include/SDL_audio.h
author Sam Lantinga <slouken@libsdl.org>
Fri, 04 Jan 2019 22:01:14 -0800
changeset 12503 806492103856
parent 12272 c68b05ca0761
child 12806 b06fa7da012b
permissions -rw-r--r--
Updated copyright for 2019
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/*
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  Simple DirectMedia Layer
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  Copyright (C) 1997-2019 Sam Lantinga <slouken@libsdl.org>
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  This software is provided 'as-is', without any express or implied
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  warranty.  In no event will the authors be held liable for any damages
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  arising from the use of this software.
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  Permission is granted to anyone to use this software for any purpose,
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  including commercial applications, and to alter it and redistribute it
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  freely, subject to the following restrictions:
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  1. The origin of this software must not be misrepresented; you must not
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     claim that you wrote the original software. If you use this software
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     in a product, an acknowledgment in the product documentation would be
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     appreciated but is not required.
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  2. Altered source versions must be plainly marked as such, and must not be
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     misrepresented as being the original software.
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  3. This notice may not be removed or altered from any source distribution.
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*/
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/**
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 *  \file SDL_audio.h
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 *
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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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 *  \brief Audio format flags.
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 *
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 *  These are what the 16 bits in SDL_AudioFormat currently mean...
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 *  (Unspecified bits are always zero).
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 *
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 *  \verbatim
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    ++-----------------------sample is signed if set
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    ||
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    ||       ++-----------sample is bigendian if set
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    ||       ||
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    ||       ||          ++---sample is float if set
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    ||       ||          ||
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    ||       ||          || +---sample bit size---+
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    ||       ||          || |                     |
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    15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
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    \endverbatim
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 *
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 *  There are macros in SDL 2.0 and later to query these bits.
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 */
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typedef Uint16 SDL_AudioFormat;
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/**
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 *  \name Audio flags
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 */
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/* @{ */
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#define SDL_AUDIO_MASK_BITSIZE       (0xFF)
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#define SDL_AUDIO_MASK_DATATYPE      (1<<8)
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#define SDL_AUDIO_MASK_ENDIAN        (1<<12)
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#define SDL_AUDIO_MASK_SIGNED        (1<<15)
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#define SDL_AUDIO_BITSIZE(x)         (x & SDL_AUDIO_MASK_BITSIZE)
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#define SDL_AUDIO_ISFLOAT(x)         (x & SDL_AUDIO_MASK_DATATYPE)
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#define SDL_AUDIO_ISBIGENDIAN(x)     (x & SDL_AUDIO_MASK_ENDIAN)
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#define SDL_AUDIO_ISSIGNED(x)        (x & SDL_AUDIO_MASK_SIGNED)
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#define SDL_AUDIO_ISINT(x)           (!SDL_AUDIO_ISFLOAT(x))
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#define SDL_AUDIO_ISLITTLEENDIAN(x)  (!SDL_AUDIO_ISBIGENDIAN(x))
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#define SDL_AUDIO_ISUNSIGNED(x)      (!SDL_AUDIO_ISSIGNED(x))
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/**
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 *  \name Audio format flags
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 *
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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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/**
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 *  \name int32 support
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 */
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/* @{ */
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#define AUDIO_S32LSB    0x8020  /**< 32-bit integer samples */
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#define AUDIO_S32MSB    0x9020  /**< As above, but big-endian byte order */
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#define AUDIO_S32       AUDIO_S32LSB
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/* @} */
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/**
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 *  \name float32 support
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 */
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/* @{ */
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#define AUDIO_F32LSB    0x8120  /**< 32-bit floating point samples */
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#define AUDIO_F32MSB    0x9120  /**< As above, but big-endian byte order */
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#define AUDIO_F32       AUDIO_F32LSB
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/* @} */
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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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#define AUDIO_S32SYS    AUDIO_S32LSB
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#define AUDIO_F32SYS    AUDIO_F32LSB
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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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#define AUDIO_S32SYS    AUDIO_S32MSB
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#define AUDIO_F32SYS    AUDIO_F32MSB
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#endif
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/* @} */
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/**
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 *  \name Allow change flags
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 *
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 *  Which audio format changes are allowed when opening a device.
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 */
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/* @{ */
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#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE    0x00000001
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#define SDL_AUDIO_ALLOW_FORMAT_CHANGE       0x00000002
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#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE     0x00000004
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#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE      0x00000008
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#define SDL_AUDIO_ALLOW_ANY_CHANGE          (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE)
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/* @} */
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/* @} *//* Audio flags */
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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 userdata An application-specific parameter saved in
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 *                  the SDL_AudioSpec structure
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 *  \param stream A pointer to the audio data buffer.
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 *  \param len    The length of that 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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 *  You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
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 *  you like. Just open your audio device with a NULL callback.
