1156 lines
47 KiB
C
Executable File
1156 lines
47 KiB
C
Executable File
/*
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Simple DirectMedia Layer
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Copyright (C) 1997-2021 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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/* !!! FIXME: several functions in here need Doxygen comments. */
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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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* Get the name of the current audio driver.
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*
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* The returned string points to internal static memory and thus never becomes
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* invalid, even if you quit the audio subsystem and initialize a new driver
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* (although such a case would return a different static string from another
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* call to this function, of course). As such, you should not modify or free
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* the returned string.
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*
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* \returns the name of the current audio driver or NULL if no driver has been
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* initialized.
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*
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* \since This function is available since SDL 2.0.0.
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*
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* \sa SDL_AudioInit
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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 is a legacy means of opening the audio device.
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*
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* This function remains for compatibility with SDL 1.2, but also because it's
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* slightly easier to use than the new functions in SDL 2.0. The new, more
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* powerful, and preferred way to do this is SDL_OpenAudioDevice().
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*
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* This function is roughly equivalent to:
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*
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* ```c++
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* SDL_OpenAudioDevice(NULL, 0, desired, obtained, SDL_AUDIO_ALLOW_ANY_CHANGE);
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* ```
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*
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* With two notable exceptions:
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*
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* - If `obtained` is NULL, we use `desired` (and allow no changes), which
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* means desired will be modified to have the correct values for silence,
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* etc, and SDL will convert any differences between your app's specific
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* request and the hardware behind the scenes.
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* - The return value is always success or failure, and not a device ID, which
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* means you can only have one device open at a time with this function.
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*
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* \param desired an SDL_AudioSpec structure representing the desired output
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* format. Please refer to the SDL_OpenAudioDevice documentation
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* for details on how to prepare this structure.
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* \param obtained an SDL_AudioSpec structure filled in with the actual
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* parameters, or NULL.
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* \returns This function opens the audio device with the desired parameters,
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* and returns 0 if successful, placing the actual hardware
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* parameters in the structure pointed to by `obtained`.
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*
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* If `obtained` is NULL, the audio data passed to the callback
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* function will be guaranteed to be in the requested format, and
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* will be automatically converted to the actual hardware audio
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* format if necessary. If `obtained` is NULL, `desired` will
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* have fields modified.
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*
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* This function returns a negative error code on failure to open the
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* audio device or failure to set up the audio thread; call
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* SDL_GetError() for more information.
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*
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* \sa SDL_CloseAudio
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* \sa SDL_LockAudio
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* \sa SDL_PauseAudio
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* \sa SDL_UnlockAudio
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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 built-in audio devices.
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*
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* This function is only valid after successfully initializing the audio
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* subsystem.
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*
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* Note that audio capture support is not implemented as of SDL 2.0.4, so the
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* `iscapture` parameter is for future expansion and should always be zero
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* for now.
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*
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* This function will return -1 if an explicit list of devices can't be
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* determined. Returning -1 is not an error. For example, if SDL is set up to
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* talk to a remote audio server, it can't list every one available on the
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* Internet, but it will still allow a specific host to be specified in
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* SDL_OpenAudioDevice().
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*
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* In many common cases, when this function returns a value <= 0, it can still
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* successfully open the default device (NULL for first argument of
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* SDL_OpenAudioDevice()).
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*
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* This function may trigger a complete redetect of available hardware. It
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* should not be called for each iteration of a loop, but rather once at the
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* start of a loop:
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*
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* ```c++
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* // Don't do this:
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* for (int i = 0; i < SDL_GetNumAudioDevices(0); i++)
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*
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* // do this instead:
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* const int count = SDL_GetNumAudioDevices(0);
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* for (int i = 0; i < count; ++i) { do_something_here(); }
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* ```
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*
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* \param iscapture zero to request playback devices, non-zero to request
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* recording devices
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* \returns the number of available devices exposed by the current driver or
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* -1 if an explicit list of devices can't be determined. A return
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* value of -1 does not necessarily mean an error condition.
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*
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* \since This function is available since SDL 2.0.0.
