mirror of
https://github.com/thunderbrewhq/thunderbrew
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706c8903a1
* chore(build): add vendored SDL 3.0.0 library * chore(build): add vendored glew-cmake-2.2.0 library * feat(console): in the presence of -opengl launch flag, change GxApi to OpenGl * feat(gx): add uncompleted CGxDeviceGLSDL targeting Windows and Linux * chore(build): change SDL3 linkage from shared (bad) to to static (good)
1496 lines
58 KiB
C
1496 lines
58 KiB
C
/*
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Simple DirectMedia Layer
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Copyright (C) 1997-2023 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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* \brief Audio functionality 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 <SDL3/SDL_stdinc.h>
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#include <SDL3/SDL_endian.h>
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#include <SDL3/SDL_error.h>
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#include <SDL3/SDL_mutex.h>
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#include <SDL3/SDL_properties.h>
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#include <SDL3/SDL_rwops.h>
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#include <SDL3/SDL_thread.h>
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#include <SDL3/SDL_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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* 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 LFE BL BR (4.1 surround)
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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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/**
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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_FLOAT (1<<8)
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#define SDL_AUDIO_MASK_BIG_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_BYTESIZE(x) (SDL_AUDIO_BITSIZE(x) / 8)
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#define SDL_AUDIO_ISFLOAT(x) ((x) & SDL_AUDIO_MASK_FLOAT)
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#define SDL_AUDIO_ISBIGENDIAN(x) ((x) & SDL_AUDIO_MASK_BIG_ENDIAN)
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#define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x))
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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_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 SDL_AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
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#define SDL_AUDIO_S8 0x8008 /**< Signed 8-bit samples */
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#define SDL_AUDIO_S16LE 0x8010 /**< Signed 16-bit samples */
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#define SDL_AUDIO_S16BE 0x9010 /**< As above, but big-endian byte order */
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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 SDL_AUDIO_S32LE 0x8020 /**< 32-bit integer samples */
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#define SDL_AUDIO_S32BE 0x9020 /**< As above, but big-endian byte order */
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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 SDL_AUDIO_F32LE 0x8120 /**< 32-bit floating point samples */
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#define SDL_AUDIO_F32BE 0x9120 /**< As above, but big-endian byte order */
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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 SDL_AUDIO_S16 SDL_AUDIO_S16LE
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#define SDL_AUDIO_S32 SDL_AUDIO_S32LE
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#define SDL_AUDIO_F32 SDL_AUDIO_F32LE
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#else
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#define SDL_AUDIO_S16 SDL_AUDIO_S16BE
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#define SDL_AUDIO_S32 SDL_AUDIO_S32BE
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#define SDL_AUDIO_F32 SDL_AUDIO_F32BE
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#endif
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/* @} */
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/* @} *//* Audio flags */
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/**
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* SDL Audio Device instance IDs.
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*/
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typedef Uint32 SDL_AudioDeviceID;
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#define SDL_AUDIO_DEVICE_DEFAULT_OUTPUT ((SDL_AudioDeviceID) 0xFFFFFFFF)
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#define SDL_AUDIO_DEVICE_DEFAULT_CAPTURE ((SDL_AudioDeviceID) 0xFFFFFFFE)
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typedef struct SDL_AudioSpec
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{
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SDL_AudioFormat format; /**< Audio data format */
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int channels; /**< Number of channels: 1 mono, 2 stereo, etc */
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int freq; /**< sample rate: sample frames per second */
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} SDL_AudioSpec;
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/* Calculate the size of each audio frame (in bytes) */
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#define SDL_AUDIO_FRAMESIZE(x) (SDL_AUDIO_BYTESIZE((x).format) * (x).channels)
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/* SDL_AudioStream is an audio conversion interface.
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- It can handle resampling data in chunks without generating
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artifacts, when it doesn't have the complete buffer available.
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- It can handle incoming data in any variable size.
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- It can handle input/output format changes on the fly.
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- You push data as you have it, and pull it when you need it
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- It can also function as a basic audio data queue even if you
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just have sound that needs to pass from one place to another.
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- You can hook callbacks up to them when more data is added or
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requested, to manage data on-the-fly.
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*/
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struct SDL_AudioStream; /* this is opaque to the outside world. */
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typedef struct SDL_AudioStream SDL_AudioStream;
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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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/**
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* Use this function to get the number of built-in audio drivers.
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*
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* This function returns a hardcoded number. This never returns a negative
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* value; if there are no drivers compiled into this build of SDL, this
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* function returns zero. The presence of a driver in this list does not mean
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* it will function, it just means SDL is capable of interacting with that
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* interface. For example, a build of SDL might have esound support, but if
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* there's no esound server available, SDL's esound driver would fail if used.
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*
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* By default, SDL tries all drivers, in its preferred order, until one is
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* found to be usable.
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*
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* \returns the number of built-in audio drivers.
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*
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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*
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* \sa SDL_GetAudioDriver
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*/
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extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
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/**
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* Use this function to get the name of a built in audio driver.
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*
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* The list of audio drivers is given in the order that they are normally
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* initialized by default; the drivers that seem more reasonable to choose
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* first (as far as the SDL developers believe) are earlier in the list.
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*
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* The names of drivers are all simple, low-ASCII identifiers, like "alsa",
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* "coreaudio" or "xaudio2". These never have Unicode characters, and are not
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* meant to be proper names.
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*
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* \param index the index of the audio driver; the value ranges from 0 to
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* SDL_GetNumAudioDrivers() - 1
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* \returns the name of the audio driver at the requested index, or NULL if an
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* invalid index was specified.
