/*

    SDL - Simple DirectMedia Layer

    Copyright (C) 1997-2012 Sam Lantinga

    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., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA

    Sam Lantinga

    slouken@libsdl.org

*/

/**

 *  @file SDL_audio.h

 *  Access to the raw audio mixing buffer for the SDL library

 */

#ifndef _SDL_audio_h

#define _SDL_audio_h

#include "SDL_stdinc.h"

#include "SDL_error.h"

#include "SDL_endian.h"

#include "SDL_mutex.h"

#include "SDL_thread.h"

#include "SDL_rwops.h"

#include "begin_code.h"

/* Set up for C function definitions, even when using C++ */

#ifdef __cplusplus

extern "C" {

#endif

/**

 * When filling in the desired audio spec structure,

 * - 'desired->freq' should be the desired audio frequency in samples-per-second.

 * - 'desired->format' should be the desired audio format.

 * - 'desired->samples' is the desired size of the audio buffer, in samples.

 *     This number should be a power of two, and may be adjusted by the audio

 *     driver to a value more suitable for the hardware.  Good values seem to

 *     range between 512 and 8096 inclusive, depending on the application and

 *     CPU speed.  Smaller values yield faster response time, but can lead

 *     to underflow if the application is doing heavy processing and cannot

 *     fill the audio buffer in time.  A stereo sample consists of both right

 *     and left channels in LR ordering.

 *     Note that the number of samples is directly related to time by the

 *     following formula:  ms = (samples*1000)/freq

 * - 'desired->size' is the size in bytes of the audio buffer, and is

 *     calculated by SDL_OpenAudio().

 * - 'desired->silence' is the value used to set the buffer to silence,

 *     and is calculated by SDL_OpenAudio().

 * - 'desired->callback' should be set to a function that will be called

 *     when the audio device is ready for more data.  It is passed a pointer

 *     to the audio buffer, and the length in bytes of the audio buffer.

 *     This function usually runs in a separate thread, and so you should

 *     protect data structures that it accesses by calling SDL_LockAudio()

 *     and SDL_UnlockAudio() in your code.

 * - 'desired->userdata' is passed as the first parameter to your callback

 *     function.

 *

 * @note The calculated values in this structure are calculated by SDL_OpenAudio()

 *

 */

typedef struct SDL_AudioSpec {

    int freq;       /**< DSP frequency -- samples per second */

    Uint16 format;      /**< Audio data format */

    Uint8  channels;    /**< Number of channels: 1 mono, 2 stereo */

    Uint8  silence;     /**< Audio buffer silence value (calculated) */

    Uint16 samples;     /**< Audio buffer size in samples (power of 2) */

    Uint16 padding;     /**< Necessary for some compile environments */

    Uint32 size;        /**< Audio buffer size in bytes (calculated) */

    /**

     *  This function is called when the audio device needs more data.

     *

     *  @param[out] stream  A pointer to the audio data buffer

     *  @param[in]  len The length of the audio buffer in bytes.

     *

     *  Once the callback returns, the buffer will no longer be valid.

     *  Stereo samples are stored in a LRLRLR ordering.

     */

    void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);

    void  *userdata;

} SDL_AudioSpec;

/**

 *  @name 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 */

#define AUDIO_U16   AUDIO_U16LSB

#define AUDIO_S16   AUDIO_S16LSB

/**

 *  @name Native audio byte ordering

 */

/*@{*/

#if SDL_BYTEORDER == SDL_LIL_ENDIAN

#define AUDIO_U16SYS    AUDIO_U16LSB

#define AUDIO_S16SYS    AUDIO_S16LSB

#else

#define AUDIO_U16SYS    AUDIO_U16MSB

#define AUDIO_S16SYS    AUDIO_S16MSB

#endif

/*@}*/

/*@}*/

/** A structure to hold a set of audio conversion filters and buffers */

typedef struct SDL_AudioCVT {

    int needed;         /**< Set to 1 if conversion possible */

    Uint16 src_format;      /**< Source audio format */

    Uint16 dst_format;      /**< Target audio format */

    double rate_incr;       /**< Rate conversion increment */

    Uint8 *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 (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);

    int filter_index;       /**< Current audio conversion function */

} SDL_AudioCVT;

/* Function prototypes */

/**

 * @name Audio Init and Quit

 * These functions are used internally, and should not be used unless you

 * have a specific need to specify the audio driver you want to use.