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*
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* \sa SDL_GetAudioDeviceName
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* \sa SDL_OpenAudioDevice
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*/
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extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
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/**
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* Get the human-readable name of a specific audio device.
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*
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* This function is only valid after successfully initializing the audio
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* subsystem. The values returned by this function reflect the latest call to
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* SDL_GetNumAudioDevices(); re-call that function to redetect available
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* hardware.
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*
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* The string returned by this function is UTF-8 encoded, read-only, and
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* managed internally. You are not to free it. If you need to keep the string
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* for any length of time, you should make your own copy of it, as it will be
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* invalid next time any of several other SDL functions are called.
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*
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* \param index the index of the audio device; valid values range from 0 to
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* SDL_GetNumAudioDevices() - 1
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* \param iscapture non-zero to query the list of recording devices, zero to
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* query the list of output devices.
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* \returns the name of the audio device at the requested index, or NULL on
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* error.
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*
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* \sa SDL_GetNumAudioDevices
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*/
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extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
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int iscapture);
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/**
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* Get the preferred audio format of a specific audio device.
|
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*
|
|
* This function is only valid after a successfully initializing the audio
|
|
* subsystem. The values returned by this function reflect the latest call to
|
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* SDL_GetNumAudioDevices(); re-call that function to redetect available
|
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* hardware.
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*
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* `spec` will be filled with the sample rate, sample format, and channel
|
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* count. All other values in the structure are filled with 0. When the
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* supported struct members are 0, SDL was unable to get the property from the
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* backend.
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*
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* \param index the index of the audio device; valid values range from 0 to
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* SDL_GetNumAudioDevices() - 1
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* \param iscapture non-zero to query the list of recording devices, zero to
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* query the list of output devices.
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* \param spec The SDL_AudioSpec to be initialized by this function.
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* \returns 0 on success, nonzero on error
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*
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* \sa SDL_GetNumAudioDevices
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*/
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extern DECLSPEC int SDLCALL SDL_GetAudioDeviceSpec(int index,
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int iscapture,
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SDL_AudioSpec *spec);
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|
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/**
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* Open a specific audio device.
|
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*
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* SDL_OpenAudio(), unlike this function, always acts on device ID 1. As such,
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* this function will never return a 1 so as not to conflict with the legacy
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* function.
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*
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* Please note that SDL 2.0 before 2.0.5 did not support recording; as such,
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* this function would fail if `iscapture` was not zero. Starting with SDL
|
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* 2.0.5, recording is implemented and this value can be non-zero.
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*
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|
* Passing in a `device` name of NULL requests the most reasonable default
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* (and is equivalent to what SDL_OpenAudio() does to choose a device). The
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* `device` name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
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|
* some drivers allow arbitrary and driver-specific strings, such as a
|
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* hostname/IP address for a remote audio server, or a filename in the
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* diskaudio driver.
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|
*
|
|
* When filling in the desired audio spec structure:
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*
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|
* - `desired->freq` should be the frequency in sample-frames-per-second (Hz).
|
|
* - `desired->format` should be the audio format (`AUDIO_S16SYS`, etc).
|
|
* - `desired->samples` is the desired size of the audio buffer, in
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* _sample frames_ (with stereo output, two samples--left and right--would
|
|
* make a single sample frame). 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 reduce
|
|
* latency, but can lead to underflow if the application is doing heavy
|
|
* processing and cannot fill the audio buffer in time. Note that the
|
|
* number of sample frames is directly related to time by the following
|
|
* formula: `ms = (sampleframes*1000)/freq`
|
|
* - `desired->size` is the size in _bytes_ of the audio buffer, and is
|
|
* calculated by SDL_OpenAudioDevice(). You don't initialize this.
|
|
* - `desired->silence` is the value used to set the buffer to silence,
|
|
* and is calculated by SDL_OpenAudioDevice(). You don't initialize this.
|
|
* - `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_LockAudioDevice()
|
|
* and SDL_UnlockAudioDevice() in your code. Alternately, you may pass a NULL
|
|
* pointer here, and call SDL_QueueAudio() with some frequency, to queue
|
|
* more audio samples to be played (or for capture devices, call
|
|
* SDL_DequeueAudio() with some frequency, to obtain audio samples).