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*
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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*
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* \sa SDL_GetNumAudioDrivers
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*/
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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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* 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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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*/
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extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
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/**
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* Get a list of currently-connected audio output devices.
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*
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* This returns of list of available devices that play sound, perhaps to
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* speakers or headphones ("output" devices). If you want devices that record
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* audio, like a microphone ("capture" devices), use
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* SDL_GetAudioCaptureDevices() instead.
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*
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* This only returns a list of physical devices; it will not have any device
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* IDs returned by SDL_OpenAudioDevice().
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*
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* \param count a pointer filled in with the number of devices returned
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* \returns a 0 terminated array of device instance IDs which should be freed
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* with SDL_free(), or NULL on error; call SDL_GetError() for more
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* details.
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*
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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*
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* \sa SDL_OpenAudioDevice
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* \sa SDL_GetAudioCaptureDevices
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*/
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extern DECLSPEC SDL_AudioDeviceID *SDLCALL SDL_GetAudioOutputDevices(int *count);
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/**
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* Get a list of currently-connected audio capture devices.
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*
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* This returns of list of available devices that record audio, like a
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* microphone ("capture" devices). If you want devices that play sound,
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* perhaps to speakers or headphones ("output" devices), use
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* SDL_GetAudioOutputDevices() instead.
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*
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* This only returns a list of physical devices; it will not have any device
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* IDs returned by SDL_OpenAudioDevice().
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*
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* \param count a pointer filled in with the number of devices returned
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* \returns a 0 terminated array of device instance IDs which should be freed
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* with SDL_free(), or NULL on error; call SDL_GetError() for more
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* details.
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*
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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*
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* \sa SDL_OpenAudioDevice
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* \sa SDL_GetAudioOutputDevices
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*/
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extern DECLSPEC SDL_AudioDeviceID *SDLCALL SDL_GetAudioCaptureDevices(int *count);
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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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* The string returned by this function is UTF-8 encoded. The caller should
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* call SDL_free on the return value when done with it.
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*
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* \param devid the instance ID of the device to query.
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* \returns the name of the audio device, or NULL on error.
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*
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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*
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* \sa SDL_GetAudioOutputDevices
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* \sa SDL_GetAudioCaptureDevices
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* \sa SDL_GetDefaultAudioInfo
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*/
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extern DECLSPEC char *SDLCALL SDL_GetAudioDeviceName(SDL_AudioDeviceID devid);
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/**
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* Get the current audio format of a specific audio device.
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*
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* For an opened device, this will report the format the device is currently
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* using. If the device isn't yet opened, this will report the device's
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* preferred format (or a reasonable default if this can't be determined).
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*
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* You may also specify SDL_AUDIO_DEVICE_DEFAULT_OUTPUT or
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* SDL_AUDIO_DEVICE_DEFAULT_CAPTURE here, which is useful for getting a
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* reasonable recommendation before opening the system-recommended default
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* device.
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*
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* You can also use this to request the current device buffer size. This is
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* specified in sample frames and represents the amount of data SDL will feed
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* to the physical hardware in each chunk. This can be converted to
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* milliseconds of audio with the following equation:
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*
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* `ms = (int) ((((Sint64) frames) * 1000) / spec.freq);`
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*
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* Buffer size is only important if you need low-level control over the audio
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* playback timing. Most apps do not need this.
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*
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* \param devid the instance ID of the device to query.
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* \param spec On return, will be filled with device details.
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* \param sample_frames Pointer to store device buffer size, in sample frames.
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* Can be NULL.
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* \returns 0 on success or a negative error code on failure; call
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* SDL_GetError() for more information.
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*
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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*/
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extern DECLSPEC int SDLCALL SDL_GetAudioDeviceFormat(SDL_AudioDeviceID devid, SDL_AudioSpec *spec, int *sample_frames);
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/**
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* Open a specific audio device.
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*
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* You can open both output and capture devices through this function. Output
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* devices will take data from bound audio streams, mix it, and send it to the
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* hardware. Capture devices will feed any bound audio streams with a copy of
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* any incoming data.
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*
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* An opened audio device starts out with no audio streams bound. To start
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* audio playing, bind a stream and supply audio data to it. Unlike SDL2,
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* there is no audio callback; you only bind audio streams and make sure they
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* have data flowing into them (however, you can simulate SDL2's semantics
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* fairly closely by using SDL_OpenAudioDeviceStream instead of this
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* function).
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*
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* If you don't care about opening a specific device, pass a `devid` of either
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* `SDL_AUDIO_DEVICE_DEFAULT_OUTPUT` or `SDL_AUDIO_DEVICE_DEFAULT_CAPTURE`. In
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* this case, SDL will try to pick the most reasonable default, and may also
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* switch between physical devices seamlessly later, if the most reasonable
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* default changes during the lifetime of this opened device (user changed the
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* default in the OS's system preferences, the default got unplugged so the
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* system jumped to a new default, the user plugged in headphones on a mobile
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* device, etc). Unless you have a good reason to choose a specific device,
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* this is probably what you want.
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*
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* You may request a specific format for the audio device, but there is no
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* promise the device will honor that request for several reasons. As such,
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* it's only meant to be a hint as to what data your app will provide. Audio
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* streams will accept data in whatever format you specify and manage
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* conversion for you as appropriate. SDL_GetAudioDeviceFormat can tell you
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* the preferred format for the device before opening and the actual format
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* the device is using after opening.
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*
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* It's legal to open the same device ID more than once; each successful open
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* will generate a new logical SDL_AudioDeviceID that is managed separately
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* from others on the same physical device. This allows libraries to open a
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* device separately from the main app and bind its own streams without
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* conflicting.
