ref: 9831e451d4dffcfe05f7d8d60e3c96c8f04d7eef
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 1.36 2003/07/07 09:48:31 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 <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 */
typedef struct {
long bytes_into_buffer;
long bytes_consumed;
long file_offset;
unsigned char *buffer;
int at_eof;
FILE *infile;
} aac_buffer;
int fill_buffer(aac_buffer *b)
{
int bread;
if (b->bytes_consumed > 0)
{
if (b->bytes_into_buffer)
{
memmove((void*)b->buffer, (void*)(b->buffer + b->bytes_consumed),
b->bytes_into_buffer*sizeof(unsigned char));
}
if (!b->at_eof)
{
bread = fread((void*)(b->buffer + b->bytes_into_buffer), 1,
b->bytes_consumed, b->infile);
if (bread != b->bytes_consumed)
b->at_eof = 1;
b->bytes_into_buffer += bread;
}
b->bytes_consumed = 0;
if (b->bytes_into_buffer > 3)
{
if (memcmp(b->buffer, "TAG", 3) == 0)
b->bytes_into_buffer = 0;
}
if (b->bytes_into_buffer > 11)
{
if (memcmp(b->buffer, "LYRICSBEGIN", 11) == 0)
b->bytes_into_buffer = 0;
}
if (b->bytes_into_buffer > 8)
{
if (memcmp(b->buffer, "APETAGEX", 8) == 0)
b->bytes_into_buffer = 0;
}
}
return 1;
}
void advance_buffer(aac_buffer *b, int bytes)
{
b->file_offset += bytes;
b->bytes_consumed = bytes;
b->bytes_into_buffer -= bytes;
}
int adts_parse(aac_buffer *b, int *bitrate, float *length)
{
static int sample_rates[] = {96000,88200,64000,48000,44100,32000,24000,22050,16000,12000,11025,8000};
int frames, frame_length;
int t_framelength = 0;
int samplerate;
float frames_per_sec, bytes_per_frame;
/* Read all frames to ensure correct time and bitrate */
for (frames = 0; /* */; frames++)
{
fill_buffer(b);
if (b->bytes_into_buffer > 7)
{
/* check syncword */
if (!((b->buffer[0] == 0xFF)&&((b->buffer[1] & 0xF6) == 0xF0)))
break;
if (frames == 0)
samplerate = sample_rates[(b->buffer[2]&0x3c)>>2];
frame_length = ((((unsigned int)b->buffer[3] & 0x3)) << 11)
| (((unsigned int)b->buffer[4]) << 3) | (b->buffer[5] >> 5);
t_framelength += frame_length;
if (frame_length > b->bytes_into_buffer)
break;
advance_buffer(b, frame_length);
} else {
break;
}
}
frames_per_sec = (float)samplerate/1024.0;
if (frames != 0)
bytes_per_frame = (float)t_framelength/(float)(frames*1000);
else
bytes_per_frame = 0;
*bitrate = (int)(8. * bytes_per_frame * frames_per_sec + 0.5);
if (frames_per_sec != 0)
*length = (float)frames/frames_per_sec;
else
*length = 1;
return 1;
}
/* globals */
char *progName;
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, " -i Shows info about the input file.\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: 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 floating point data.\n");
fprintf(stderr, " 5: 64 bit floating point data.\n");
fprintf(stderr, " 6: 16 bit PCM data (dithered).\n");
fprintf(stderr, " 7: 16 bit PCM data (dithered with LIGHT noise shaping).\n");
fprintf(stderr, " 8: 16 bit PCM data (dithered with MEDIUM noise shaping).\n");
fprintf(stderr, " 9: 16 bit PCM data (dithered with HEAVY noise shaping).\n");
fprintf(stderr, " -s X Force the samplerate to X (for RAW files).\n");
fprintf(stderr, " -l X Set object type. Supported object types:\n");
fprintf(stderr, " 0: Main object type.\n");
fprintf(stderr, " 1: LC (Low Complexity) object type.\n");
fprintf(stderr, " 3: LTP (Long Term Prediction) object type.\n");
fprintf(stderr, " 23: LD (Low Delay) object type.\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 object_type, int outputFormat, int fileType,
int infoOnly)
{
int tagsize;
unsigned long samplerate;
unsigned char channels;
void *sample_buffer;
audio_file *aufile;
faacDecHandle hDecoder;
faacDecFrameInfo frameInfo;
faacDecConfigurationPtr config;
char percents[200];
int percent, old_percent = -1;
int bread, fileread;
int header_type = 0;
int bitrate = 0;
float length = 0;
int first_time = 1;
aac_buffer b;
memset(&b, 0, sizeof(aac_buffer));
b.infile = fopen(aacfile, "rb");
if (b.infile == NULL)
