include/SDL_audio.h
author Ozkan Sezer <sezeroz@gmail.com>
Sat, 24 Mar 2018 22:27:31 +0300
branchSDL-1.2
changeset 11965 afbe1aac2dd1
parent 6137 4720145f848b
permissions -rw-r--r--
SDL_endian.h: add Watcom inline asm for SDL_Swap16 and SDL_Swap32
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/*
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    SDL - Simple DirectMedia Layer
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    Copyright (C) 1997-2012 Sam Lantinga
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    This library is free software; you can redistribute it and/or
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    modify it under the terms of the GNU Lesser General Public
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    License as published by the Free Software Foundation; either
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    version 2.1 of the License, or (at your option) any later version.
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    This library is distributed in the hope that it will be useful,
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    but WITHOUT ANY WARRANTY; without even the implied warranty of
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    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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    Lesser General Public License for more details.
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    You should have received a copy of the GNU Lesser General Public
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    License along with this library; if not, write to the Free Software
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    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
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    Sam Lantinga
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    slouken@libsdl.org
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*/
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/**
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 *  @file SDL_audio.h
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 *  Access to the raw audio mixing buffer for the SDL library
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 */
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#ifndef _SDL_audio_h
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#define _SDL_audio_h
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#include "SDL_stdinc.h"
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#include "SDL_error.h"
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#include "SDL_endian.h"
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#include "SDL_mutex.h"
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#include "SDL_thread.h"
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#include "SDL_rwops.h"
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#include "begin_code.h"
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/* Set up for C function definitions, even when using C++ */
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#ifdef __cplusplus
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extern "C" {
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#endif
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/**
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 * When filling in the desired audio spec structure,
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 * - 'desired->freq' should be the desired audio frequency in samples-per-second.
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 * - 'desired->format' should be the desired audio format.
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 * - 'desired->samples' is the desired size of the audio buffer, in samples.
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 *     This number should be a power of two, and may be adjusted by the audio
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 *     driver to a value more suitable for the hardware.  Good values seem to
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 *     range between 512 and 8096 inclusive, depending on the application and
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 *     CPU speed.  Smaller values yield faster response time, but can lead
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 *     to underflow if the application is doing heavy processing and cannot
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 *     fill the audio buffer in time.  A stereo sample consists of both right
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 *     and left channels in LR ordering.
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 *     Note that the number of samples is directly related to time by the
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 *     following formula:  ms = (samples*1000)/freq
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 * - 'desired->size' is the size in bytes of the audio buffer, and is
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 *     calculated by SDL_OpenAudio().
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 * - 'desired->silence' is the value used to set the buffer to silence,
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 *     and is calculated by SDL_OpenAudio().
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 * - 'desired->callback' should be set to a function that will be called
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 *     when the audio device is ready for more data.  It is passed a pointer
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 *     to the audio buffer, and the length in bytes of the audio buffer.
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 *     This function usually runs in a separate thread, and so you should
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 *     protect data structures that it accesses by calling SDL_LockAudio()
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 *     and SDL_UnlockAudio() in your code.
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 * - 'desired->userdata' is passed as the first parameter to your callback
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 *     function.
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 *
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 * @note The calculated values in this structure are calculated by SDL_OpenAudio()
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 *
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 */
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typedef struct SDL_AudioSpec {
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	int freq;		/**< DSP frequency -- samples per second */
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	Uint16 format;		/**< Audio data format */
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	Uint8  channels;	/**< Number of channels: 1 mono, 2 stereo */
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	Uint8  silence;		/**< Audio buffer silence value (calculated) */
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	Uint16 samples;		/**< Audio buffer size in samples (power of 2) */
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	Uint16 padding;		/**< Necessary for some compile environments */
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	Uint32 size;		/**< Audio buffer size in bytes (calculated) */
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	/**
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	 *  This function is called when the audio device needs more data.
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	 *
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	 *  @param[out] stream	A pointer to the audio data buffer
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	 *  @param[in]  len	The length of the audio buffer in bytes.
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	 *
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	 *  Once the callback returns, the buffer will no longer be valid.
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	 *  Stereo samples are stored in a LRLRLR ordering.
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	 */
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	void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
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	void  *userdata;
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} SDL_AudioSpec;
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/**
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 *  @name Audio format flags
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 *  defaults to LSB byte order
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 */
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/*@{*/
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#define AUDIO_U8	0x0008	/**< Unsigned 8-bit samples */
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#define AUDIO_S8	0x8008	/**< Signed 8-bit samples */
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#define AUDIO_U16LSB	0x0010	/**< Unsigned 16-bit samples */
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#define AUDIO_S16LSB	0x8010	/**< Signed 16-bit samples */
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#define AUDIO_U16MSB	0x1010	/**< As above, but big-endian byte order */
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#define AUDIO_S16MSB	0x9010	/**< As above, but big-endian byte order */
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#define AUDIO_U16	AUDIO_U16LSB
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#define AUDIO_S16	AUDIO_S16LSB
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/**
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 *  @name Native audio byte ordering
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 */
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/*@{*/
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#if SDL_BYTEORDER == SDL_LIL_ENDIAN
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#define AUDIO_U16SYS	AUDIO_U16LSB
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#define AUDIO_S16SYS	AUDIO_S16LSB
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#else
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#define AUDIO_U16SYS	AUDIO_U16MSB
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#define AUDIO_S16SYS	AUDIO_S16MSB
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#endif
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/*@}*/
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/*@}*/
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/** A structure to hold a set of audio conversion filters and buffers */
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typedef struct SDL_AudioCVT {
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	int needed;			/**< Set to 1 if conversion possible */
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	Uint16 src_format;		/**< Source audio format */
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	Uint16 dst_format;		/**< Target audio format */
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	double rate_incr;		/**< Rate conversion increment */
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	Uint8 *buf;			/**< Buffer to hold entire audio data */
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	int    len;			/**< Length of original audio buffer */
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	int    len_cvt;			/**< Length of converted audio buffer */
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	int    len_mult;		/**< buffer must be len*len_mult big */
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	double len_ratio; 	/**< Given len, final size is len*len_ratio */
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	void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
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	int filter_index;		/**< Current audio conversion function */
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} SDL_AudioCVT;
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/* Function prototypes */
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/**
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 * @name Audio Init and Quit
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 * These functions are used internally, and should not be used unless you
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 * have a specific need to specify the audio driver you want to use.