|
|
* - `desired->userdata` is passed as the first parameter to your callback
|
|
* function. If you passed a NULL callback, this value is ignored.
|
|
*
|
|
* `allowed_changes` can have the following flags OR'd together:
|
|
*
|
|
* - `SDL_AUDIO_ALLOW_FREQUENCY_CHANGE`
|
|
* - `SDL_AUDIO_ALLOW_FORMAT_CHANGE`
|
|
* - `SDL_AUDIO_ALLOW_CHANNELS_CHANGE`
|
|
* - `SDL_AUDIO_ALLOW_ANY_CHANGE`
|
|
*
|
|
* These flags specify how SDL should behave when a device cannot offer a
|
|
* specific feature. If the application requests a feature that the hardware
|
|
* doesn't offer, SDL will always try to get the closest equivalent.
|
|
*
|
|
* For example, if you ask for float32 audio format, but the sound card only
|
|
* supports int16, SDL will set the hardware to int16. If you had set
|
|
* SDL_AUDIO_ALLOW_FORMAT_CHANGE, SDL will change the format in the
|
|
* `obtained` structure. If that flag was *not* set, SDL will prepare to
|
|
* convert your callback's float32 audio to int16 before feeding it to the
|
|
* hardware and will keep the originally requested format in the `obtained`
|
|
* structure.
|
|
*
|
|
* If your application can only handle one specific data format, pass a zero
|
|
* for `allowed_changes` and let SDL transparently handle any differences.
|
|
*
|
|
* An opened audio device starts out paused, and should be enabled for playing
|
|
* by calling SDL_PauseAudioDevice(devid, 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.
|
|
*
|
|
* The audio callback runs in a separate thread in most cases; you can prevent
|
|
* race conditions between your callback and other threads without fully
|
|
* pausing playback with SDL_LockAudioDevice(). For more information about the
|
|
* callback, see SDL_AudioSpec.
|
|
*
|
|
* \param device a UTF-8 string reported by SDL_GetAudioDeviceName() or a
|
|
* driver-specific name as appropriate. NULL requests the most
|
|
* reasonable default device.
|
|
* \param iscapture non-zero to specify a device should be opened for
|
|
* recording, not playback
|
|
* \param desired an SDL_AudioSpec structure representing the desired output
|
|
* format; see SDL_OpenAudio() for more information
|
|
* \param obtained an SDL_AudioSpec structure filled in with the actual output
|
|
* format; see SDL_OpenAudio() for more information
|
|
* \param allowed_changes 0, or one or more flags OR'd together
|
|
* \returns a valid device ID that is > 0 on success or 0 on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* For compatibility with SDL 1.2, this will never return 1, since
|
|
* SDL reserves that ID for the legacy SDL_OpenAudio() function.
|
|
*
|
|
* \since This function is available since SDL 2.0.0.
|
|
*
|
|
* \sa SDL_CloseAudioDevice
|
|
* \sa SDL_GetAudioDeviceName
|
|
* \sa SDL_LockAudioDevice
|
|
* \sa SDL_OpenAudio
|
|
* \sa SDL_PauseAudioDevice
|
|
* \sa SDL_UnlockAudioDevice
|
|
*/
|
|
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 */
|
|
|
|
/**
|
|
* Load the audio data of a WAVE file into memory.
|
|
*
|
|
* Loading a WAVE file requires `src`, `spec`, `audio_buf` and `audio_len`
|
|
* to be valid pointers. The entire data portion of the file is then loaded
|
|
* into memory and decoded if necessary.
|
|
*
|
|
* If `freesrc` is non-zero, the data source gets automatically closed and
|
|
* freed before the function returns.
|
|
*
|
|
* Supported formats are RIFF WAVE files with the formats PCM (8, 16, 24, and
|
|
* 32 bits), IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits),
|
|
* and A-law and mu-law (8 bits). Other formats are currently unsupported and
|
|
* cause an error.
