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/**
* Parts of this file are copied from OpenAL source code.
* Copyright (C) Loki Games, licensed under the LGPL.
*
* This code has since been (more or less) modified to be a part of:
* OpenAL++ - an object oriented toolkit for spatial sound
* Copyright (C) 2002 VRlab, Ume University
*
* OpenAL++ was created using the libraries:
* OpenAL (http://www.openal.org),
* PortAudio (http://www.portaudio.com/), and
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
*/
#ifndef AUDIOCONVERT_H_INCLUDED
#define AUDIOCONVERT_H_INCLUDED
#include "openalpp/export.h"
#include <AL/al.h>
#include <stdlib.h>
#include <string.h>
#include "openalpp/error.h"
namespace openalpp {
/**
* Class for converting audio.
*/
class OPENALPP_API AudioConvert {
unsigned short channels_,bits_;
unsigned int frequency_;
ALenum format_;
public:
/**
* Constructor.
* @param format is the (OpenAL) format that data will be converted to.
* @param frequency is the frequency the data will be converted to.
*/
AudioConvert(ALenum format,unsigned int frequency);
/**
* Apply the conversion to data.
* @param data is the data to convert.
* @param format is the (OpenAL) format of the data.
* @param frequency is the frequency of the data.
* @param size is the size of the data. It will be updated to the new size.
*/
void *apply(void *data,ALenum format,unsigned int frequency,unsigned int &size);
};
typedef struct _acAudioCVT {
int needed; /* Set to 1 if conversion possible */
ALushort src_format; /* Source audio format */
ALushort dst_format; /* Target audio format */
double rate_incr; /* Rate conversion increment */
void *buf; /* Buffer to hold entire audio data */
int len; /* Length of original audio buffer */
int len_cvt; /* Length of converted audio buffer */
int len_mult; /* buffer must be len*len_mult big */
double len_ratio; /* Given len, final size is len*len_ratio */
void (*filters[10])(struct _acAudioCVT *cvt, ALushort format);
int filter_index; /* Current audio conversion function */
} acAudioCVT;
/* Audio format flags (defaults to LSB byte order) */
#define AUDIO_U8 0x0008 /* Unsigned 8-bit samples */
#define AUDIO_S8 0x8008 /* Signed 8-bit samples */
#define AUDIO_U16LSB 0x0010 /* Unsigned 16-bit samples */
#define AUDIO_S16LSB 0x8010 /* Signed 16-bit samples */
#define AUDIO_U16MSB 0x1010 /* As above, but big-endian byte order */
#define AUDIO_S16MSB 0x9010 /* As above, but big-endian byte order */
/* Native audio byte ordering */
#ifndef WORDS_BIGENDIAN
#define AUDIO_U16 AUDIO_U16LSB
#define AUDIO_S16 AUDIO_S16LSB
#define swap16le(x) (x)
#define swap32le(x) (x)
#define swap16be(x) swap16(x)
#define swap32be(x) swap32(x)
#else
#define AUDIO_U16 AUDIO_U16MSB
#define AUDIO_S16 AUDIO_S16MSB
#define swap16le(x) swap16(x)
#define swap32le(x) swap32(x)
#define swap16be(x) (x)
#define swap32be(x) (x)
#endif
#define _al_ALCHANNELS(fmt) ((fmt==AL_FORMAT_MONO16||fmt==AL_FORMAT_MONO8)?1:2)
#define DATA 0x61746164 /* "data" */
#define FACT 0x74636166 /* "fact" */
#define LIST 0x5453494c /* "LIST" */
#define RIFF 0x46464952
#define WAVE 0x45564157
#define FMT 0x20746D66
#define AL_FORMAT_IMA_ADPCM_MONO16_EXT 0x10000
#define AL_FORMAT_IMA_ADPCM_STEREO16_EXT 0x10001
#define AL_FORMAT_WAVE_EXT 0x10002
#define NELEMS(x) ((sizeof x) / (sizeof *x))
#define PCM_CODE 0x0001
#define MS_ADPCM_CODE 0x0002
#define IMA_ADPCM_CODE 0x0011
#define MS_ADPCM_max ((1<<(16-1))-1)
#define MS_ADPCM_min -(1<<(16-1))
typedef struct Chunk {
ALuint magic;
ALuint length;
void *data;
} Chunk;
struct MS_ADPCM_decodestate_FULL {
ALubyte hPredictor;
ALushort iDelta;
ALshort iSamp1;
ALshort iSamp2;
};
typedef struct WaveFMT {
ALushort encoding;
ALushort channels; /* 1 = mono, 2 = stereo */
ALuint frequency; /* One of 11025, 22050, or 44100 Hz */
ALuint byterate; /* Average bytes per second */
ALushort blockalign; /* Bytes per sample block */
ALushort bitspersample;
} alWaveFMT_LOKI;
typedef struct IMA_ADPCM_decodestate_s {
ALint valprev; /* Previous output value */
ALbyte index; /* Index into stepsize table */
} alIMAADPCM_decodestate_LOKI;
typedef struct IMA_ADPCM_decoder {
alWaveFMT_LOKI wavefmt;
ALushort wSamplesPerBlock;
alIMAADPCM_decodestate_LOKI state[2];
} alIMAADPCM_state_LOKI;
/*
void *AudioConvert(ALvoid *data,
ALenum f_format, ALuint f_size, ALuint f_freq,
ALenum t_format, ALuint t_freq, ALuint *retsize);
*/
}
#endif
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