ref: c0ae72652fc9619e8b1e8f365ab977614179779a
dir: /aacDECdrop/audio.c/
/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003 M. Bakker (mbakker(at)nero.com), Ahead Software AG
**
** 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.
**
** Commercial non-GPL licensing of this software is also possible.
** For more info contact Ahead Software through [email protected].
**
** $Id: audio.c,v 1.10 2003/07/29 08:20:11 menno Exp $
**/
#ifdef _WIN32
#include <io.h>
#endif
#include <stdlib.h>
#include <stdio.h>
#include <fcntl.h>
#include <math.h>
#include <faad.h>
#include "audio.h"
audio_file *open_audio_file(char *infile, int samplerate, int channels,
int outputFormat, int fileType)
{
audio_file *aufile = malloc(sizeof(audio_file));
aufile->outputFormat = outputFormat;
aufile->samplerate = samplerate;
aufile->channels = channels;
aufile->total_samples = 0;
aufile->fileType = fileType;
switch (outputFormat)
{
case FAAD_FMT_16BIT:
case FAAD_FMT_16BIT_DITHER:
case FAAD_FMT_16BIT_L_SHAPE:
case FAAD_FMT_16BIT_M_SHAPE:
case FAAD_FMT_16BIT_H_SHAPE:
aufile->bits_per_sample = 16;
break;
case FAAD_FMT_24BIT:
aufile->bits_per_sample = 24;
break;
case FAAD_FMT_32BIT:
case FAAD_FMT_FLOAT:
aufile->bits_per_sample = 32;
break;
default:
if (aufile) free(aufile);
return NULL;
}
#ifdef _WIN32
if(infile[0] == '-')
{
setmode(fileno(stdout), O_BINARY);
}
#endif
aufile->sndfile = fopen(infile, "wb");
if (aufile->sndfile == NULL)
{
if (aufile) free(aufile);
return NULL;
}
if (aufile->fileType == OUTPUT_WAV)
{
write_wav_header(aufile);
}
return aufile;
}
int write_audio_file(audio_file *aufile, void *sample_buffer, int samples)
{
switch (aufile->outputFormat)
{
case FAAD_FMT_16BIT:
case FAAD_FMT_16BIT_DITHER:
case FAAD_FMT_16BIT_L_SHAPE:
case FAAD_FMT_16BIT_M_SHAPE:
case FAAD_FMT_16BIT_H_SHAPE:
return write_audio_16bit(aufile, sample_buffer, samples);
case FAAD_FMT_24BIT:
return write_audio_24bit(aufile, sample_buffer, samples);
case FAAD_FMT_32BIT:
return write_audio_32bit(aufile, sample_buffer, samples);
case FAAD_FMT_FLOAT:
return write_audio_float(aufile, sample_buffer, samples);
default:
return 0;
}
return 0;
}
void close_audio_file(audio_file *aufile)
{
if (aufile->fileType == OUTPUT_WAV)
{
fseek(aufile->sndfile, 0, SEEK_SET);
write_wav_header(aufile);
}
fclose(aufile->sndfile);
if (aufile) free(aufile);
}
static int write_wav_header(audio_file *aufile)
{
unsigned char header[44];
unsigned char* p = header;
unsigned int bytes = (aufile->bits_per_sample + 7) / 8;
float data_size = (float)bytes * aufile->total_samples;
unsigned long word32;
*p++ = 'R'; *p++ = 'I'; *p++ = 'F'; *p++ = 'F';
word32 = (data_size + (44 - 8) < (float)MAXWAVESIZE) ?
(unsigned long)data_size + (44 - 8) : (unsigned long)MAXWAVESIZE;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
*p++ = 'W'; *p++ = 'A'; *p++ = 'V'; *p++ = 'E';
*p++ = 'f'; *p++ = 'm'; *p++ = 't'; *p++ = ' ';
*p++ = 0x10; *p++ = 0x00; *p++ = 0x00; *p++ = 0x00;
if (aufile->outputFormat == FAAD_FMT_FLOAT)
{
*p++ = 0x03; *p++ = 0x00;
} else {
*p++ = 0x01; *p++ = 0x00;
}
*p++ = (unsigned char)(aufile->channels >> 0);
*p++ = (unsigned char)(aufile->channels >> 8);
word32 = (unsigned long)(aufile->samplerate + 0.5);
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
word32 = aufile->samplerate * bytes * aufile->channels;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
word32 = bytes * aufile->channels;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(aufile->bits_per_sample >> 0);
*p++ = (unsigned char)(aufile->bits_per_sample >> 8);
*p++ = 'd'; *p++ = 'a'; *p++ = 't'; *p++ = 'a';
word32 = data_size < MAXWAVESIZE ?