|
|
*
|
|
* If this function succeeds, the pointer returned by it is equal to `spec`
|
|
* and the pointer to the audio data allocated by the function is written to
|
|
* `audio_buf` and its length in bytes to `audio_len`. The SDL_AudioSpec
|
|
* members `freq`, `channels`, and `format` are set to the values of the
|
|
* audio data in the buffer. The `samples` member is set to a sane default
|
|
* and all others are set to zero.
|
|
*
|
|
* It's necessary to use SDL_FreeWAV() to free the audio data returned in
|
|
* `audio_buf` when it is no longer used.
|
|
*
|
|
* Because of the underspecification of the .WAV format, there are many
|
|
* problematic files in the wild that cause issues with strict decoders. To
|
|
* provide compatibility with these files, this decoder is lenient in regards
|
|
* to the truncation of the file, the fact chunk, and the size of the RIFF
|
|
* chunk. The hints `SDL_HINT_WAVE_RIFF_CHUNK_SIZE`, `SDL_HINT_WAVE_TRUNCATION`,
|
|
* and `SDL_HINT_WAVE_FACT_CHUNK` can be used to tune the behavior of the
|
|
* loading process.
|
|
*
|
|
* Any file that is invalid (due to truncation, corruption, or wrong values in
|
|
* the headers), too big, or unsupported causes an error. Additionally, any
|
|
* critical I/O error from the data source will terminate the loading process
|
|
* with an error. The function returns NULL on error and in all cases (with the
|
|
* exception of `src` being NULL), an appropriate error message will be set.
|
|
*
|
|
* It is required that the data source supports seeking.
|
|
*
|
|
* Example:
|
|
* ```c++
|
|
* SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, &spec, &buf, &len);
|
|
* ```
|
|
*
|
|
* Note that the SDL_LoadWAV macro does this same thing for you, but in a less
|
|
* messy way:
|
|
*
|
|
* ```c++
|
|
* SDL_LoadWAV("sample.wav", &spec, &buf, &len);
|
|
* ```
|
|
*
|
|
* \param src The data source for the WAVE data
|
|
* \param freesrc If non-zero, SDL will _always_ free the data source
|
|
* \param spec An SDL_AudioSpec that will be filled in with the wave file's
|
|
* format details
|
|
* \param audio_buf A pointer filled with the audio data, allocated by the function.
|
|
* \param audio_len A pointer filled with the length of the audio data buffer in bytes
|
|
* \returns This function, if successfully called, returns `spec`, which will
|
|
* be filled with the audio data format of the wave source data.
|
|
* `audio_buf` will be filled with a pointer to an allocated buffer
|
|
* containing the audio data, and `audio_len` is filled with the
|
|
* length of that audio buffer in bytes.
|
|
*
|
|
* This function returns NULL if the .WAV file cannot be opened, uses
|
|
* an unknown data format, or is corrupt; call SDL_GetError() for
|
|
* more information.
|
|
*
|
|
* When the application is done with the data returned in
|
|
* `audio_buf`, it should call SDL_FreeWAV() to dispose of it.
|
|
*
|
|
* \sa SDL_FreeWAV
|
|
* \sa SDL_LoadWAV
|
|
*/
|
|
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)
|
|
|
|
/**
|
|
* Free data previously allocated with SDL_LoadWAV() or SDL_LoadWAV_RW().
|
|
*
|
|
* After a WAVE file has been opened with SDL_LoadWAV() or SDL_LoadWAV_RW()
|
|
* its data can eventually be freed with SDL_FreeWAV(). It is safe to call
|
|
* this function with a NULL pointer.
|
|
*
|
|
* \param audio_buf a pointer to the buffer created by SDL_LoadWAV() or
|
|
* SDL_LoadWAV_RW()
|
|
*
|
|
* \sa SDL_LoadWAV
|
|
* \sa SDL_LoadWAV_RW
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
|
|
|
|
/**
|
|
* Initialize an SDL_AudioCVT structure for conversion.
|
|
*
|
|
* Before an SDL_AudioCVT structure can be used to convert audio data it must
|
|
* be initialized with source and destination information.
