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author | Warren Dukes <warren.dukes@gmail.com> | 2004-03-22 04:20:19 +0000 |
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committer | Warren Dukes <warren.dukes@gmail.com> | 2004-03-22 04:20:19 +0000 |
commit | da969faed174f975841920332df2a3d551c4b3ea (patch) | |
tree | 54ca537990ec414037f59565c855178cfc797f8a /src/aac_decode.c | |
parent | 145f0b62d26974c5e15ad65c902045fe671a9d05 (diff) | |
download | mpd-da969faed174f975841920332df2a3d551c4b3ea.tar.gz mpd-da969faed174f975841920332df2a3d551c4b3ea.tar.xz mpd-da969faed174f975841920332df2a3d551c4b3ea.zip |
implement acc-raw decoding, no seeking for aac
git-svn-id: https://svn.musicpd.org/mpd/trunk@357 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src/aac_decode.c')
-rw-r--r-- | src/aac_decode.c | 122 |
1 files changed, 60 insertions, 62 deletions
diff --git a/src/aac_decode.c b/src/aac_decode.c index 40e1217e4..f049d0771 100644 --- a/src/aac_decode.c +++ b/src/aac_decode.c @@ -132,17 +132,18 @@ int adtsParse(AacBuffer * b, float * length) { return 1; } -int initAacBuffer(char * file, AacBuffer * b, float * length) { +void initAacBuffer(FILE * fp, AacBuffer * b, float * length, + size_t * retFileread, size_t * retTagsize) +{ size_t fileread; size_t bread; size_t tagsize; - *length = -1; + if(length) *length = -1; memset(b,0,sizeof(AacBuffer)); - b->infile = fopen(file,"r"); - if(b->infile == NULL) return -1; + b->infile = fp; fseek(b->infile,0,SEEK_END); fileread = ftell(b->infile); @@ -169,6 +170,11 @@ int initAacBuffer(char * file, AacBuffer * b, float * length) { fillAacBuffer(b); } + if(retFileread) *retFileread = fileread; + if(retTagsize) *retTagsize = tagsize; + + if(!length) return; + if((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) { adtsParse(b, length); fseek(b->infile, tagsize, SEEK_SET); @@ -193,26 +199,53 @@ int initAacBuffer(char * file, AacBuffer * b, float * length) { *length = fileread; if(*length!=0 && bitRate!=0) *length = *length*8.0/bitRate; } - - if(*length<0) { - fclose(b->infile); - if(b->buffer) free(b->buffer); - return -1; - } - - return 0; } -int getAacTotalTime(char * file) { +float getAacFloatTotalTime(char * file) { AacBuffer b; float length; + size_t fileread, tagsize; + faacDecHandle decoder; + faacDecConfigurationPtr config; + unsigned long sampleRate; + unsigned char channels; + FILE * fp = fopen(file,"r"); - if(initAacBuffer(file,&b,&length) < 0) return -1; + if(fp==NULL) return -1; + + initAacBuffer(fp,&b,&length,&fileread,&tagsize); + + if(length < 0) { + decoder = faacDecOpen(); + + config = faacDecGetCurrentConfiguration(decoder); + config->outputFormat = FAAD_FMT_16BIT; + faacDecSetConfiguration(decoder,config); + + fillAacBuffer(&b); + if(faacDecInit(decoder,b.buffer,b.bytesIntoBuffer, + &sampleRate,&channels) >= 0 && + sampleRate > 0 && channels > 0) + { + length = 0; + } + + faacDecClose(decoder); + } if(b.buffer) free(b.buffer); fclose(b.infile); - return (int)(length+0.5); + return length; +} + +int getAacTotalTime(char * file) { + int time = -1; + float length; + + if((length = getAacFloatTotalTime(file))>=0) time = length+0.5; + + return time; } @@ -235,13 +268,15 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { int seekPositionFound = 0;*/ mpd_uint16 bitRate = 0; AacBuffer b; + FILE * fp; - printf("aac_decode!\n"); + if((totalTime = getAacFloatTotalTime(dc->file)) < 0) return -1; - if(initAacBuffer(dc->file,&b,&totalTime) < 0) { - ERROR("Not AAC file no ADTS or ADIF headers found.\n"); - return -1; - } + fp = fopen(dc->file,"r"); + + if(fp==NULL) return -1; + + initAacBuffer(fp,&b,NULL,NULL,NULL); decoder = faacDecOpen(); @@ -255,8 +290,6 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { #endif faacDecSetConfiguration(decoder,config); - af->bits = 16; - fillAacBuffer(&b); if((bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer, &sampleRate,&channels)) < 0) @@ -270,8 +303,9 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { af->sampleRate = sampleRate; af->channels = channels; + af->bits = 16; - cb->totalTime = totalTime+0.5; + cb->totalTime = totalTime; dc->state = DECODE_STATE_DECODE; dc->start = 0; @@ -280,44 +314,8 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { advanceAacBuffer(&b,bread); fillAacBuffer(&b); - /*seekTable = malloc(sizeof(float)*numSamples);*/ - do { - /*if(dc->seek && seekTableEnd>1 && - seekTable[seekTableEnd]>=dc->seekWhere) - { - int i = 2; - while(seekTable[i]<dc->seekWhere) i++; - sampleId = i-1; - time = seekTable[sampleId]; - } - - if(sampleId>seekTableEnd) { - seekTable[sampleId] = time; - seekTableEnd = sampleId; - } - - if(sampleId==0) dur = 0; - if(offset>dur) dur = 0; - else dur-=offset; - time+=((float)dur)/scale; - - if(dc->seek && time>dc->seekWhere) seekPositionFound = 1; - - if(dc->seek && seekPositionFound) { - seekPositionFound = 0; - chunkLen = 0; - cb->end = 0; - cb->wrap = 0; - dc->seek = 0; - } - - if(dc->seek) continue;*/ - if(dc->seek) { - /*chunkLen = 0; - cb->wrap = 0; - cb->end = 0;*/ dc->seekError = 1; dc->seek = 0; } @@ -335,9 +333,10 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { if(sampleCount>0) { bitRate = frameInfo.bytesconsumed*8.0* - frameInfo.channels*sampleRate/ + frameInfo.channels*frameInfo.samplerate/ frameInfo.samples/1024+0.5; - time+= (float)(frameInfo.samples)/channels/sampleRate; + time+= (float)(frameInfo.samples)/frameInfo.channels/ + frameInfo.samplerate; } sampleBufferLen = sampleCount*2; @@ -392,7 +391,6 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { chunkLen = 0; } - /*free(seekTable);*/ faacDecClose(decoder); fclose(b.infile); if(b.buffer) free(b.buffer); |