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 */
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typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
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                                            int len);
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/**
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 *  The calculated values in this structure are calculated by SDL_OpenAudio().
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 *
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 *  For multi-channel audio, the default SDL channel mapping is:
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 *  2:  FL FR                       (stereo)
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 *  3:  FL FR LFE                   (2.1 surround)
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 *  4:  FL FR BL BR                 (quad)
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 *  5:  FL FR FC BL BR              (quad + center)
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 *  6:  FL FR FC LFE SL SR          (5.1 surround - last two can also be BL BR)
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 *  7:  FL FR FC LFE BC SL SR       (6.1 surround)
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 *  8:  FL FR FC LFE BL BR SL SR    (7.1 surround)
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 */
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typedef struct SDL_AudioSpec
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{
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    int freq;                   /**< DSP frequency -- samples per second */
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    SDL_AudioFormat 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 sample FRAMES (total samples divided by channel count) */
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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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    SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
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    void *userdata;             /**< Userdata passed to callback (ignored for NULL callbacks). */
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} SDL_AudioSpec;
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struct SDL_AudioCVT;
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typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
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                                          SDL_AudioFormat format);
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/**
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 *  \brief Upper limit of filters in SDL_AudioCVT
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 *
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 *  The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
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 *  currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
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 *  one of which is the terminating NULL pointer.
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 */
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#define SDL_AUDIOCVT_MAX_FILTERS 9
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/**
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 *  \struct SDL_AudioCVT
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 *  \brief A structure to hold a set of audio conversion filters and buffers.
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 *
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 *  Note that various parts of the conversion pipeline can take advantage
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 *  of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
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 *  you to pass it aligned data, but can possibly run much faster if you
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 *  set both its (buf) field to a pointer that is aligned to 16 bytes, and its
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 *  (len) field to something that's a multiple of 16, if possible.
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 */
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#ifdef __GNUC__
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/* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
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   pad it out to 88 bytes to guarantee ABI compatibility between compilers.
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   vvv
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   The next time we rev the ABI, make sure to size the ints and add padding.
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*/
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#define SDL_AUDIOCVT_PACKED __attribute__((packed))
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#else
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#define SDL_AUDIOCVT_PACKED
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#endif
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/* */
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typedef struct SDL_AudioCVT
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{
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    int needed;                 /**< Set to 1 if conversion possible */
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    SDL_AudioFormat src_format; /**< Source audio format */
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    SDL_AudioFormat 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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    SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
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    int filter_index;           /**< Current audio conversion function */
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} SDL_AUDIOCVT_PACKED SDL_AudioCVT;
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/* Function prototypes */
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/**
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 *  \name Driver discovery functions
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 *
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 *  These functions return the list of built in audio drivers, in the
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 *  order that they are normally initialized by default.
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 */
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/* @{ */
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extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
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extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
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/* @} */
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/**
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 *  \name Initialization and cleanup
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 *
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 *  \internal These functions are used internally, and should not be used unless
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 *            you have a specific need to specify the audio driver you want to
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 *            use.  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 returns the name of the current audio driver, or NULL
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 *  if no driver has been initialized.
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 */
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extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
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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 \c obtained.  If \c 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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 *  When filling in the desired audio spec structure,
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 *    - \c desired->freq should be the desired audio frequency in samples-per-
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 *      second.
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 *    - \c desired->format should be the desired audio format.
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 *    - \c desired->samples is the desired size of the audio buffer, in
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 *      samples.  This number should be a power of two, and may be adjusted by
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 *      the audio driver to a value more suitable for the hardware.  Good values
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 *      seem to range between 512 and 8096 inclusive, depending on the
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 *      application and CPU speed.  Smaller values yield faster response time,
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 *      but can lead to underflow if the application is doing heavy processing
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 *      and cannot fill the audio buffer in time.  A stereo sample consists of
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 *      both right 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:  \code ms = (samples*1000)/freq \endcode
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 *    - \c 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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 *    - \c 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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 *    - \c 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. Alternately, you may pass a NULL
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 *      pointer here, and call SDL_QueueAudio() with some frequency, to queue
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 *      more audio samples to be played (or for capture devices, call
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 *      SDL_DequeueAudio() with some frequency, to obtain audio samples).
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 *    - \c desired->userdata is passed as the first parameter to your callback
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 *      function. If you passed a NULL callback, this value is ignored.
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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 \c 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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extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
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                                          SDL_AudioSpec * obtained);
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/**
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 *  SDL Audio Device IDs.
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 *
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 *  A successful call to SDL_OpenAudio() is always device id 1, and legacy
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 *  SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
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 *  always returns devices >= 2 on success. The legacy calls are good both
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 *  for backwards compatibility and when you don't care about multiple,
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 *  specific, or capture devices.
