
 OSStatus err;
 UInt32 size, maxFrames, i_param_size;
@@ -396,20 +396,20 @@ int b_alive;
 	desc.componentManufacturer = kAudioUnitManufacturer_Apple;
 	desc.componentFlags = 0;
 	desc.componentFlagsMask = 0;
-				
+
 	comp = FindNextComponent(NULL, &desc);  //Finds an component that meets the desc spec's
 	if (comp == NULL) {
 		ao_msg(MSGT_AO, MSGL_WARN, "Unable to find Output Unit component\n");
 		goto err_out;
 	}
-		
+
 	err = OpenAComponent(comp, &(ao->theOutputUnit));  //gains access to the services provided by the component
 	if (err) {
 		ao_msg(MSGT_AO, MSGL_WARN, "Unable to open Output Unit component: [%4.4s]\n", (char *)&err);
 		goto err_out;
 	}
 
-	// Initialize AudioUnit 
+	// Initialize AudioUnit
 	err = AudioUnitInitialize(ao->theOutputUnit);
 	if (err) {
 		ao_msg(MSGT_AO, MSGL_WARN, "Unable to initialize Output Unit component: [%4.4s]\n", (char *)&err);
@@ -426,7 +426,7 @@ int b_alive;
 
 	size = sizeof(UInt32);
 	err = AudioUnitGetProperty(ao->theOutputUnit, kAudioDevicePropertyBufferSize, kAudioUnitScope_Input, 0, &maxFrames, &size);
-	
+
 	if (err)
 	{
 		ao_msg(MSGT_AO,MSGL_WARN, "AudioUnitGetProperty returned [%4.4s] when getting kAudioDevicePropertyBufferSize\n", (char *)&err);
@@ -434,7 +434,7 @@ int b_alive;
 	}
 
 	ao->chunk_size = maxFrames;//*inDesc.mBytesPerFrame;
-    
+
 	ao_data.samplerate = inDesc.mSampleRate;
 	ao_data.channels = inDesc.mChannelsPerFrame;
     ao_data.bps = ao_data.samplerate * inDesc.mBytesPerFrame;
@@ -444,7 +444,7 @@ int b_alive;
 	ao->num_chunks = (ao_data.bps+ao->chunk_size-1)/ao->chunk_size;
     ao->buffer_len = ao->num_chunks * ao->chunk_size;
     ao->buffer = av_fifo_alloc(ao->buffer_len);
-	
+
 	ao_msg(MSGT_AO,MSGL_V, "using %5d chunks of %d bytes (buffer len %d bytes)\n", (int)ao->num_chunks, (int)ao->chunk_size, (int)ao->buffer_len);
 
     renderCallback.inputProc = theRenderProc;
@@ -456,7 +456,7 @@ int b_alive;
 	}
 
 	reset();
-    
+
     return CONTROL_OK;
 
 err_out2:
@@ -467,7 +467,7 @@ err_out:
     av_fifo_free(ao->buffer);
     free(ao);
     ao = NULL;
-    return CONTROL_FALSE; 
+    return CONTROL_FALSE;
 }
 
 /*****************************************************************************
@@ -734,7 +734,7 @@ err_out:
     av_fifo_free(ao->buffer);
     free(ao);
     ao = NULL;
-    return CONTROL_FALSE; 
+    return CONTROL_FALSE;
 }
 
 /*****************************************************************************
@@ -943,7 +943,7 @@ static OSStatus RenderCallbackSPDIF( AudioDeviceID inDevice,
 
 
 static int play(void* output_samples,int num_bytes,int flags)
-{  
+{
     int wrote, b_digital;
 
     // Check whether we need to reset the digital output stream.
diff --git a/libao2/ao_dsound.c b/libao2/ao_dsound.c
index 729084b145..4197c30e27 100644
--- a/libao2/ao_dsound.c
+++ b/libao2/ao_dsound.c
@@ -113,17 +113,17 @@ static const int channel_mask[] = {
 };
 
 static HINSTANCE hdsound_dll = NULL;      ///handle to the dll
-static LPDIRECTSOUND hds = NULL;          ///direct sound object 
+static LPDIRECTSOUND hds = NULL;          ///direct sound object
 static LPDIRECTSOUNDBUFFER hdspribuf = NULL; ///primary direct sound buffer
 static LPDIRECTSOUNDBUFFER hdsbuf = NULL; ///secondary direct sound buffer (stream buffer)
-static int buffer_size = 0;               ///size in bytes of the direct sound buffer   
+static int buffer_size = 0;               ///size in bytes of the direct sound buffer
 static int write_offset = 0;              ///offset of the write cursor in the direct sound buffer
 static int min_free_space = 0;            ///if the free space is below this value get_space() will return 0
                                           ///there will always be at least this amout of free space to prevent
                                           ///get_space() from returning wrong values when buffer is 100% full.
                                           ///will be replaced with nBlockAlign in init()
 static int device_num = 0;                ///wanted device number
-static GUID device;                       ///guid of the device 
+static GUID device;                       ///guid of the device
 
 /***************************************************************************************/
 
@@ -221,17 +221,17 @@ static int InitDirectSound(void)
 
