ref: 2cff3a9f96eabd25287392db3454bdb40d14f4b2
dir: /frontend/main.c/
/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003 M. Bakker, Ahead Software AG, http://www.nero.com
**
** 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.
**
** Any non-GPL usage of this software or parts of this software is strictly
** forbidden.
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Ahead Software through Mpeg4AAClicense@nero.com.
**
** $Id: main.c,v 1.49 2003/09/30 08:07:47 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;
}
static int adts_sample_rates[] = {96000,88200,64000,48000,44100,32000,24000,22050,16000,12000,11025,8000,7350,0,0,0};
int adts_parse(aac_buffer *b, int *bitrate, float *length)
{
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 = adts_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/1024.0f;
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;
}
/* MicroSoft channel definitions */
#define SPEAKER_FRONT_LEFT 0x1
#define SPEAKER_FRONT_RIGHT 0x2
#define SPEAKER_FRONT_CENTER 0x4
#define SPEAKER_LOW_FREQUENCY 0x8
#define SPEAKER_BACK_LEFT 0x10
#define SPEAKER_BACK_RIGHT 0x20
#define SPEAKER_FRONT_LEFT_OF_CENTER 0x40
#define SPEAKER_FRONT_RIGHT_OF_CENTER 0x80
#define SPEAKER_BACK_CENTER 0x100
#define SPEAKER_SIDE_LEFT 0x200
#define SPEAKER_SIDE_RIGHT 0x400
#define SPEAKER_TOP_CENTER 0x800
#define SPEAKER_TOP_FRONT_LEFT 0x1000
#define SPEAKER_TOP_FRONT_CENTER 0x2000
#define SPEAKER_TOP_FRONT_RIGHT 0x4000
#define SPEAKER_TOP_BACK_LEFT 0x8000
#define SPEAKER_TOP_BACK_CENTER 0x10000
#define SPEAKER_TOP_BACK_RIGHT 0x20000
#define SPEAKER_RESERVED 0x80000000
long aacChannelConfig2wavexChannelMask(faacDecFrameInfo *hInfo)
{
if (hInfo->channels == 6 && hInfo->num_lfe_channels)
{
return SPEAKER_FRONT_LEFT + SPEAKER_FRONT_RIGHT +
SPEAKER_FRONT_CENTER + SPEAKER_LOW_FREQUENCY +
SPEAKER_BACK_LEFT + SPEAKER_BACK_RIGHT;
} else {
return 0;
}
}
char *position2string(int position)
{
switch (position)
{
case FRONT_CHANNEL_CENTER: return "Center front";
case FRONT_CHANNEL_LEFT: return "Left front";
case FRONT_CHANNEL_RIGHT: return "Right front";
case SIDE_CHANNEL_LEFT: return "Left side";
case SIDE_CHANNEL_RIGHT: return "Right side";
case BACK_CHANNEL_LEFT: return "Left back";
case BACK_CHANNEL_RIGHT: return "Right back";
case BACK_CHANNEL_CENTER: return "Center back";
case LFE_CHANNEL: return "LFE";
case UNKNOWN_CHANNEL: return "Unknown";
default: return "";
}
return "";
}
void print_channel_info(faacDecFrameInfo *frameInfo)
{
/* print some channel info */
int i;
long channelMask = aacChannelConfig2wavexChannelMask(frameInfo);
fprintf(stderr, " ---------------------\n");
if (frameInfo->num_lfe_channels > 0)
{
fprintf(stderr, " | Config: %2d.%d Ch |", frameInfo->channels-frameInfo->num_lfe_channels, frameInfo->num_lfe_channels);
} else {
fprintf(stderr, " | Config: %2d Ch |", frameInfo->channels);
}
if (channelMask)
fprintf(stderr, " WARNING: channels are reordered according to\n");
else
fprintf(stderr, "\n");
fprintf(stderr, " ---------------------");
if (channelMask)
fprintf(stderr, " MS defaults defined in WAVE_FORMAT_EXTENSIBLE\n");
else
fprintf(stderr, "\n");
fprintf(stderr, " | Ch | Position |\n");
fprintf(stderr, " ---------------------\n");
for (i = 0; i < frameInfo->channels; i++)
{
fprintf(stderr, " | %.2d | %-14s |\n", i, position2string((int)frameInfo->channel_position[i]));
}
fprintf(stderr, " ---------------------\n");
fprintf(stderr, "\n");
}
int FindAdtsSRIndex(int sr)
{
int i;
for (i = 0; i < 16; i++)
{
if (sr == adts_sample_rates[i])
return i;
}
return 16 - 1;
}
unsigned char *MakeAdtsHeader(int *dataSize, faacDecFrameInfo *hInfo, int old_format)
{
unsigned char *data;
int profile = (hInfo->object_type - 1) & 0x3;
int sr_index = ((hInfo->sbr == SBR_UPSAMPLED) || (hInfo->sbr == NO_SBR_UPSAMPLED)) ?
