ref: 5a241403dd9338b0b46ef16ae88f7abde9b5dfdc
dir: /aacDECdrop/decode.c/
/*
** FAAD - Freeware Advanced Audio Decoder
** Copyright (C) 2002 M. Bakker
**
** 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.
**
** $Id: decode.c,v 1.5 2002/08/07 08:37:15 menno Exp $
** $Id: decode.c,v 1.5 2002/08/07 08:37:15 menno Exp $
**/
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#else
#include <time.h>
#endif
#include <stdio.h>
#include <stdlib.h>
#include <faad.h>
#include <sndfile.h>
#include <mp4.h>
#include "audio.h"
#include "decode.h"
#include "wave_out.h"
#ifndef min
#define min(a,b) ( (a) < (b) ? (a) : (b) )
#endif
#define MAX_CHANNELS 6 /* make this higher to support files with
more channels */
/* FAAD file buffering routines */
/* declare buffering variables */
#define DEC_BUFF_VARS \
int fileread, bytesconsumed, k; \
int buffercount = 0, buffer_index = 0; \
unsigned char *buffer;
/* initialise buffering */
#define INIT_BUFF(file) \
fseek(file, 0, SEEK_END); \
fileread = ftell(file); \
fseek(file, 0, SEEK_SET); \
buffer = (unsigned char*)malloc(FAAD_MIN_STREAMSIZE*MAX_CHANNELS); \
memset(buffer, 0, FAAD_MIN_STREAMSIZE*MAX_CHANNELS); \
fread(buffer, 1, FAAD_MIN_STREAMSIZE*MAX_CHANNELS, file);
/* skip bytes in buffer */
#define UPDATE_BUFF_SKIP(bytes) \
fseek(infile, bytes, SEEK_SET); \
buffer_index += bytes; \
buffercount = 0; \
fread(buffer, 1, FAAD_MIN_STREAMSIZE*MAX_CHANNELS, infile);
/* update buffer */
#define UPDATE_BUFF_READ \
if (bytesconsumed > 0) { \
for (k = 0; k < (FAAD_MIN_STREAMSIZE*MAX_CHANNELS - bytesconsumed); k++) \
buffer[k] = buffer[k + bytesconsumed]; \
fread(buffer + (FAAD_MIN_STREAMSIZE*MAX_CHANNELS) - bytesconsumed, 1, bytesconsumed, infile); \
bytesconsumed = 0; \
}
/* update buffer indices after faacDecDecode */
#define UPDATE_BUFF_IDX(frame) \
bytesconsumed += frame.bytesconsumed; \
buffer_index += frame.bytesconsumed;
/* true if decoding has to stop because of EOF */
#define IS_FILE_END buffer_index >= fileread
/* end buffering */
#define END_BUFF if (buffer) free(buffer);
/* globals */
char *progName;
extern int stop_decoding;
int id3v2_tag(unsigned char *buffer)
{
if (strncmp(buffer, "ID3", 3) == 0) {
unsigned long tagsize;
/* high bit is not used */
tagsize = (buffer[6] << 21) | (buffer[7] << 14) |
(buffer[8] << 7) | (buffer[9] << 0);
tagsize += 10;
return tagsize;
} else {
return 0;
}
}
char *file_ext[] =
{
NULL,
".wav",
".aif",
".au",
".au",
NULL
};
int decodeAACfile(char *sndfile, int def_srate, aac_dec_opt *opt)
{
int tagsize;
unsigned long samplerate;
unsigned char channels;
void *sample_buffer;
FILE *infile;
audio_file *aufile;
faacDecHandle hDecoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
int first_time = 1;
int i;
/* declare variables for buffering */
DEC_BUFF_VARS
infile = fopen(opt->filename, "rb");
if (infile == NULL)
{
/* unable to open file */
error_handler("Error opening file: %s\n", opt->filename);
return 1;
}
INIT_BUFF(infile)
tagsize = id3v2_tag(buffer);
if (tagsize)
{
UPDATE_BUFF_SKIP(tagsize)
}
hDecoder = faacDecOpen();
/* Set the default object type and samplerate */
/* This is useful for RAW AAC files */
config = faacDecGetCurrentConfiguration(hDecoder);
if (def_srate)
config->defSampleRate = def_srate;
config->defObjectType = opt->object_type;
config->outputFormat = opt->output_format;
faacDecSetConfiguration(hDecoder, config);
if((bytesconsumed = faacDecInit(hDecoder, buffer, &samplerate, &channels)) < 0)
{
/* If some error initializing occured, skip the file */
error_handler("Error initializing decoder library.\n");
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return 1;
}
buffer_index += bytesconsumed;
do
{
/* update buffer */
UPDATE_BUFF_READ
sample_buffer = faacDecDecode(hDecoder, &frameInfo, buffer);
/* update buffer indices */
UPDATE_BUFF_IDX(frameInfo)
if (frameInfo.error > 0)
{
error_handler("Error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
}
opt->progress_update((long)fileread, buffer_index);
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{
if(opt->decode_mode == 0)
{
if (Set_WIN_Params (INVALID_FILEDESC, samplerate, SAMPLE_SIZE,
frameInfo.channels) < 0)
{
error_handler("\nCan't access %s\n", "WAVE OUT");
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return (0);
}
}
else
{
aufile = open_audio_file(sndfile, samplerate, frameInfo.channels,
opt->output_format, opt->file_type);
if (aufile == NULL)
{
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return 0;
}
}
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{
if(opt->decode_mode == 0)
WIN_Play_Samples((short*)sample_buffer, frameInfo.channels*frameInfo.samples);
else