 	// initialize directsound
     HRESULT (WINAPI *OurDirectSoundCreate)(LPGUID, LPDIRECTSOUND *, LPUNKNOWN);
-	HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);   
+	HRESULT (WINAPI *OurDirectSoundEnumerate)(LPDSENUMCALLBACKA, LPVOID);
 	int device_index=0;
 	opt_t subopts[] = {
 	  {"device", OPT_ARG_INT, &device_num,NULL},
 	  {NULL}
-	}; 
+	};
 	if (subopt_parse(ao_subdevice, subopts) != 0) {
 		print_help();
 		return 0;
 	}
-    
+
 	hdsound_dll = LoadLibrary("DSOUND.DLL");
 	if (hdsound_dll == NULL) {
 		mp_msg(MSGT_AO, MSGL_ERR, "ao_dsound: cannot load DSOUND.DLL\n");
@@ -245,7 +245,7 @@ static int InitDirectSound(void)
 		FreeLibrary(hdsound_dll);
 		return 0;
 	}
-    
+
 	// Enumerate all directsound devices
 	mp_msg(MSGT_AO, MSGL_V,"ao_dsound: Output Devices:\n");
 	OurDirectSoundEnumerate(DirectSoundEnum,&device_index);
@@ -310,22 +310,22 @@ static void DestroyBuffer(void)
 static int write_buffer(unsigned char *data, int len)
 {
   HRESULT res;
-  LPVOID lpvPtr1; 
-  DWORD dwBytes1; 
-  LPVOID lpvPtr2; 
-  DWORD dwBytes2; 
-	
+  LPVOID lpvPtr1;
+  DWORD dwBytes1;
+  LPVOID lpvPtr2;
+  DWORD dwBytes2;
+
   // Lock the buffer
-  res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0); 
-  // If the buffer was lost, restore and retry lock. 
-  if (DSERR_BUFFERLOST == res) 
-  { 
+  res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
+  // If the buffer was lost, restore and retry lock.
+  if (DSERR_BUFFERLOST == res)
+  {
     IDirectSoundBuffer_Restore(hdsbuf);
 	res = IDirectSoundBuffer_Lock(hdsbuf,write_offset, len, &lpvPtr1, &dwBytes1, &lpvPtr2, &dwBytes2, 0);
   }
- 
-  
-  if (SUCCEEDED(res)) 
+
+
+  if (SUCCEEDED(res))
   {
   	if( (ao_data.channels == 6) && (ao_data.format!=AF_FORMAT_AC3) ) {
   	    // reorder channels while writing to pointers.
@@ -354,27 +354,27 @@ static int write_buffer(unsigned char *data, int len)
   	    write_offset+=dwBytes1+dwBytes2;
   	    if(write_offset>=buffer_size)write_offset=dwBytes2;
   	} else {
-  	    // Write to pointers without reordering. 
+  	    // Write to pointers without reordering.
 	fast_memcpy(lpvPtr1,data,dwBytes1);
     if (NULL != lpvPtr2 )fast_memcpy(lpvPtr2,data+dwBytes1,dwBytes2);
 	write_offset+=dwBytes1+dwBytes2;
     if(write_offset>=buffer_size)write_offset=dwBytes2;
   	}
-	
-   // Release the data back to DirectSound. 
+
+   // Release the data back to DirectSound.
     res = IDirectSoundBuffer_Unlock(hdsbuf,lpvPtr1,dwBytes1,lpvPtr2,dwBytes2);
-    if (SUCCEEDED(res)) 
-    { 
-	  // Success. 
+    if (SUCCEEDED(res))
+    {
+	  // Success.
 	  DWORD status;
 	  IDirectSoundBuffer_GetStatus(hdsbuf, &status);
       if (!(status & DSBSTATUS_PLAYING)){
 	    res = IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
 	  }
-	  return dwBytes1+dwBytes2; 
-    } 
-  } 
-  // Lock, Unlock, or Restore failed. 
+	  return dwBytes1+dwBytes2;
+    }
+  }
+  // Lock, Unlock, or Restore failed.
   return 0;
 }
 
@@ -408,7 +408,7 @@ static int control(int cmd, void *arg)
 	return -1;
 }
 
-/** 
+/**
 \brief setup sound device
 \param rate samplerate
 \param channels number of channels
@@ -436,7 +436,7 @@ static int init(int rate, int channels, int format, int flags)
 		default:
 			mp_msg(MSGT_AO, MSGL_V,"ao_dsound: format %s not supported defaulting to Signed 16-bit Little-Endian\n",af_fmt2str_short(format));
 			format=AF_FORMAT_S16_LE;
-	}   	
+	}
 	//fill global ao_data
 	ao_data.channels = channels;
 	ao_data.samplerate = rate;
@@ -493,7 +493,7 @@ static int init(int rate, int channels, int format, int flags)
 	ao_data.outburst = wformat.Format.nBlockAlign * 512;
 
 	// create primary buffer and set its format
-    
+
 	res = IDirectSound_CreateSoundBuffer( hds, &dsbpridesc, &hdspribuf, NULL );
 	if ( res != DS_OK ) {
 		UninitDirectSound();
@@ -553,7 +553,7 @@ static void audio_resume(void)
 	IDirectSoundBuffer_Play(hdsbuf, 0, 0, DSBPLAY_LOOPING);
 }
 
-/** 
+/**
 \brief close audio device
 \param immed stop playback immediately
 */
@@ -579,7 +579,7 @@ static int get_space(void)
 	int space;
 	DWORD play_offset;
 	IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
-	space=buffer_size-(write_offset-play_offset);                                             
+	space=buffer_size-(write_offset-play_offset);
 	//                |                                                      | <-- const --> |                |                 |
 	//                buffer start                                           play_cursor     write_cursor     write_offset      buffer end
 	// play_cursor is the actual postion of the play cursor
@@ -601,10 +601,10 @@ static int play(void* data, int len, int flags)
 {
 	DWORD play_offset;
 	int space;
-  
+
 	// make sure we have enough space to write data
 	IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
-	space=buffer_size-(write_offset-play_offset);                                             
+	space=buffer_size-(write_offset-play_offset);
 	if(space > buffer_size)space -= buffer_size; // write_offset < play_offset
 	if(space < len) len = space;
 
