aac decoding!

need to work on seeking for AAC
also, don't reset cb->begin on seek to 0,
instead just set cb->end=cb->begin, works much better for
disabling seeking (like ADIF AAC's)

git-svn-id: https://svn.musicpd.org/mpd/trunk@356 09075e82-0dd4-0310-85a5-a0d7c8717e4f

src/aac_decode.c

@@ -2,8 +2,6 @@
  * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
  * This project's homepage is: http://www.musicpd.org
  * 
- * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
- * 
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License as published by
  * the Free Software Foundation; either version 2 of the License, or
@@ -196,7 +194,11 @@ int initAacBuffer(char * file, AacBuffer * b, float * length) {
 		if(*length!=0 && bitRate!=0) *length = *length*8.0/bitRate;
 	}
 
-	if(*length<0) return -1;
+	if(*length<0) {
+		fclose(b->infile);
+		if(b->buffer) free(b->buffer);
+		return -1;
+	}
 
 	return 0;
 }
@@ -215,59 +217,29 @@ int getAacTotalTime(char * file) {
 
 
 int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
-	/*FILE * fh;
-	mp4ff_t * mp4fh;
-	mp4ff_callback_t * mp4cb; 
-	int32_t track;
 	float time;
-	int32_t scale;
+	float totalTime;
 	faacDecHandle decoder;
 	faacDecFrameInfo frameInfo;
 	faacDecConfigurationPtr config;
-	unsigned char * mp4Buffer;
+	size_t bread;
-	int mp4BufferSize;
 	unsigned long sampleRate;
 	unsigned char channels;
-	long sampleId;
-	long numSamples;
 	int eof = 0;
-	long dur;
 	unsigned int sampleCount;
 	char * sampleBuffer;
 	size_t sampleBufferLen;
-	unsigned int initial = 1;
 	int chunkLen = 0;
-	float * seekTable;
+	/*float * seekTable;
 	long seekTableEnd = -1;
-	int seekPositionFound = 0;
+	int seekPositionFound = 0;*/
-	long offset;
 	mpd_uint16 bitRate = 0;
+	AacBuffer b;
 
-	fh = fopen(dc->file,"r");
+	printf("aac_decode!\n");
-	if(!fh) {
-		ERROR("failed to open %s\n",dc->file);
-		return -1;
-	}
 
-	mp4cb = malloc(sizeof(mp4ff_callback_t));
+	if(initAacBuffer(dc->file,&b,&totalTime) < 0) {
-	mp4cb->read = mp4_readCallback;
+		ERROR("Not AAC file no ADTS or ADIF headers found.\n");
-	mp4cb->seek = mp4_seekCallback;
-	mp4cb->user_data = fh;
-
-	mp4fh = mp4ff_open_read(mp4cb);
-	if(!mp4fh) {
-		ERROR("Input does not appear to be a mp4 stream.\n");
-		free(mp4cb);
-		fclose(fh);
-		return -1;
-	}
-
-	track = mp4_getAACTrack(mp4fh);
-	if(track < 0) {
-		ERROR("No AAC track found in mp4 stream.\n");
-		mp4ff_close(mp4fh);
-		fclose(fh);
-		free(mp4cb);
 		return -1;
 	}
 
@@ -285,48 +257,33 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 
 	af->bits = 16;
 
-	mp4Buffer = NULL;
+	fillAacBuffer(&b);
-	mp4BufferSize = 0;
+	if((bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
-	mp4ff_get_decoder_config(mp4fh,track,&mp4Buffer,&mp4BufferSize);
+			&sampleRate,&channels)) < 0)
-
-	if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
-			< 0)
 	{
-		ERROR("Error initializing AAC decoder library.\n");
+		ERROR("Error not a AAC stream.\n");
 		faacDecClose(decoder);
-		mp4ff_close(mp4fh);
+		fclose(b.infile);
-		free(mp4cb);
+		if(b.buffer) free(b.buffer);
-		fclose(fh);
 		return -1;
 	}
 
 	af->sampleRate = sampleRate;
 	af->channels = channels;
-	time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
-	scale = mp4ff_time_scale(mp4fh,track);
 
-	if(mp4Buffer) free(mp4Buffer);
+	cb->totalTime = totalTime+0.5;
-
-	if(scale < 0) {
-		ERROR("Error getting audio format of mp4 AAC track.\n");
-		faacDecClose(decoder);
-		mp4ff_close(mp4fh);
-		fclose(fh);
-		free(mp4cb);
-		return -1;
-	}
-	cb->totalTime = ((float)time)/scale;
-
-	numSamples = mp4ff_num_samples(mp4fh,track);
 
 	dc->state = DECODE_STATE_DECODE;
 	dc->start = 0;
 	time = 0.0;
 
