Remove trailing whitespace from all files (#497)

dsp/mpeg/README.txt

@@ -37,10 +37,10 @@ This library provides several interfaces to load, demux and decode MPEG video
 and audio data. A high-level API combines the demuxer, video & audio decoders
 in an easy to use wrapper.
 
-Lower-level APIs for accessing the demuxer, video decoder and audio decoder, 
+Lower-level APIs for accessing the demuxer, video decoder and audio decoder,
 as well as providing different data sources are also available.
 
-Interfaces are written in an object orientet style, meaning you create object 
+Interfaces are written in an object orientet style, meaning you create object
 instances via various different constructor functions (plm_*create()),
 do some work on them and later dispose them via plm_*destroy().
 
@@ -51,11 +51,11 @@ plm_video_*  -- the MPEG1 Video ("mpeg1") decoder
 plm_audio_*  -- the MPEG1 Audio Layer II ("mp2") decoder
 
 
-This library uses malloc(), realloc() and free() to manage memory. Typically 
+This library uses malloc(), realloc() and free() to manage memory. Typically
 all allocation happens up-front when creating the interface. However, the
 default buffer size may be too small for certain inputs. In these cases plmpeg
 will realloc() the buffer with a larger size whenever needed. You can configure
-the default buffer size by defining PLM_BUFFER_DEFAULT_SIZE *before* 
+the default buffer size by defining PLM_BUFFER_DEFAULT_SIZE *before*
 including this library.
 
 With the high-level interface you have two options to decode video & audio:
@@ -85,7 +85,7 @@ mat4 rec601 = mat4(
 gl_FragColor = vec4(y, cb, cr, 1.0) * rec601;
 
 Audio data is decoded into a struct with either one single float array with the
-samples for the left and right channel interleaved, or if the 
+samples for the left and right channel interleaved, or if the
 PLM_AUDIO_SEPARATE_CHANNELS is defined *before* including this library, into
 two separate float arrays - one for each channel.
 

dsp/mpeg/demux.c

@@ -115,8 +115,8 @@ plm_packet_t *plm_demux_decode(plm_demux_t *self) {
 	do {
 		code = plm_buffer_next_start_code(self->buffer);
 		if (
-			code == PLM_DEMUX_PACKET_VIDEO_1 || 
-			code == PLM_DEMUX_PACKET_PRIVATE || 
+			code == PLM_DEMUX_PACKET_VIDEO_1 ||
+			code == PLM_DEMUX_PACKET_PRIVATE ||
 			(code >= PLM_DEMUX_PACKET_AUDIO_1 && code <= PLM_DEMUX_PACKET_AUDIO_4)
 		) {
 			return plm_demux_decode_packet(self, code);

dsp/mpeg/mp2.c

@@ -273,7 +273,7 @@ typedef plm_audio_t plm_audio_t;
 int plm_audio_decode_header(plm_audio_t *self);
 void plm_audio_decode_frame(plm_audio_t *self);
 const plm_quantizer_spec_t *plm_audio_read_allocation(plm_audio_t *self, int sb, int tab3);
-void plm_audio_read_samples(plm_audio_t *self, int ch, int sb, int part); 
+void plm_audio_read_samples(plm_audio_t *self, int ch, int sb, int part);
 void plm_audio_matrix_transform(int s[32][3], int ss, float *d, int dp);
 
 plm_audio_t *plm_audio_create_with_buffer(plm_buffer_t *buffer, int destroy_when_done) {
@@ -345,7 +345,7 @@ plm_samples_t *plm_audio_decode(plm_audio_t *self) {
 	self->samples.time = self->time;
 
 	self->samples_decoded += PLM_AUDIO_SAMPLES_PER_FRAME;
-	self->time = (double)self->samples_decoded / 
+	self->time = (double)self->samples_decoded /
 		(double)PLM_AUDIO_SAMPLE_RATE[self->samplerate_index];
 
 	return &self->samples;
@@ -402,7 +402,7 @@ int plm_audio_decode_header(plm_audio_t *self) {
 		plm_buffer_skip(self->buffer, 16);
 	}
 
-	// Compute frame size, check if we have enough data to decode the whole 
+	// Compute frame size, check if we have enough data to decode the whole
 	// frame.
 	int bitrate = PLM_AUDIO_BIT_RATE[self->bitrate_index];
 	int samplerate = PLM_AUDIO_SAMPLE_RATE[self->samplerate_index];
@@ -561,7 +561,7 @@ void plm_audio_decode_frame(plm_audio_t *self) {
 						}
 					#else
 						for (int j = 0; j < 32; j++) {
-							self->samples.interleaved[((out_pos + j) << 1) + ch] = 
+							self->samples.interleaved[((out_pos + j) << 1) + ch] =
 								self->U[j] / 2147418112.0f;
 						}
 					#endif

dsp/mpeg/plm.c

@@ -94,7 +94,7 @@ plm_t *plm_create_with_buffer(plm_buffer_t *buffer, int destroy_when_done) {
 
 	self->demux = plm_demux_create(buffer, destroy_when_done);
 
-	// In theory we should check plm_demux_get_num_video_streams() and 
+	// In theory we should check plm_demux_get_num_video_streams() and
 	// plm_demux_get_num_audio_streams() here, but older files typically
 	// do not specify these correctly. So we just assume we have a video and
 	// audio stream and create the decoders.
@@ -159,7 +159,7 @@ double plm_get_framerate(plm_t *self) {
 
 int plm_get_num_audio_streams(plm_t *self) {
 	// Some files do not specify the number of audio streams in the system header.
-	// If the reported number of streams is 0, we check if we have a samplerate, 
+	// If the reported number of streams is 0, we check if we have a samplerate,
 	// indicating at least one audio stream.
 	int num_streams = plm_demux_get_num_audio_streams(self->demux);
 	return num_streams == 0 && plm_get_samplerate(self) ? 1 : num_streams;