|
|
*
|
|
* This function will zero out every field of the SDL_AudioCVT, so it must be
|
|
* called before the application fills in the final buffer information.
|
|
*
|
|
* Once this function has returned successfully, and reported that a
|
|
* conversion is necessary, the application fills in the rest of the fields in
|
|
* SDL_AudioCVT, now that it knows how large a buffer it needs to allocate,
|
|
* and then can call SDL_ConvertAudio() to complete the conversion.
|
|
*
|
|
* \param cvt an SDL_AudioCVT structure filled in with audio conversion
|
|
* information
|
|
* \param src_format the source format of the audio data; for more info see
|
|
* SDL_AudioFormat
|
|
* \param src_channels the number of channels in the source
|
|
* \param src_rate the frequency (sample-frames-per-second) of the source
|
|
* \param dst_format the destination format of the audio data; for more info
|
|
* see SDL_AudioFormat
|
|
* \param dst_channels the number of channels in the destination
|
|
* \param dst_rate the frequency (sample-frames-per-second) of the
|
|
* destination
|
|
* \returns 1 if the audio filter is prepared, 0 if no conversion is needed,
|
|
* or a negative error code on failure; call SDL_GetError() for more
|
|
* information.
|
|
*
|
|
* \sa SDL_ConvertAudio
|
|
*/
|
|
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);
|
|
|
|
/**
|
|
* Convert audio data to a desired audio format.
|
|
*
|
|
* This function does the actual audio data conversion, after the application
|
|
* has called SDL_BuildAudioCVT() to prepare the conversion information and
|
|
* then filled in the buffer details.
|
|
*
|
|
* Once the application has initialized the `cvt` structure using
|
|
* SDL_BuildAudioCVT(), allocated an audio buffer and filled it with audio
|
|
* data in the source format, this function will convert the buffer, in-place,
|
|
* to the desired format.
|
|
*
|
|
* The data conversion may go through several passes; any given pass may
|
|
* possibly temporarily increase the size of the data. For example, SDL might
|
|
* expand 16-bit data to 32 bits before resampling to a lower frequency,
|
|
* shrinking the data size after having grown it briefly. Since the supplied
|
|
* buffer will be both the source and destination, converting as necessary
|
|
* in-place, the application must allocate a buffer that will fully contain
|
|
* the data during its largest conversion pass. After SDL_BuildAudioCVT()
|
|
* returns, the application should set the `cvt->len` field to the size, in
|
|
* bytes, of the source data, and allocate a buffer that is
|
|
* `cvt->len * cvt->len_mult` bytes long for the `buf` field.
|
|
*
|
|
* The source data should be copied into this buffer before the call to
|
|
* SDL_ConvertAudio(). Upon successful return, this buffer will contain the
|
|
* converted audio, and `cvt->len_cvt` will be the size of the converted data,
|
|
* in bytes. Any bytes in the buffer past `cvt->len_cvt` are undefined once
|
|
* this function returns.
|
|
*
|
|
* \param cvt an SDL_AudioCVT structure that was previously set up by
|
|
* SDL_BuildAudioCVT().
|
|
* \returns 0 if the conversion was completed successfully or a negative error
|
|
* code on failure; call SDL_GetError() for more information.
|
|
*
|
|
* \sa SDL_BuildAudioCVT
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
|
|
|
|
/* SDL_AudioStream is a new audio conversion interface.
|
|
The benefits vs SDL_AudioCVT:
|
|
- it can handle resampling data in chunks without generating
|
|
artifacts, when it doesn't have the complete buffer available.
|
|
- it can handle incoming data in any variable size.
|
|
- You push data as you have it, and pull it when you need it
|
|
*/
|
|
/* this is opaque to the outside world. */
|
|
struct _SDL_AudioStream;
|
|
typedef struct _SDL_AudioStream SDL_AudioStream;
|
|
|
|
/**
|
|
* Create a new audio stream.
|
|
*
|
|
* \param src_format The format of the source audio
|
|
* \param src_channels The number of channels of the source audio
|
|
* \param src_rate The sampling rate of the source audio
|
|
* \param dst_format The format of the desired audio output
|
|
* \param dst_channels The number of channels of the desired audio output
|
|
* \param dst_rate The sampling rate of the desired audio output
|
|
* \returns 0 on success, or -1 on error.
