/*

    SDL - Simple DirectMedia Layer

    Copyright (C) 1997-2009 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

/* *INDENT-OFF* */

extern "C" {

/* *INDENT-ON* */

#endif

/**

 *  \brief Audio format flags.

 *  

 *  These are what the 16 bits in SDL_AudioFormat currently mean...

 *  (Unspecified bits are always zero).

 *  

 *  \verbatim

    ++-----------------------sample is signed if set

    ||

    ||       ++-----------sample is bigendian if set

    ||       ||

    ||       ||          ++---sample is float if set

    ||       ||          ||

    ||       ||          || +---sample bit size---+

    ||       ||          || |                     |

    15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00

    \endverbatim

 *  

 *  There are macros in SDL 1.3 and later to query these bits.

 */

typedef Uint16 SDL_AudioFormat;

/**

 *  \name Audio flags

 */

/*@{*/

#define SDL_AUDIO_MASK_BITSIZE       (0xFF)

#define SDL_AUDIO_MASK_DATATYPE      (1<<8)

#define SDL_AUDIO_MASK_ENDIAN        (1<<12)

#define SDL_AUDIO_MASK_SIGNED        (1<<15)

#define SDL_AUDIO_BITSIZE(x)         (x & SDL_AUDIO_MASK_BITSIZE)

#define SDL_AUDIO_ISFLOAT(x)         (x & SDL_AUDIO_MASK_DATATYPE)

#define SDL_AUDIO_ISBIGENDIAN(x)     (x & SDL_AUDIO_MASK_ENDIAN)

#define SDL_AUDIO_ISSIGNED(x)        (x & SDL_AUDIO_MASK_SIGNED)

#define SDL_AUDIO_ISINT(x)           (!SDL_AUDIO_ISFLOAT(x))

#define SDL_AUDIO_ISLITTLEENDIAN(x)  (!SDL_AUDIO_ISBIGENDIAN(x))

#define SDL_AUDIO_ISUNSIGNED(x)      (!SDL_AUDIO_ISSIGNED(x))

/** 

 *  \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 int32 support

 *  

 *  New to SDL 1.3.

 */

/*@{*/

#define AUDIO_S32LSB	0x8020  /**< 32-bit integer samples */

#define AUDIO_S32MSB	0x9020  /**< As above, but big-endian byte order */

#define AUDIO_S32	AUDIO_S32LSB

/*@}*/

/**

 *  \name float32 support

 *  

 *  New to SDL 1.3.

 */

/*@{*/

#define AUDIO_F32LSB	0x8120  /**< 32-bit floating point samples */

#define AUDIO_F32MSB	0x9120  /**< As above, but big-endian byte order */

#define AUDIO_F32	AUDIO_F32LSB

/*@}*/

/**

 *  \name Native audio byte ordering

 */

/*@{*/

#if SDL_BYTEORDER == SDL_LIL_ENDIAN

#define AUDIO_U16SYS	AUDIO_U16LSB

#define AUDIO_S16SYS	AUDIO_S16LSB

#define AUDIO_S32SYS	AUDIO_S32LSB

#define AUDIO_F32SYS	AUDIO_F32LSB

#else

#define AUDIO_U16SYS	AUDIO_U16MSB

#define AUDIO_S16SYS	AUDIO_S16MSB

#define AUDIO_S32SYS	AUDIO_S32MSB

#define AUDIO_F32SYS	AUDIO_F32MSB

#endif

/*@}*/

/** 

 *  \name Allow change flags

 *  

 *  Which audio format changes are allowed when opening a device.

 */

/*@{*/

#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE    0x00000001

#define SDL_AUDIO_ALLOW_FORMAT_CHANGE       0x00000002

#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE     0x00000004

#define SDL_AUDIO_ALLOW_ANY_CHANGE          (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE)

/*@}*/

/*@}*//*Audio flags*/

/**

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

 *

 *  \param userdata An application-specific parameter saved in

 *                  the SDL_AudioSpec structure

 *  \param stream A pointer to the audio data buffer.

 *  \param len    The length of that buffer in bytes.

 *

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

 *  Stereo samples are stored in a LRLRLR ordering.

 */

typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,

                                            int len);

/**

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

 */

typedef struct SDL_AudioSpec

{

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

    SDL_AudioFormat 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) */

    SDL_AudioCallback callback;

    void *userdata;

} SDL_AudioSpec;

struct SDL_AudioCVT;

typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,

                                          SDL_AudioFormat format);

/**

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

 */

typedef struct SDL_AudioCVT

{

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

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

    SDL_AudioFormat 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 */

    SDL_AudioFilter filters[10];        /**< Filter list */

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

} SDL_AudioCVT;

/* Function prototypes */

/**

 *  \name Driver discovery functions

 *  

 *  These functions return the list of built in audio drivers, in the

 *  order that they are normally initialized by default.

 */

/*@{*/

extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);

extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);

/*@}*/

/**

 *  \name Initialization and cleanup

 *  

 *  \internal 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 returns the name of the current audio driver, or NULL

 *  if no driver has been initialized.