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 */
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typedef Uint32 SDL_AudioDeviceID;
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/**
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 *  Get the number of available devices exposed by the current driver.
icculus@1964
   334
 *  Only valid after a successfully initializing the audio subsystem.
icculus@2049
   335
 *  Returns -1 if an explicit list of devices can't be determined; this is
icculus@2049
   336
 *  not an error. For example, if SDL is set up to talk to a remote audio
icculus@2049
   337
 *  server, it can't list every one available on the Internet, but it will
icculus@2049
   338
 *  still allow a specific host to be specified to SDL_OpenAudioDevice().
slouken@7191
   339
 *
slouken@3407
   340
 *  In many common cases, when this function returns a value <= 0, it can still
icculus@2049
   341
 *  successfully open the default device (NULL for first argument of
icculus@2049
   342
 *  SDL_OpenAudioDevice()).
icculus@1964
   343
 */
icculus@1964
   344
extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
icculus@1964
   345
slouken@3407
   346
/**
slouken@3407
   347
 *  Get the human-readable name of a specific audio device.
icculus@1964
   348
 *  Must be a value between 0 and (number of audio devices-1).
icculus@1964
   349
 *  Only valid after a successfully initializing the audio subsystem.
icculus@2049
   350
 *  The values returned by this function reflect the latest call to
icculus@2049
   351
 *  SDL_GetNumAudioDevices(); recall that function to redetect available
icculus@2049
   352
 *  hardware.
slouken@7191
   353
 *
slouken@3407
   354
 *  The string returned by this function is UTF-8 encoded, read-only, and
icculus@2049
   355
 *  managed internally. You are not to free it. If you need to keep the
icculus@2049
   356
 *  string for any length of time, you should make your own copy of it, as it
icculus@2049
   357
 *  will be invalid next time any of several other SDL functions is called.
icculus@1964
   358
 */
icculus@2049
   359
extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
icculus@2049
   360
                                                           int iscapture);
icculus@1964
   361
icculus@1964
   362
slouken@3407
   363
/**
slouken@3407
   364
 *  Open a specific audio device. Passing in a device name of NULL requests
icculus@2049
   365
 *  the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
slouken@7191
   366
 *
slouken@3407
   367
 *  The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
icculus@2049
   368
 *  some drivers allow arbitrary and driver-specific strings, such as a
icculus@2049
   369
 *  hostname/IP address for a remote audio server, or a filename in the
icculus@2049
   370
 *  diskaudio driver.
slouken@7191
   371
 *
slouken@3407
   372
 *  \return 0 on error, a valid device ID that is >= 2 on success.
slouken@7191
   373
 *
icculus@2049
   374
 *  SDL_OpenAudio(), unlike this function, always acts on device ID 1.
icculus@1964
   375
 */
slouken@1967
   376
extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
slouken@1967
   377
                                                              *device,
slouken@1967
   378
                                                              int iscapture,
slouken@1967
   379
                                                              const
slouken@1967
   380
                                                              SDL_AudioSpec *
slouken@1967
   381
                                                              desired,
slouken@1967
   382
                                                              SDL_AudioSpec *
slouken@2866
   383
                                                              obtained,
slouken@2866
   384
                                                              int
slouken@2866
   385
                                                              allowed_changes);
icculus@1964
   386
icculus@1964
   387
icculus@1964
   388
slouken@3407
   389
/**
slouken@3407
   390
 *  \name Audio state
slouken@7191
   391
 *
slouken@3407
   392
 *  Get the current audio state.
slouken@0
   393
 */
gabomdq@7678
   394
/* @{ */
slouken@1895
   395
typedef enum
slouken@1895
   396
{
slouken@1895
   397
    SDL_AUDIO_STOPPED = 0,
slouken@1895
   398
    SDL_AUDIO_PLAYING,
slouken@1895
   399
    SDL_AUDIO_PAUSED
slouken@3537
   400
} SDL_AudioStatus;
slouken@3537
   401
extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
slouken@0
   402
slouken@3537
   403
extern DECLSPEC SDL_AudioStatus SDLCALL
slouken@1967
   404
SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
gabomdq@7678
   405
/* @} *//* Audio State */
icculus@1964
   406
slouken@3407
   407
/**
slouken@3407
   408
 *  \name Pause audio functions
slouken@7191
   409
 *
slouken@3407
   410
 *  These functions pause and unpause the audio callback processing.
slouken@3407
   411
 *  They should be called with a parameter of 0 after opening the audio
slouken@3407
   412
 *  device to start playing sound.  This is so you can safely initialize
slouken@3407
   413
 *  data for your callback function after opening the audio device.
slouken@3407
   414
 *  Silence will be written to the audio device during the pause.