|
|
*
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
|
|
const Uint8 src_channels,
|
|
const int src_rate,
|
|
const SDL_AudioFormat dst_format,
|
|
const Uint8 dst_channels,
|
|
const int dst_rate);
|
|
|
|
/**
|
|
* Add data to be converted/resampled to the stream.
|
|
*
|
|
* \param stream The stream the audio data is being added to
|
|
* \param buf A pointer to the audio data to add
|
|
* \param len The number of bytes to write to the stream
|
|
* \returns 0 on success, or -1 on error.
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
|
|
|
|
/**
|
|
* Get converted/resampled data from the stream
|
|
*
|
|
* \param stream The stream the audio is being requested from
|
|
* \param buf A buffer to fill with audio data
|
|
* \param len The maximum number of bytes to fill
|
|
* \returns the number of bytes read from the stream, or -1 on error
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
|
|
|
|
/**
|
|
* Get the number of converted/resampled bytes available. The stream may be
|
|
* buffering data behind the scenes until it has enough to resample
|
|
* correctly, so this number might be lower than what you expect, or even
|
|
* be zero. Add more data or flush the stream if you need the data now.
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Tell the stream that you're done sending data, and anything being buffered
|
|
* should be converted/resampled and made available immediately.
|
|
*
|
|
* It is legal to add more data to a stream after flushing, but there will
|
|
* be audio gaps in the output. Generally this is intended to signal the
|
|
* end of input, so the complete output becomes available.
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamClear
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Clear any pending data in the stream without converting it
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_FreeAudioStream
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Free an audio stream
|
|
*
|
|
* \sa SDL_NewAudioStream
|
|
* \sa SDL_AudioStreamPut
|
|
* \sa SDL_AudioStreamGet
|
|
* \sa SDL_AudioStreamAvailable
|
|
* \sa SDL_AudioStreamFlush
|
|
* \sa SDL_AudioStreamClear
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
|
|
|
|
#define SDL_MIX_MAXVOLUME 128
|
|
/**
|
|
* This function is a legacy means of mixing audio.
|
|
*
|
|
* This function is equivalent to calling
|
|
*
|
|
* ```c++
|
|
* SDL_MixAudioFormat(dst, src, format, len, volume);
|
|
* ```
|
|
*
|
|
* where `format` is the obtained format of the audio device from the legacy
|
|
* SDL_OpenAudio() function.
|
|
*
|
|
* \param dst the destination for the mixed audio
|
|
* \param src the source audio buffer to be mixed
|
|
* \param len the length of the audio buffer in bytes
|
|
* \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
|
|
* for full audio volume
|
|
*
|
|
* \sa SDL_MixAudioFormat
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
|
|
Uint32 len, int volume);
|
|
|
|
/**
|
|
* Mix audio data in a specified format.
|
|
*
|
|
* This takes an audio buffer `src` of `len` bytes of `format` data and
|
|
* mixes it into `dst`, performing addition, volume adjustment, and overflow
|
|
* clipping. The buffer pointed to by `dst` must also be `len` bytes of
|
|
* `format` data.
|
|
*
|
|
* This is provided for convenience -- you can mix your own audio data.
|
|
*
|
|
* Do not use this function for mixing together more than two streams of
|
|
* sample data. The output from repeated application of this function may be
|
|
* distorted by clipping, because there is no accumulator with greater range
|
|
* than the input (not to mention this being an inefficient way of doing it).
|
|
*
|
|
* It is a common misconception that this function is required to write audio
|
|
* data to an output stream in an audio callback. While you can do that,
|
|
* SDL_MixAudioFormat() is really only needed when you're mixing a single
|
|
* audio stream with a volume adjustment.
