ref: 36ac51109224df99ad0e96609d37d126864f51ff
dir: /frontend/main.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: main.c,v 1.8 2002/01/19 09:39:00 menno Exp $
**/
#ifdef _WIN32
#define WIN32_LEAN_AND_MEAN
#include <windows.h>
#else
#include <time.h>
#endif
#include <stdio.h>
#include <stdlib.h>
#include <getopt.h>
#include <faad.h>
#include <sndfile.h>
#include <mp4.h>
#include "audio.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;
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",
".pcm",
NULL
};
void usage(void)
{
fprintf(stderr, "\nUsage:\n");
fprintf(stderr, "%s [options] infile.aac\n", progName);
fprintf(stderr, "Options:\n");
fprintf(stderr, " -h Shows this help screen.\n");
fprintf(stderr, " -o X Set output filename.\n");
fprintf(stderr, " -f X Set output format. Valid values for X are:\n");
fprintf(stderr, " 1: Microsoft WAV format (default).\n");
fprintf(stderr, " 2: Apple/SGI AIFF format.\n");
fprintf(stderr, " 3: Sun/NeXT AU format.\n");
fprintf(stderr, " 4: DEC AU format.\n");
fprintf(stderr, " 5: RAW PCM data.\n");
fprintf(stderr, " -b X Set output sample format. Valid values for X are:\n");
fprintf(stderr, " 1: 16 bit PCM data (default).\n");
fprintf(stderr, " 2: 24 bit PCM data.\n");
fprintf(stderr, " 3: 32 bit PCM data.\n");
fprintf(stderr, " 4: 32 bit floats.\n");
fprintf(stderr, " -s X Force the samplerate to X (for RAW files).\n");
fprintf(stderr, " -l Use Long Term Prediction (for RAW files).\n");
fprintf(stderr, " -w Write output to stdio instead of a file.\n");
fprintf(stderr, "Example:\n");
fprintf(stderr, " faad infile.aac\n");
fprintf(stderr, " faad infile.mp4\n");
fprintf(stderr, " faad -o outfile.wav infile.aac\n");
fprintf(stderr, " faad -w infile.aac > outfile.wav\n");
return;
}
int decodeAACfile(char *aacfile, char *sndfile, int to_stdout,
int def_srate, int use_ltp, int outputFormat, int fileType)
{
int tagsize;
unsigned long samplerate, channels;
void *sample_buffer;
FILE *infile;
audio_file *aufile;
faacDecHandle hDecoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
char percents[200];
int percent, old_percent = -1;
int first_time = 1;
/* declare variables for buffering */
DEC_BUFF_VARS
infile = fopen(aacfile, "rb");
if (infile == NULL)
{
/* unable to open file */
fprintf(stderr, "Error opening file: %s\n", aacfile);
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;
if (use_ltp)
config->defObjectType = LTP;
config->outputFormat = outputFormat;
faacDecSetConfiguration(hDecoder, config);
if((bytesconsumed = faacDecInit(hDecoder, buffer, &samplerate,
&channels)) < 0)
{
/* If some error initializing occured, skip the file */
fprintf(stderr, "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)
{
fprintf(stderr, "Error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
}
percent = min((int)(buffer_index*100)/fileread, 100);
if (percent > old_percent)
{
old_percent = percent;
sprintf(percents, "%d%% decoding %s.", percent, aacfile);
fprintf(stderr, "%s\r", percents);
#ifdef _WIN32
SetConsoleTitle(percents);
#endif
}
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{
if(!to_stdout)
{
aufile = open_audio_file(sndfile, samplerate, frameInfo.channels,
outputFormat, fileType);
} else {
setmode(fileno(stdout), O_BINARY);
aufile = open_audio_file("-", samplerate, frameInfo.channels,
outputFormat, fileType);
}
if (aufile == NULL)
{
END_BUFF
faacDecClose(hDecoder);
fclose(infile);
return 0;
}
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{
write_audio_file(aufile, sample_buffer, frameInfo.samples);
}
if (buffer_index >= fileread)
sample_buffer = NULL; /* to make sure it stops now */
} while (sample_buffer != NULL);
faacDecClose(hDecoder);
fclose(infile);
close_audio_file(aufile);
END_BUFF
return frameInfo.error;
}
int GetAACTrack(MP4FileHandle *infile)
{
/* find AAC track */
int i;
int numTracks = MP4GetNumberOfTracks(infile, NULL);
for (i = 0; i < numTracks; i++)
{
MP4TrackId trackId = MP4FindTrackId(infile, i, NULL);
const char* trackType = MP4GetTrackType(infile, trackId);
if (!strcmp(trackType, MP4_AUDIO_TRACK_TYPE))
{
int type = MP4GetTrackAudioType(infile, trackId);
switch (type)
{
case MP4_MPEG4_AUDIO_TYPE:
case MP4_MPEG2_AAC_MAIN_AUDIO_TYPE:
case MP4_MPEG2_AAC_LC_AUDIO_TYPE:
case MP4_MPEG2_AAC_SSR_AUDIO_TYPE:
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 *mp4file, char *sndfile, int to_stdout,
int outputFormat, int fileType)
{
int track;
unsigned long samplerate, channels;
void *sample_buffer;
MP4FileHandle *infile;
MP4SampleId sampleId, numSamples;
audio_file *aufile;
faacDecHandle hDecoder;
faacDecConfigurationPtr config;
faacDecFrameInfo frameInfo;
unsigned char *buffer;
int buffer_size;
char percents[200];
int percent, old_percent = -1;
int first_time = 1;