slouken@0
   415
 */
gabomdq@7678
   416
/* @{ */
slouken@337
   417
extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
icculus@1964
   418
extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
icculus@1964
   419
                                                  int pause_on);
gabomdq@7678
   420
/* @} *//* Pause audio functions */
slouken@0
   421
slouken@3407
   422
/**
slouken@3407
   423
 *  This function loads a WAVE from the data source, automatically freeing
slouken@3407
   424
 *  that source if \c freesrc is non-zero.  For example, to load a WAVE file,
slouken@3407
   425
 *  you could do:
slouken@3407
   426
 *  \code
slouken@7191
   427
 *      SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
slouken@3407
   428
 *  \endcode
slouken@0
   429
 *
slouken@3407
   430
 *  If this function succeeds, it returns the given SDL_AudioSpec,
slouken@3407
   431
 *  filled with the audio data format of the wave data, and sets
slouken@3407
   432
 *  \c *audio_buf to a malloc()'d buffer containing the audio data,
slouken@3407
   433
 *  and sets \c *audio_len to the length of that audio buffer, in bytes.
slouken@7191
   434
 *  You need to free the audio buffer with SDL_FreeWAV() when you are
slouken@3407
   435
 *  done with it.
slouken@0
   436
 *
slouken@7191
   437
 *  This function returns NULL and sets the SDL error message if the
slouken@7191
   438
 *  wave file cannot be opened, uses an unknown data format, or is
slouken@3407
   439
 *  corrupt.  Currently raw and MS-ADPCM WAVE files are supported.
slouken@0
   440
 */
slouken@1895
   441
extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
slouken@1895
   442
                                                      int freesrc,
slouken@1895
   443
                                                      SDL_AudioSpec * spec,
slouken@1895
   444
                                                      Uint8 ** audio_buf,
slouken@1895
   445
                                                      Uint32 * audio_len);
slouken@0
   446
slouken@7191
   447
/**
slouken@3407
   448
 *  Loads a WAV from a file.
slouken@3407
   449
 *  Compatibility convenience function.
slouken@3407
   450
 */
slouken@0
   451
#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
slouken@7191
   452
    SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
slouken@0
   453
slouken@3407
   454
/**
slouken@3407
   455
 *  This function frees data previously allocated with SDL_LoadWAV_RW()
slouken@0
   456
 */
slouken@1895
   457
extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
slouken@0
   458
slouken@3407
   459
/**
slouken@3407
   460
 *  This function takes a source format and rate and a destination format
slouken@3407
   461
 *  and rate, and initializes the \c cvt structure with information needed
slouken@3407
   462
 *  by SDL_ConvertAudio() to convert a buffer of audio data from one format
icculus@11319
   463
 *  to the other. An unsupported format causes an error and -1 will be returned.
slouken@7191
   464
 *
icculus@11319
   465
 *  \return 0 if no conversion is needed, 1 if the audio filter is set up,
icculus@11319
   466
 *  or -1 on error.
slouken@0
   467
 */
slouken@1895
   468
extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
icculus@1983
   469
                                              SDL_AudioFormat src_format,
slouken@1895
   470
                                              Uint8 src_channels,
slouken@1895
   471
                                              int src_rate,
icculus@1983
   472
                                              SDL_AudioFormat dst_format,
slouken@1895
   473
                                              Uint8 dst_channels,
slouken@1895
   474
                                              int dst_rate);
slouken@0
   475
slouken@3407
   476
/**
slouken@3407
   477
 *  Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
slouken@3407
   478
 *  created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of
slouken@3407
   479
 *  audio data in the source format, this function will convert it in-place
slouken@3407
   480
 *  to the desired format.
slouken@7191
   481
 *
slouken@3407
   482
 *  The data conversion may expand the size of the audio data, so the buffer
slouken@3407
   483
 *  \c cvt->buf should be allocated after the \c cvt structure is initialized by
slouken@3407
   484
 *  SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
icculus@11319
   485
 *
icculus@11319
   486
 *  \return 0 on success or -1 if \c cvt->buf is NULL.
slouken@0
   487
 */
slouken@1895
   488
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
slouken@0
   489
slouken@11631
   490
/* SDL_AudioStream is a new audio conversion interface.
slouken@11631
   491
   The benefits vs SDL_AudioCVT:
slouken@11631
   492
    - it can handle resampling data in chunks without generating
slouken@11631
   493
      artifacts, when it doesn't have the complete buffer available.
slouken@11631
   494
    - it can handle incoming data in any variable size.
slouken@11631
   495
    - You push data as you have it, and pull it when you need it
slouken@11631
   496
 */
slouken@11631
   497
/* this is opaque to the outside world. */
slouken@11631
   498
struct _SDL_AudioStream;
slouken@11631
   499
typedef struct _SDL_AudioStream SDL_AudioStream;
slouken@11631
   500
slouken@11631
   501
/**
slouken@11631
   502
 *  Create a new audio stream
slouken@11631
   503
 *
slouken@11631
   504
 *  \param src_format The format of the source audio
slouken@11631
   505
 *  \param src_channels The number of channels of the source audio
slouken@11631
   506
 *  \param src_rate The sampling rate of the source audio
slouken@11631
   507
 *  \param dst_format The format of the desired audio output
slouken@11631
   508
 *  \param dst_channels The number of channels of the desired audio output
slouken@11631
   509
 *  \param dst_rate The sampling rate of the desired audio output
slouken@11631
   510
 *  \return 0 on success, or -1 on error.
