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-rw-r--r--src/aac_decode.c39
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;