(unsigned long)data_size : (unsigned long)MAXWAVESIZE;
*p++ = (unsigned char)(word32 >> 0);
*p++ = (unsigned char)(word32 >> 8);
*p++ = (unsigned char)(word32 >> 16);
*p++ = (unsigned char)(word32 >> 24);
return fwrite(header, sizeof(header), 1, aufile->sndfile);
}
static int write_audio_16bit(audio_file *aufile, void *sample_buffer,
unsigned int samples)
{
int ret;
unsigned int i;
short *sample_buffer16 = (short*)sample_buffer;
char *data = malloc(samples*aufile->bits_per_sample*sizeof(char)/8);
aufile->total_samples += samples;
for (i = 0; i < samples; i++)
{
data[i*2] = (char)(sample_buffer16[i] & 0xFF);
data[i*2+1] = (char)((sample_buffer16[i] >> 8) & 0xFF);
}
ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile);
if (data) free(data);
return ret;
}
static int write_audio_24bit(audio_file *aufile, void *sample_buffer,
unsigned int samples)
{
int ret;
unsigned int i;
long *sample_buffer24 = (long*)sample_buffer;
char *data = malloc(samples*aufile->bits_per_sample*sizeof(char)/8);
aufile->total_samples += samples;
for (i = 0; i < samples; i++)
{
data[i*3] = (char)(sample_buffer24[i] & 0xFF);
data[i*3+1] = (char)((sample_buffer24[i] >> 8) & 0xFF);
data[i*3+2] = (char)((sample_buffer24[i] >> 16) & 0xFF);
}
ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile);
if (data) free(data);
return ret;
}
static int write_audio_32bit(audio_file *aufile, void *sample_buffer,
unsigned int samples)
{
int ret;
unsigned int i;
long *sample_buffer32 = (long*)sample_buffer;
char *data = malloc(samples*aufile->bits_per_sample*sizeof(char)/8);
aufile->total_samples += samples;
for (i = 0; i < samples; i++)
{
data[i*4] = (char)(sample_buffer32[i] & 0xFF);
data[i*4+1] = (char)((sample_buffer32[i] >> 8) & 0xFF);
data[i*4+2] = (char)((sample_buffer32[i] >> 16) & 0xFF);
data[i*4+3] = (char)((sample_buffer32[i] >> 24) & 0xFF);
}
ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile);
if (data) free(data);
return ret;
}
static int write_audio_float(audio_file *aufile, void *sample_buffer,
unsigned int samples)
{
int ret;
unsigned int i;
float *sample_buffer_f = (float*)sample_buffer;
unsigned char *data = malloc(samples*aufile->bits_per_sample*sizeof(char)/8);
aufile->total_samples += samples;
for (i = 0; i < samples; i++)
{
int exponent, mantissa, negative = 0 ;
float in = sample_buffer_f[i];
data[i*4] = 0; data[i*4+1] = 0; data[i*4+2] = 0; data[i*4+3] = 0;
if (in == 0.0)
continue;
if (in < 0.0)
{
in *= -1.0;
negative = 1;
}
in = (float)frexp(in, &exponent);
exponent += 126;
in *= (float)0x1000000;
mantissa = (((int)in) & 0x7FFFFF);
if (negative)
data[i*4+3] |= 0x80;
if (exponent & 0x01)
data[i*4+2] |= 0x80;
data[i*4] = mantissa & 0xFF;
data[i*4+1] = (mantissa >> 8) & 0xFF;
data[i*4+2] |= (mantissa >> 16) & 0x7F;
data[i*4+3] |= (exponent >> 1) & 0x7F;
}
ret = fwrite(data, samples, aufile->bits_per_sample/8, aufile->sndfile);
if (data) free(data);
return ret;
}