/* 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
*
* libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
*
* 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 "mp4_decode.h"
#ifdef HAVE_FAAD
#include "command.h"
#include "utils.h"
#include "audio.h"
#include "log.h"
#include "pcm_utils.h"
#include "mp4ff/mp4ff.h"
#include <stdio.h>
#include <unistd.h>
#include <stdlib.h>
#include <string.h>
#include <faad.h>
int mp4_getAACTrack(mp4ff_t *infile) {
/* find AAC track */
int i, rc;
int numTracks = mp4ff_total_tracks(infile);
for (i = 0; i < numTracks; i++) {
unsigned char *buff = NULL;
int buff_size = 0;
mp4AudioSpecificConfig mp4ASC;
mp4ff_get_decoder_config(infile, i, &buff, &buff_size);
if (buff) {
rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
free(buff);
if (rc < 0) continue;
return i;
}
}
/* can't decode this */
return -1;
}
uint32_t mp4_readCallback(void *user_data, void *buffer, uint32_t length) {
return fread(buffer, 1, length, (FILE*)user_data);
}
uint32_t mp4_seekCallback(void *user_data, uint64_t position) {
return fseek((FILE*)user_data, position, SEEK_SET);
}
int mp4_decode(Buffer * cb, AudioFormat * af, DecoderControl * dc)
{
FILE * fh;
mp4ff_t * mp4fh;
mp4ff_callback_t * mp4cb;
int32_t track;
int32_t 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;
fh = fopen(dc->file,"r");
if(!fh) {
ERROR("failed to open %s\n",dc->file);
return -1;
}
mp4cb = malloc(sizeof(mp4ff_callback_t));
mp4cb->read = mp4_readCallback;
mp4cb->seek = mp4_seekCallback;
mp4cb->user_data = fh;
mp4fh = mp4ff_open_read(mp4cb);
if(!mp4fh) {
ERROR("Input does not appear to be a mp4 stream.\n");
free(mp4cb);
fclose(fh);
return -1;
}
track = mp4_getAACTrack(mp4fh);
if(track < 0) {
ERROR("No AAC track found in mp4 stream.\n");
mp4ff_close(mp4fh);
fclose(fh);
free(mp4cb);
return -1;
}
decoder = faacDecOpen();
config = faacDecGetCurrentConfiguration(decoder);
config->outputFormat = FAAD_FMT_16BIT;
config->downMatrix = 1;
config->dontUpSampleImplicitSBR = 1;
faacDecSetConfiguration(decoder,config);
af->bits = 16;
mp4Buffer = NULL;
mp4BufferSize = 0;
mp4ff_get_decoder_config(mp4fh,track,&mp4Buffer,&mp4BufferSize);
if(faacDecInit2(decoder,mp4Buffer,mp4BufferSize,&sampleRate,&channels)
< 0)
{
ERROR("Error initializing AAC decoder library.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
free(mp4cb);
fclose(fh);
return -1;
}
af->sampleRate = sampleRate;
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(scale < 0) {
ERROR("Error getting audio format of mp4 AAC track.\n");
faacDecClose(decoder);
mp4ff_close(mp4fh);
fclose(fh);
free(mp4cb);
return -1;
}
cb->totalTime = ((float)time)/scale;
numSamples = mp4ff_num_samples(mp4fh,track);
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;
for(sampleId=0; sampleId<numSamples && !eof; sampleId++) {
if(dc->seek) {
cb->end = 0;
cb->wrap = 0;
//#warning implement seeking here!
dc->seek = 0;
}
dur = mp4ff_get_sample_duration(mp4fh,track,sampleId);
rc = mp4ff_read_sample(mp4fh,track,sampleId,&mp4Buffer,
&mp4BufferSize);
if(rc==0) eof = 1;
else {
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;
}
}
if(sampleCount>0) initial =0;
sampleBufferLen = sampleCount*2;
sampleBuffer+=delay*2;
while(sampleBufferLen > 0) {
size_t size = sampleBufferLen>
CHUNK_SIZE?
CHUNK_SIZE:
sampleBufferLen;
while(cb->begin==cb->end && cb->wrap &&
!dc->stop && !dc->seek)
{
usleep(10000);
}
if(dc->stop) {
eof = 1;
break;
}
else if(dc->seek) break;
#ifdef WORDS_BIGENDIAN
pcm_changeBufferEndianness(sampleBuffer,
size,af->bits);
#endif
memcpy(cb->chunks+cb->end*CHUNK_SIZE,
sampleBuffer,size);
cb->chunkSize[cb->end] = size;
//#warning implement time for AAC
cb->times[cb->end] = 0;
++cb->end;
if(cb->end>=buffered_chunks) {
cb->end = 0;
cb->wrap = 1;
}
}
}
}
if(dc->seek) dc->seek = 0;
if(dc->stop) {
dc->state = DECODE_STATE_STOP;
dc->stop = 0;
}
else dc->state = DECODE_STATE_STOP;
}
faacDecClose(decoder);
mp4ff_close(mp4fh);
fclose(fh);
free(mp4cb);
return 0;
}
#endif /* HAVE_FAAD */