FindAdtsSRIndex(hInfo->samplerate / 2) : FindAdtsSRIndex(hInfo->samplerate);
int skip = (old_format) ? 8 : 7;
int framesize = skip + hInfo->bytesconsumed;
if (hInfo->header_type == ADTS)
framesize -= skip;
*dataSize = 7;
data = malloc(*dataSize * sizeof(unsigned char));
memset(data, 0, *dataSize * sizeof(unsigned char));
data[0] += 0xFF; /* 8b: syncword */
data[1] += 0xF0; /* 4b: syncword */
/* 1b: mpeg id = 0 */
/* 2b: layer = 0 */
data[1] += 1; /* 1b: protection absent */
data[2] += ((profile << 6) & 0xC0); /* 2b: profile */
data[2] += ((sr_index << 2) & 0x3C); /* 4b: sampling_frequency_index */
/* 1b: private = 0 */
data[2] += ((hInfo->channels >> 2) & 0x1); /* 1b: channel_configuration */
data[3] += ((hInfo->channels << 6) & 0xC0); /* 2b: channel_configuration */
/* 1b: original */
/* 1b: home */
/* 1b: copyright_id */
/* 1b: copyright_id_start */
data[3] += ((framesize >> 11) & 0x3); /* 2b: aac_frame_length */
data[4] += ((framesize >> 3) & 0xFF); /* 8b: aac_frame_length */
data[5] += ((framesize << 5) & 0xE0); /* 3b: aac_frame_length */
data[5] += ((0x7FF >> 6) & 0x1F); /* 5b: adts_buffer_fullness */
data[6] += ((0x7FF << 2) & 0x3F); /* 6b: adts_buffer_fullness */
/* 2b: num_raw_data_blocks */
return data;
}
/* globals */
char *progName;
char *file_ext[] =
{
NULL,
".wav",
".aif",
".au",
".au",
".pcm",
NULL
};
void usage(void)
{
fprintf(stdout, "\nUsage:\n");
fprintf(stdout, "%s [options] infile.aac\n", progName);
fprintf(stdout, "Options:\n");
fprintf(stdout, " -h Shows this help screen.\n");
fprintf(stdout, " -i Shows info about the input file.\n");
fprintf(stdout, " -a X Write MPEG-4 AAC ADTS output file.\n");
fprintf(stdout, " -t Assume old ADTS format.\n");
fprintf(stdout, " -o X Set output filename.\n");
fprintf(stdout, " -f X Set output format. Valid values for X are:\n");
fprintf(stdout, " 1: Microsoft WAV format (default).\n");
fprintf(stdout, " 2: RAW PCM data.\n");
fprintf(stdout, " -b X Set output sample format. Valid values for X are:\n");
fprintf(stdout, " 1: 16 bit PCM data (default).\n");
fprintf(stdout, " 2: 24 bit PCM data.\n");
fprintf(stdout, " 3: 32 bit PCM data.\n");
fprintf(stdout, " 4: 32 bit floating point data.\n");
fprintf(stdout, " 5: 64 bit floating point data.\n");
fprintf(stdout, " 6: 16 bit PCM data (dithered).\n");
fprintf(stdout, " 7: 16 bit PCM data (dithered with LIGHT noise shaping).\n");
fprintf(stdout, " 8: 16 bit PCM data (dithered with MEDIUM noise shaping).\n");
fprintf(stdout, " 9: 16 bit PCM data (dithered with HEAVY noise shaping).\n");
fprintf(stdout, " -s X Force the samplerate to X (for RAW files).\n");
fprintf(stdout, " -l X Set object type. Supported object types:\n");
fprintf(stdout, " 1: Main object type.\n");
fprintf(stdout, " 2: LC (Low Complexity) object type.\n");
fprintf(stdout, " 3: LTP (Long Term Prediction) object type.\n");
fprintf(stdout, " 23: LD (Low Delay) object type.\n");
fprintf(stdout, " -d Down matrix 5.1 to 2 channels\n");
fprintf(stdout, " -w Write output to stdio instead of a file.\n");
fprintf(stdout, " -g Disable gapless decoding.\n");
fprintf(stdout, "Example:\n");
fprintf(stdout, " %s infile.aac\n", progName);
fprintf(stdout, " %s infile.mp4\n", progName);
fprintf(stdout, " %s -o outfile.wav infile.aac\n", progName);
fprintf(stdout, " %s -w infile.aac > outfile.wav\n", progName);
fprintf(stdout, " %s -a outfile.aac infile.aac\n", progName);
return;
}
int decodeAACfile(char *aacfile, char *sndfile, char *adts_fn, int to_stdout,
int def_srate, int object_type, int outputFormat, int fileType,
int downMatrix, int infoOnly, int adts_out, int old_format)
{
int tagsize;
unsigned long samplerate;
unsigned char channels;
void *sample_buffer;
audio_file *aufile;
FILE *adtsFile;
unsigned char *adtsData;
int adtsDataSize;
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));
if (adts_out)
{
adtsFile = fopen(adts_fn, "wb");