@@ -622,7 +622,7 @@ static float get_delay(void)
 	DWORD play_offset;
 	int space;
 	IDirectSoundBuffer_GetCurrentPosition(hdsbuf,&play_offset,NULL);
-	space=play_offset-write_offset;                                             
+	space=play_offset-write_offset;
 	if(space <= 0)space += buffer_size;
 	return (float)(buffer_size - space) / (float)ao_data.bps;
 }
diff --git a/libao2/ao_dxr2.c b/libao2/ao_dxr2.c
index 51cd1a0d01..e8aa65aab5 100644
--- a/libao2/ao_dxr2.c
+++ b/libao2/ao_dxr2.c
@@ -67,7 +67,7 @@ static int control(int cmd,void *arg){
       ao_control_vol_t* vol = (ao_control_vol_t*)arg;
       // We need this trick because the volume stepping is often too small
       diff = ((vol->left+vol->right) / 2 - (volume*19.0/100.0)) * 19.0 / 100.0;
-      v.arg = volume + (diff > 0 ? ceil(diff) : floor(diff)); 
+      v.arg = volume + (diff > 0 ? ceil(diff) : floor(diff));
       if(v.arg > 19) v.arg = 19;
       if(v.arg < 0) v.arg = 0;
       if(v.arg != volume) {
@@ -95,7 +95,7 @@ static int init(int rate,int channels,int format,int flags){
 	  return 0;
 
         last_freq_id = -1;
-        
+
 	ao_data.outburst=2048;
 	ao_data.samplerate=rate;
 	ao_data.channels=channels;
@@ -178,11 +178,11 @@ static int get_space(void){
 static void dxr2_send_lpcm_packet(unsigned char* data,int len,int id,unsigned int timestamp,int freq_id)
 {
   int write_dxr2(const unsigned char *data, int len);
-  
+
   if(dxr2_fd < 0) {
     mp_msg(MSGT_AO,MSGL_ERR,"DXR2 fd is not valid\n");
     return;
-  }    
+  }
 
   if(last_freq_id != freq_id) {
     ioctl(dxr2_fd, DXR2_IOC_SET_AUDIO_SAMPLE_FREQUENCY, &freq_id);
@@ -220,4 +220,3 @@ static float get_delay(void){
 
     return 0.0;
 }
-
diff --git a/libao2/ao_esd.c b/libao2/ao_esd.c
index 7a9ad96170..28a0009b80 100644
--- a/libao2/ao_esd.c
+++ b/libao2/ao_esd.c
@@ -122,7 +122,7 @@ static int control(int cmd, void *arg)
 	    vol_cache_time = now;
 	}
 	esd_free_all_info(esd_i);
-	
+
 	return CONTROL_OK;
 
     case AOCONTROL_SET_VOLUME:
@@ -236,19 +236,19 @@ static int init(int rate_hz, int channels, int format, int flags)
 #ifdef CONFIG_ESD_LATENCY
     esd_latency = esd_get_latency(esd_fd);
 #else
-    esd_latency = ((channels == 1 ? 2 : 1) * ESD_DEFAULT_RATE * 
+    esd_latency = ((channels == 1 ? 2 : 1) * ESD_DEFAULT_RATE *
 		   (ESD_BUF_SIZE + 64 * (4.0f / bytes_per_sample))
-		   ) / rate_hz;  
-    esd_latency += ESD_BUF_SIZE * 2; 
+		   ) / rate_hz;
+    esd_latency += ESD_BUF_SIZE * 2;
 #endif
     if(esd_latency > 0) {
 	lag_serv = (esd_latency * 4.0f) / (bytes_per_sample * rate_hz);
 	lag_seconds = lag_net + lag_serv;
 	audio_delay += lag_seconds;
-	mp_tmsg(MSGT_AO, MSGL_INFO,"[AO ESD] latency: [server: %0.2fs, net: %0.2fs] (adjust %0.2fs)\n", 
+	mp_tmsg(MSGT_AO, MSGL_INFO,"[AO ESD] latency: [server: %0.2fs, net: %0.2fs] (adjust %0.2fs)\n",
 	       lag_serv, lag_net, lag_seconds);
     }
-    
+
     esd_play_fd = esd_play_stream_fallback(esd_fmt, rate_hz,
 					   server, ESD_CLIENT_NAME);
     if (esd_play_fd < 0) {
@@ -333,7 +333,7 @@ static int play(void* data, int len, int flags)
 	 */
 	n = write(esd_play_fd, (char*)data + offs, ESD_BUF_SIZE);
 	if ( n < 0 ) {
-	    if ( errno != EAGAIN ) 
+	    if ( errno != EAGAIN )
 		dprintf("esd play: write failed: %s\n", strerror(errno));
 	    break;
 	} else if ( n != ESD_BUF_SIZE ) {
@@ -343,13 +343,13 @@ static int play(void* data, int len, int flags)
 	    nwritten += n;
     }
 #endif
-	
+
     if (nwritten > 0) {
 	if (!esd_play_start.tv_sec)
 	    gettimeofday(&esd_play_start, NULL);
 	nsamples = nwritten / esd_bytes_per_sample;
 	esd_samples_written += nsamples;
- 
+
 	dprintf("esd play: %d %lu\n", nsamples, esd_samples_written);
     } else {
 	dprintf("esd play: blocked / %lu\n", esd_samples_written);
@@ -395,7 +395,7 @@ static void reset(void)
 {
 #ifdef	__svr4__
     /* throw away data buffered in the esd connection */
-    if (ioctl(esd_play_fd, I_FLUSH, FLUSHW)) 
+    if (ioctl(esd_play_fd, I_FLUSH, FLUSHW))
 	perror("I_FLUSH");
 #endif
 }
@@ -411,7 +411,7 @@ static int get_space(void)
     float current_delay;
     int space;
 