{
/* unable to open file */
fprintf(stderr, "Error opening file: %s\n", aacfile);
return 1;
}
fseek(b.infile, 0, SEEK_END);
fileread = ftell(b.infile);
fseek(b.infile, 0, SEEK_SET);
if (!(b.buffer = (unsigned char*)malloc(FAAD_MIN_STREAMSIZE*MAX_CHANNELS)))
{
fprintf(stderr, "Memory allocation error\n");
return 0;
}
memset(b.buffer, 0, FAAD_MIN_STREAMSIZE*MAX_CHANNELS);
bread = fread(b.buffer, 1, FAAD_MIN_STREAMSIZE*MAX_CHANNELS, b.infile);
b.bytes_into_buffer = bread;
b.bytes_consumed = 0;
b.file_offset = 0;
if (bread != 768*6)
b.at_eof = 1;
tagsize = 0;
if (!memcmp(b.buffer, "ID3", 3))
{
/* high bit is not used */
tagsize = (b.buffer[6] << 21) | (b.buffer[7] << 14) |
(b.buffer[8] << 7) | (b.buffer[9] << 0);
tagsize += 10;
advance_buffer(&b, tagsize);
fill_buffer(&b);
}
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 = object_type;
config->outputFormat = outputFormat;
faacDecSetConfiguration(hDecoder, config);
/* get AAC infos for printing */
header_type = 0;
if ((b.buffer[0] == 0xFF) && ((b.buffer[1] & 0xF6) == 0xF0))
{
adts_parse(&b, &bitrate, &length);
fseek(b.infile, tagsize, SEEK_SET);
bread = fread(b.buffer, 1, FAAD_MIN_STREAMSIZE*MAX_CHANNELS, b.infile);
if (bread != FAAD_MIN_STREAMSIZE*MAX_CHANNELS)
b.at_eof = 1;
else
b.at_eof = 0;
b.bytes_into_buffer = bread;
b.bytes_consumed = 0;
b.file_offset = tagsize;
header_type = 1;
} else if (memcmp(b.buffer, "ADIF", 4) == 0) {
int skip_size = (b.buffer[4] & 0x80) ? 9 : 0;
bitrate = ((unsigned int)(b.buffer[4 + skip_size] & 0x0F)<<19) |
((unsigned int)b.buffer[5 + skip_size]<<11) |
((unsigned int)b.buffer[6 + skip_size]<<3) |
((unsigned int)b.buffer[7 + skip_size] & 0xE0);
length = (float)fileread;
if (length != 0)
{
length = ((float)length*8.)/((float)bitrate) + 0.5;
}
bitrate = (int)((float)bitrate/1000.0 + 0.5);
header_type = 2;
}
fill_buffer(&b);
if ((bread = faacDecInit(hDecoder, b.buffer,
b.bytes_into_buffer, &samplerate, &channels)) < 0)
{
/* If some error initializing occured, skip the file */
fprintf(stderr, "Error initializing decoder library.\n");
if (b.buffer)
free(b.buffer);
faacDecClose(hDecoder);
fclose(b.infile);
return 1;
}
advance_buffer(&b, bread);
fill_buffer(&b);
/* print AAC file info */
fprintf(stderr, "AAC file info:\n");
switch (header_type)
{
case 0:
fprintf(stderr, "RAW\n\n");
break;
case 1:
fprintf(stderr, "ADTS, %.3f sec, %d kbps, %d Hz\n\n",
length, bitrate, samplerate);
break;
case 2:
fprintf(stderr, "ADIF, %.3f sec, %d kbps, %d Hz\n\n",
length, bitrate, samplerate);
break;
}
if (infoOnly)
{
faacDecClose(hDecoder);
fclose(b.infile);
if (!first_time)
close_audio_file(aufile);
if (b.buffer)
free(b.buffer);
return 0;
}
do
{
sample_buffer = faacDecDecode(hDecoder, &frameInfo,
b.buffer, b.bytes_into_buffer);
/* update buffer indices */
advance_buffer(&b, frameInfo.bytesconsumed);
if (frameInfo.error > 0)
{
fprintf(stderr, "Error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
}
percent = min((int)(b.file_offset*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, frameInfo.samplerate, frameInfo.channels,
outputFormat, fileType);
} else {
aufile = open_audio_file("-", frameInfo.samplerate, frameInfo.channels,
outputFormat, fileType);
}
if (aufile == NULL)
{
if (b.buffer)
free(b.buffer);
faacDecClose(hDecoder);
fclose(b.infile);
return 0;
}
first_time = 0;
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0))
{
write_audio_file(aufile, sample_buffer, frameInfo.samples);
}
/* fill buffer */
fill_buffer(&b);
if (b.bytes_into_buffer == 0)
sample_buffer = NULL; /* to make sure it stops now */
} while (sample_buffer != NULL);
faacDecClose(hDecoder);
fclose(b.infile);
if (!first_time)
close_audio_file(aufile);
if (b.buffer)
free(b.buffer);
return frameInfo.error;
}
int GetAACTrack(MP4FileHandle infile)
{
/* find AAC track */
unsigned short i;
int 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;
mp4AudioSpecificConfig mp4ASC;
