diff options
Diffstat (limited to 'src/aac_decode.c')
| -rw-r--r-- | src/aac_decode.c | 39 |
1 files changed, 27 insertions, 12 deletions
diff --git a/src/aac_decode.c b/src/aac_decode.c index c4d8dcf4d..03f291fcc 100644 --- a/src/aac_decode.c +++ b/src/aac_decode.c @@ -210,6 +210,7 @@ float getAacFloatTotalTime(char * file) { unsigned long sampleRate; unsigned char channels; FILE * fp = fopen(file,"r"); + size_t bread; if(fp==NULL) return -1; @@ -223,12 +224,13 @@ float getAacFloatTotalTime(char * file) { faacDecSetConfiguration(decoder,config); fillAacBuffer(&b); - if(faacDecInit(decoder,b.buffer,b.bytesIntoBuffer, - &sampleRate,&channels) >= 0 && - sampleRate > 0 && channels > 0) - { - length = 0; - } +#ifdef HAVE_FAAD_BUFLEN_FUNCS + bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer, + &sampleRate,&channels); +#else + bread = faacDecInit(decoder,b.buffer,&sampleRate,&channels); +#endif + if(bread >= 0 && sampleRate > 0 && channels > 0) length = 0; faacDecClose(decoder); } @@ -291,9 +293,14 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { faacDecSetConfiguration(decoder,config); fillAacBuffer(&b); - if((bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer, - &sampleRate,&channels)) < 0) - { + +#ifdef HAVE_FAAD_BUFLEN_FUNCS + bread = faacDecInit(decoder,b.buffer,b.bytesIntoBuffer, + &sampleRate,&channels); +#else + bread = faacDecInit(decoder,b.buffer,&sampleRate,&channels); +#endif + if(bread < 0) { ERROR("Error not a AAC stream.\n"); faacDecClose(decoder); fclose(b.infile); @@ -317,8 +324,12 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { break; } +#ifdef HAVE_FAAD_BUFLEN_FUNCS sampleBuffer = faacDecDecode(decoder,&frameInfo,b.buffer, b.bytesIntoBuffer); +#else + sampleBuffer = faacDecDecode(decoder,&frameInfo,b.buffer); +#endif if(frameInfo.error > 0) { ERROR("error decoding AAC file: %s\n",dc->file); @@ -328,9 +339,13 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { break; } +#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE + sampleRate = frameInfo.samplerate; +#endif + if(dc->start) { af->channels = frameInfo.channels; - af->sampleRate = frameInfo.samplerate; + af->sampleRate = sampleRate; dc->state = DECODE_STATE_DECODE; dc->start = 0; } @@ -341,10 +356,10 @@ int aac_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) { if(sampleCount>0) { bitRate = frameInfo.bytesconsumed*8.0* - frameInfo.channels*frameInfo.samplerate/ + frameInfo.channels*sampleRate/ frameInfo.samples/1024+0.5; time+= (float)(frameInfo.samples)/frameInfo.channels/ - frameInfo.samplerate; + sampleRate; } sampleBufferLen = sampleCount*2; |