 */

extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);

/**

 *  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 \c obtained.  If \c 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.

 *  

 *  When filling in the desired audio spec structure,

 *    - \c desired->freq should be the desired audio frequency in samples-per-

 *      second.

 *    - \c desired->format should be the desired audio format.

 *    - \c 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:  \code ms = (samples*1000)/freq \endcode

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

 *      calculated by SDL_OpenAudio().

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

 *      and is calculated by SDL_OpenAudio().

 *    - \c 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.

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

 *      function.

 *  

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

 *  be enabled for playing by calling \c 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.

 */

extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,

                                          SDL_AudioSpec * obtained);

/**

 *  SDL Audio Device IDs.

 *  

 *  A successful call to SDL_OpenAudio() is always device id 1, and legacy

 *  SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls

 *  always returns devices >= 2 on success. The legacy calls are good both

 *  for backwards compatibility and when you don't care about multiple,

 *  specific, or capture devices.

 */

typedef Uint32 SDL_AudioDeviceID;

/**

 *  Get the number of available devices exposed by the current driver.

 *  Only valid after a successfully initializing the audio subsystem.

 *  Returns -1 if an explicit list of devices can't be determined; this is

 *  not an error. For example, if SDL is set up to talk to a remote audio

 *  server, it can't list every one available on the Internet, but it will

 *  still allow a specific host to be specified to SDL_OpenAudioDevice().

 *  

 *  In many common cases, when this function returns a value <= 0, it can still

 *  successfully open the default device (NULL for first argument of

 *  SDL_OpenAudioDevice()).

 */

extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);

/**

 *  Get the human-readable name of a specific audio device.

 *  Must be a value between 0 and (number of audio devices-1).

 *  Only valid after a successfully initializing the audio subsystem.

 *  The values returned by this function reflect the latest call to

 *  SDL_GetNumAudioDevices(); recall that function to redetect available

 *  hardware.

 *  

 *  The string returned by this function is UTF-8 encoded, read-only, and

 *  managed internally. You are not to free it. If you need to keep the

 *  string for any length of time, you should make your own copy of it, as it

 *  will be invalid next time any of several other SDL functions is called.

 */

extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,

                                                           int iscapture);

/**

 *  Open a specific audio device. Passing in a device name of NULL requests

 *  the most reasonable default (and is equivalent to calling SDL_OpenAudio()).

 *  

 *  The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but

 *  some drivers allow arbitrary and driver-specific strings, such as a

 *  hostname/IP address for a remote audio server, or a filename in the

 *  diskaudio driver.

 *  

 *  \return 0 on error, a valid device ID that is >= 2 on success.

 *  

 *  SDL_OpenAudio(), unlike this function, always acts on device ID 1.

 */

extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char

                                                              *device,

                                                              int iscapture,

                                                              const

                                                              SDL_AudioSpec *

                                                              desired,

                                                              SDL_AudioSpec *

                                                              obtained,

                                                              int

                                                              allowed_changes);

/**

 *  \name Audio state

 *  

 *  Get the current audio state.

 */

/*@{*/

typedef enum

{

    SDL_AUDIO_STOPPED = 0,

    SDL_AUDIO_PLAYING,

    SDL_AUDIO_PAUSED

} SDL_AudioStatus;

extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);

extern DECLSPEC SDL_AudioStatus SDLCALL

SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);

/*@}*//*Audio State*/

/**

 *  \name Pause audio functions

 *  

 *  These functions pause and unpause the audio callback processing.

 *  They 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);

extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,

                                                  int pause_on);

/*@}*//*Pause audio functions*/

/**

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

 *  that source if \c 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

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

 *  and sets \c *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);

/** 

 *  Loads a WAV from a file.

 *  Compatibility convenience function.

 */

#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 \c cvt structure with information needed

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

 *  to the other.

 *  

 *  \return -1 if the format conversion is not supported, 0 if there's

 *  no conversion needed, or 1 if the audio filter is set up.

 */

extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,

                                              SDL_AudioFormat src_format,

                                              Uint8 src_channels,

                                              int src_rate,

                                              SDL_AudioFormat dst_format,

                                              Uint8 dst_channels,

                                              int dst_rate);

/**

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

 *  created an audio buffer \c cvt->buf, and filled it with \c 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

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

 *  SDL_BuildAudioCVT(), and should be \c 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);

/**

 *  This works like SDL_MixAudio(), but you specify the audio format instead of

 *  using the format of audio device 1. Thus it can be used when no audio

 *  device is open at all.

 */

extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,

                                                const Uint8 * src,

                                                SDL_AudioFormat format,

                                                Uint32 len, int volume);

/**

 *  \name Audio lock functions

 *  

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

 *  During a SDL_LockAudio()/SDL_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_LockAudioDevice(SDL_AudioDeviceID dev);

extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);

extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);

/*@}*//*Audio lock functions*/

/**

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