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mp4/aac cleanups 

git-svn-id: https://svn.musicpd.org/mpd/trunk@276 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes 2004-03-18 16:31:29 +00:00
parent f409d85bbd
commit ad94c1dcf3
4 changed files with 82 additions and 107 deletions
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8
src/decode.c
View File

@ -38,6 +38,9 @@
#ifdef HAVE_AUDIOFILE
#include "audiofile_decode.h"
#endif
#ifdef HAVE_FAAD
#include "mp4_decode.h"
#endif
#include <signal.h>
#include <sys/types.h>
@ -213,6 +216,11 @@ int decoderInit(PlayerControl * pc, Buffer * cb, AudioFormat *af,
dc->error = mp3_decode(cb,af,dc);
break;
#endif
#ifdef HAVE_FAAD
case DECODE_TYPE_MP4:
dc->error = mp4_decode(cb,af,dc);
break;
#endif
#ifdef HAVE_OGG
case DECODE_TYPE_OGG:
dc->error = ogg_decode(cb,af,dc);

6
src/libid3tag/config.h.in
View File

@ -72,6 +72,8 @@
/* Define to empty if `const' does not conform to ANSI C. */
#undef const
/* Define as `__inline' if that's what the C compiler calls it, or to nothing
if it is not supported. */
/* Define to `__inline__' or `__inline' if that's what the C compiler
calls it, or to nothing if 'inline' is not supported under any name. */
#ifndef __cplusplus
#undef inline
#endif

6
src/libmad/config.h.in
View File

@ -125,9 +125,11 @@
/* Define to empty if `const' does not conform to ANSI C. */
#undef const
/* Define as `__inline' if that's what the C compiler calls it, or to nothing
if it is not supported. */
/* Define to `__inline__' or `__inline' if that's what the C compiler
calls it, or to nothing if 'inline' is not supported under any name. */
#ifndef __cplusplus
#undef inline
#endif
/* Define to `int' if <sys/types.h> does not define. */
#undef pid_t

93
src/mp4_decode.c
View File

@ -36,6 +36,8 @@
#include <string.h>
#include <faad.h>
/* all code here is either based on or copied from FAAD2's frontend code */
int mp4_getAACTrack(mp4ff_t *infile) {
/* find AAC track */
int i, rc;
@ -75,20 +77,25 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
mp4ff_t * mp4fh;
mp4ff_callback_t * mp4cb;
int32_t track;
int32_t time;
float time;
int32_t scale;
faacDecHandle decoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
mp4AudioSpecificConfig mp4ASC;
unsigned char * mp4Buffer;
int mp4BufferSize;
unsigned int frameSize;
unsigned int useAacLength;
unsigned long sampleRate;
unsigned char channels;
long sampleId;
long numSamples;
int eof = 0;
int rc;
long dur;
unsigned int sampleCount;
char * sampleBuffer;
unsigned int initial = 1;
size_t sampleBufferLen;
fh = fopen(dc->file,"r");
if(!fh) {
@ -147,16 +154,8 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
af->channels = channels;
time = mp4ff_get_track_duration_use_offsets(mp4fh,track);
scale = mp4ff_time_scale(mp4fh,track);
frameSize = 1024;
useAacLength = 0;
if(mp4Buffer) {
if(AudioSpecificConfig(mp4Buffer,mp4BufferSize,&mp4ASC) >= 0) {
if(mp4ASC.frameLengthFlag==1) frameSize = 960;
if(mp4ASC.sbr_present_flag==1) frameSize*= 2;
}
free(mp4Buffer);
}
if(mp4Buffer) free(mp4Buffer);
if(scale < 0) {
ERROR("Error getting audio format of mp4 AAC track.\n");
@ -172,21 +171,13 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
dc->state = DECODE_STATE_DECODE;
dc->start = 0;
{
int eof = 0;
int rc;
long dur;
unsigned int sampleCount;
unsigned int delay = 0;
char * sampleBuffer;
unsigned int initial = 1;
size_t sampleBufferLen;
time = 0.0;
for(sampleId=0; sampleId<numSamples && !eof; sampleId++) {
if(dc->seek) {
cb->end = 0;
cb->wrap = 0;
//#warning implement seeking here!
#warning implement seeking here!
dc->seek = 0;
}
@ -194,47 +185,22 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
rc = mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
&mp4BufferSize);
if(rc==0) eof = 1;
else {
sampleBuffer = faacDecDecode(decoder,
&frameInfo,
mp4Buffer,
if(rc==0) {
eof = 1;
break;
}
sampleBuffer = faacDecDecode(decoder,&frameInfo,mp4Buffer,
mp4BufferSize);
if(mp4Buffer) free(mp4Buffer);
if(sampleId==0) dur = 0;
if(useAacLength || scale!=sampleRate) {
sampleCount = frameInfo.samples;
}
else {
sampleCount = (unsigned long)(dur *
frameInfo.channels);
if(!useAacLength && !initial &&
(sampleId < numSamples/2) &&
(sampleCount!=
frameInfo.samples))
{
useAacLength = 1;
sampleCount = frameInfo.samples;
}
if(initial && (sampleCount < frameSize*
frameInfo.channels) &&
(frameInfo.samples >
sampleCount))
{
delay = frameInfo.samples -
sampleCount;
}
}
time+=((float)dur)/scale;
sampleCount = (unsigned long)(dur*channels);
if(sampleCount>0) initial =0;
sampleBufferLen = sampleCount*2;
sampleBuffer+=delay*2;
while(sampleBufferLen > 0) {
size_t size = sampleBufferLen>
CHUNK_SIZE?
CHUNK_SIZE:
size_t size = sampleBufferLen>CHUNK_SIZE ? CHUNK_SIZE:
sampleBufferLen;
while(cb->begin==cb->end && cb->wrap &&
!dc->stop && !dc->seek)
@ -248,15 +214,14 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
else if(dc->seek) break;
#ifdef WORDS_BIGENDIAN
pcm_changeBufferEndianness(sampleBuffer,
size,af->bits);
pcm_changeBufferEndianness(sampleBuffer,size,af->bits);
#endif
memcpy(cb->chunks+cb->end*CHUNK_SIZE,
sampleBuffer,size);
sampleBufferLen-=size;
memcpy(cb->chunks+cb->end*CHUNK_SIZE,sampleBuffer,size);
cb->chunkSize[cb->end] = size;
sampleBuffer+=size;
//#warning implement time for AAC
cb->times[cb->end] = 0;
cb->times[cb->end] = time;
++cb->end;
@ -266,7 +231,6 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
}
}
}
}
if(dc->seek) dc->seek = 0;
@ -275,7 +239,6 @@ int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
dc->stop = 0;
}
else dc->state = DECODE_STATE_STOP;
}
faacDecClose(decoder);
mp4ff_close(mp4fh);