if (adtsFile == NULL)
{
fprintf(stderr, "Error opening file: %s\n", adts_fn);
return 1;
}
}
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 != FAAD_MIN_STREAMSIZE*MAX_CHANNELS)
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;
config->downMatrix = downMatrix;
config->useOldADTSFormat = old_format;
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.f)/((float)bitrate) + 0.5f;
}
bitrate = (int)((float)bitrate/1000.0f + 0.5f);
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, "%s file info:\n", aacfile);
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 (b.buffer)
free(b.buffer);
return 0;
}
do
{
sample_buffer = faacDecDecode(hDecoder, &frameInfo,
b.buffer, b.bytes_into_buffer);
if (adts_out == 1)
{
int skip = (old_format) ? 8 : 7;
adtsData = MakeAdtsHeader(&adtsDataSize, &frameInfo, old_format);
/* write the adts header */
fwrite(adtsData, 1, adtsDataSize, adtsFile);
/* write the frame data */
if (frameInfo.header_type == ADTS)
fwrite(b.buffer + skip, 1, frameInfo.bytesconsumed - skip, adtsFile);
else
fwrite(b.buffer, 1, frameInfo.bytesconsumed, adtsFile);
}
/* update buffer indices */
advance_buffer(&b, frameInfo.bytesconsumed);
if (frameInfo.error > 0)
{
fprintf(stderr, "Error: %s\n",
faacDecGetErrorMessage(frameInfo.error));
}
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{
/* print some channel info */
print_channel_info(&frameInfo);
if (!adts_out)
{
/* open output file */
if (!to_stdout)
{
aufile = open_audio_file(sndfile, frameInfo.samplerate, frameInfo.channels,
outputFormat, fileType, aacChannelConfig2wavexChannelMask(&frameInfo));
} else {
aufile = open_audio_file("-", frameInfo.samplerate, frameInfo.channels,
outputFormat, fileType, aacChannelConfig2wavexChannelMask(&frameInfo));
}
if (aufile == NULL)
{
if (b.buffer)
free(b.buffer);
faacDecClose(hDecoder);
fclose(b.infile);
return 0;
}
} else {
fprintf(stderr, "Writing output MPEG-4 AAC ADTS file.\n\n");
}
first_time = 0;
}
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
}
if ((frameInfo.error == 0) && (frameInfo.samples > 0) && (!adts_out))
{
write_audio_file(aufile, sample_buffer, frameInfo.samples, 0);
}
/* 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);
if (adts_out == 1)
{
fclose(adtsFile);
}
fclose(b.infile);
if (!first_time && !adts_out)
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, char *adts_fn, int to_stdout,
int outputFormat, int fileType, int downMatrix, int noGapless,
int infoOnly, int adts_out)
{
int track;
unsigned long samplerate;
unsigned char channels;
void *sample_buffer;
MP4FileHandle infile;
MP4SampleId sampleId, numSamples;
audio_file *aufile;
FILE *adtsFile;
unsigned char *adtsData;
int adtsDataSize;
faacDecHandle hDecoder;
faacDecConfigurationPtr config;
faacDecFrameInfo frameInfo;
unsigned char *buffer;
int buffer_size;
char percents[200];
int percent, old_percent = -1;
int first_time = 1;
/* for gapless decoding */
unsigned int useAacLength = 1;
unsigned int framesize;
unsigned int initial = 1;
unsigned long timescale;
hDecoder = faacDecOpen();
/* Set configuration */
config = faacDecGetCurrentConfiguration(hDecoder);
config->outputFormat = outputFormat;
config->downMatrix = downMatrix;
faacDecSetConfiguration(hDecoder, config);
if (adts_out)
{
adtsFile = fopen(adts_fn, "wb");
if (adtsFile == NULL)
{
fprintf(stderr, "Error opening file: %s\n", adts_fn);
return 1;
}
}
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 */
fprintf(stderr, "%s file info:\n", mp4file);
{
char *file_info = MP4Info(infile, MP4_INVALID_TRACK_ID);
fprintf(stderr, "%s\n", file_info);
free(file_info);
}
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 (!noGapless)
{
mp4AudioSpecificConfig mp4ASC;
timescale = MP4GetTrackTimeScale(infile, track);
framesize = 1024;
useAacLength = 0;
if (buffer)
{
if (AudioSpecificConfig(buffer, buffer_size, &mp4ASC) >= 0)
{