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*
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* It is also legal to open a device ID returned by a previous call to this
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* function; doing so just creates another logical device on the same physical
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* device. This may be useful for making logical groupings of audio streams.
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*
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* This function returns the opened device ID on success. This is a new,
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* unique SDL_AudioDeviceID that represents a logical device.
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*
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* Some backends might offer arbitrary devices (for example, a networked audio
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* protocol that can connect to an arbitrary server). For these, as a change
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* from SDL2, you should open a default device ID and use an SDL hint to
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* specify the target if you care, or otherwise let the backend figure out a
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* reasonable default. Most backends don't offer anything like this, and often
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* this would be an end user setting an environment variable for their custom
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* need, and not something an application should specifically manage.
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*
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* When done with an audio device, possibly at the end of the app's life, one
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* should call SDL_CloseAudioDevice() on the returned device id.
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*
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* \param devid the device instance id to open, or
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* SDL_AUDIO_DEVICE_DEFAULT_OUTPUT or
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* SDL_AUDIO_DEVICE_DEFAULT_CAPTURE for the most reasonable
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* default device.
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* \param spec the requested device configuration. Can be NULL to use
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* reasonable defaults.
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* \returns The device ID on success, 0 on error; call SDL_GetError() for more
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* information.
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*
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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*
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* \sa SDL_CloseAudioDevice
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* \sa SDL_GetAudioDeviceFormat
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*/
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extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(SDL_AudioDeviceID devid, const SDL_AudioSpec *spec);
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/**
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* Use this function to pause audio playback on a specified device.
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*
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* This function pauses audio processing for a given device. Any bound audio
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* streams will not progress, and no audio will be generated. Pausing one
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* device does not prevent other unpaused devices from running.
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*
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* Unlike in SDL2, audio devices start in an _unpaused_ state, since an app
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* has to bind a stream before any audio will flow. Pausing a paused device is
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* a legal no-op.
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*
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* Pausing a device can be useful to halt all audio without unbinding all the
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* audio streams. This might be useful while a game is paused, or a level is
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* loading, etc.
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*
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* Physical devices can not be paused or unpaused, only logical devices
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* created through SDL_OpenAudioDevice() can be.
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*
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* \param dev a device opened by SDL_OpenAudioDevice()
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* \returns 0 on success or a negative error code on failure; call
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* SDL_GetError() for more information.
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*
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* \threadsafety It is safe to call this function from any thread.
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*
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* \since This function is available since SDL 3.0.0.
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*
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* \sa SDL_ResumeAudioDevice
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* \sa SDL_AudioDevicePaused
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*/
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extern DECLSPEC int SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev);
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/**
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* Use this function to unpause audio playback on a specified device.
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*
|
|
* This function unpauses audio processing for a given device that has
|
|
* previously been paused with SDL_PauseAudioDevice(). Once unpaused, any
|
|
* bound audio streams will begin to progress again, and audio can be
|
|
* generated.
|
|
*
|
|
* Unlike in SDL2, audio devices start in an _unpaused_ state, since an app
|
|
* has to bind a stream before any audio will flow. Unpausing an unpaused
|
|
* device is a legal no-op.
|
|
*
|
|
* Physical devices can not be paused or unpaused, only logical devices
|
|
* created through SDL_OpenAudioDevice() can be.
|
|
*
|
|
* \param dev a device opened by SDL_OpenAudioDevice()
|
|
* \returns 0 on success or a negative error code on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_AudioDevicePaused
|
|
* \sa SDL_PauseAudioDevice
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_ResumeAudioDevice(SDL_AudioDeviceID dev);
|
|
|
|
/**
|
|
* Use this function to query if an audio device is paused.
|
|
*
|
|
* Unlike in SDL2, audio devices start in an _unpaused_ state, since an app
|
|
* has to bind a stream before any audio will flow.
|
|
*
|
|
* Physical devices can not be paused or unpaused, only logical devices
|
|
* created through SDL_OpenAudioDevice() can be. Physical and invalid device
|
|
* IDs will report themselves as unpaused here.
|
|
*
|
|
* \param dev a device opened by SDL_OpenAudioDevice()
|
|
* \returns SDL_TRUE if device is valid and paused, SDL_FALSE otherwise.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_PauseAudioDevice
|
|
* \sa SDL_ResumeAudioDevice
|
|
*/
|
|
extern DECLSPEC SDL_bool SDLCALL SDL_AudioDevicePaused(SDL_AudioDeviceID dev);
|
|
|
|
/**
|
|
* Close a previously-opened audio device.
|
|
*
|
|
* The application should close open audio devices once they are no longer
|
|
* needed.
|
|
*
|
|
* This function may block briefly while pending audio data is played by the
|
|
* hardware, so that applications don't drop the last buffer of data they
|
|
* supplied if terminating immediately afterwards.
|
|
*
|
|
* \param devid an audio device id previously returned by
|
|
* SDL_OpenAudioDevice()
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_OpenAudioDevice
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID devid);
|
|
|
|
/**
|
|
* Bind a list of audio streams to an audio device.
|
|
*
|
|
* Audio data will flow through any bound streams. For an output device, data
|
|
* for all bound streams will be mixed together and fed to the device. For a
|
|
* capture device, a copy of recorded data will be provided to each bound
|
|
* stream.
|
|
*
|
|
* Audio streams can only be bound to an open device. This operation is
|
|
* atomic--all streams bound in the same call will start processing at the
|
|
* same time, so they can stay in sync. Also: either all streams will be bound
|
|
* or none of them will be.
|
|
*
|
|
* It is an error to bind an already-bound stream; it must be explicitly
|
|
* unbound first.
