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authorWarren Dukes <warren.dukes@gmail.com>2004-03-22 02:44:22 +0000
committerWarren Dukes <warren.dukes@gmail.com>2004-03-22 02:44:22 +0000
commit145f0b62d26974c5e15ad65c902045fe671a9d05 (patch)
treef86617eeae1a30f1e97a35507b93756a99cf0b29 /src/aac_decode.c
parent4c1eb9225d5a741e1234d48eb38a8df3da908259 (diff)
downloadmpd-145f0b62d26974c5e15ad65c902045fe671a9d05.tar.gz
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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
Diffstat (limited to 'src/aac_decode.c')
-rw-r--r--src/aac_decode.c149
1 files changed, 50 insertions, 99 deletions
diff --git a/src/aac_decode.c b/src/aac_decode.c
index 26e430d06..40e1217e4 100644
--- a/src/aac_decode.c
+++ b/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;
- int mp4BufferSize;
+ size_t bread;
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;
- long offset;
+ int seekPositionFound = 0;*/
mpd_uint16 bitRate = 0;
+ AacBuffer b;
- fh = fopen(dc->file,"r");
- if(!fh) {
- ERROR("failed to open %s\n",dc->file);
- return -1;
- }
-
- mp4cb = malloc(sizeof(mp4ff_callback_t));
- mp4cb->read = mp4_readCallback;
- 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;
- }
+ printf("aac_decode!\n");
- track = mp4_getAACTrack(mp4fh);
- if(track < 0) {
- ERROR("No AAC track found in mp4 stream.\n");
- mp4ff_close(mp4fh);
- fclose(fh);
- free(mp4cb);
+ if(initAacBuffer(dc->file,&b,&totalTime) < 0) {
+ ERROR("Not AAC file no ADTS or ADIF headers found.\n");
return -1;
}
@@ -285,48 +257,33 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
af->bits = 16;
- mp4Buffer = NULL;
- mp4BufferSize = 0;
- mp4ff_get_decoder_config(mp4fh,track,&mp4Buffer,&mp4BufferSize);
-
- if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
- < 0)
+ fillAacBuffer(&b);
+ if((bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer,
+ &sampleRate,&channels)) < 0)
{
- ERROR("Error initializing AAC decoder library.\n");
+ ERROR("Error not a AAC stream.\n");
faacDecClose(decoder);
- mp4ff_close(mp4fh);
- free(mp4cb);
- fclose(fh);
+ fclose(b.infile);
+ if(b.buffer) free(b.buffer);
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);
-
- 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);
+ cb->totalTime = totalTime+0.5;
dc->state = DECODE_STATE_DECODE;
dc->start = 0;
time = 0.0;
- seekTable = malloc(sizeof(float)*numSamples);
+ advanceAacBuffer(&b,bread);
+ fillAacBuffer(&b);
+
+ /*seekTable = malloc(sizeof(float)*numSamples);*/
- for(sampleId=0; sampleId<numSamples && !eof; sampleId++) {
- 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(mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
- &mp4BufferSize) == 0)
- {
- eof = 1;
- continue;
+ if(dc->seek) continue;*/
+
+ if(dc->seek) {
+ /*chunkLen = 0;
+ cb->wrap = 0;
+ cb->end = 0;*/
+ 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) {
- dur = frameInfo.samples;
- offset = 0;
- }
-
- sampleCount = (unsigned long)(dur*channels);
+ sampleCount = (unsigned long)(frameInfo.samples);
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(fh);
- free(mp4cb);
+ fclose(b.infile);
+ if(b.buffer) free(b.buffer);
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;
}