some more work on organizing code for resampling/audioFormat conversion

src/aac_decode.c

@@ -251,7 +251,7 @@ int getAacTotalTime(char * file) {
 }
 
 
-int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
+int aac_decode(OutputBuffer * cb, DecoderControl * dc) {
 	float time;
 	float totalTime;
 	faacDecHandle decoder;
@@ -306,9 +306,9 @@ int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 		return -1;
 	}
 
-	af->bits = 16;
+	dc->audioFormat.bits = 16;
 
-	cb->totalTime = totalTime;
+	dc->totalTime = totalTime;
 
 	time = 0.0;
 
@@ -342,8 +342,10 @@ int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 #endif
 
 		if(dc->start) {
-			af->channels = frameInfo.channels;
-			af->sampleRate = sampleRate;
+			dc->audioFormat.channels = frameInfo.channels;
+			dc->audioFormat.sampleRate = sampleRate;
+                        getOutputAudioFormat(&(dc->audioFormat),
+                                        &(cb->audioFormat));
 			dc->state = DECODE_STATE_DECODE;
 			dc->start = 0;
 		}

src/aac_decode.h

@@ -27,7 +27,7 @@
 
 int getAacTotalTime(char * file);
 
-int aac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int aac_decode(OutputBuffer * cb, DecoderControl * dc);
 
 #endif /* HAVE_FAAD */
 

src/audiofile_decode.c

@@ -51,8 +51,7 @@ int getAudiofileTotalTime(char * file)
 	return time;
 }
 
-int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
-{
+int audiofile_decode(OutputBuffer * cb, DecoderControl * dc) {
 	int fs, frame_count;
 	AFfilehandle af_fp;
 	int bits;
@@ -71,19 +70,20 @@ int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
 	}
 
 	afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
-	af->bits = bits;
-	af->sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK);
-	af->channels = afGetChannels(af_fp,AF_DEFAULT_TRACK);
+	dc->audioFormat.bits = bits;
+	dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK);
+	dc->audioFormat.channels = afGetChannels(af_fp,AF_DEFAULT_TRACK);
+        getOutputAudioFormat(&(dc->audioFormat),&(cb->audioFormat));
 	
 	frame_count = afGetFrameCount(af_fp,AF_DEFAULT_TRACK);
 	
-	cb->totalTime = ((float)frame_count/(float)af->sampleRate);
+	dc->totalTime = ((float)frame_count/(float)dc->audioFormat.sampleRate);
 
-	bitRate = st.st_size*8.0/cb->totalTime/1000.0+0.5;
+	bitRate = st.st_size*8.0/dc->totalTime/1000.0+0.5;
 	
-	if (af->bits != 8 && af->bits != 16) {
+	if (dc->audioFormat.bits != 8 && dc->audioFormat.bits != 16) {
 		ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
-			dc->file,af->bits);
+			dc->file,dc->audioFormat.bits);
 		afCloseFile(af_fp);
 		return -1;
 	}
@@ -100,7 +100,8 @@ int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
 			if(dc->seek) {
 				cb->end = cb->begin;
 				cb->wrap = 0;
-				current = dc->seekWhere * af->sampleRate;
+				current = dc->seekWhere * 
+                                                dc->audioFormat.sampleRate;
 				afSeekFrame(af_fp, AF_DEFAULT_TRACK,current);
 				
 				dc->seek = 0;
@@ -112,7 +113,7 @@ int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
 				current += ret;
 				sendDataToOutputBuffer(cb,dc,chunk,ret*fs,
 					(float)current /
-						(float)af->sampleRate,
+					(float)dc->audioFormat.sampleRate,
 					bitRate);
 				if(dc->stop) break;
 				else if(dc->seek) continue;

src/audiofile_decode.h

@@ -27,7 +27,7 @@
 
 #include "playerData.h"
 