slouken@11631
   511
 *
slouken@11631
   512
 *  \sa SDL_AudioStreamPut
slouken@11631
   513
 *  \sa SDL_AudioStreamGet
slouken@11631
   514
 *  \sa SDL_AudioStreamAvailable
slouken@11639
   515
 *  \sa SDL_AudioStreamFlush
slouken@11631
   516
 *  \sa SDL_AudioStreamClear
slouken@11631
   517
 *  \sa SDL_FreeAudioStream
slouken@11631
   518
 */
slouken@11631
   519
extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
slouken@11631
   520
                                           const Uint8 src_channels,
slouken@11631
   521
                                           const int src_rate,
slouken@11631
   522
                                           const SDL_AudioFormat dst_format,
slouken@11631
   523
                                           const Uint8 dst_channels,
slouken@11631
   524
                                           const int dst_rate);
slouken@11631
   525
slouken@11631
   526
/**
slouken@11631
   527
 *  Add data to be converted/resampled to the stream
slouken@11631
   528
 *
slouken@11631
   529
 *  \param stream The stream the audio data is being added to
slouken@11631
   530
 *  \param buf A pointer to the audio data to add
icculus@11748
   531
 *  \param len The number of bytes to write to the stream
slouken@11631
   532
 *  \return 0 on success, or -1 on error.
slouken@11631
   533
 *
slouken@11631
   534
 *  \sa SDL_NewAudioStream
slouken@11631
   535
 *  \sa SDL_AudioStreamGet
slouken@11631
   536
 *  \sa SDL_AudioStreamAvailable
slouken@11639
   537
 *  \sa SDL_AudioStreamFlush
slouken@11631
   538
 *  \sa SDL_AudioStreamClear
slouken@11631
   539
 *  \sa SDL_FreeAudioStream
slouken@11631
   540
 */
slouken@11631
   541
extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
slouken@11631
   542
slouken@11631
   543
/**
slouken@11631
   544
 *  Get converted/resampled data from the stream
slouken@11631
   545
 *
slouken@11631
   546
 *  \param stream The stream the audio is being requested from
slouken@11631
   547
 *  \param buf A buffer to fill with audio data
slouken@11631
   548
 *  \param len The maximum number of bytes to fill
slouken@11631
   549
 *  \return The number of bytes read from the stream, or -1 on error
slouken@11631
   550
 *
slouken@11631
   551
 *  \sa SDL_NewAudioStream
slouken@11631
   552
 *  \sa SDL_AudioStreamPut
slouken@11631
   553
 *  \sa SDL_AudioStreamAvailable
slouken@11639
   554
 *  \sa SDL_AudioStreamFlush
slouken@11631
   555
 *  \sa SDL_AudioStreamClear
slouken@11631
   556
 *  \sa SDL_FreeAudioStream
slouken@11631
   557
 */
slouken@11631
   558
extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
slouken@11631
   559
slouken@11631
   560
/**
icculus@11636
   561
 * Get the number of converted/resampled bytes available. The stream may be
icculus@11636
   562
 *  buffering data behind the scenes until it has enough to resample
icculus@11636
   563
 *  correctly, so this number might be lower than what you expect, or even
icculus@11636
   564
 *  be zero. Add more data or flush the stream if you need the data now.
icculus@11636
   565
 *
icculus@11636
   566
 *  \sa SDL_NewAudioStream
icculus@11636
   567
 *  \sa SDL_AudioStreamPut
icculus@11636
   568
 *  \sa SDL_AudioStreamGet
slouken@11639
   569
 *  \sa SDL_AudioStreamFlush
icculus@11636
   570
 *  \sa SDL_AudioStreamClear
icculus@11636
   571
 *  \sa SDL_FreeAudioStream
icculus@11636
   572
 */
icculus@11636
   573
extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
icculus@11636
   574
icculus@11636
   575
/**
icculus@11636
   576
 * Tell the stream that you're done sending data, and anything being buffered
icculus@11636
   577
 *  should be converted/resampled and made available immediately.
icculus@11636
   578
 *
icculus@11636
   579
 * It is legal to add more data to a stream after flushing, but there will
icculus@11636
   580
 *  be audio gaps in the output. Generally this is intended to signal the
icculus@11636
   581
 *  end of input, so the complete output becomes available.