-    /* 
+    /*
      * Don't buffer too much data in the esd daemon.
      *
      * If we send too much, esd will block in write()s to the sound
@@ -461,7 +461,7 @@ static float get_delay(void)
     gettimeofday(&now, NULL);
     play_time  =  now.tv_sec  - esd_play_start.tv_sec;
     play_time += (now.tv_usec - esd_play_start.tv_usec) / 1000000.;
-    
+
     /* dprintf("esd delay: %f %f\n", play_time, buffered_samples_time); */
 
     if (play_time > buffered_samples_time) {
diff --git a/libao2/ao_ivtv.c b/libao2/ao_ivtv.c
index 22e2f49a74..6b6bb8816c 100644
--- a/libao2/ao_ivtv.c
+++ b/libao2/ao_ivtv.c
@@ -41,7 +41,7 @@
 
 static int freq = 0;
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
   "IVTV MPEG Audio Decoder output",
   "ivtv",
@@ -73,7 +73,7 @@ init (int rate, int channels, int format, int flags)
             "AO: [ivtv] can only handle MPEG audio streams.\n");
     return 0;
   }
-  
+
   ao_data.outburst = 2048;
   ao_data.samplerate = rate;
   ao_data.channels = channels;
@@ -132,11 +132,11 @@ get_space (void)
   x = (float) (vo_pts - ao_data.pts) / 90000.0;
   if (x <= 0)
     return 0;
-  
+
   y  = freq * 4 * x;
   y /= ao_data.outburst;
   y *= ao_data.outburst;
-  
+
   if (y > 32000)
     y = 32000;
 
@@ -148,7 +148,7 @@ static int
 play (void *data, int len, int flags)
 {
   int ivtv_write (const unsigned char *data, int len);
-  
+
   if (ao_data.format != AF_FORMAT_MPEG2)
     return 0;
 
diff --git a/libao2/ao_jack.c b/libao2/ao_jack.c
index 8ee5550602..14bb1c90dd 100644
--- a/libao2/ao_jack.c
+++ b/libao2/ao_jack.c
@@ -40,7 +40,7 @@
 
 #include <jack/jack.h>
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
   "JACK audio output",
   "jack",
@@ -360,4 +360,3 @@ static float get_delay(void) {
   }
   return (float)buffered / (float)ao_data.bps + in_jack;
 }
-
diff --git a/libao2/ao_mpegpes.c b/libao2/ao_mpegpes.c
index e8c0f01d14..096f7bade5 100644
--- a/libao2/ao_mpegpes.c
+++ b/libao2/ao_mpegpes.c
@@ -63,7 +63,7 @@ int vo_mpegpes_fd2 = -1;
 
 #include <errno.h>
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
 #ifdef CONFIG_DVB
 	"DVB audio output",
@@ -194,7 +194,7 @@ static int preinit(const char *arg)
 #ifdef CONFIG_DVB
 	if(!ao_file)
 		return init_device(card);
-#else	
+#else
 	if(!ao_file)
 		return vo_mpegpes_fd;	//video fd
 #endif
diff --git a/libao2/ao_nas.c b/libao2/ao_nas.c
index e5e675be3f..fb49c5e60e 100644
--- a/libao2/ao_nas.c
+++ b/libao2/ao_nas.c
@@ -119,7 +119,7 @@ static const char* nas_state(unsigned int state) {
 	return nas_states[state];
 }
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
 	"NAS audio output",
 	"nas",
@@ -201,7 +201,7 @@ static int nas_readBuffer(struct ao_nas_data *nas_data, unsigned int num)
 	 * Now write the new buffer to the network.
 	 */
 	AuWriteElement(nas_data->aud, nas_data->flow, 0, num, nas_data->server_buffer, AuFalse, &as);
-	if (as != AuSuccess) 
+	if (as != AuSuccess)
 		nas_print_error(nas_data->aud, "nas_readBuffer(): AuWriteElement", as);
 
 	return num;
@@ -229,7 +229,7 @@ static int nas_empty_event_queue(struct ao_nas_data *nas_data)
 {
 	AuEvent ev;
 	int result = 0;
-	
+
 	while (AuScanForTypedEvent(nas_data->aud, AuEventsQueuedAfterFlush,
 				   AuTrue, AuEventTypeElementNotify, &ev)) {
 		AuDispatchEvent(nas_data->aud, &ev);
@@ -462,7 +462,7 @@ static int init(int rate,int channels,int format,int flags)
 	mp_msg(MSGT_AO, MSGL_V, "ao_nas: init(): Using audioserver %s\n", server);
 
 	nas_data->aud = AuOpenServer(server, 0, NULL, 0, NULL, NULL);
-	if (!nas_data->aud) { 
+	if (!nas_data->aud) {
 		mp_msg(MSGT_AO, MSGL_ERR, "ao_nas: init(): Can't open nas audio server -> nosound\n");
 		return 0;
 	}
@@ -571,7 +571,7 @@ static void audio_resume(void)
 static int get_space(void)
 {
 	int result;
-	
+
 	mp_msg(MSGT_AO, MSGL_DBG3, "ao_nas: get_space()\n");
 
 	pthread_mutex_lock(&nas_data->buffer_mutex);
@@ -631,7 +631,7 @@ static int play(void* data,int len,int flags)
 static float get_delay(void)
 {
 	float result;
-	
+
 	mp_msg(MSGT_AO, MSGL_DBG3, "ao_nas: get_delay()\n");
 