 * You should normally use SDL_Init() or SDL_InitSubSystem().

 */

/*@{*/

extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);

extern DECLSPEC void SDLCALL SDL_AudioQuit(void);

/*@}*/

/**

 * This function fills the given character buffer with the name of the

 * current audio driver, and returns a pointer to it if the audio driver has

 * been initialized.  It returns NULL if no driver has been initialized.

 */

extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);

/**

 * This function opens the audio device with the desired parameters, and

 * returns 0 if successful, placing the actual hardware parameters in the

 * structure pointed to by 'obtained'.  If 'obtained' is NULL, the audio

 * data passed to the callback function will be guaranteed to be in the

 * requested format, and will be automatically converted to the hardware

 * audio format if necessary.  This function returns -1 if it failed

 * to open the audio device, or couldn't set up the audio thread.

 *

 * The audio device starts out playing silence when it's opened, and should

 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready

 * for your audio callback function to be called.  Since the audio driver

 * may modify the requested size of the audio buffer, you should allocate

 * any local mixing buffers after you open the audio device.

 *

 * @sa SDL_AudioSpec

 */

extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);

typedef enum {

    SDL_AUDIO_STOPPED = 0,

    SDL_AUDIO_PLAYING,

    SDL_AUDIO_PAUSED

} SDL_audiostatus;

/** Get the current audio state */

extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);

/**

 * This function pauses and unpauses the audio callback processing.

 * It should be called with a parameter of 0 after opening the audio

 * device to start playing sound.  This is so you can safely initialize

 * data for your callback function after opening the audio device.

 * Silence will be written to the audio device during the pause.

 */

extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);

/**

 * This function loads a WAVE from the data source, automatically freeing

 * that source if 'freesrc' is non-zero.  For example, to load a WAVE file,

 * you could do:

 *  @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode

 *

 * If this function succeeds, it returns the given SDL_AudioSpec,

 * filled with the audio data format of the wave data, and sets

 * 'audio_buf' to a malloc()'d buffer containing the audio data,

 * and sets 'audio_len' to the length of that audio buffer, in bytes.

 * You need to free the audio buffer with SDL_FreeWAV() when you are

 * done with it.

 *

 * This function returns NULL and sets the SDL error message if the

 * wave file cannot be opened, uses an unknown data format, or is

 * corrupt.  Currently raw and MS-ADPCM WAVE files are supported.

 */

extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);

/** Compatibility convenience function -- loads a WAV from a file */

#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \

    SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)

/**

 * This function frees data previously allocated with SDL_LoadWAV_RW()

 */

extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);

/**

 * This function takes a source format and rate and a destination format

 * and rate, and initializes the 'cvt' structure with information needed

 * by SDL_ConvertAudio() to convert a buffer of audio data from one format

 * to the other.

 *

 * @return This function returns 0, or -1 if there was an error.

 */

extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,

        Uint16 src_format, Uint8 src_channels, int src_rate,

        Uint16 dst_format, Uint8 dst_channels, int dst_rate);

/**

 * Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),

 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of

 * audio data in the source format, this function will convert it in-place

 * to the desired format.

 * The data conversion may expand the size of the audio data, so the buffer

 * cvt->buf should be allocated after the cvt structure is initialized by

 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.

 */

extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);

#define SDL_MIX_MAXVOLUME 128

/**

 * This takes two audio buffers of the playing audio format and mixes

 * them, performing addition, volume adjustment, and overflow clipping.

 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME

 * for full audio volume.  Note this does not change hardware volume.

 * This is provided for convenience -- you can mix your own audio data.

 */

extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);

/**

 * @name Audio Locks

 * The lock manipulated by these functions protects the callback function.

 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the

 * callback function is not running.  Do not call these from the callback

 * function or you will cause deadlock.

 */

/*@{*/

extern DECLSPEC void SDLCALL SDL_LockAudio(void);

extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);

/*@}*/

/**

 * This function shuts down audio processing and closes the audio device.

 */

extern DECLSPEC void SDLCALL SDL_CloseAudio(void);

/* Ends C function definitions when using C++ */

#ifdef __cplusplus

}

#endif

#include "close_code.h"

#endif /* _SDL_audio_h */