|
|
*
|
|
* Binding a stream to a device will set its output format for output devices,
|
|
* and its input format for capture devices, so they match the device's
|
|
* settings. The caller is welcome to change the other end of the stream's
|
|
* format at any time.
|
|
*
|
|
* \param devid an audio device to bind a stream to.
|
|
* \param streams an array of audio streams to unbind.
|
|
* \param num_streams Number streams listed in the `streams` array.
|
|
* \returns 0 on success, -1 on error; call SDL_GetError() for more
|
|
* information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_BindAudioStreams
|
|
* \sa SDL_UnbindAudioStreams
|
|
* \sa SDL_UnbindAudioStream
|
|
* \sa SDL_GetAudioStreamDevice
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_BindAudioStreams(SDL_AudioDeviceID devid, SDL_AudioStream **streams, int num_streams);
|
|
|
|
/**
|
|
* Bind a single audio stream to an audio device.
|
|
*
|
|
* This is a convenience function, equivalent to calling
|
|
* `SDL_BindAudioStreams(devid, &stream, 1)`.
|
|
*
|
|
* \param devid an audio device to bind a stream to.
|
|
* \param stream an audio stream to bind to a device.
|
|
* \returns 0 on success, -1 on error; call SDL_GetError() for more
|
|
* information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_BindAudioStreams
|
|
* \sa SDL_UnbindAudioStreams
|
|
* \sa SDL_UnbindAudioStream
|
|
* \sa SDL_GetAudioStreamDevice
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_BindAudioStream(SDL_AudioDeviceID devid, SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Unbind a list of audio streams from their audio devices.
|
|
*
|
|
* The streams being unbound do not all have to be on the same device. All
|
|
* streams on the same device will be unbound atomically (data will stop
|
|
* flowing through them all unbound streams on the same device at the same
|
|
* time).
|
|
*
|
|
* Unbinding a stream that isn't bound to a device is a legal no-op.
|
|
*
|
|
* \param streams an array of audio streams to unbind.
|
|
* \param num_streams Number streams listed in the `streams` array.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_BindAudioStreams
|
|
* \sa SDL_BindAudioStream
|
|
* \sa SDL_UnbindAudioStream
|
|
* \sa SDL_GetAudioStreamDevice
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_UnbindAudioStreams(SDL_AudioStream **streams, int num_streams);
|
|
|
|
/**
|
|
* Unbind a single audio stream from its audio device.
|
|
*
|
|
* This is a convenience function, equivalent to calling
|
|
* `SDL_UnbindAudioStreams(&stream, 1)`.
|
|
*
|
|
* \param stream an audio stream to unbind from a device.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_BindAudioStream
|
|
* \sa SDL_BindAudioStreams
|
|
* \sa SDL_UnbindAudioStreams
|
|
* \sa SDL_GetAudioStreamDevice
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_UnbindAudioStream(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Query an audio stream for its currently-bound device.
|
|
*
|
|
* This reports the audio device that an audio stream is currently bound to.
|
|
*
|
|
* If not bound, or invalid, this returns zero, which is not a valid device
|
|
* ID.
|
|
*
|
|
* \param stream the audio stream to query.
|
|
* \returns The bound audio device, or 0 if not bound or invalid.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_BindAudioStream
|
|
* \sa SDL_BindAudioStreams
|
|
* \sa SDL_UnbindAudioStream
|
|
* \sa SDL_UnbindAudioStreams
|
|
*/
|
|
extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_GetAudioStreamDevice(SDL_AudioStream *stream);
|
|
|
|
|
|
/**
|
|
* Create a new audio stream.
|
|
*
|
|
* \param src_spec The format details of the input audio
|
|
* \param dst_spec The format details of the output audio
|
|
* \returns a new audio stream on success, or NULL on failure.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_PutAudioStreamData
|
|
* \sa SDL_GetAudioStreamData
|
|
* \sa SDL_GetAudioStreamAvailable
|
|
* \sa SDL_FlushAudioStream
|
|
* \sa SDL_ClearAudioStream
|
|
* \sa SDL_ChangeAudioStreamOutput
|
|
* \sa SDL_DestroyAudioStream
|
|
*/
|
|
extern DECLSPEC SDL_AudioStream *SDLCALL SDL_CreateAudioStream(const SDL_AudioSpec *src_spec, const SDL_AudioSpec *dst_spec);
|
|
|
|
/**
|
|
* Get the properties associated with an audio stream.
|
|
*
|
|
* \param stream the SDL_AudioStream to query
|
|
* \returns a valid property ID on success or 0 on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_GetProperty
|
|
* \sa SDL_SetProperty
|
|
*/
|
|
extern DECLSPEC SDL_PropertiesID SDLCALL SDL_GetAudioStreamProperties(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Query the current format of an audio stream.
|
|
*
|
|
* \param stream the SDL_AudioStream to query.
|
|
* \param src_spec Where to store the input audio format; ignored if NULL.
|
|
* \param dst_spec Where to store the output audio format; ignored if NULL.
|
|
* \returns 0 on success, or -1 on error.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread, as it holds
|
|
* a stream-specific mutex while running.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_GetAudioStreamFormat(SDL_AudioStream *stream,
|
|
SDL_AudioSpec *src_spec,
|
|
SDL_AudioSpec *dst_spec);
|
|
|
|
/**
|
|
* Change the input and output formats of an audio stream.
|
|
*
|
|
* Future calls to and SDL_GetAudioStreamAvailable and SDL_GetAudioStreamData
|
|
* will reflect the new format, and future calls to SDL_PutAudioStreamData
|
|
* must provide data in the new input formats.
|
|
*
|
|
* \param stream The stream the format is being changed
|
|
* \param src_spec The new format of the audio input; if NULL, it is not
|
|
* changed.