 	pthread_mutex_lock(&nas_data->buffer_mutex);
diff --git a/libao2/ao_null.c b/libao2/ao_null.c
index 20b715fb96..17b3bc7e4e 100644
--- a/libao2/ao_null.c
+++ b/libao2/ao_null.c
@@ -27,7 +27,7 @@
 #include "audio_out.h"
 #include "audio_out_internal.h"
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
 	"Null audio output",
 	"null",
@@ -41,14 +41,14 @@ struct	timeval last_tv;
 int	buffer;
 
 static void drain(void){
- 
+
     struct timeval now_tv;
     int temp, temp2;
 
     gettimeofday(&now_tv, 0);
     temp = now_tv.tv_sec - last_tv.tv_sec;
     temp *= ao_data.bps;
-    
+
     temp2 = now_tv.tv_usec - last_tv.tv_usec;
     temp2 /= 1000;
     temp2 *= ao_data.bps;
diff --git a/libao2/ao_openal.c b/libao2/ao_openal.c
index c80c49b27a..b00a77364a 100644
--- a/libao2/ao_openal.c
+++ b/libao2/ao_openal.c
@@ -42,7 +42,7 @@
 #include "osdep/timer.h"
 #include "subopt-helper.h"
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
   "OpenAL audio output",
   "openal",
@@ -251,4 +251,3 @@ static float get_delay(void) {
   alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
   return queued * CHUNK_SIZE / 2 / (float)ao_data.samplerate;
 }
-
diff --git a/libao2/ao_oss.c b/libao2/ao_oss.c
index c331f8331b..c16d55c8df 100644
--- a/libao2/ao_oss.c
+++ b/libao2/ao_oss.c
@@ -48,7 +48,7 @@
 #include "audio_out.h"
 #include "audio_out_internal.h"
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
 	"OSS/ioctl audio output",
 	"oss",
@@ -228,7 +228,7 @@ static int control(int cmd,void *arg){
 
 	    if(ao_data.format == AF_FORMAT_AC3)
 		return CONTROL_TRUE;
-    
+
 	    if ((fd = open(oss_mixer_device, O_RDONLY)) > 0)
 	    {
 		ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
@@ -289,17 +289,17 @@ static int init(int rate,int channels,int format,int flags){
     oss_mixer_device=mdev;
   else
     oss_mixer_device=PATH_DEV_MIXER;
-  
+
   if(mchan){
     int fd, devs, i;
-    
+
     if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){
       mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't open mixer device %s: %s\n",
         oss_mixer_device, strerror(errno));
     }else{
       ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs);
       close(fd);
-      
+
       for (i=0; i<SOUND_MIXER_NRDEVICES; i++){
         if(!strcasecmp(mixer_channels[i], mchan)){
           if(!(devs & (1 << i))){
@@ -337,19 +337,19 @@ static int init(int rate,int channels,int format,int flags){
   if(fcntl(audio_fd, F_SETFL, 0) < 0) {
    mp_tmsg(MSGT_AO,MSGL_ERR,"[AO OSS] audio_setup: Can't make file descriptor blocking: %s\n", strerror(errno));
    return 0;
-  }  
+  }
 #endif
 
 #if defined(FD_CLOEXEC) && defined(F_SETFD)
   fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
 #endif
-  
+
   if(format == AF_FORMAT_AC3) {
     ao_data.samplerate=rate;
     ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
   }
 
-ac3_retry:  
+ac3_retry:
   ao_data.format=format;
   oss_format=format2oss(format);
   if (oss_format == -1) {
@@ -377,7 +377,7 @@ ac3_retry:
 
   mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n",
     af_fmt2str_short(ao_data.format), af_fmt2str_short(format));
-  
+
   ao_data.channels = channels;
   if(format != AF_FORMAT_AC3) {
     // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
diff --git a/libao2/ao_pcm.c b/libao2/ao_pcm.c
index 8940107eec..8be3bdabb5 100644
--- a/libao2/ao_pcm.c
+++ b/libao2/ao_pcm.c
@@ -35,7 +35,7 @@
 #include "help_mp.h"
 
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
     "RAW PCM/WAVE file writer audio output",
     "pcm",
@@ -149,13 +149,13 @@ static int init(int rate,int channels,int format,int flags){
     wavhdr.bytes_per_second = le2me_32(ao_data.bps);
     wavhdr.bits = le2me_16(bits);
     wavhdr.block_align = le2me_16(ao_data.channels * (bits / 8));
-    
+
     wavhdr.data = le2me_32(WAV_ID_DATA);
     wavhdr.data_length=le2me_32(0x7ffff000);
     wavhdr.file_length = wavhdr.data_length + sizeof(wavhdr) - 8;
 
-    mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\nPCM: Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename, 
-           (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate, 
+    mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] File: %s (%s)\nPCM: Samplerate: %iHz Channels: %s Format %s\n", ao_outputfilename,
+           (ao_pcm_waveheader?"WAVE":"RAW PCM"), rate,
            (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
     mp_tmsg(MSGT_AO, MSGL_INFO, "[AO PCM] Info: Faster dumping is achieved with -vc null -vo null -ao pcm:fast\n[AO PCM] Info: To write WAVE files use -ao pcm:waveheader (default).\n");
 
@@ -166,14 +166,14 @@ static int init(int rate,int channels,int format,int flags){
         }
         return 1;
     }
-    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO PCM] Failed to open %s for writing!\n", 
+    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO PCM] Failed to open %s for writing!\n",
                ao_outputfilename);
     return 0;
 }
 