-int audiofile_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int audiofile_decode(OutputBuffer * cb, DecoderControl * dc);
 
 int getAudiofileTotalTime(char * file);
 

src/decode.c

@@ -111,9 +111,7 @@ int calculateCrossFadeChunks(PlayerControl * pc, AudioFormat * af) {
 	return (int)chunks;
 }
 
-int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
-		OutputBuffer * cb) 
-{
+int waitOnDecode(PlayerControl * pc, DecoderControl * dc, OutputBuffer * cb) {
 	while(decode_pid && *decode_pid>0 && dc->start) my_usleep(1000);
 
 	if(dc->start || dc->error!=DECODE_ERROR_NOERROR) {
@@ -124,7 +122,7 @@ int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
 		return -1;
 	}
 
-	if(openAudioDevice(af)<0) {
+	if(openAudioDevice(&(cb->audioFormat))<0) {
 		strncpy(pc->erroredFile,pc->file,MAXPATHLEN);
 		pc->erroredFile[MAXPATHLEN] = '\0';
 		pc->error = PLAYER_ERROR_AUDIO;
@@ -132,19 +130,17 @@ int waitOnDecode(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
 		return -1;
 	}
 
-	pc->totalTime = cb->totalTime;
+	pc->totalTime = dc->totalTime;
 	pc->elapsedTime = 0;
 	pc->bitRate = 0;
-	pc->sampleRate = af->sampleRate;
-	pc->bits = af->bits;
-	pc->channels = af->channels;
+	pc->sampleRate = dc->audioFormat.sampleRate;
+	pc->bits = dc->audioFormat.bits;
+	pc->channels = dc->audioFormat.channels;
 
 	return 0;
 }
 
-void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc, 
-		OutputBuffer * cb) 
-{
+void decodeSeek(PlayerControl * pc, DecoderControl * dc, OutputBuffer * cb) {
 	if(decode_pid && *decode_pid>0) {
 		cb->next = -1;
 		if(dc->state!=DECODE_STATE_DECODE || dc->error || 
@@ -156,7 +152,7 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
 			cb->wrap = 0;
 			dc->error = 0;
 			dc->start = 1;
-			waitOnDecode(pc,af,dc,cb);
+			waitOnDecode(pc,dc,cb);
 		}
 		if(*decode_pid>0 && dc->state==DECODE_STATE_DECODE) {
 			dc->seekWhere = pc->seekWhere > pc->totalTime-0.1 ?
@@ -205,7 +201,7 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
 	} \
 	if(pc->seek) { \
 		pc->totalPlayTime+= pc->elapsedTime-pc->beginTime; \
-		decodeSeek(pc,af,dc,cb); \
+		decodeSeek(pc,dc,cb); \
 		pc->beginTime = pc->elapsedTime; \
 		doCrossFade = 0; \
 		nextChunk =  -1; \
@@ -217,8 +213,8 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
 		return; \
 	}
 
-int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af, 
-			DecoderControl * dc) {
+int decoderInit(PlayerControl * pc, OutputBuffer * cb, DecoderControl * dc) {
+			
 	int pid;
 	int ret; 
 	decode_pid = &(pc->decode_pid);
@@ -237,30 +233,30 @@ int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af,
 				switch(pc->decodeType) {
 #ifdef HAVE_MAD
 				case DECODE_TYPE_MP3:
-					ret = mp3_decode(cb,af,dc);
+					ret = mp3_decode(cb,dc);
 					break;
 #endif
 #ifdef HAVE_FAAD
 				case DECODE_TYPE_AAC:
-					ret = aac_decode(cb,af,dc);
+					ret = aac_decode(cb,dc);
 					break;
 				case DECODE_TYPE_MP4:
-					ret = mp4_decode(cb,af,dc);
+					ret = mp4_decode(cb,dc);
 					break;
 #endif
 #ifdef HAVE_OGG
 				case DECODE_TYPE_OGG:
-					ret = ogg_decode(cb,af,dc);
+					ret = ogg_decode(cb,dc);
 					break;
 #endif
 #ifdef HAVE_FLAC
 				case DECODE_TYPE_FLAC:
-					ret = flac_decode(cb,af,dc);
+					ret = flac_decode(cb,dc);
 					break;
 #endif
 #ifdef HAVE_AUDIOFILE
 				case DECODE_TYPE_AUDIOFILE:
-					ret = audiofile_decode(cb,af,dc);
+					ret = audiofile_decode(cb,dc);
 					break;
 #endif
 				default:
@@ -313,7 +309,6 @@ int decoderInit(PlayerControl * pc, OutputBuffer * cb, AudioFormat *af,
 void decode() {
 	OutputBuffer * cb;
 	PlayerControl * pc;
-	AudioFormat * af;
 	DecoderControl * dc;
 