|
|
* \param dst_spec The new format of the audio output; if NULL, it is not
|
|
* changed.
|
|
* \returns 0 on success, or -1 on error.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread, as it holds
|
|
* a stream-specific mutex while running.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_GetAudioStreamFormat
|
|
* \sa SDL_PutAudioStreamData
|
|
* \sa SDL_GetAudioStreamData
|
|
* \sa SDL_GetAudioStreamAvailable
|
|
* \sa SDL_SetAudioStreamFrequencyRatio
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_SetAudioStreamFormat(SDL_AudioStream *stream,
|
|
const SDL_AudioSpec *src_spec,
|
|
const SDL_AudioSpec *dst_spec);
|
|
|
|
/**
|
|
* Get the frequency ratio of an audio stream.
|
|
*
|
|
* \param stream the SDL_AudioStream to query.
|
|
* \returns the frequency ratio of the stream, or 0.0 on error
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread, as it holds
|
|
* a stream-specific mutex while running.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_SetAudioStreamFrequencyRatio
|
|
*/
|
|
extern DECLSPEC float SDLCALL SDL_GetAudioStreamFrequencyRatio(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Change the frequency ratio of an audio stream.
|
|
*
|
|
* The frequency ratio is used to adjust the rate at which input data is
|
|
* consumed. Changing this effectively modifies the speed and pitch of the
|
|
* audio. A value greater than 1.0 will play the audio faster, and at a higher
|
|
* pitch. A value less than 1.0 will play the audio slower, and at a lower
|
|
* pitch.
|
|
*
|
|
* This is applied during SDL_GetAudioStreamData, and can be continuously
|
|
* changed to create various effects.
|
|
*
|
|
* \param stream The stream the frequency ratio is being changed
|
|
* \param ratio The frequency ratio. 1.0 is normal speed. Must be between 0.01
|
|
* and 100.
|
|
* \returns 0 on success, or -1 on error.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread, as it holds
|
|
* a stream-specific mutex while running.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_GetAudioStreamFrequencyRatio
|
|
* \sa SDL_SetAudioStreamFormat
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_SetAudioStreamFrequencyRatio(SDL_AudioStream *stream, float ratio);
|
|
|
|
/**
|
|
* Add data to be converted/resampled to the stream.
|
|
*
|
|
* This data must match the format/channels/samplerate specified in the latest
|
|
* call to SDL_SetAudioStreamFormat, or the format specified when creating the
|
|
* stream if it hasn't been changed.
|
|
*
|
|
* Note that this call simply queues unconverted data for later. This is
|
|
* different than SDL2, where data was converted during the Put call and the
|
|
* Get call would just dequeue the previously-converted data.
|
|
*
|
|
* \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 a negative error code on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread, but if the
|
|
* stream has a callback set, the caller might need to manage
|
|
* extra locking.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_CreateAudioStream
|
|
* \sa SDL_GetAudioStreamData
|
|
* \sa SDL_GetAudioStreamAvailable
|
|
* \sa SDL_FlushAudioStream
|
|
* \sa SDL_ClearAudioStream
|
|
* \sa SDL_DestroyAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len);
|
|
|
|
/**
|
|
* Get converted/resampled data from the stream.
|
|
*
|
|
* The input/output data format/channels/samplerate is specified when creating
|
|
* the stream, and can be changed after creation by calling
|
|
* SDL_SetAudioStreamFormat.
|
|
*
|
|
* Note that any conversion and resampling necessary is done during this call,
|
|
* and SDL_PutAudioStreamData simply queues unconverted data for later. This
|
|
* is different than SDL2, where that work was done while inputting new data
|
|
* to the stream and requesting the output just copied the converted data.
|
|
*
|
|
* \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
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread, but if the
|
|
* stream has a callback set, the caller might need to manage
|
|
* extra locking.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_CreateAudioStream
|
|
* \sa SDL_PutAudioStreamData
|
|
* \sa SDL_GetAudioStreamAvailable
|
|
* \sa SDL_SetAudioStreamFormat
|
|
* \sa SDL_FlushAudioStream
|
|
* \sa SDL_ClearAudioStream
|
|
* \sa SDL_DestroyAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_GetAudioStreamData(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.
|
|
*
|
|
* If the stream has so much data that it would overflow an int, the return
|
|
* value is clamped to a maximum value, but no queued data is lost; if there
|
|
* are gigabytes of data queued, the app might need to read some of it with
|
|
* SDL_GetAudioStreamData before this function's return value is no longer
|
|
* clamped.
|
|
*
|
|
* \param stream The audio stream to query
|
|
* \returns the number of converted/resampled bytes available.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_CreateAudioStream
|
|
* \sa SDL_PutAudioStreamData
|
|
* \sa SDL_GetAudioStreamData
|
|
* \sa SDL_FlushAudioStream
|
|
* \sa SDL_ClearAudioStream
|
|
* \sa SDL_DestroyAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_GetAudioStreamAvailable(SDL_AudioStream *stream);
|
|
|
|
|
|
/**
|
|
* Get the number of bytes currently queued.
|
|
*
|
|
* Note that audio streams can change their input format at any time, even if
|
|
* there is still data queued in a different format, so the returned byte
|
|
* count will not necessarily match the number of _sample frames_ available.
|
|
* Users of this API should be aware of format changes they make when feeding
|
|
* a stream and plan accordingly.
|
|
*
|
|
* Queued data is not converted until it is consumed by
|
|
* SDL_GetAudioStreamData, so this value should be representative of the exact
|
|
* data that was put into the stream.