 // close audio device
 static void uninit(int immed){
-    
+
     if(ao_pcm_waveheader){ /* Rewrite wave header */
         if (fseek(fp, 0, SEEK_SET) != 0)
             mp_msg(MSGT_AO, MSGL_ERR, "Could not seek to start, WAV size headers not updated!\n");
@@ -232,7 +232,7 @@ static int play(void* data,int len,int flags){
         buffer[i] = le2me_16(buffer[i]);
       }
     }
-#endif 
+#endif
 
     if (ao_data.channels == 6 || ao_data.channels == 5) {
         int frame_size = le2me_16(wavhdr.bits) / 8;
@@ -248,7 +248,7 @@ static int play(void* data,int len,int flags){
 
     if(ao_pcm_waveheader)
         data_length += len;
-    
+
     return len;
 }
 
@@ -257,9 +257,3 @@ static float get_delay(void){
 
     return 0.0;
 }
-
-
-
-
-
-
diff --git a/libao2/ao_sdl.c b/libao2/ao_sdl.c
index 66d37eae77..b5c56d9f33 100644
--- a/libao2/ao_sdl.c
+++ b/libao2/ao_sdl.c
@@ -36,7 +36,7 @@
 
 #include "libavutil/fifo.h"
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
 	"SDLlib audio output",
 	"sdl",
@@ -129,7 +129,7 @@ static int init(int rate,int channels,int format,int flags){
 
 	/* SDL Audio Specifications */
 	SDL_AudioSpec aspec, obtained;
-	
+
 	/* Allocate ring-buffer memory */
 	buffer = av_fifo_alloc(BUFFSIZE);
 
@@ -147,7 +147,7 @@ static int init(int rate,int channels,int format,int flags){
 	ao_data.bps=channels*rate;
 	if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8)
 	  ao_data.bps*=2;
-	
+
 	/* The desired audio format (see SDL_AudioSpec) */
 	switch(format) {
 	    case AF_FORMAT_U8:
@@ -200,7 +200,7 @@ void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer s
 	if(SDL_OpenAudio(&aspec, &obtained) < 0) {
         	mp_tmsg(MSGT_AO,MSGL_ERR,"[AO SDL] Unable to open audio: %s\n", SDL_GetError());
         	return 0;
-	} 
+	}
 
 	/* did we got what we wanted ? */
 	ao_data.channels=obtained.channels;
@@ -233,7 +233,7 @@ void callback(void *userdata, Uint8 *stream, int len); userdata is the pointer s
 	mp_msg(MSGT_AO,MSGL_V,"SDL: buf size = %d\n",obtained.size);
 	ao_data.buffersize=obtained.size;
 	ao_data.outburst = CHUNK_SIZE;
-	
+
 	/* unsilence audio, if callback is ready */
 	SDL_PauseAudio(0);
 
@@ -253,7 +253,7 @@ static void uninit(int immed){
 // stop playing and empty buffers (for seeking/pause)
 static void reset(void){
 
-	//printf("SDL: reset called!\n");	
+	//printf("SDL: reset called!\n");
 
 	SDL_PauseAudio(1);
 	/* Reset ring-buffer state */
@@ -265,15 +265,15 @@ static void reset(void){
 static void audio_pause(void)
 {
 
-	//printf("SDL: audio_pause called!\n");	
+	//printf("SDL: audio_pause called!\n");
 	SDL_PauseAudio(1);
-	
+
 }
 
 // resume playing, after audio_pause()
 static void audio_resume(void)
 {
-	//printf("SDL: audio_resume called!\n");	
+	//printf("SDL: audio_resume called!\n");
 	SDL_PauseAudio(0);
 }
 
@@ -290,12 +290,12 @@ static int play(void* data,int len,int flags){
 
 	if (!(flags & AOPLAY_FINAL_CHUNK))
 	len = (len/ao_data.outburst)*ao_data.outburst;
-#if 0	
+#if 0
 	int ret;
 
 	/* Audio locking prohibits call of outputaudio */
 	SDL_LockAudio();
-	// copy audio stream into ring-buffer 
+	// copy audio stream into ring-buffer
 	ret = write_buffer(data, len);
 	SDL_UnlockAudio();
 
@@ -310,9 +310,3 @@ static float get_delay(void){
     int buffered = av_fifo_size(buffer); // could be less
     return (float)(buffered + ao_data.buffersize)/(float)ao_data.bps;
 }
-
-
-
-
-
-
diff --git a/libao2/ao_sgi.c b/libao2/ao_sgi.c
index 8a2899d915..71ce784657 100644
--- a/libao2/ao_sgi.c
+++ b/libao2/ao_sgi.c
@@ -32,7 +32,7 @@
 #include "help_mp.h"
 #include "libaf/af_format.h"
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
 	"sgi audio output",
 	"sgi",
@@ -116,9 +116,9 @@ static int fmt2sgial(int *format, int *width) {
 
 // to set/get/query special features/parameters
 static int control(int cmd, void *arg){
-  
+
   mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] control.\n");
-  
+
   switch(cmd) {
   case AOCONTROL_QUERY_FORMAT:
     /* Do not reject any format: return the closest matching
@@ -139,9 +139,9 @@ static int init(int rate, int channels, int format, int flags) {
   smpfmt = fmt2sgial(&format, &smpwidth);
 
   mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: Samplerate: %iHz Channels: %s Format %s\n", rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format));
-  
+
   { /* from /usr/share/src/dmedia/audio/setrate.c */
-  
+
     double frate, realrate;
     ALpv x[2];
 
