general: whitespace cleanup

src/AudioCompress/compress.c

@@ -16,16 +16,16 @@
 struct Compressor {
         //! The compressor's preferences
         struct CompressorConfig prefs;
-        
+
         //! History of the peak values
         int *peaks;
-                
+
         //! History of the gain values
         int *gain;
-        
+
         //! History of clip amounts
         int *clipped;
-        
+
         unsigned int pos;
         unsigned int bufsz;
 };
@@ -41,9 +41,9 @@ struct Compressor *Compressor_new(unsigned int history)
         obj->peaks = obj->gain = obj->clipped = NULL;
 	obj->bufsz = 0;
         obj->pos = 0;
-        
+
         Compressor_setHistory(obj, history);
-        
+
         return obj;
 }
 
@@ -70,7 +70,7 @@ void Compressor_setHistory(struct Compressor *obj, unsigned int history)
 {
 	if (!history)
                 history = BUCKETS;
-        
+
         obj->peaks = resizeArray(obj->peaks, history, obj->bufsz);
         obj->gain = resizeArray(obj->gain, history, obj->bufsz);
         obj->clipped = resizeArray(obj->clipped, history, obj->bufsz);
@@ -82,7 +82,7 @@ struct CompressorConfig *Compressor_getConfig(struct Compressor *obj)
         return &obj->prefs;
 }
 
-void Compressor_Process_int16(struct Compressor *obj, int16_t *audio, 
+void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
                               unsigned int count)
 {
         struct CompressorConfig *prefs = Compressor_getConfig(obj);
@@ -97,7 +97,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
         int *clipped = obj->clipped + slot;
         unsigned int ramp = count;
         int delta;
-        
+
 	ap = audio;
 	for (i = 0; i < count; i++)
 	{
@@ -124,15 +124,15 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
 
 	//! Determine target gain
 	newGain = (1 << 10)*prefs->target/peakVal;
-        
+
         //! Adjust the gain with inertia from the previous gain value
-        newGain = (curGain*((1 << prefs->smooth) - 1) + newGain) 
+        newGain = (curGain*((1 << prefs->smooth) - 1) + newGain)
                 >> prefs->smooth;
-        
+
         //! Make sure it's no more than the maximum gain value
         if (newGain > (prefs->maxgain << 10))
                 newGain = prefs->maxgain << 10;
-        
+
         //! Make sure it's no less than 1:1
 	if (newGain < (1 << 10))
 		newGain = 1 << 10;
@@ -144,7 +144,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
                 //! Truncate the ramp time
                 ramp = peakPos;
         }
-        
+
         //! Record the new gain
         obj->gain[slot] = newGain;
 

src/decoder/audiofile_decoder_plugin.c

@@ -244,7 +244,7 @@ static const char *const audiofile_suffixes[] = {
 static const char *const audiofile_mime_types[] = {
 	"audio/x-wav",
 	"audio/x-aiff",
-	NULL 
+	NULL
 };
 
 const struct decoder_plugin audiofile_decoder_plugin = {

src/encoder/flac_encoder.c

@@ -55,7 +55,7 @@ static bool
 flac_encoder_configure(struct flac_encoder *encoder,
 		const struct config_param *param, G_GNUC_UNUSED GError **error)
 {
-	encoder->compression = config_get_block_unsigned(param, 
+	encoder->compression = config_get_block_unsigned(param,
 						"compression", 5);
 
 	return true;
@@ -218,7 +218,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
 
 		if (init_status != FLAC__STREAM_ENCODER_OK) {
 			g_set_error(error, flac_encoder_quark(), 0,
-			    "failed to initialize encoder: %s\n", 
+			    "failed to initialize encoder: %s\n",
 			    FLAC__StreamEncoderStateString[init_status]);
 			flac_encoder_close(_encoder);
 			return false;
@@ -234,7 +234,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
 
 		if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
 			g_set_error(error, flac_encoder_quark(), 0,
-			    "failed to initialize encoder: %s\n", 
+			    "failed to initialize encoder: %s\n",
 			    FLAC__StreamEncoderInitStatusString[init_status]);
 			flac_encoder_close(_encoder);
 			return false;

src/encoder/wave_encoder.c

@@ -58,7 +58,7 @@ wave_encoder_quark(void)
 }
 
 static void
-fill_wave_header(struct wave_header *header, int channels, int bits, 
+fill_wave_header(struct wave_header *header, int channels, int bits,
 		int freq, int block_size)
 {
 	int data_size = 0x0FFFFFFF;
@@ -142,7 +142,7 @@ wave_encoder_open(struct encoder *_encoder,
 	buffer = pcm_buffer_get(&encoder->buffer, sizeof(struct wave_header) );
 
 	/* create PCM wave header in initial buffer */
-	fill_wave_header((struct wave_header *) buffer, 
+	fill_wave_header((struct wave_header *) buffer,
 			audio_format->channels,
 			 encoder->bits,
 			audio_format->sample_rate,

src/mixer/winmm_mixer_plugin.c

@@ -58,11 +58,11 @@ winmm_mixer_init(void *ao, G_GNUC_UNUSED const struct config_param *param,
 		 G_GNUC_UNUSED GError **error_r)
 {
 	assert(ao != NULL);
-	
+
 	struct winmm_mixer *wm = g_new(struct winmm_mixer, 1);
 	mixer_init(&wm->base, &winmm_mixer_plugin);
 	wm->output = (struct winmm_output *) ao;
-	
+
 	return &wm->base;
 }
 
@@ -79,13 +79,13 @@ winmm_mixer_get_volume(struct mixer *mixer, GError **error_r)
 	DWORD volume;
 	HWAVEOUT handle = winmm_output_get_handle(wm->output);
 	MMRESULT result = waveOutGetVolume(handle, &volume);
-	
+
 	if (result != MMSYSERR_NOERROR) {
 		g_set_error(error_r, 0, winmm_mixer_quark(),
 			    "Failed to get winmm volume");
 		return -1;
 	}
-	
+
 	return winmm_volume_decode(volume);
 }
 
@@ -102,7 +102,7 @@ winmm_mixer_set_volume(struct mixer *mixer, unsigned volume, GError **error_r)
 			    "Failed to set winmm volume");
 		return false;
 	}
-	
+
 	return true;
 }
 

src/output/httpd_internal.h

@@ -111,7 +111,7 @@ struct httpd_output {
 	char buffer[32768];
 
 	/**
-	 * The maximum and current number of clients connected 
+	 * The maximum and current number of clients connected
 	 * at the same time.
 	 */
 	guint clients_max, clients_cnt;

src/output/mvp_plugin.c

@@ -17,7 +17,7 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-/* 
+/*
  * Media MVP audio output based on code from MVPMC project:
  * http://mvpmc.sourceforge.net/
  */

src/output/oss_plugin.c

@@ -347,7 +347,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format,
 		case SUCCESS:
 			if (!audio_valid_sample_rate(sample_rate))
 				break;
-		
+
 			audio_format->sample_rate = sample_rate;
 			return true;
 

src/pcm_byteswap.c

@@ -49,7 +49,7 @@ const int16_t *pcm_byteswap_16(struct pcm_buffer *buffer,
 
 static inline uint32_t swab32(uint32_t x)
 {
-	return (x << 24) | 
+	return (x << 24) |
 		((x & 0xff00) << 8) |
 		((x & 0xff0000) >> 8) |
 		(x >> 24);