hDecoder = faacDecOpen();
infile = MP4Read(mp4file, 0);
if (!infile)
{
/* unable to open file */
fprintf(stderr, "Error opening file: %s\n", mp4file);
return 1;
}
if ((track = GetAACTrack(infile)) < 0)
{
fprintf(stderr, "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 */
fprintf(stderr, "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 acces unit from MP4 file */
buffer = NULL;
buffer_size = 0;
rc = MP4ReadSample(infile, track, sampleId, &buffer, &buffer_size,
NULL, NULL, NULL, NULL);
if (rc == 0)
{
fprintf(stderr, "Reading from MP4 file failed.\n");
faacDecClose(hDecoder);
MP4Close(infile);
return 1;
}
sample_buffer = faacDecDecode(hDecoder, &frameInfo, buffer);
if (buffer) free(buffer);
percent = min((int)(sampleId*100)/numSamples, 100);
if (percent > old_percent)
{
old_percent = percent;
sprintf(percents, "%d%% decoding %s.", percent, mp4file);
fprintf(stderr, "%s\r", percents);
#ifdef _WIN32
SetConsoleTitle(percents);
#endif
}
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{
if(!to_stdout)
{
aufile = open_audio_file(sndfile, samplerate, frameInfo.channels,
outputFormat, fileType);
} else {
setmode(fileno(stdout), O_BINARY);
aufile = open_audio_file("-", samplerate, frameInfo.channels,
outputFormat, fileType);
}
if (aufile == NULL)
{
faacDecClose(hDecoder);
MP4Close(infile);
return 0;
}
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{
write_audio_file(aufile, sample_buffer, frameInfo.samples);
}
if (frameInfo.error > 0)
{
fprintf(stderr, "Error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
break;
}
}
faacDecClose(hDecoder);
MP4Close(infile);
close_audio_file(aufile);
return frameInfo.error;
}
int main(int argc, char *argv[])
{
int result;
int writeToStdio = 0;
int use_ltp = 0;
int def_srate = 0;
int format = 1;
int outputFormat = FAAD_FMT_16BIT;
int outfile_set = 0;
int showHelp = 0;
int mp4file = 0;
char *fnp;
char aacFileName[255];
char audioFileName[255];
FILE *infile;
/* System dependant types */
#ifdef _WIN32
long begin, end;
#else
clock_t begin;
#endif
fprintf(stderr, "FAAD (Freeware AAC Decoder) Compiled on: " __DATE__ "\n");
fprintf(stderr, "Copyright: M. Bakker\n");
fprintf(stderr, " http://www.audiocoding.com\n\n");
/* begin process command line */
progName = argv[0];
while (1) {
int c = -1;
int option_index = 0;
static struct option long_options[] = {
{ "outfile", 0, 0, 'o' },
{ "format", 0, 0, 'f' },
{ "bits", 0, 0, 'b' },
{ "samplerate", 0, 0, 's' },
{ "ltp", 0, 0, 'l' },
{ "stdio", 0, 0, 'w' },
{ "help", 0, 0, 'h' }
};
c = getopt_long(argc, argv, "o:s:f:b:lwh",
long_options, &option_index);
if (c == -1)
break;
switch (c) {
case 'o':
if (optarg) {
char dr[255];
if (sscanf(optarg, "%s", dr) < 1) {
outfile_set = 0;
} else {
outfile_set = 1;
strcpy(audioFileName, dr);
}
}
break;
case 's':
if (optarg) {
char dr[10];
if (sscanf(optarg, "%s", dr) < 1) {
def_srate = 0;
} else {
def_srate = atoi(dr);
}
}
break;
case 'f':
if (optarg) {
char dr[10];
if (sscanf(optarg, "%s", dr) < 1) {
format = 1;
} else {
format = atoi(dr);
if ((format < 1) || (format > 5))
showHelp = 1;
}
}
break;
case 'b':
if (optarg) {
char dr[10];
if (sscanf(optarg, "%s", dr) < 1) {
outputFormat = FAAD_FMT_16BIT; /* just use default */
} else {
outputFormat = atoi(dr);
if ((outputFormat < 1) || (outputFormat > 4))
showHelp = 1;
}
}
break;
case 'l':
use_ltp = 1;
break;
case 'w':
writeToStdio = 1;
break;
case 'h':
showHelp = 1;
break;
default:
break;
}
}
/* check that we have at least two non-option arguments */
/* Print help if requested */
if (((argc - optind) < 1) || showHelp)
{
usage();
return 1;
}
/* point to the specified file name */
strcpy(aacFileName, argv[optind]);
#ifdef _WIN32
begin = GetTickCount();
#else
begin = clock();
#endif
/* Only calculate the path and open the file for writing if
we are not writing to stdout.
*/
if(!writeToStdio && !outfile_set)
{
strcpy(audioFileName, aacFileName);
fnp = (char *)strrchr(audioFileName,'.');
if (fnp)
fnp[0] = '\0';
strcat(audioFileName, file_ext[format]);
}
fnp = (char *)strrchr(aacFileName, '.');
if (!stricmp(fnp, ".MP4"))
mp4file = 1;
if (mp4file)
{
result = decodeMP4file(aacFileName, audioFileName, writeToStdio,
outputFormat, format);
} else {
result = decodeAACfile(aacFileName, audioFileName, writeToStdio,
def_srate, use_ltp, outputFormat, format);
}
if (!result)
{
#ifdef _WIN32
end = GetTickCount();
fprintf(stderr, "Decoding %s took: %d sec.\n", aacFileName,
(end-begin)/1000);
SetConsoleTitle("FAAD");
#else
/* clock() grabs time since the start of the app but when we decode
multiple files, each file has its own starttime (begin).
*/
fprintf(stderr, "Decoding %s took: %5.2f sec.\n", aacFileName,
(float)(clock() - begin)/(float)CLOCKS_PER_SEC);
#endif
}
return 0;
}