-	seekTable = malloc(sizeof(float)*numSamples);
+	advanceAacBuffer(&b,bread);
+	fillAacBuffer(&b);
 
-	for(sampleId=0; sampleId<numSamples && !eof; sampleId++) {
+	/*seekTable = malloc(sizeof(float)*numSamples);*/
-		if(dc->seek && seekTableEnd>1 && 
+
+	do {
+		/*if(dc->seek && seekTableEnd>1 && 
 				seekTable[seekTableEnd]>=dc->seekWhere)
 		{
 			int i = 2;
@@ -335,9 +292,6 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 			time = seekTable[sampleId];
 		}
 
-		dur = mp4ff_get_sample_duration(mp4fh,track,sampleId);
-		offset = mp4ff_get_sample_offset(mp4fh,track,sampleId);
-
 		if(sampleId>seekTableEnd) {
 			seekTable[sampleId] = time;
 			seekTableEnd = sampleId;
@@ -358,42 +312,36 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 			dc->seek = 0;
 		}
 
-		if(dc->seek) continue;
+		if(dc->seek) continue;*/
-		
+
-		if(mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
+		if(dc->seek) {
-				&mp4BufferSize) == 0)
+			/*chunkLen = 0;
-		{
+			cb->wrap = 0;
-			eof = 1;
+			cb->end = 0;*/
-			continue;
+			dc->seekError = 1;
-		}
+			dc->seek = 0;
+		}
+		
+		sampleBuffer = faacDecDecode(decoder,&frameInfo,b.buffer,
+						b.bytesIntoBuffer);
+		advanceAacBuffer(&b,frameInfo.bytesconsumed);
 
-		sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer,
-						mp4BufferSize);
-		if(mp4Buffer) free(mp4Buffer);
 		if(frameInfo.error > 0) {
 			eof = 1;
 			break;
 		}
 
-		if(channels*(dur+offset) > frameInfo.samples) {
+		sampleCount = (unsigned long)(frameInfo.samples);
-			dur = frameInfo.samples;
-			offset = 0;
-		}
-
-		sampleCount = (unsigned long)(dur*channels);
 
 		if(sampleCount>0) {
-			initial =0;
 			bitRate = frameInfo.bytesconsumed*8.0*
-				frameInfo.channels*scale/
+				frameInfo.channels*sampleRate/
 				frameInfo.samples/1024+0.5;
+			time+= (float)(frameInfo.samples)/channels/sampleRate;
 		}
 			
-
 		sampleBufferLen = sampleCount*2;
 
-		sampleBuffer+=offset*channels*2;
-
 		while(sampleBufferLen>0 && !dc->seek) {
 			size_t size = sampleBufferLen>CHUNK_SIZE-chunkLen ? 
 							CHUNK_SIZE-chunkLen:
@@ -427,7 +375,11 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 				}
 			}
 		}
-	}
+
+		fillAacBuffer(&b);
+
+		if(b.bytesIntoBuffer==0) eof = 1;
+	} while (!eof);
 
 	if(!dc->stop && !dc->seek && chunkLen>0) {
 		cb->chunkSize[cb->end] = chunkLen;
@@ -440,11 +392,10 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 		chunkLen = 0;
 	}
 
-	free(seekTable);
+	/*free(seekTable);*/
 	faacDecClose(decoder);
-	mp4ff_close(mp4fh);
+	fclose(b.infile);
-	fclose(fh);
+	if(b.buffer) free(b.buffer);
-	free(mp4cb);
 
 	if(dc->seek) dc->seek = 0;
 
@@ -452,7 +403,7 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 		dc->state = DECODE_STATE_STOP;
 		dc->stop = 0;
 	}
-	else dc->state = DECODE_STATE_STOP;*/
+	else dc->state = DECODE_STATE_STOP;
 
 	return 0;
 }

src/aac_decode.h

@@ -2,8 +2,6 @@
  * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
  * This project's homepage is: http://www.musicpd.org
  * 
- * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
- * 
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License as published by
  * the Free Software Foundation; either version 2 of the License, or

src/audiofile_decode.c

@@ -86,7 +86,7 @@ int audiofile_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
 
 		while(!eof) {
 			if(dc->seek) {
-				cb->end = 0;
+				cb->end = cb->begin;
 				cb->wrap = 0;
 				current = dc->seekWhere * af->sampleRate;
 				afSeekFrame(af_fp, AF_DEFAULT_TRACK,current);

src/decode.c

@@ -40,6 +40,7 @@
 #endif
 #ifdef HAVE_FAAD
 #include "mp4_decode.h"
+#include "aac_decode.h"
 #endif
 