@@ -152,9 +152,9 @@ static int init(int rate, int channels, int format, int flags) {
 	return 0;
       }
     }
-    
+
     frate = rate;
-   
+
     x[0].param = AL_RATE;
     x[0].value.ll = alDoubleToFixed(rate);
     x[1].param = AL_MASTER_CLOCK;
@@ -163,7 +163,7 @@ static int init(int rate, int channels, int format, int flags) {
     if (alSetParams(rv,x, 2)<0) {
       mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: setparams failed: %s\nCould not set desired samplerate.\n", alGetErrorString(oserror()));
     }
-    
+
     if (x[0].sizeOut < 0) {
       mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: AL_RATE was not accepted on the given resource.\n");
     }
@@ -171,14 +171,14 @@ static int init(int rate, int channels, int format, int flags) {
     if (alGetParams(rv,x, 1)<0) {
       mp_tmsg(MSGT_AO, MSGL_WARN, "[AO SGI] init: getparams failed: %s\n", alGetErrorString(oserror()));
     }
-    
+
     realrate = alFixedToDouble(x[0].value.ll);
     if (frate != realrate) {
       mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] init: samplerate is now %lf (desired rate is %lf)\n", realrate, frate);
-    } 
+    }
     sample_rate = (int)realrate;
   }
-  
+
   bytes_per_frame = channels * smpwidth;
 
   ao_data.samplerate = sample_rate;
@@ -187,14 +187,14 @@ static int init(int rate, int channels, int format, int flags) {
   ao_data.bps = sample_rate * bytes_per_frame;
   ao_data.buffersize=131072;
   ao_data.outburst = ao_data.buffersize/16;
-  
+
   ao_config = alNewConfig();
-  
+
   if (!ao_config) {
     mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
     return 0;
   }
-  
+
   if(alSetChannels(ao_config, channels) < 0 ||
      alSetWidth(ao_config, smpwidth) < 0 ||
      alSetSampFmt(ao_config, smpfmt) < 0 ||
@@ -203,17 +203,17 @@ static int init(int rate, int channels, int format, int flags) {
     mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: %s\n", alGetErrorString(oserror()));
     return 0;
   }
-  
+
   ao_port = alOpenPort("mplayer", "w", ao_config);
-  
+
   if (!ao_port) {
     mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SGI] init: Unable to open audio channel: %s\n", alGetErrorString(oserror()));
     return 0;
   }
-  
+
   // printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config);
   queue_size = alGetQueueSize(ao_config);
-  return 1;  
+  return 1;
 
 }
 
@@ -231,26 +231,26 @@ static void uninit(int immed) {
 
   if (ao_port) {
     if (!immed)
-    while(alGetFilled(ao_port) > 0) sginap(1);  
+    while(alGetFilled(ao_port) > 0) sginap(1);
     alClosePort(ao_port);
     ao_port = NULL;
   }
-	
+
 }
 
 // stop playing and empty buffers (for seeking/pause)
 static void reset(void) {
-  
+
   mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] reset: ...\n");
-  
+
   alDiscardFrames(ao_port, queue_size);
 }
 
 // stop playing, keep buffers (for pause)
 static void audio_pause(void) {
-    
+
   mp_tmsg(MSGT_AO, MSGL_INFO, "[AO SGI] audio_pause: ...\n");
-    
+
 }
 
 // resume playing, after audio_pause()
@@ -262,12 +262,12 @@ static void audio_resume(void) {
 
 // return: how many bytes can be played without blocking
 static int get_space(void) {
-  
+
   // printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_data.outburst);
   // printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port));
-  
+
   return alGetFillable(ao_port) * bytes_per_frame;
-    
+
 }
 
 
@@ -275,7 +275,7 @@ static int get_space(void) {
 // it should round it down to outburst*n
 // return: number of bytes played
 static int play(void* data, int len, int flags) {
-    
+
   /* Always process data in quadword-aligned chunks (64-bits). */
   const int plen = len / (sizeof(uint64_t) * bytes_per_frame);
   const int framecount = plen * sizeof(uint64_t);
@@ -294,22 +294,16 @@ static int play(void* data, int len, int flags) {
   alWriteFrames(ao_port, data, framecount);
 
   return framecount * bytes_per_frame;
-  
+
 }
 
 // return: delay in seconds between first and last sample in buffer
 static float get_delay(void){
-  
+
   // printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize);
-  
+
   // return  (float)queue_size/((float)sample_rate);
   const int outstanding = alGetFilled(ao_port);
   return (float)((outstanding < 0) ? queue_size : outstanding) /
     ((float)sample_rate);
 }
-
-
-
-
-
-
diff --git a/libao2/ao_sun.c b/libao2/ao_sun.c
index d2da673177..72353bc736 100644
--- a/libao2/ao_sun.c
+++ b/libao2/ao_sun.c
@@ -49,7 +49,7 @@
 #include "mp_msg.h"
 #include "help_mp.h"
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
     "Sun audio output",
     "sun",
@@ -142,7 +142,7 @@ static int realtime_samplecounter_available(char *dev)
     silence = calloc(1, len);
     if (silence == NULL)
 	goto error;
-    
+
     if ((fd = open(dev, O_WRONLY)) < 0)
 	goto error;
 
@@ -157,7 +157,7 @@ static int realtime_samplecounter_available(char *dev)
 	    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] rtsc: SETINFO failed.\n");
 	goto error;
     }
-    
+
     if (write(fd, silence, len) != len) {
 	if ( mp_msg_test(MSGT_AO,MSGL_V) )
 	    mp_tmsg(MSGT_AO, MSGL_ERR, "[AO SUN] rtsc: write failed.\n");
@@ -215,7 +215,7 @@ static int realtime_samplecounter_available(char *dev)
      * sample counter increment from the soundcard driver of less than
      * 2000 samples,  we assume that the driver provides a useable realtime
      * sample counter in the AUDIO_INFO play.samples field.  Timing based
-     * on sample counts should be much more accurate than counting whole 
+     * on sample counts should be much more accurate than counting whole
      * 16kbyte chunks.
      */
     if (min_increment < 2000)
@@ -225,7 +225,7 @@ static int realtime_samplecounter_available(char *dev)
 	mp_msg(MSGT_AO,MSGL_V,"ao_sun: minimum sample counter increment per 10msec interval: %d\n"
 	       "\t%susing sample counter based timing code\n",
 	       min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not ");
-    
+
 