slouken@11631
   582
 *
slouken@11631
   583
 *  \sa SDL_NewAudioStream
slouken@11631
   584
 *  \sa SDL_AudioStreamPut
slouken@11631
   585
 *  \sa SDL_AudioStreamGet
slouken@11639
   586
 *  \sa SDL_AudioStreamAvailable
slouken@11631
   587
 *  \sa SDL_AudioStreamClear
slouken@11631
   588
 *  \sa SDL_FreeAudioStream
slouken@11631
   589
 */
icculus@11636
   590
extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
slouken@11631
   591
slouken@11631
   592
/**
slouken@11631
   593
 *  Clear any pending data in the stream without converting it
slouken@11631
   594
 *
slouken@11631
   595
 *  \sa SDL_NewAudioStream
slouken@11631
   596
 *  \sa SDL_AudioStreamPut
slouken@11631
   597
 *  \sa SDL_AudioStreamGet
slouken@11631
   598
 *  \sa SDL_AudioStreamAvailable
slouken@11639
   599
 *  \sa SDL_AudioStreamFlush
slouken@11631
   600
 *  \sa SDL_FreeAudioStream
slouken@11631
   601
 */
slouken@11631
   602
extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
slouken@11631
   603
slouken@11631
   604
/**
slouken@11631
   605
 * Free an audio stream
slouken@11631
   606
 *
slouken@11631
   607
 *  \sa SDL_NewAudioStream
slouken@11631
   608
 *  \sa SDL_AudioStreamPut
slouken@11631
   609
 *  \sa SDL_AudioStreamGet
slouken@11631
   610
 *  \sa SDL_AudioStreamAvailable
slouken@11639
   611
 *  \sa SDL_AudioStreamFlush
slouken@11631
   612
 *  \sa SDL_AudioStreamClear
slouken@11631
   613
 */
slouken@11631
   614
extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
slouken@11631
   615
slouken@3407
   616
#define SDL_MIX_MAXVOLUME 128
slouken@3407
   617
/**
slouken@3407
   618
 *  This takes two audio buffers of the playing audio format and mixes
slouken@3407
   619
 *  them, performing addition, volume adjustment, and overflow clipping.
slouken@3407
   620
 *  The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME
slouken@3407
   621
 *  for full audio volume.  Note this does not change hardware volume.
slouken@3407
   622
 *  This is provided for convenience -- you can mix your own audio data.
slouken@0
   623
 */
slouken@1895
   624
extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
slouken@1895
   625
                                          Uint32 len, int volume);
slouken@0
   626
slouken@3407
   627
/**
slouken@3407
   628
 *  This works like SDL_MixAudio(), but you specify the audio format instead of
icculus@1964
   629
 *  using the format of audio device 1. Thus it can be used when no audio
icculus@1964
   630
 *  device is open at all.
icculus@1964
   631
 */
slouken@1967
   632
extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
slouken@1967
   633
                                                const Uint8 * src,
icculus@1964
   634
                                                SDL_AudioFormat format,
icculus@1964
   635
                                                Uint32 len, int volume);
icculus@1964
   636
slouken@3407
   637
/**
icculus@9012
   638
 *  Queue more audio on non-callback devices.
icculus@9012
   639
 *
icculus@10262
   640
 *  (If you are looking to retrieve queued audio from a non-callback capture
icculus@10262
   641
 *  device, you want SDL_DequeueAudio() instead. This will return -1 to
icculus@10262
   642
 *  signify an error if you use it with capture devices.)
icculus@10262
   643
 *
icculus@9012
   644
 *  SDL offers two ways to feed audio to the device: you can either supply a
icculus@9012
   645
 *  callback that SDL triggers with some frequency to obtain more audio
icculus@9012
   646
 *  (pull method), or you can supply no callback, and then SDL will expect
icculus@9012
   647
 *  you to supply data at regular intervals (push method) with this function.
icculus@9012
   648
 *
icculus@9012
   649
 *  There are no limits on the amount of data you can queue, short of
icculus@9012
   650
 *  exhaustion of address space. Queued data will drain to the device as
icculus@9012
   651
 *  necessary without further intervention from you. If the device needs
icculus@9012
   652
 *  audio but there is not enough queued, it will play silence to make up
icculus@9012
   653
 *  the difference. This means you will have skips in your audio playback
icculus@9012
   654
 *  if you aren't routinely queueing sufficient data.
icculus@9012
   655
 *
icculus@9012
   656
 *  This function copies the supplied data, so you are safe to free it when
icculus@9012
   657
 *  the function returns. This function is thread-safe, but queueing to the
icculus@9012
   658
 *  same device from two threads at once does not promise which buffer will
icculus@9012
   659
 *  be queued first.
icculus@9012
   660
 *
icculus@9012
   661
 *  You may not queue audio on a device that is using an application-supplied
icculus@9012
   662
 *  callback; doing so returns an error. You have to use the audio callback
icculus@9012
   663
 *  or queue audio with this function, but not both.