|
|
*
|
|
* \param dst the destination for the mixed audio
|
|
* \param src the source audio buffer to be mixed
|
|
* \param format the SDL_AudioFormat structure representing the desired audio
|
|
* format
|
|
* \param len the length of the audio buffer in bytes
|
|
* \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
|
|
* for full audio volume
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
|
|
const Uint8 * src,
|
|
SDL_AudioFormat format,
|
|
Uint32 len, int volume);
|
|
|
|
/**
|
|
* Queue more audio on non-callback devices.
|
|
*
|
|
* If you are looking to retrieve queued audio from a non-callback capture
|
|
* device, you want SDL_DequeueAudio() instead. SDL_QueueAudio() will return
|
|
* -1 to signify an error if you use it with capture devices.
|
|
*
|
|
* SDL offers two ways to feed audio to the device: you can either supply a
|
|
* callback that SDL triggers with some frequency to obtain more audio (pull
|
|
* method), or you can supply no callback, and then SDL will expect you to
|
|
* supply data at regular intervals (push method) with this function.
|
|
*
|
|
* There are no limits on the amount of data you can queue, short of
|
|
* exhaustion of address space. Queued data will drain to the device as
|
|
* necessary without further intervention from you. If the device needs audio
|
|
* but there is not enough queued, it will play silence to make up the
|
|
* difference. This means you will have skips in your audio playback if you
|
|
* aren't routinely queueing sufficient data.
|
|
*
|
|
* This function copies the supplied data, so you are safe to free it when the
|
|
* function returns. This function is thread-safe, but queueing to the same
|
|
* device from two threads at once does not promise which buffer will be
|
|
* queued first.
|
|
*
|
|
* You may not queue audio on a device that is using an application-supplied
|
|
* callback; doing so returns an error. You have to use the audio callback or
|
|
* queue audio with this function, but not both.
|
|
*
|
|
* You should not call SDL_LockAudio() on the device before queueing; SDL
|
|
* handles locking internally for this function.
|
|
*
|
|
* \param dev the device ID to which we will queue audio
|
|
* \param data the data to queue to the device for later playback
|
|
* \param len the number of bytes (not samples!) to which `data` points
|
|
* \returns 0 on success or a negative error code on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \since This function is available since SDL 2.0.4.
|
|
*
|
|
* \sa SDL_ClearQueuedAudio
|
|
* \sa SDL_GetQueuedAudioSize
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
|
|
|
|
/**
|
|
* Dequeue more audio on non-callback devices.
|
|
*
|
|
* If you are looking to queue audio for output on a non-callback playback
|
|
* device, you want SDL_QueueAudio() instead. SDL_DequeueAudio() will always
|
|
* return 0 if you use it with playback devices.
|
|
*
|
|
* SDL offers two ways to retrieve audio from a capture device: you can either
|
|
* supply a callback that SDL triggers with some frequency as the device
|
|
* records more audio data, (push method), or you can supply no callback, and
|
|
* then SDL will expect you to retrieve data at regular intervals (pull
|
|
* method) with this function.
|
|
*
|
|
* There are no limits on the amount of data you can queue, short of
|
|
* exhaustion of address space. Data from the device will keep queuing as
|
|
* necessary without further intervention from you. This means you will
|
|
* eventually run out of memory if you aren't routinely dequeueing data.
|
|
*
|
|
* Capture devices will not queue data when paused; if you are expecting to
|
|
* not need captured audio for some length of time, use SDL_PauseAudioDevice()
|
|
* to stop the capture device from queueing more data. This can be useful
|
|
* during, say, level loading times. When unpaused, capture devices will start
|
|
* queueing data from that point, having flushed any capturable data available
|
|
* while paused.
|
|
*
|
|
* This function is thread-safe, but dequeueing from the same device from two
|
|
* threads at once does not promise which thread will dequeue data first.
|
|
*
|
|
* You may not dequeue audio from a device that is using an
|
|
* application-supplied callback; doing so returns an error. You have to use
|
|
* the audio callback, or dequeue audio with this function, but not both.
|
|
*
|
|
* You should not call SDL_LockAudio() on the device before dequeueing; SDL
|
|
* handles locking internally for this function.