 #include <signal.h>
@@ -159,11 +160,11 @@ void decodeSeek(PlayerControl * pc, AudioFormat * af, DecoderControl * dc,
 						pc->totalTime-0.1 : 
 						pc->seekWhere;
 			dc->seekWhere = 0 > dc->seekWhere ? 0 : dc->seekWhere;
-			cb->begin = 0;
+			dc->seekError = 0;
 			dc->seek = 1;
-			pc->elapsedTime = dc->seekWhere;
 			pc->bitRate = 0;
 			while(*decode_pid>0 && dc->seek) usleep(1000);
+			if(dc->seekError) pc->elapsedTime = dc->seekWhere;
 		}
 	}
 	pc->seek = 0;
@@ -229,6 +230,9 @@ int decoderInit(PlayerControl * pc, Buffer * cb, AudioFormat *af,
 					break;
 #endif
 #ifdef HAVE_FAAD
+				case DECODE_TYPE_AAC:
+					dc->error = aac_decode(cb,af,dc);
+					break;
 				case DECODE_TYPE_MP4:
 					dc->error = mp4_decode(cb,af,dc);
 					break;

src/decode.h

@@ -21,6 +21,8 @@
 
 #include "../config.h"
 
+#include "mpd_types.h"
+
 #include <stdio.h>
 #include <sys/param.h>
 
@@ -38,11 +40,12 @@
 #define DECODE_ERROR_UNKTYPE	1
 
 typedef struct _DecoderControl {
-	int state;
+	mpd_sint8 state;
-	int stop;
+	mpd_sint8 stop;
-	int start;
+	mpd_sint8 start;
-	int error;
+	mpd_uint16 error;
-	int seek;
+	mpd_sint8 seek;
+	mpd_sint8 seekError;
 	double seekWhere;
 	char file[MAXPATHLEN+1];
 } DecoderControl;

src/flac_decode.c

@@ -104,7 +104,7 @@ void flacPlayFile(char *file, Buffer * cb, AudioFormat * af,
 		if(dc->seek) {
 			FLAC__uint64 sampleToSeek = dc->seekWhere*
 					af->sampleRate+0.5;
-			cb->end = 0;
+			cb->end = cb->begin;
 			cb->wrap = 0;
 			if(FLAC__file_decoder_seek_absolute(flacDec,
 						sampleToSeek))

src/mp3_decode.c

@@ -487,7 +487,7 @@ int mp3Read(mp3DecodeData * data, Buffer * cb, DecoderControl * dc) {
 		if(dc->seek) {
 			long i = 0;
 			cb->wrap = 0;
-			cb->end = 0;
+			cb->end = cb->begin;
 			data->muteFrame = 1;
 			while(i<data->highestFrame && dc->seekWhere >
 					((float)mad_timer_count(data->times[i],

src/mp4_decode.c

@@ -2,8 +2,6 @@
  * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
  * This project's homepage is: http://www.musicpd.org
  * 
- * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
- * 
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License as published by
  * the Free Software Foundation; either version 2 of the License, or
@@ -149,7 +147,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 	if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
 			< 0)
 	{
-		ERROR("Error initializing AAC decoder library.\n");
+		ERROR("Error not a AAC stream.\n");
 		faacDecClose(decoder);
 		mp4ff_close(mp4fh);
 		free(mp4cb);
@@ -210,7 +208,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 		if(dc->seek && seekPositionFound) {
 			seekPositionFound = 0;
 			chunkLen = 0;
-			cb->end = 0;
+			cb->end = cb->begin;
 			cb->wrap = 0;
 			dc->seek = 0;
 		}
@@ -233,7 +231,7 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
 		}
 
 		if(channels*(dur+offset) > frameInfo.samples) {
-			dur = frameInfo.samples;
+			dur = frameInfo.samples/channels;
 			offset = 0;
 		}
 

src/mp4_decode.h

@@ -2,8 +2,6 @@
  * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
  * This project's homepage is: http://www.musicpd.org
  * 
- * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
- * 
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License as published by
  * the Free Software Foundation; either version 2 of the License, or

src/ogg_decode.c

@@ -95,7 +95,7 @@ int ogg_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
 
 		while(!eof) {
 			if(dc->seek) {
-				cb->end = 0;
+				cb->end = cb->begin;
 				cb->wrap = 0;
 				chunkpos = 0;
 				ov_time_seek_page(&vf,dc->seekWhere);