|
|
*
|
|
* If the stream has so much data that it would overflow an int, the return
|
|
* value is clamped to a maximum value, but no queued data is lost; if there
|
|
* are gigabytes of data queued, the app might need to read some of it with
|
|
* SDL_GetAudioStreamData before this function's return value is no longer
|
|
* clamped.
|
|
*
|
|
* \param stream The audio stream to query
|
|
* \returns the number of bytes queued.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_PutAudioStreamData
|
|
* \sa SDL_GetAudioStreamData
|
|
* \sa SDL_ClearAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_GetAudioStreamQueued(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.
|
|
*
|
|
* \param stream The audio stream to flush
|
|
* \returns 0 on success or a negative error code on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_CreateAudioStream
|
|
* \sa SDL_PutAudioStreamData
|
|
* \sa SDL_GetAudioStreamData
|
|
* \sa SDL_GetAudioStreamAvailable
|
|
* \sa SDL_ClearAudioStream
|
|
* \sa SDL_DestroyAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_FlushAudioStream(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Clear any pending data in the stream without converting it
|
|
*
|
|
* \param stream The audio stream to clear
|
|
* \returns 0 on success or a negative error code on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_CreateAudioStream
|
|
* \sa SDL_PutAudioStreamData
|
|
* \sa SDL_GetAudioStreamData
|
|
* \sa SDL_GetAudioStreamAvailable
|
|
* \sa SDL_FlushAudioStream
|
|
* \sa SDL_DestroyAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_ClearAudioStream(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* Lock an audio stream for serialized access.
|
|
*
|
|
* Each SDL_AudioStream has an internal mutex it uses to protect its data
|
|
* structures from threading conflicts. This function allows an app to lock
|
|
* that mutex, which could be useful if registering callbacks on this stream.
|
|
*
|
|
* One does not need to lock a stream to use in it most cases, as the stream
|
|
* manages this lock internally. However, this lock is held during callbacks,
|
|
* which may run from arbitrary threads at any time, so if an app needs to
|
|
* protect shared data during those callbacks, locking the stream guarantees
|
|
* that the callback is not running while the lock is held.
|
|
*
|
|
* As this is just a wrapper over SDL_LockMutex for an internal lock, it has
|
|
* all the same attributes (recursive locks are allowed, etc).
|
|
*
|
|
* \param stream The audio stream to lock.
|
|
* \returns 0 on success or a negative error code on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_UnlockAudioStream
|
|
* \sa SDL_SetAudioStreamPutCallback
|
|
* \sa SDL_SetAudioStreamGetCallback
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_LockAudioStream(SDL_AudioStream *stream);
|
|
|
|
|
|
/**
|
|
* Unlock an audio stream for serialized access.
|
|
*
|
|
* This unlocks an audio stream after a call to SDL_LockAudioStream.
|
|
*
|
|
* \param stream The audio stream to unlock.
|
|
* \returns 0 on success or a negative error code on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \threadsafety You should only call this from the same thread that
|
|
* previously called SDL_LockAudioStream.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_LockAudioStream
|
|
* \sa SDL_SetAudioStreamPutCallback
|
|
* \sa SDL_SetAudioStreamGetCallback
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_UnlockAudioStream(SDL_AudioStream *stream);
|
|
|
|
/**
|
|
* A callback that fires when data passes through an SDL_AudioStream.
|
|
*
|
|
* Apps can (optionally) register a callback with an audio stream that
|
|
* is called when data is added with SDL_PutAudioStreamData, or requested
|
|
* with SDL_GetAudioStreamData. These callbacks may run from any
|
|
* thread, so if you need to protect shared data, you should use
|
|
* SDL_LockAudioStream to serialize access; this lock will be held by
|
|
* before your callback is called, so your callback does not need to
|
|
* manage the lock explicitly.
|
|
*
|
|
* Two values are offered here: one is the amount of additional data needed
|
|
* to satisfy the immediate request (which might be zero if the stream
|
|
* already has enough data queued) and the other is the total amount
|
|
* being requested. In a Get call triggering a Put callback, these
|
|
* values can be different. In a Put call triggering a Get callback,
|
|
* these values are always the same.
|
|
*
|
|
* Byte counts might be slightly overestimated due to buffering or
|
|
* resampling, and may change from call to call.
|
|
*
|
|
* \param stream The SDL audio stream associated with this callback.
|
|
* \param additional_amount The amount of data, in bytes, that is needed right now.
|
|
* \param total_amount The total amount of data requested, in bytes, that is requested or available.
|
|
* \param userdata An opaque pointer provided by the app for their personal use.
|
|
*/
|
|
typedef void (SDLCALL *SDL_AudioStreamCallback)(void *userdata, SDL_AudioStream *stream, int additional_amount, int total_amount);
|
|
|
|
/**
|
|
* Set a callback that runs when data is requested from an audio stream.
|
|
*
|
|
* This callback is called _before_ data is obtained from the stream, giving
|
|
* the callback the chance to add more on-demand.
|
|
*
|
|
* The callback can (optionally) call SDL_PutAudioStreamData() to add more
|
|
* audio to the stream during this call; if needed, the request that triggered
|
|
* this callback will obtain the new data immediately.
|
|
*
|
|
* The callback's `approx_request` argument is roughly how many bytes of
|
|
* _unconverted_ data (in the stream's input format) is needed by the caller,
|
|
* although this may overestimate a little for safety. This takes into account
|
|
* how much is already in the stream and only asks for any extra necessary to
|
|
* resolve the request, which means the callback may be asked for zero bytes,
|
|
* and a different amount on each call.
|
|
*
|
|
* The callback is not required to supply exact amounts; it is allowed to
|
|
* supply too much or too little or none at all. The caller will get what's
|
|
* available, up to the amount they requested, regardless of this callback's
|
|
* outcome.