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 * You should normally use SDL_Init() or SDL_InitSubSystem().
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 */
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/*@{*/
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extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
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extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
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/*@}*/
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/**
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 * This function fills the given character buffer with the name of the
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 * current audio driver, and returns a pointer to it if the audio driver has
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 * been initialized.  It returns NULL if no driver has been initialized.
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 */
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extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);
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/**
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 * This function opens the audio device with the desired parameters, and
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 * returns 0 if successful, placing the actual hardware parameters in the
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 * structure pointed to by 'obtained'.  If 'obtained' is NULL, the audio
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 * data passed to the callback function will be guaranteed to be in the
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 * requested format, and will be automatically converted to the hardware
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 * audio format if necessary.  This function returns -1 if it failed 
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 * to open the audio device, or couldn't set up the audio thread.
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 *
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 * The audio device starts out playing silence when it's opened, and should
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 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready
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 * for your audio callback function to be called.  Since the audio driver
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 * may modify the requested size of the audio buffer, you should allocate
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 * any local mixing buffers after you open the audio device.
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 *
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 * @sa SDL_AudioSpec
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 */
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extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);
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typedef enum {
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	SDL_AUDIO_STOPPED = 0,
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	SDL_AUDIO_PLAYING,
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	SDL_AUDIO_PAUSED
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} SDL_audiostatus;
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/** Get the current audio state */
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extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);
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/**
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 * This function pauses and unpauses the audio callback processing.
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 * It should be called with a parameter of 0 after opening the audio
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 * device to start playing sound.  This is so you can safely initialize
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 * data for your callback function after opening the audio device.
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 * Silence will be written to the audio device during the pause.
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 */
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extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
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/**
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 * This function loads a WAVE from the data source, automatically freeing
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 * that source if 'freesrc' is non-zero.  For example, to load a WAVE file,
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 * you could do:
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 *	@code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode
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 *
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 * If this function succeeds, it returns the given SDL_AudioSpec,
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 * filled with the audio data format of the wave data, and sets
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 * 'audio_buf' to a malloc()'d buffer containing the audio data,
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 * and sets 'audio_len' to the length of that audio buffer, in bytes.
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 * You need to free the audio buffer with SDL_FreeWAV() when you are 
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 * done with it.
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 *
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 * This function returns NULL and sets the SDL error message if the 
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 * wave file cannot be opened, uses an unknown data format, or is 
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 * corrupt.  Currently raw and MS-ADPCM WAVE files are supported.
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 */
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extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
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/** Compatibility convenience function -- loads a WAV from a file */
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#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
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	SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
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/**
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 * This function frees data previously allocated with SDL_LoadWAV_RW()
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 */
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extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);
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/**
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 * This function takes a source format and rate and a destination format
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 * and rate, and initializes the 'cvt' structure with information needed
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 * by SDL_ConvertAudio() to convert a buffer of audio data from one format
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 * to the other.
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 *
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 * @return This function returns 0, or -1 if there was an error.
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 */
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extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
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		Uint16 src_format, Uint8 src_channels, int src_rate,
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		Uint16 dst_format, Uint8 dst_channels, int dst_rate);
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/**
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 * Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),
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 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of
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 * audio data in the source format, this function will convert it in-place
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 * to the desired format.
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 * The data conversion may expand the size of the audio data, so the buffer
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 * cvt->buf should be allocated after the cvt structure is initialized by
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 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.
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 */
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extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);
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#define SDL_MIX_MAXVOLUME 128
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/**
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 * This takes two audio buffers of the playing audio format and mixes
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 * them, performing addition, volume adjustment, and overflow clipping.
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 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
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 * for full audio volume.  Note this does not change hardware volume.
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 * This is provided for convenience -- you can mix your own audio data.
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 */
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extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);
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/**
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 * @name Audio Locks
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 * The lock manipulated by these functions protects the callback function.
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 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the
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 * callback function is not running.  Do not call these from the callback
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 * function or you will cause deadlock.
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 */
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/*@{*/
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extern DECLSPEC void SDLCALL SDL_LockAudio(void);
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extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
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/*@}*/
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/**
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 * This function shuts down audio processing and closes the audio device.
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 */
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extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
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/* Ends C function definitions when using C++ */
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#ifdef __cplusplus
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}
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#endif
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#include "close_code.h"
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#endif /* _SDL_audio_h */