 	cb = &(getPlayerData()->buffer);
@@ -323,13 +318,12 @@ void decode() {
 	cb->wrap = 0;
 	pc = &(getPlayerData()->playerControl);
 	dc = &(getPlayerData()->decoderControl);
-	af = &(getPlayerData()->audioFormat);
 	dc->error = 0;
 	dc->start = 1;
 	cb->next = -1;
 
 	if(decode_pid==NULL || *decode_pid<=0) {
-		if(decoderInit(pc,cb,af,dc)<0) return;
+		if(decoderInit(pc,cb,dc)<0) return;
 	}
 
 	{
@@ -343,7 +337,7 @@ void decode() {
 		int nextChunk = -1;
 		int test;
 
-		if(waitOnDecode(pc,af,dc,cb)<0) return;
+		if(waitOnDecode(pc,dc,cb)<0) return;
 
 		pc->state = PLAYER_STATE_PLAY;
 		pc->play = 0;
@@ -371,12 +365,13 @@ void decode() {
 			}
 			if(cb->next>=0 && doCrossFade==0 && !dc->start) {
 				nextChunk = -1;
-				if(isCurrentAudioFormat(af)) {
+				if(isCurrentAudioFormat(&(cb->audioFormat))) {
 					doCrossFade = 1;
 					crossFadeChunks = 
-						calculateCrossFadeChunks(pc,af);
+						calculateCrossFadeChunks(pc,
+                                                        &(cb->audioFormat));
 					if(!crossFadeChunks ||
-						pc->crossFade>=cb->totalTime) 
+						pc->crossFade>=dc->totalTime) 
 					{
 						doCrossFade = -1;
 					}
@@ -415,7 +410,7 @@ void decode() {
 								cb->begin],
 							cb->chunkSize[
 								nextChunk],
-							af,
+							&(cb->audioFormat),
 							((float)fadePosition)/
 							crossFadeChunks);
 						if(cb->chunkSize[nextChunk]>
@@ -440,7 +435,7 @@ void decode() {
 				pcm_volumeChange(cb->chunks+cb->begin*
 					CHUNK_SIZE,
 					cb->chunkSize[cb->begin],
-					af,
+					&(cb->audioFormat),
 					pc->softwareVolume);
 				if(playAudio(cb->chunks+cb->begin*CHUNK_SIZE,
 					cb->chunkSize[cb->begin])<0) 
@@ -485,7 +480,7 @@ void decode() {
 				}
 				else {
 					cb->next = -1;
-					if(waitOnDecode(pc,af,dc,cb)<0) return;
+					if(waitOnDecode(pc,dc,cb)<0) return;
 					nextChunk = -1;
 					doCrossFade = 0;
 					crossFadeChunks = 0;

src/decode.h

@@ -22,6 +22,7 @@
 #include "../config.h"
 
 #include "mpd_types.h"
+#include "audio.h"
 