|
|
*
|
|
* Clearing or flushing an audio stream does not call this callback.
|
|
*
|
|
* This function obtains the stream's lock, which means any existing callback
|
|
* (get or put) in progress will finish running before setting the new
|
|
* callback.
|
|
*
|
|
* Setting a NULL function turns off the callback.
|
|
*
|
|
* \param stream the audio stream to set the new callback on.
|
|
* \param callback the new callback function to call when data is added to the
|
|
* stream.
|
|
* \param userdata an opaque pointer provided to the callback for its own
|
|
* personal use.
|
|
* \returns 0 on success, -1 on error. This only fails if `stream` is NULL.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_SetAudioStreamPutCallback
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_SetAudioStreamGetCallback(SDL_AudioStream *stream, SDL_AudioStreamCallback callback, void *userdata);
|
|
|
|
/**
|
|
* Set a callback that runs when data is added to an audio stream.
|
|
*
|
|
* This callback is called _after_ the data is added to the stream, giving the
|
|
* callback the chance to obtain it immediately.
|
|
*
|
|
* The callback can (optionally) call SDL_GetAudioStreamData() to obtain audio
|
|
* from the stream during this call.
|
|
*
|
|
* The callback's `approx_request` argument is how many bytes of _converted_
|
|
* data (in the stream's output format) was provided by the caller, although
|
|
* this may underestimate a little for safety. This value might be less than
|
|
* what is currently available in the stream, if data was already there, and
|
|
* might be less than the caller provided if the stream needs to keep a buffer
|
|
* to aid in resampling. Which means the callback may be provided with zero
|
|
* bytes, and a different amount on each call.
|
|
*
|
|
* The callback may call SDL_GetAudioStreamAvailable to see the total amount
|
|
* currently available to read from the stream, instead of the total provided
|
|
* by the current call.
|
|
*
|
|
* The callback is not required to obtain all data. It is allowed to read less
|
|
* or none at all. Anything not read now simply remains in the stream for
|
|
* later access.
|
|
*
|
|
* Clearing or flushing an audio stream does not call this callback.
|
|
*
|
|
* This function obtains the stream's lock, which means any existing callback
|
|
* (get or put) in progress will finish running before setting the new
|
|
* callback.
|
|
*
|
|
* Setting a NULL function turns off the callback.
|
|
*
|
|
* \param stream the audio stream to set the new callback on.
|
|
* \param callback the new callback function to call when data is added to the
|
|
* stream.
|
|
* \param userdata an opaque pointer provided to the callback for its own
|
|
* personal use.
|
|
* \returns 0 on success, -1 on error. This only fails if `stream` is NULL.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_SetAudioStreamGetCallback
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_SetAudioStreamPutCallback(SDL_AudioStream *stream, SDL_AudioStreamCallback callback, void *userdata);
|
|
|
|
|
|
/**
|
|
* Free an audio stream
|
|
*
|
|
* \param stream The audio stream to free
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_CreateAudioStream
|
|
* \sa SDL_PutAudioStreamData
|
|
* \sa SDL_GetAudioStreamData
|
|
* \sa SDL_GetAudioStreamAvailable
|
|
* \sa SDL_FlushAudioStream
|
|
* \sa SDL_ClearAudioStream
|
|
*/
|
|
extern DECLSPEC void SDLCALL SDL_DestroyAudioStream(SDL_AudioStream *stream);
|
|
|
|
|
|
/**
|
|
* Convenience function for straightforward audio init for the common case.
|
|
*
|
|
* If all your app intends to do is provide a single source of PCM audio, this
|
|
* function allows you to do all your audio setup in a single call.
|
|
*
|
|
* This is intended to be a clean means to migrate apps from SDL2.
|
|
*
|
|
* This function will open an audio device, create a stream and bind it.
|
|
* Unlike other methods of setup, the audio device will be closed when this
|
|
* stream is destroyed, so the app can treat the returned SDL_AudioStream as
|
|
* the only object needed to manage audio playback.
|
|
*
|
|
* Also unlike other functions, the audio device begins paused. This is to map
|
|
* more closely to SDL2-style behavior, and since there is no extra step here
|
|
* to bind a stream to begin audio flowing. The audio device should be resumed
|
|
* with SDL_ResumeAudioDevice(SDL_GetAudioStreamDevice(stream));
|
|
*
|
|
* This function works with both playback and capture devices.
|
|
*
|
|
* The `spec` parameter represents the app's side of the audio stream. That
|
|
* is, for recording audio, this will be the output format, and for playing
|
|
* audio, this will be the input format.
|
|
*
|
|
* If you don't care about opening a specific audio device, you can (and
|
|
* probably _should_), use SDL_AUDIO_DEVICE_DEFAULT_OUTPUT for playback and
|
|
* SDL_AUDIO_DEVICE_DEFAULT_CAPTURE for recording.
|
|
*
|
|
* One can optionally provide a callback function; if NULL, the app is
|
|
* expected to queue audio data for playback (or unqueue audio data if
|
|
* capturing). Otherwise, the callback will begin to fire once the device is
|
|
* unpaused.
|
|
*
|
|
* \param devid an audio device to open, or SDL_AUDIO_DEVICE_DEFAULT_OUTPUT or
|
|
* SDL_AUDIO_DEVICE_DEFAULT_CAPTURE.
|
|
* \param spec the audio stream's data format. Required.
|
|
* \param callback A callback where the app will provide new data for
|
|
* playback, or receive new data for capture. Can be NULL, in
|
|
* which case the app will need to call SDL_PutAudioStreamData
|
|
* or SDL_GetAudioStreamData as necessary.
|
|
* \param userdata App-controlled pointer passed to callback. Can be NULL.