icculus@9012
   664
 *
icculus@9012
   665
 *  You should not call SDL_LockAudio() on the device before queueing; SDL
icculus@9012
   666
 *  handles locking internally for this function.
icculus@9012
   667
 *
icculus@9012
   668
 *  \param dev The device ID to which we will queue audio.
icculus@9012
   669
 *  \param data The data to queue to the device for later playback.
icculus@9012
   670
 *  \param len The number of bytes (not samples!) to which (data) points.
slouken@11631
   671
 *  \return 0 on success, or -1 on error.
icculus@9012
   672
 *
icculus@9012
   673
 *  \sa SDL_GetQueuedAudioSize
icculus@9012
   674
 *  \sa SDL_ClearQueuedAudio
icculus@9012
   675
 */
icculus@9012
   676
extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
icculus@9012
   677
icculus@9012
   678
/**
icculus@10262
   679
 *  Dequeue more audio on non-callback devices.
icculus@10262
   680
 *
icculus@10262
   681
 *  (If you are looking to queue audio for output on a non-callback playback
icculus@10262
   682
 *  device, you want SDL_QueueAudio() instead. This will always return 0
icculus@10262
   683
 *  if you use it with playback devices.)
icculus@10262
   684
 *
icculus@10262
   685
 *  SDL offers two ways to retrieve audio from a capture device: you can
icculus@10262
   686
 *  either supply a callback that SDL triggers with some frequency as the
icculus@10262
   687
 *  device records more audio data, (push method), or you can supply no
icculus@10262
   688
 *  callback, and then SDL will expect you to retrieve data at regular
icculus@10262
   689
 *  intervals (pull method) with this function.
icculus@10262
   690
 *
icculus@10262
   691
 *  There are no limits on the amount of data you can queue, short of
icculus@10262
   692
 *  exhaustion of address space. Data from the device will keep queuing as
icculus@10262
   693
 *  necessary without further intervention from you. This means you will
icculus@10262
   694
 *  eventually run out of memory if you aren't routinely dequeueing data.
icculus@10262
   695
 *
icculus@10262
   696
 *  Capture devices will not queue data when paused; if you are expecting
icculus@10262
   697
 *  to not need captured audio for some length of time, use
icculus@10262
   698
 *  SDL_PauseAudioDevice() to stop the capture device from queueing more
icculus@10262
   699
 *  data. This can be useful during, say, level loading times. When
icculus@10262
   700
 *  unpaused, capture devices will start queueing data from that point,
icculus@10262
   701
 *  having flushed any capturable data available while paused.
icculus@10262
   702
 *
icculus@10262
   703
 *  This function is thread-safe, but dequeueing from the same device from
icculus@10262
   704
 *  two threads at once does not promise which thread will dequeued data
icculus@10262
   705
 *  first.
icculus@10262
   706
 *
icculus@10262
   707
 *  You may not dequeue audio from a device that is using an
icculus@10262
   708
 *  application-supplied callback; doing so returns an error. You have to use
icculus@10262
   709
 *  the audio callback, or dequeue audio with this function, but not both.
icculus@10262
   710
 *
icculus@10262
   711
 *  You should not call SDL_LockAudio() on the device before queueing; SDL
icculus@10262
   712
 *  handles locking internally for this function.
icculus@10262
   713
 *
icculus@10262
   714
 *  \param dev The device ID from which we will dequeue audio.
icculus@10262
   715
 *  \param data A pointer into where audio data should be copied.
icculus@10262
   716
 *  \param len The number of bytes (not samples!) to which (data) points.
icculus@10262
   717
 *  \return number of bytes dequeued, which could be less than requested.
icculus@10262
   718
 *
icculus@10262
   719
 *  \sa SDL_GetQueuedAudioSize
icculus@10262
   720
 *  \sa SDL_ClearQueuedAudio
icculus@10262
   721
 */
icculus@10262
   722
extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
icculus@10262
   723
icculus@10262
   724
/**
icculus@9012
   725
 *  Get the number of bytes of still-queued audio.
icculus@9012
   726
 *
icculus@10262
   727
 *  For playback device:
icculus@9012
   728
 *
icculus@10262
   729
 *    This is the number of bytes that have been queued for playback with
icculus@10262
   730
 *    SDL_QueueAudio(), but have not yet been sent to the hardware. This
icculus@10262
   731
 *    number may shrink at any time, so this only informs of pending data.
icculus@10262
   732
 *
icculus@10262
   733
 *    Once we've sent it to the hardware, this function can not decide the
icculus@10262
   734
 *    exact byte boundary of what has been played. It's possible that we just
icculus@10262
   735
 *    gave the hardware several kilobytes right before you called this
icculus@10262
   736
 *    function, but it hasn't played any of it yet, or maybe half of it, etc.