if (mp4ASC.frameLengthFlag == 1) framesize = 960;
if (mp4ASC.sbr_present_flag == 1) framesize *= 2;
}
}
}
if (buffer) free(buffer);
numSamples = MP4GetTrackNumberOfSamples(infile, track);
for (sampleId = 1; sampleId <= numSamples; sampleId++)
{
int rc;
MP4Duration dur;
unsigned int sample_count;
unsigned int delay = 0;
/* get acces unit from MP4 file */
buffer = NULL;
buffer_size = 0;
rc = MP4ReadSample(infile, track, sampleId, &buffer, &buffer_size,
NULL, &dur, 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 (adts_out == 1)
{
adtsData = MakeAdtsHeader(&adtsDataSize, &frameInfo, 0);
/* write the adts header */
fwrite(adtsData, 1, adtsDataSize, adtsFile);
fwrite(buffer, 1, frameInfo.bytesconsumed, adtsFile);
}
if (buffer) free(buffer);
if (!noGapless)
{
if (useAacLength || (timescale != samplerate)) {
sample_count = frameInfo.samples;
} else {
sample_count = (unsigned int)(dur * frameInfo.channels);
if (!useAacLength && !initial && (sampleId < numSamples/2) && (sample_count != frameInfo.samples))
{
fprintf(stderr, "MP4 seems to have incorrect frame duration, using values from AAC data.\n");
useAacLength = 1;
}
}
if (initial && (sample_count < framesize*frameInfo.channels))
delay = frameInfo.samples - sample_count;
}
else
{
sample_count = frameInfo.samples;
}
/* open the sound file now that the number of channels are known */
if (first_time && !frameInfo.error)
{
/* print some channel info */
print_channel_info(&frameInfo);
if (!adts_out)
{
/* open output file */
if(!to_stdout)
{
aufile = open_audio_file(sndfile, frameInfo.samplerate, frameInfo.channels,
outputFormat, fileType, aacChannelConfig2wavexChannelMask(&frameInfo));
} else {
#ifdef _WIN32
setmode(fileno(stdout), O_BINARY);
#endif
aufile = open_audio_file("-", frameInfo.samplerate, frameInfo.channels,
outputFormat, fileType, aacChannelConfig2wavexChannelMask(&frameInfo));
}
if (aufile == NULL)
{
faacDecClose(hDecoder);
MP4Close(infile);
return 0;
}
}
first_time = 0;
}
if (sample_count > 0) initial = 0;
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
}
if ((frameInfo.error == 0) && (sample_count > 0) && (!adts_out))
{
write_audio_file(aufile, sample_buffer, sample_count, delay);
}
if (frameInfo.error > 0)
{
fprintf(stderr, "Warning: %s\n",
faacDecGetErrorMessage(frameInfo.error));
}
}
faacDecClose(hDecoder);
if (adts_out == 1)
{
fclose(adtsFile);
}
MP4Close(infile);
if (!first_time && !adts_out)
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 downMatrix = 0;
int format = 1;
int outputFormat = FAAD_FMT_16BIT;
int outfile_set = 0;
int adts_out = 0;
int old_format = 0;
int showHelp = 0;
int mp4file = 0;
int noGapless = 0;
char *fnp;
char aacFileName[255];
char audioFileName[255];
char adtsFileName[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' },
{ "adtsout", 0, 0, 'a' },
{ "oldformat", 0, 0, 't' },
{ "format", 0, 0, 'f' },
{ "bits", 0, 0, 'b' },
{ "samplerate", 0, 0, 's' },
{ "objecttype", 0, 0, 'l' },
{ "downmix", 0, 0, 'd' },
{ "info", 0, 0, 'i' },
{ "stdio", 0, 0, 'w' },
{ "stdio", 0, 0, 'g' },
{ "help", 0, 0, 'h' }
};
c = getopt_long(argc, argv, "o:a:s:f:b:l:wgdhit",
long_options, &option_index);
if (c == -1)
break;
switch (c) {
case 'o':
if (optarg)
{
outfile_set = 1;
strcpy(audioFileName, optarg);
}
break;
case 'a':
if (optarg)
{
adts_out = 1;
strcpy(adtsFileName, 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 't':
old_format = 1;
break;
case 'd':
downMatrix = 1;
break;
case 'w':
writeToStdio = 1;
break;
case 'g':
noGapless = 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, adtsFileName, writeToStdio,
outputFormat, format, downMatrix, noGapless, infoOnly, adts_out);
} else {
result = decodeAACfile(aacFileName, audioFileName, adtsFileName, writeToStdio,
def_srate, object_type, outputFormat, format, downMatrix, infoOnly, adts_out,
old_format);
}
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;
}