 #include <stdio.h>
 #include <sys/param.h>
@@ -48,8 +49,10 @@ typedef struct _DecoderControl {
 	volatile mpd_sint8 seek;
 	volatile mpd_sint8 seekError;
 	volatile mpd_sint8 cycleLogFiles;
-	double seekWhere;
+	volatile double seekWhere;
 	char file[MAXPATHLEN+1];
+        AudioFormat audioFormat;
+        volatile float totalTime;
 } DecoderControl;
 
 void decodeSigHandler(int sig);

src/flac_decode.c

@@ -26,6 +26,7 @@
 #include "inputStream.h"
 #include "outputBuffer.h"
 #include "replayGain.h"
+#include "audio.h"
 
 #include <stdio.h>
 #include <string.h>
@@ -40,7 +41,6 @@ typedef struct {
 	int bitRate;
 	FLAC__uint64 position;
 	OutputBuffer * cb;
-	AudioFormat * af;
 	DecoderControl * dc;
         InputStream inStream;
         float replayGainScale;
@@ -67,7 +67,7 @@ FLAC__SeekableStreamDecoderLengthStatus flacLength(
                 const FLAC__SeekableStreamDecoder *, FLAC__uint64 *, void *);
 FLAC__bool flacEOF(const FLAC__SeekableStreamDecoder *, void *);
 
-int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl *dc) {
+int flac_decode(OutputBuffer * cb, DecoderControl *dc) {
 	FLAC__SeekableStreamDecoder * flacDec;
 	FlacData data;
 	int status = 1;
@@ -77,7 +77,6 @@ int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl *dc) {
 	data.position = 0;
 	data.bitRate = 0;
 	data.cb = cb;
-	data.af = af;
 	data.dc = dc;
         data.replayGainScale = 1.0;
 
@@ -146,14 +145,14 @@ int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl *dc) {
 		}
 		if(dc->seek) {
 			FLAC__uint64 sampleToSeek = dc->seekWhere*
-					af->sampleRate+0.5;
+					dc->audioFormat.sampleRate+0.5;
 			cb->end = cb->begin;
 			cb->wrap = 0;
 			if(FLAC__seekable_stream_decoder_seek_absolute(flacDec,
 						sampleToSeek))
 			{
 				data.time = ((float)sampleToSeek)/
-					af->sampleRate;
+					dc->audioFormat.sampleRate;
 				data.position = 0;
 			}
 			dc->seek = 0;
@@ -354,13 +353,17 @@ void flacMetadata(const FLAC__SeekableStreamDecoder *dec,
 
         switch(block->type) {
         case FLAC__METADATA_TYPE_STREAMINFO:
-		data->af->bits = block->data.stream_info.bits_per_sample;
-		data->af->bits = 16;
-		data->af->sampleRate = block->data.stream_info.sample_rate;
-		data->af->channels = block->data.stream_info.channels;
-		data->cb->totalTime = 
+		data->dc->audioFormat.bits = 
+                                block->data.stream_info.bits_per_sample;
+		data->dc->audioFormat.sampleRate = 
+                                block->data.stream_info.sample_rate;
+		data->dc->audioFormat.channels = 
+                                block->data.stream_info.channels;
+		data->dc->totalTime = 
                                 ((float)block->data.stream_info.total_samples)/
-			        data->af->sampleRate;
+			        data->dc->audioFormat.sampleRate;
+                getOutputAudioFormat(&(data->dc->audioFormat),
+                                &(data->cb->audioFormat));
                 break;
         case FLAC__METADATA_TYPE_VORBIS_COMMENT:
                 flacParseReplayGain(block,data);
@@ -370,7 +373,7 @@ void flacMetadata(const FLAC__SeekableStreamDecoder *dec,
 }
 
 int flacSendChunk(FlacData * data) {
-        doReplayGain(data->chunk,data->chunk_length,data->af,
+        doReplayGain(data->chunk,data->chunk_length,&(data->dc->audioFormat),
                         data->replayGainScale);
 
