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/*
** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
** Copyright (C) 2003-2005 M. Bakker, Nero 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.
**
** The "appropriate copyright message" mentioned in section 2c of the GPLv2
** must read: "Code from FAAD2 is copyright (c) Nero AG, www.nero.com"
**
** Commercial non-GPL licensing of this software is possible.
** For more info contact Nero AG through Mpeg4AAClicense@nero.com.
**
** $Id: audio.h,v 1.19 2007/11/01 12:33:29 menno Exp $
**/
#ifndef AUDIO_H_INCLUDED
#define AUDIO_H_INCLUDED
#ifdef __cplusplus
extern "C" {
#endif
#define MAXWAVESIZE 4294967040LU
#define OUTPUT_WAV 1
#define OUTPUT_RAW 2
typedef struct
{
int toStdio;
int outputFormat;
FILE *sndfile;
unsigned int fileType;
unsigned long samplerate;
unsigned int bits_per_sample;
unsigned int channels;
unsigned long total_samples;
long channelMask;
} audio_file;
audio_file *open_audio_file(char *infile, int samplerate, int channels,
int outputFormat, int fileType, long channelMask);
int write_audio_file(audio_file *aufile, void *sample_buffer, int samples, int offset);
void close_audio_file(audio_file *aufile);
static int write_wav_header(audio_file *aufile);
static int write_wav_extensible_header(audio_file *aufile, long channelMask);
static int write_audio_16bit(audio_file *aufile, void *sample_buffer,
unsigned int samples);
static int write_audio_24bit(audio_file *aufile, void *sample_buffer,
unsigned int samples);
static int write_audio_32bit(audio_file *aufile, void *sample_buffer,
unsigned int samples);
static int write_audio_float(audio_file *aufile, void *sample_buffer,
unsigned int samples);
#ifdef __cplusplus
}
#endif
#endif
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