icculus@10262
   737
 *
icculus@10262
   738
 *  For capture devices:
icculus@10262
   739
 *
icculus@10262
   740
 *    This is the number of bytes that have been captured by the device and
icculus@10262
   741
 *    are waiting for you to dequeue. This number may grow at any time, so
icculus@10262
   742
 *    this only informs of the lower-bound of available data.
icculus@9012
   743
 *
icculus@9012
   744
 *  You may not queue audio on a device that is using an application-supplied
icculus@9012
   745
 *  callback; calling this function on such a device always returns 0.
icculus@10262
   746
 *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
icculus@10262
   747
 *  the audio callback, but not both.
icculus@9012
   748
 *
icculus@9012
   749
 *  You should not call SDL_LockAudio() on the device before querying; SDL
icculus@9012
   750
 *  handles locking internally for this function.
icculus@9012
   751
 *
icculus@9012
   752
 *  \param dev The device ID of which we will query queued audio size.
icculus@9012
   753
 *  \return Number of bytes (not samples!) of queued audio.
icculus@9012
   754
 *
icculus@9012
   755
 *  \sa SDL_QueueAudio
icculus@9012
   756
 *  \sa SDL_ClearQueuedAudio
icculus@9012
   757
 */
icculus@9012
   758
extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
icculus@9012
   759
icculus@9012
   760
/**
icculus@10262
   761
 *  Drop any queued audio data. For playback devices, this is any queued data
icculus@10262
   762
 *  still waiting to be submitted to the hardware. For capture devices, this
icculus@10262
   763
 *  is any data that was queued by the device that hasn't yet been dequeued by
icculus@10262
   764
 *  the application.
icculus@9012
   765
 *
icculus@10262
   766
 *  Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
icculus@10262
   767
 *  playback devices, the hardware will start playing silence if more audio
icculus@10262
   768
 *  isn't queued. Unpaused capture devices will start filling the queue again
icculus@10262
   769
 *  as soon as they have more data available (which, depending on the state
icculus@10262
   770
 *  of the hardware and the thread, could be before this function call
icculus@10262
   771
 *  returns!).
icculus@9012
   772
 *
icculus@9012
   773
 *  This will not prevent playback of queued audio that's already been sent
icculus@9012
   774
 *  to the hardware, as we can not undo that, so expect there to be some
icculus@9012
   775
 *  fraction of a second of audio that might still be heard. This can be
icculus@9012
   776
 *  useful if you want to, say, drop any pending music during a level change
icculus@9012
   777
 *  in your game.
icculus@9012
   778
 *
icculus@9012
   779
 *  You may not queue audio on a device that is using an application-supplied
icculus@9012
   780
 *  callback; calling this function on such a device is always a no-op.
icculus@10262
   781
 *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
icculus@10262
   782
 *  the audio callback, but not both.
icculus@9012
   783
 *
icculus@9012
   784
 *  You should not call SDL_LockAudio() on the device before clearing the
icculus@9012
   785
 *  queue; SDL handles locking internally for this function.
icculus@9012
   786
 *
icculus@9012
   787
 *  This function always succeeds and thus returns void.
icculus@9012
   788
 *
icculus@9012
   789
 *  \param dev The device ID of which to clear the audio queue.
icculus@9012
   790
 *
icculus@9012
   791
 *  \sa SDL_QueueAudio
icculus@9012
   792
 *  \sa SDL_GetQueuedAudioSize
icculus@9012
   793
 */
icculus@9012
   794
extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
icculus@9012
   795
icculus@9012
   796
icculus@9012
   797
/**
slouken@3407
   798
 *  \name Audio lock functions
slouken@7191
   799
 *
slouken@3407
   800
 *  The lock manipulated by these functions protects the callback function.
slouken@7191
   801
 *  During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
slouken@3407
   802
 *  the callback function is not running.  Do not call these from the callback
slouken@3407
   803
 *  function or you will cause deadlock.
slouken@0
   804
 */
gabomdq@7678
   805
/* @{ */
slouken@337
   806
extern DECLSPEC void SDLCALL SDL_LockAudio(void);
icculus@1964
   807
extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
slouken@337
   808
extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
icculus@1964
   809
extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
gabomdq@7678
   810
/* @} *//* Audio lock functions */
slouken@0
   811
slouken@3407
   812
/**
slouken@3407
   813
 *  This function shuts down audio processing and closes the audio device.
slouken@0
   814
 */
slouken@337
   815
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
icculus@1964
   816
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
icculus@1964
   817
slouken@0
   818
/* Ends C function definitions when using C++ */
slouken@0
   819
#ifdef __cplusplus
slouken@0
   820
}
slouken@0
   821
#endif
slouken@0
   822
#include "close_code.h"
slouken@0
   823
slouken@10638
   824
#endif /* SDL_audio_h_ */
slouken@1895
   825
slouken@1895
   826
/* vi: set ts=4 sw=4 expandtab: */