 	switch(sendDataToOutputBuffer(data->cb,data->dc,data->chunk,
@@ -413,7 +416,7 @@ FLAC__StreamDecoderWriteStatus flacWrite(const FLAC__SeekableStreamDecoder *dec,
 				c_chan++, d_samp++) {
 			u16 = buf[c_chan][c_samp];
 			uc = (unsigned char *)&u16;
-			for(i=0;i<(data->af->bits/8);i++) {
+			for(i=0;i<(data->dc->audioFormat.bits/8);i++) {
 				if(data->chunk_length>=CHUNK_SIZE) {
 					if(flacSendChunk(data)<0) {
 						return FLAC__STREAM_DECODER_WRITE_STATUS_ABORT;

src/flac_decode.h

@@ -25,7 +25,7 @@
 
 #include <stdio.h>
 
-int flac_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int flac_decode(OutputBuffer * cb, DecoderControl * dc);
 
 #endif
 /* vim:set shiftwidth=8 tabstop=8 expandtab: */

src/mp3_decode.c

@@ -523,7 +523,7 @@ void initAudioFormatFromMp3DecodeData(mp3DecodeData * data, AudioFormat * af) {
 	af->channels = MAD_NCHANNELS(&(data->frame).header);
 }
 
-int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
+int mp3_decode(OutputBuffer * cb, DecoderControl * dc) {
 	mp3DecodeData data;
 
 	if(openMp3(dc->file,&data) < 0) {
@@ -531,8 +531,10 @@ int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 		return -1;
 	}
 
-	initAudioFormatFromMp3DecodeData(&data,af);
-	cb->totalTime = data.totalTime;
+	initAudioFormatFromMp3DecodeData(&data, &(dc->audioFormat));
+        getOutputAudioFormat(&(dc->audioFormat), &(cb->audioFormat));
+        
+	dc->totalTime = data.totalTime;
 	dc->start = 0;
 	dc->state = DECODE_STATE_DECODE;
 

src/mp3_decode.h

@@ -28,7 +28,7 @@
 /* this is primarily used in tag.c */
 int getMp3TotalTime(char * file);
 
-int mp3_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int mp3_decode(OutputBuffer * cb, DecoderControl * dc);
 
 #endif
 

src/mp4_decode.c

@@ -84,7 +84,7 @@ uint32_t mp4_inputStreamSeekCallback(void *inStream, uint64_t position) {
 }       
 		    
 
-int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
+int mp4_decode(OutputBuffer * cb, DecoderControl * dc) {
 	mp4ff_t * mp4fh;
 	mp4ff_callback_t * mp4cb; 
 	int32_t track;
@@ -152,7 +152,7 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 #endif
 	faacDecSetConfiguration(decoder,config);
 
-	af->bits = 16;
+	dc->audioFormat.bits = 16;
 
 	mp4Buffer = NULL;
 	mp4BufferSize = 0;
@@ -169,8 +169,8 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 		return -1;
 	}
 
-	af->sampleRate = sampleRate;
-	af->channels = channels;
+	dc->audioFormat.sampleRate = sampleRate;
+	dc->audioFormat.channels = channels;
 	time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
 	scale = mp4ff_time_scale(mp4fh,track);
 
@@ -184,7 +184,7 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 		free(mp4cb);
 		return -1;
 	}
-	cb->totalTime = ((float)time)/scale;
+	dc->totalTime = ((float)time)/scale;
 
 	numSamples = mp4ff_num_samples(mp4fh,track);
 
@@ -255,8 +255,10 @@ int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc) {
 #ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
 			scale = frameInfo.samplerate;
 #endif
-			af->sampleRate = scale;
-			af->channels = frameInfo.channels;
+			dc->audioFormat.sampleRate = scale;
+			dc->audioFormat.channels = frameInfo.channels;
+                        getOutputAudioFormat(&(dc->audioFormat),
+                                        &(cb->audioFormat));
 			dc->state = DECODE_STATE_DECODE;
 			dc->start = 0;
 		}

src/mp4_decode.h

@@ -29,7 +29,7 @@
 
 int mp4_getAACTrack(mp4ff_t *infile);
 