 error:
     if (silence != NULL) free(silence);
@@ -272,7 +272,7 @@ find_close_samplerate_match(int dev, unsigned sample_rate)
 
     if (sr->flags & MIXER_SR_LIMITS) {
 	/*
-	 * HW can playback any rate between 
+	 * HW can playback any rate between
 	 * sr->samp_rates[0] .. sr->samp_rates[1]
 	 */
 	free(sr);
@@ -318,7 +318,7 @@ find_close_samplerate_match(int dev, unsigned sample_rate)
     for (i = 0; audiocs_rates[i]; i++) {
 	err = abs(audiocs_rates[i] - sample_rate);
 	if (err == 0) {
-	    /* 
+	    /*
 	     * exact supported sample rate match, no need to
 	     * retry something elise
 	     */
@@ -361,7 +361,7 @@ find_highest_samplerate(int dev)
 
     if (sr->flags & MIXER_SR_LIMITS) {
 	/*
-	 * HW can playback any rate between 
+	 * HW can playback any rate between
 	 * sr->samp_rates[0] .. sr->samp_rates[1]
 	 */
 	max_rate = sr->samp_rates[1];
@@ -435,7 +435,7 @@ static int control(int cmd,void *arg){
 	    }
 	    close( fd );
 	    return CONTROL_OK;
-	}	
+	}
 	return CONTROL_ERROR;
     }
     case AOCONTROL_SET_VOLUME:
@@ -466,7 +466,7 @@ static int control(int cmd,void *arg){
 	    ioctl( fd,AUDIO_SETINFO,&info );
 	    close( fd );
 	    return CONTROL_OK;
-	}	
+	}
 	return CONTROL_ERROR;
     }
     }
@@ -516,7 +516,7 @@ static int init(int rate,int channels,int format,int flags){
 
 	if (pass & 1) {
 	    /*
-	     * on some sun audio drivers, 8-bit unsigned LINEAR8 encoding is 
+	     * on some sun audio drivers, 8-bit unsigned LINEAR8 encoding is
 	     * not supported, but 8-bit signed encoding is.
 	     *
 	     * Try S8, and if it works, use our own U8->S8 conversion before
@@ -539,7 +539,7 @@ static int init(int rate,int channels,int format,int flags){
 	     * supported rates,  use the fixed supported rate instead.
 	     */
 	    if (!(info.play.sample_rate =
-		  find_close_samplerate_match(audio_fd, rate))) 
+		  find_close_samplerate_match(audio_fd, rate)))
 	      continue;
 
 	    /*
@@ -694,4 +694,3 @@ static float get_delay(void){
 	return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec;
 #endif
 }
-
diff --git a/libao2/ao_v4l2.c b/libao2/ao_v4l2.c
index c1e0671835..82c53a226a 100644
--- a/libao2/ao_v4l2.c
+++ b/libao2/ao_v4l2.c
@@ -38,7 +38,7 @@
 
 static int freq = 0;
 
-static const ao_info_t info = 
+static const ao_info_t info =
 {
   "V4L2 MPEG Audio Decoder output",
   "v4l2",
@@ -70,7 +70,7 @@ init (int rate, int channels, int format, int flags)
             "AO: [v4l2] can only handle MPEG audio streams.\n");
     return 0;
   }
-  
+
   ao_data.outburst = 2048;
   ao_data.samplerate = rate;
   ao_data.channels = channels;
@@ -129,11 +129,11 @@ get_space (void)
   x = (float) (vo_pts - ao_data.pts) / 90000.0;
   if (x <= 0)
     return 0;
-  
+
   y  = freq * 4 * x;
   y /= ao_data.outburst;
   y *= ao_data.outburst;
-  
+
   if (y > 32000)
     y = 32000;
 
@@ -145,7 +145,7 @@ static int
 play (void *data, int len, int flags)
 {
   int v4l2_write (const unsigned char *data, int len);
-  
+
   if (ao_data.format != AF_FORMAT_MPEG2)
     return 0;
 
diff --git a/libao2/ao_win32.c b/libao2/ao_win32.c
index b2ba0fc573..a0475455ec 100644
--- a/libao2/ao_win32.c
+++ b/libao2/ao_win32.c
@@ -200,7 +200,7 @@ static int init(int rate,int channels,int format,int flags)
     }
 
     wformat.Format.nAvgBytesPerSec = wformat.Format.nSamplesPerSec * wformat.Format.nBlockAlign;
- 
+
     //open sound device
     //WAVE_MAPPER always points to the default wave device on the system
     result = waveOutOpen(&hWaveOut,WAVE_MAPPER,(WAVEFORMATEX*)&wformat,(DWORD_PTR)waveOutProc,0,CALLBACK_FUNCTION);
diff --git a/libao2/audio_out.c b/libao2/audio_out.c
index ab40121d76..5ceafff73f 100644
--- a/libao2/audio_out.c
+++ b/libao2/audio_out.c
@@ -188,4 +188,3 @@ const ao_functions_t* init_best_audio_out(char** ao_list,int use_plugin,int rate
     }
     return NULL;
 }
-
diff --git a/libao2/audio_out.h b/libao2/audio_out.h
index b7f51e0ed8..e483a88422 100644
--- a/libao2/audio_out.h
+++ b/libao2/audio_out.h
@@ -52,8 +52,8 @@ typedef struct ao_data {
   int channels;
   int format;
   int bps;
-  int outburst; 	  
-  int buffersize;         
+  int outburst;
+  int buffersize;
   int pts;
 } ao_data_t;