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authorWarren Dukes <warren.dukes@gmail.com>2004-03-25 01:08:13 +0000
committerWarren Dukes <warren.dukes@gmail.com>2004-03-25 01:08:13 +0000
commit1de1bfe314bb8c48b3e508f5001927988588fcf6 (patch)
treee6ce56cbabdcd4420b16c4dd9e9707205c5619f1 /src
parent63a62a45142380d67db3d2d5cb57891b2c9371e7 (diff)
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fix and detect differences between faad2 1.1 and 2.0
git-svn-id: https://svn.musicpd.org/mpd/trunk@459 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r--src/aac_decode.c39
-rw-r--r--src/mp4_decode.c20
-rw-r--r--src/mp4ff/mp4ff_int_types.h9
3 files changed, 54 insertions, 14 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;
diff --git a/src/mp4_decode.c b/src/mp4_decode.c
index 6ecce1853..a697a8b3d 100644
--- a/src/mp4_decode.c
+++ b/src/mp4_decode.c
@@ -44,12 +44,24 @@ int mp4_getAACTrack(mp4ff_t *infile) {
for (i = 0; i < numTracks; i++) {
unsigned char *buff = NULL;
int buff_size = 0;
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
mp4AudioSpecificConfig mp4ASC;
+#else
+ unsigned long dummy1_32;
+ unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
+ dummy7_8, dummy8_8;
+#endif
mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
if (buff) {
+#ifdef HAVE_MP4AUDIOSPECIFICCONFIG
rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
+#else
+ rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8,
+ &dummy3_8, &dummy4_8, &dummy5_8,
+ &dummy6_8, &dummy7_8, &dummy8_8);
+#endif
free(buff);
if (rc < 0) continue;
return i;
@@ -220,8 +232,12 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
continue;
}
+#ifdef HAVE_FAAD_BUFLEN_FUNCS
sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer,
mp4BufferSize);
+#else
+ sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer);
+#endif
if(mp4Buffer) free(mp4Buffer);
if(frameInfo.error > 0) {
@@ -234,9 +250,11 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc) {
if(dc->start) {
channels = frameInfo.channels;
+#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
scale = frameInfo.samplerate;
+#endif
+ af->sampleRate = scale;
af->channels = frameInfo.channels;
- af->sampleRate = frameInfo.samplerate;
dc->state = DECODE_STATE_DECODE;
dc->start = 0;
}
diff --git a/src/mp4ff/mp4ff_int_types.h b/src/mp4ff/mp4ff_int_types.h
index 8dbf7bc2e..88651a81c 100644
--- a/src/mp4ff/mp4ff_int_types.h
+++ b/src/mp4ff/mp4ff_int_types.h
@@ -15,9 +15,16 @@ typedef unsigned __int64 uint64_t;
#else
-#include <stdint.h>
+#include "../../config.h"
+#ifdef HAVE_STDINT_H
+#include <stdint.h>
+#else
+#ifdef HAVE_INTTYPES_H
+#include <inttypes.h>
+#endif
#endif
+#endif
#endif