write_audio_file(aufile, sample_buffer, frameInfo.samples);
}
if (buffer_index >= fileread)
sample_buffer = NULL; /* to make sure it stops now */
if(stop_decoding)
break;
} while (sample_buffer != NULL);
faacDecClose(hDecoder);
fclose(infile);
if(opt->decode_mode == 0)
WIN_Audio_close();
else
{
if (!first_time)
close_audio_file(aufile);
}
END_BUFF
return frameInfo.error;
}
int GetAACTrack(MP4FileHandle infile)
{
/* find AAC track */
int i, rc;
int numTracks = MP4GetNumberOfTracks(infile, NULL, /* subType */ 0);
for (i = 0; i < numTracks; i++)
{
MP4TrackId trackId = MP4FindTrackId(infile, i, NULL, /* subType */ 0);
const char* trackType = MP4GetTrackType(infile, trackId);
if (!strcmp(trackType, MP4_AUDIO_TRACK_TYPE))
{
unsigned char *buff = NULL;
int buff_size = 0;
unsigned long dummy1_32;
unsigned char dummy2_8, dummy3_8, dummy4_8, dummy5_8, dummy6_8,
dummy7_8, dummy8_8;
MP4GetTrackESConfiguration(infile, trackId, &buff, &buff_size);
if (buff)
{
rc = AudioSpecificConfig(buff, &dummy1_32, &dummy2_8, &dummy3_8, &dummy4_8,
&dummy5_8, &dummy6_8, &dummy7_8, &dummy8_8);
free(buff);
if (rc < 0)
return -1;
return trackId;
}
}
}
/* can't decode this */
return -1;
}
unsigned long srates[] =
{
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000,
12000, 11025, 8000
};
int decodeMP4file(char *sndfile, aac_dec_opt *opt)
{
int track;
unsigned long samplerate;
unsigned char channels;
void *sample_buffer;
MP4FileHandle infile;
MP4SampleId sampleId, numSamples;
audio_file *aufile;
faacDecHandle hDecoder;
faacDecConfigurationPtr config;
faacDecFrameInfo frameInfo;
unsigned char *buffer;
int buffer_size;
int first_time = 1;
hDecoder = faacDecOpen();
infile = MP4Read(opt->filename, 0);
if (!infile)
{
/* unable to open file */
error_handler("Error opening file: %s\n", opt->filename);
return 1;
}
if ((track = GetAACTrack(infile)) < 0)
{
error_handler("Unable to find correct AAC sound track in the MP4 file.\n");
MP4Close(infile);
return 1;
}
buffer = NULL;
buffer_size = 0;
MP4GetTrackESConfiguration(infile, track, &buffer, &buffer_size);
if(faacDecInit2(hDecoder, buffer, buffer_size, &samplerate, &channels) < 0)
{
/* If some error initializing occured, skip the file */
error_handler("Error initializing decoder library.\n");
faacDecClose(hDecoder);
MP4Close(infile);
return 1;
}
if (buffer)
free(buffer);
numSamples = MP4GetTrackNumberOfSamples(infile, track);
for (sampleId = 1; sampleId <= numSamples; sampleId++)
{
int rc;
/* get access unit from MP4 file */
buffer = NULL;
buffer_size = 0;
rc = MP4ReadSample(infile, track, sampleId, &buffer, &buffer_size, NULL, NULL, NULL, NULL);
if (rc == 0)
{
error_handler("Reading from MP4 file failed.\n");
faacDecClose(hDecoder);
MP4Close(infile);
return 1;
}
sample_buffer = faacDecDecode(hDecoder, &frameInfo, buffer);
if (buffer)
free(buffer);
opt->progress_update((long)numSamples, sampleId);
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{
if(opt->decode_mode == 0)
{
if (Set_WIN_Params (INVALID_FILEDESC, samplerate, SAMPLE_SIZE,
frameInfo.channels) < 0)
{
error_handler("\nCan't access %s\n", "WAVE OUT");
faacDecClose(hDecoder);
MP4Close(infile);
return (0);
}
}
else
{
aufile = open_audio_file(sndfile, samplerate, frameInfo.channels,
opt->output_format, opt->file_type);
if (aufile == NULL)
{
faacDecClose(hDecoder);
MP4Close(infile);
return 0;
}
}
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{
if(opt->decode_mode == 0)
WIN_Play_Samples((short*)sample_buffer, frameInfo.channels*frameInfo.samples);
else
write_audio_file(aufile, sample_buffer, frameInfo.samples);
}
if (frameInfo.error > 0)
{
error_handler("Error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
break;
}
if(stop_decoding)
break;
}
faacDecClose(hDecoder);
MP4Close(infile);
if(opt->decode_mode == 0)
WIN_Audio_close();
else
{
if (!first_time)
close_audio_file(aufile);
}
return frameInfo.error;
}
int str_no_case_comp(char const *str1, char const *str2, unsigned long len)
{
signed int c1 = 0, c2 = 0;
while (len--)
{
c1 = tolower(*str1++);
c2 = tolower(*str2++);
if (c1 == 0 || c1 != c2)
break;
}
return c1 - c2;
}
int aac_decode(aac_dec_opt *opt)
{
int result;
int def_srate = 0;
int outfile_set = 0;
int mp4file = 0;
char *fnp;
char audioFileName[MAX_PATH];
/* point to the specified file name */
strcpy(audioFileName, opt->filename);
fnp = (char *)strrchr(audioFileName,'.');
if (fnp)
fnp[0] = '\0';
strcat(audioFileName, file_ext[opt->file_type]);
fnp = (char *)strrchr(opt->filename, '.');
if (!str_no_case_comp(fnp, ".MP4", 4))
mp4file = 1;
if (mp4file)
{
result = decodeMP4file(audioFileName, opt);
}
else
{
result = decodeAACfile(audioFileName, def_srate, opt);
}
return 0;
}