-int mp4_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int mp4_decode(OutputBuffer * cb, DecoderControl * dc);
 
 uint32_t mp4_inputStreamReadCallback(void *inStream, void *buffer, 
 	uint32_t length);

src/ogg_decode.c

@@ -142,7 +142,7 @@ float ogg_getReplayGainScale(char ** comments) {
         return 1.0;
 }
 
-int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
+int ogg_decode(OutputBuffer * cb, DecoderControl * dc)
 {
 	OggVorbis_File vf;
 	ov_callbacks callbacks;
@@ -167,12 +167,13 @@ int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
 	
 	{
 		vorbis_info *vi=ov_info(&vf,-1);
-		af->bits = 16;
-		af->channels = vi->channels;
-		af->sampleRate = vi->rate;
+		dc->audioFormat.bits = 16;
+		dc->audioFormat.channels = vi->channels;
+		dc->audioFormat.sampleRate = vi->rate;
+                getOutputAudioFormat(&(dc->audioFormat),&(cb->audioFormat));
 	}
 
-	cb->totalTime = ov_time_total(&vf,-1);
+	dc->totalTime = ov_time_total(&vf,-1);
 	dc->state = DECODE_STATE_DECODE;
 	dc->start = 0;
 
@@ -203,7 +204,8 @@ int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc)
 				if((test = ov_bitrate_instant(&vf))>0) {
 					bitRate = test/1000;
 				}
-                                doReplayGain(chunk,ret,af,replayGainScale);
+                                doReplayGain(chunk,ret,&(dc->audioFormat),
+                                                replayGainScale);
 				sendDataToOutputBuffer(cb,dc,chunk,ret,
 					ov_time_tell(&vf),bitRate);
 				if(dc->stop) break;

src/ogg_decode.h

@@ -25,7 +25,7 @@
 
 #include <stdio.h>
 
-int ogg_decode(OutputBuffer * cb, AudioFormat * af, DecoderControl * dc);
+int ogg_decode(OutputBuffer * cb, DecoderControl * dc);
 
 int getOggTotalTime(char * file);
 

src/outputBuffer.h

@@ -21,6 +21,7 @@
 
 #include "mpd_types.h"
 #include "decode.h"
+#include "audio.h"
 
 #define OUTPUT_BUFFER_DC_STOP   -1
 #define OUTPUT_BUFFER_DC_SEEK   -2
@@ -34,7 +35,7 @@ typedef struct _OutputBuffer {
 	mpd_sint16 volatile end;
 	mpd_sint16 volatile next;
 	mpd_sint8 volatile wrap;
-	float totalTime;
+        AudioFormat audioFormat;
 } OutputBuffer;
 
 void flushOutputBuffer(OutputBuffer * cb);

src/pcm_utils.h

@@ -33,5 +33,8 @@ void pcm_volumeChange(char * buffer, int bufferSize, AudioFormat * format,
 void pcm_mix(char * buffer1, char * buffer2, size_t bufferSize1, 
 		size_t bufferSize2, AudioFormat * format, float portion1);
 
+void pmc_convertAudioFormat(AudioFormat * inFormat, char * inBuffer, size_t
+                inSize, size_t * inLeft, AudioFormat * outFormat, 
+                char * outBuffer, size_t outSize, size_t * outLeft);
 #endif
 /* vim:set shiftwidth=4 tabstop=8 expandtab: */

src/playerData.h

@@ -35,7 +35,6 @@ extern int buffered_chunks;
 
 typedef struct _PlayerData {
 	OutputBuffer buffer;
-	AudioFormat audioFormat;
 	PlayerControl playerControl;
 	DecoderControl decoderControl;
 } PlayerData;