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aac_plugin

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git-svn-id: https://svn.musicpd.org/mpd/trunk@1250 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
Warren Dukes
2004-05-31 02:56:14 +00:00
parent 3aba9b2a66
commit 9f0cbe9e49
4 changed files with 49 additions and 64 deletions
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438
src/inputPlugins/aac_plugin.c Normal file
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/* the Music Player Daemon (MPD)
* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
* This project's homepage is: http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "../inputPlugin.h"
#ifdef HAVE_FAAD
#define AAC_MAX_CHANNELS 6
#include "../utils.h"
#include "../audio.h"
#include "../log.h"
#include "../inputStream.h"
#include "../outputBuffer.h"
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <string.h>
#include <faad.h>
/* all code here is either based on or copied from FAAD2's frontend code */
typedef struct {
InputStream * inStream;
long bytesIntoBuffer;
long bytesConsumed;
long fileOffset;
unsigned char *buffer;
int atEof;
} AacBuffer;
void fillAacBuffer(AacBuffer *b) {
if(b->bytesConsumed > 0) {
int bread;
if(b->bytesIntoBuffer) {
memmove((void *)b->buffer,(void*)(b->buffer+
b->bytesConsumed),b->bytesIntoBuffer);
}
if(!b->atEof) {
bread = readFromInputStream(b->inStream,
(void *)(b->buffer+b->bytesIntoBuffer),
1,b->bytesConsumed);
if(bread!=b->bytesConsumed) b->atEof = 1;
b->bytesIntoBuffer+=bread;
}
b->bytesConsumed = 0;
if(b->bytesIntoBuffer > 3) {
if(memcmp(b->buffer,"TAG",3)==0) b->bytesIntoBuffer = 0;
}
if(b->bytesIntoBuffer > 11) {
if(memcmp(b->buffer,"LYRICSBEGIN",11)==0) {
b->bytesIntoBuffer = 0;
}
}
if(b->bytesIntoBuffer > 8) {
if(memcmp(b->buffer,"APETAGEX",8)==0) {
b->bytesIntoBuffer = 0;
}
}
}
}
void advanceAacBuffer(AacBuffer * b, int bytes) {
b->fileOffset+=bytes;
b->bytesConsumed = bytes;
b->bytesIntoBuffer-=bytes;
}
static int adtsSampleRates[] = {96000,88200,64000,48000,44100,32000,24000,22050,
16000,12000,11025,8000,7350,0,0,0};
int adtsParse(AacBuffer * b, float * length) {
int frames, frameLength;
int tFrameLength = 0;
int sampleRate = 0;
float framesPerSec, bytesPerFrame;
/* Read all frames to ensure correct time and bitrate */
for(frames = 0; ;frames++) {
fillAacBuffer(b);
if(b->bytesIntoBuffer > 7) {
/* check syncword */
if (!((b->buffer[0] == 0xFF) &&
((b->buffer[1] & 0xF6) == 0xF0)))
{
break;
}
if(frames==0) {
sampleRate = adtsSampleRates[
(b->buffer[2]&0x3c)>>2];
}
frameLength = ((((unsigned int)b->buffer[3] & 0x3))
<< 11) | (((unsigned int)b->buffer[4])
<< 3) | (b->buffer[5] >> 5);
tFrameLength+=frameLength;
if(frameLength > b->bytesIntoBuffer) break;
advanceAacBuffer(b,frameLength);
}
else break;
}
framesPerSec = (float)sampleRate/1024.0;
if(frames!=0) {
bytesPerFrame = (float)tFrameLength/(float)(frames*1000);
}
else bytesPerFrame = 0;
if(framesPerSec!=0) *length = (float)frames/framesPerSec;
return 1;
}
void initAacBuffer(InputStream * inStream, AacBuffer * b, float * length,
size_t * retFileread, size_t * retTagsize)
{
size_t fileread;
size_t bread;
size_t tagsize;
if(length) *length = -1;
memset(b,0,sizeof(AacBuffer));
b->inStream = inStream;
fileread = inStream->size;
b->buffer = malloc(FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
memset(b->buffer,0,FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
bread = readFromInputStream(inStream,b->buffer,1,
FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
b->bytesIntoBuffer = bread;
b->bytesConsumed = 0;
b->fileOffset = 0;
if(bread!=FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS) b->atEof = 1;
tagsize = 0;
if(!memcmp(b->buffer,"ID3",3)) {
tagsize = (b->buffer[6] << 21) | (b->buffer[7] << 14) |
(b->buffer[8] << 7) | (b->buffer[9] << 0);
tagsize+=10;
advanceAacBuffer(b,tagsize);
fillAacBuffer(b);
}
if(retFileread) *retFileread = fileread;
if(retTagsize) *retTagsize = tagsize;
if(length==NULL) return;
if((b->buffer[0] == 0xFF) && ((b->buffer[1] & 0xF6) == 0xF0)) {
adtsParse(b, length);
seekInputStream(b->inStream, tagsize, SEEK_SET);
bread = readFromInputStream(b->inStream, b->buffer, 1,
FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS);