|
|
*
|
|
* \param dev the device ID from which we will dequeue audio
|
|
* \param data a pointer into where audio data should be copied
|
|
* \param len the number of bytes (not samples!) to which (data) points
|
|
* \returns number of bytes dequeued, which could be less than requested; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \since This function is available since SDL 2.0.5.
|
|
*
|
|
* \sa SDL_ClearQueuedAudio
|
|
* \sa SDL_GetQueuedAudioSize
|
|
*/
|
|
extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
|
|
|
|
/**
|
|
* Get the number of bytes of still-queued audio.
|
|
*
|
|
* For playback devices: this is the number of bytes that have been queued
|
|
* for playback with SDL_QueueAudio(), but have not yet been sent to the
|
|
* hardware.
|
|
*
|
|
* Once we've sent it to the hardware, this function can not decide the exact
|
|
* byte boundary of what has been played. It's possible that we just gave the
|
|
* hardware several kilobytes right before you called this function, but it
|
|
* hasn't played any of it yet, or maybe half of it, etc.
|
|
*
|
|
* For capture devices, this is the number of bytes that have been captured by
|
|
* the device and are waiting for you to dequeue. This number may grow at any
|
|
* time, so this only informs of the lower-bound of available data.
|
|
*
|
|
* You may not queue or dequeue audio on a device that is using an
|
|
* application-supplied callback; calling this function on such a device
|
|
* always returns 0. You have to use the audio callback or queue audio, but
|
|
* not both.
|
|
*
|
|
* You should not call SDL_LockAudio() on the device before querying; SDL
|
|
* handles locking internally for this function.
|
|
*
|
|
* \param dev the device ID of which we will query queued audio size
|
|
* \returns the number of bytes (not samples!) of queued audio.
|
|
*
|
|
* \since This function is available since SDL 2.0.4.
|
|
*
|
|
* \sa SDL_ClearQueuedAudio
|
|
* \sa SDL_QueueAudio
|
|
* \sa SDL_DequeueAudio
|
|
*/
|
|
extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
|
|
|
|
/**
|
|
* Drop any queued audio data waiting to be sent to the hardware.
|
|
*
|
|
* Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
|
|
* output devices, the hardware will start playing silence if more audio isn't
|
|
* queued. For capture devices, the hardware will start filling the empty
|
|
* queue with new data if the capture device isn't paused.
|
|
*
|
|
* This will not prevent playback of queued audio that's already been sent to
|
|
* the hardware, as we can not undo that, so expect there to be some fraction
|
|
* of a second of audio that might still be heard. This can be useful if you
|
|
* want to, say, drop any pending music or any unprocessed microphone input
|
|
* during a level change in your game.
|
|
*
|
|
* You may not queue or dequeue audio on a device that is using an
|
|
* application-supplied callback; calling this function on such a device
|
|
* always returns 0. You have to use the audio callback or queue audio, but
|
|
* not both.
|
|
*
|
|
* You should not call SDL_LockAudio() on the device before clearing the
|
|
* queue; SDL handles locking internally for this function.
|
|
*
|
|
* This function always succeeds and thus returns void.
|
|
*
|
|
* \param dev the device ID of which to clear the audio queue
|
|
*
|
|
* \since This function is available since SDL 2.0.4.
|
|
*
|
|
* \sa SDL_GetQueuedAudioSize
|
|
* \sa SDL_QueueAudio
|
|
* \sa SDL_DequeueAudio
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
|
|
|
|
|
|
/**
|
|
* \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 is a legacy means of closing the audio device.
|
|
*
|
|
* This function is equivalent to calling
|
|
*
|
|
* ```c++
|
|
* SDL_CloseAudioDevice(1);
|
|
* ```
|
|
*
|
|
* and is only useful if you used the legacy SDL_OpenAudio() function.
|
|
*
|
|
* \sa SDL_OpenAudio
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
|
|
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
|
|
|
|
/* Ends C function definitions when using C++ */
|
|
#ifdef __cplusplus
|
|
}
|
|
#endif
|
|
#include "close_code.h"
|
|
|
|
#endif /* SDL_audio_h_ */
|
|
|
|
/* vi: set ts=4 sw=4 expandtab: */
|