MP4GetTrackESConfiguration(infile, trackId, &buff, &buff_size);
if (buff)
{
rc = AudioSpecificConfig(buff, buff_size, &mp4ASC);
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 *mp4file, char *sndfile, int to_stdout,
int outputFormat, int fileType, int infoOnly)
{
int track;
unsigned int tracks;
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;
char percents[200];
int percent, old_percent = -1;
int first_time = 1;
hDecoder = faacDecOpen();
/* Set configuration */
config = faacDecGetCurrentConfiguration(hDecoder);
config->outputFormat = outputFormat;
faacDecSetConfiguration(hDecoder, config);
infile = MP4Read(mp4file, 0);
if (!infile)
{
/* unable to open file */
fprintf(stderr, "Error opening file: %s\n", mp4file);
return 1;
}
/* print some mp4 file info */
tracks = MP4GetNumberOfTracks(infile, NULL, 0);
if (tracks > 0)
{
char *file_info;
unsigned int i;
fprintf(stderr, "MP4 file info:\n");
for (i = 0; i < tracks; i++)
{
file_info = MP4Info(infile, i+1);
if (file_info)
{
fprintf(stderr, "Track: %s", file_info);
free(file_info);
}
}
fprintf(stderr, "\n");
}
if (infoOnly)
{
faacDecClose(hDecoder);
MP4Close(infile);
return 0;
}
if ((track = GetAACTrack(infile)) < 0)
{
fprintf(stderr, "Unable to find correct AAC sound track in the MP4 file.\n");
faacDecClose(hDecoder);
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, buffer_size);
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, frameInfo.samplerate, frameInfo.channels,
outputFormat, fileType);
} else {
#ifdef _WIN32
setmode(fileno(stdout), O_BINARY);
#endif
aufile = open_audio_file("-", frameInfo.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, "Warning: %s\n",
faacDecGetErrorMessage(frameInfo.error));
}
}
faacDecClose(hDecoder);
MP4Close(infile);
if (!first_time)
close_audio_file(aufile);
return frameInfo.error;
}
int main(int argc, char *argv[])
{
int result;
int infoOnly = 0;
int writeToStdio = 0;
int object_type = LC;
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];
MP4FileHandle infile;
/* System dependant types */
#ifdef _WIN32
long begin, end;
#else
clock_t begin;
#endif
unsigned long cap = faacDecGetCapabilities();
fprintf(stderr, " ****** FAAD2 (Freeware AAC Decoder) V%s ******\n\n", FAAD2_VERSION);
fprintf(stderr, " Build: %s\n", __DATE__);
fprintf(stderr, " Copyright: M. Bakker\n");
fprintf(stderr, " Ahead Software AG\n");
fprintf(stderr, " http://www.audiocoding.com\n");
if (cap & FIXED_POINT_CAP)
fprintf(stderr, " Fixed point version\n");
else
fprintf(stderr, " Floating point version\n");
fprintf(stderr, "\n");
fprintf(stderr, " ****************************************************\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' },
{ "objecttype", 0, 0, 'l' },
{ "info", 0, 0, 'i' },
{ "stdio", 0, 0, 'w' },
{ "help", 0, 0, 'h' }
};
c = getopt_long(argc, argv, "o:s:f:b:l:whi",
long_options, &option_index);
if (c == -1)
break;
switch (c) {
case 'o':
if (optarg) {
outfile_set = 1;
strcpy(audioFileName, optarg);
}
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 > 2))
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 > 9))
showHelp = 1;
}
}
break;
case 'l':
if (optarg) {
char dr[10];
if (sscanf(optarg, "%s", dr) < 1) {
object_type = LC; /* default */
} else {
object_type = atoi(dr);
if ((object_type != LC) &&
(object_type != MAIN) &&
(object_type != LTP) &&
(object_type != LD))
{
showHelp = 1;
}
}
}
break;
case 'w':
writeToStdio = 1;
break;
case 'i':
infoOnly = 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]);
}
mp4file = 1;
infile = MP4Read(aacFileName, 0);
if (!infile)
mp4file = 0;
if (infile) MP4Close(infile);
if (mp4file)
{
result = decodeMP4file(aacFileName, audioFileName, writeToStdio,
outputFormat, format, infoOnly);
} else {
result = decodeAACfile(aacFileName, audioFileName, writeToStdio,
def_srate, object_type, outputFormat, format, infoOnly);
}
if (!result && !infoOnly)
{
#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;
}