if(bread != FAAD_MIN_STREAMSIZE*AAC_MAX_CHANNELS) b->atEof = 1;
else b->atEof = 0;
b->bytesIntoBuffer = bread;
b->bytesConsumed = 0;
b->fileOffset = tagsize;
}
else if(memcmp(b->buffer,"ADIF",4) == 0) {
int bitRate;
int skipSize = (b->buffer[4] & 0x80) ? 9 : 0;
bitRate = ((unsigned int)(b->buffer[4 + skipSize] & 0x0F)<<19) |
((unsigned int)b->buffer[5 + skipSize]<<11) |
((unsigned int)b->buffer[6 + skipSize]<<3) |
((unsigned int)b->buffer[7 + skipSize] & 0xE0);
*length = fileread;
if(*length!=0 && bitRate!=0) *length = *length*8.0/bitRate;
}
}
float getAacFloatTotalTime(char * file) {
AacBuffer b;
float length;
size_t fileread, tagsize;
faacDecHandle decoder;
faacDecConfigurationPtr config;
unsigned long sampleRate;
unsigned char channels;
InputStream inStream;
size_t bread;
if(openInputStream(&inStream,file) < 0) return -1;
initAacBuffer(&inStream,&b,&length,&fileread,&tagsize);
if(length < 0) {
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
faacDecSetConfiguration(decoder,config);
fillAacBuffer(&b);
#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);
}
if(b.buffer) free(b.buffer);
closeInputStream(&inStream);
return length;
}
int getAacTotalTime(char * file) {
int time = -1;
float length;
if((length = getAacFloatTotalTime(file))>=0) time = length+0.5;
return time;
}
int aac_decode(OutputBuffer * cb, DecoderControl * dc) {
float time;
float totalTime;
faacDecHandle decoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
size_t bread;
unsigned long sampleRate;
unsigned char channels;
int eof = 0;
unsigned int sampleCount;
char * sampleBuffer;
size_t sampleBufferLen;
/*float * seekTable;
long seekTableEnd = -1;
int seekPositionFound = 0;*/
mpd_uint16 bitRate = 0;
AacBuffer b;
InputStream inStream;
if((totalTime = getAacFloatTotalTime(dc->file)) < 0) return -1;
if(openInputStream(&inStream,dc->file) < 0) return -1;
initAacBuffer(&inStream,&b,NULL,NULL,NULL);
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
#ifdef HAVE_FAACDECCONFIGURATION_DOWNMATRIX
config->downMatrix = 1;
#endif
#ifdef HAVE_FAACDECCONFIGURATION_DONTUPSAMPLEIMPLICITSBR
config->dontUpSampleImplicitSBR = 0;
#endif
faacDecSetConfiguration(decoder,config);
fillAacBuffer(&b);
#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);
closeInputStream(b.inStream);
if(b.buffer) free(b.buffer);
return -1;
}
dc->audioFormat.bits = 16;
dc->totalTime = totalTime;
time = 0.0;
advanceAacBuffer(&b,bread);
while(!eof) {
fillAacBuffer(&b);
if(b.bytesIntoBuffer==0) {
eof = 1;
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);
ERROR("faad2 error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
eof = 1;
break;
}
#ifdef HAVE_FAACDECFRAMEINFO_SAMPLERATE
sampleRate = frameInfo.samplerate;
#endif
if(dc->state != DECODE_STATE_DECODE) {
dc->audioFormat.channels = frameInfo.channels;
dc->audioFormat.sampleRate = sampleRate;
getOutputAudioFormat(&(dc->audioFormat),
&(cb->audioFormat));
dc->state = DECODE_STATE_DECODE;
}
advanceAacBuffer(&b,frameInfo.bytesconsumed);
sampleCount = (unsigned long)(frameInfo.samples);
if(sampleCount>0) {
bitRate = frameInfo.bytesconsumed*8.0*
frameInfo.channels*sampleRate/
frameInfo.samples/1000+0.5;
time+= (float)(frameInfo.samples)/frameInfo.channels/
sampleRate;
}
sampleBufferLen = sampleCount*2;
sendDataToOutputBuffer(cb, NULL, dc, 0, sampleBuffer,
sampleBufferLen, time, bitRate);
if(dc->seek) {
dc->seekError = 1;
dc->seek = 0;
}
else if(dc->stop) {
eof = 1;
break;
}
} while (!eof);
flushOutputBuffer(cb);
faacDecClose(decoder);
closeInputStream(b.inStream);
if(b.buffer) free(b.buffer);
if(dc->state != DECODE_STATE_DECODE) return -1;
if(dc->seek) {
dc->seekError = 1;
dc->seek = 0;
}
if(dc->stop) {
dc->state = DECODE_STATE_STOP;
dc->stop = 0;
}
else dc->state = DECODE_STATE_STOP;
return 0;
}
MpdTag * aacTagDup(char * file) {
MpdTag * ret = NULL;
int time;
time = getAacTotalTime(file);
if(time>=0) {
if((ret = id3Dup(file))==NULL) ret = newMpdTag();
ret->time = time;
}
return ret;
}
char * aacSuffixes[] = {"aac", NULL};
InputPlugin aacPlugin =
{
"aac",
NULL,
aac_decode,
aacTagDup,
INPUT_PLUGIN_STREAM_FILE,
aacSuffixes,
NULL
};
#else
InputPlugin aacPlugin =
{
NULL,
NULL,
NULL,
NULL,
0,
NULL,
NULL,
};
#endif /* HAVE_FAAD */