|
|
* Ignored if callback is NULL.
|
|
* \returns an audio stream on success, ready to use. NULL on error; call
|
|
* SDL_GetError() for more information. When done with this stream,
|
|
* call SDL_DestroyAudioStream to free resources and close the
|
|
* device.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_GetAudioStreamDevice
|
|
* \sa SDL_ResumeAudioDevice
|
|
*/
|
|
extern DECLSPEC SDL_AudioStream *SDLCALL SDL_OpenAudioDeviceStream(SDL_AudioDeviceID devid, const SDL_AudioSpec *spec, SDL_AudioStreamCallback callback, void *userdata);
|
|
|
|
|
|
/**
|
|
* A callback that fires when data is about to be fed to an audio device.
|
|
*
|
|
* This is useful for accessing the final mix, perhaps for writing a
|
|
* visualizer or applying a final effect to the audio data before playback.
|
|
*
|
|
* \sa SDL_SetAudioDevicePostmixCallback
|
|
*/
|
|
typedef void (SDLCALL *SDL_AudioPostmixCallback)(void *userdata, const SDL_AudioSpec *spec, float *buffer, int buflen);
|
|
|
|
/**
|
|
* Set a callback that fires when data is about to be fed to an audio device.
|
|
*
|
|
* This is useful for accessing the final mix, perhaps for writing a
|
|
* visualizer or applying a final effect to the audio data before playback.
|
|
*
|
|
* The buffer is the final mix of all bound audio streams on an opened device;
|
|
* this callback will fire regularly for any device that is both opened and
|
|
* unpaused. If there is no new data to mix, either because no streams are
|
|
* bound to the device or all the streams are empty, this callback will still
|
|
* fire with the entire buffer set to silence.
|
|
*
|
|
* This callback is allowed to make changes to the data; the contents of the
|
|
* buffer after this call is what is ultimately passed along to the hardware.
|
|
*
|
|
* The callback is always provided the data in float format (values from -1.0f
|
|
* to 1.0f), but the number of channels or sample rate may be different than
|
|
* the format the app requested when opening the device; SDL might have had to
|
|
* manage a conversion behind the scenes, or the playback might have jumped to
|
|
* new physical hardware when a system default changed, etc. These details may
|
|
* change between calls. Accordingly, the size of the buffer might change
|
|
* between calls as well.
|
|
*
|
|
* This callback can run at any time, and from any thread; if you need to
|
|
* serialize access to your app's data, you should provide and use a mutex or
|
|
* other synchronization device.
|
|
*
|
|
* All of this to say: there are specific needs this callback can fulfill, but
|
|
* it is not the simplest interface. Apps should generally provide audio in
|
|
* their preferred format through an SDL_AudioStream and let SDL handle the
|
|
* difference.
|
|
*
|
|
* This function is extremely time-sensitive; the callback should do the least
|
|
* amount of work possible and return as quickly as it can. The longer the
|
|
* callback runs, the higher the risk of audio dropouts or other problems.
|
|
*
|
|
* This function will block until the audio device is in between iterations,
|
|
* so any existing callback that might be running will finish before this
|
|
* function sets the new callback and returns.
|
|
*
|
|
* Setting a NULL callback function disables any previously-set callback.
|
|
*
|
|
* \param devid The ID of an opened audio device.
|
|
* \param callback A callback function to be called. Can be NULL.
|
|
* \param userdata App-controlled pointer passed to callback. Can be NULL.
|
|
* \returns zero on success, -1 on error; call SDL_GetError() for more
|
|
* information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_SetAudioPostmixCallback(SDL_AudioDeviceID devid, SDL_AudioPostmixCallback callback, void *userdata);
|
|
|
|
|
|
/**
|
|
* 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.
|
|
*
|
|
* 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 return value is zero 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.
|
|
*
|
|
* It's necessary to use SDL_free() 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 function 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 SDL_TRUE, calls SDL_RWclose() on `src` before returning,
|
|
* even in the case of an error
|
|
* \param spec A pointer to an SDL_AudioSpec that will be set to the WAVE
|
|
* data's format details on successful return
|
|
* \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 0. `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 -1 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_free() to dispose of it.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_free
|
|
* \sa SDL_LoadWAV
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, SDL_bool freesrc,
|
|
SDL_AudioSpec * spec, Uint8 ** audio_buf,
|
|
Uint32 * audio_len);
|
|
|
|
/**
|
|
* Loads a WAV from a file path.
|
|
*
|
|
* This is a convenience function that is effectively the same as:
|
|
*
|
|
* ```c
|
|
* SDL_LoadWAV_RW(SDL_RWFromFile(path, "rb"), 1, spec, audio_buf, audio_len);
|
|
* ```
|
|
*
|
|
* Note that in SDL2, this was a preprocessor macro and not a real function.
|
|
*
|
|
* \param path The file path of the WAV file to open.
|
|
* \param spec A pointer to an SDL_AudioSpec that will be set to the WAVE
|
|
* data's format details on successful return.
|
|
* \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 0. `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 -1 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_free() to dispose of it.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_free
|
|
* \sa SDL_LoadWAV_RW
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_LoadWAV(const char *path, SDL_AudioSpec * spec,
|
|
Uint8 ** audio_buf, Uint32 * audio_len);
|
|
|
|
|
|
|
|
#define SDL_MIX_MAXVOLUME 128
|
|
|
|
/**
|
|
* 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
|
|
* \returns 0 on success or a negative error code on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_MixAudioFormat(Uint8 * dst,
|
|
const Uint8 * src,
|
|
SDL_AudioFormat format,
|
|
Uint32 len, int volume);
|
|
|
|
/**
|
|
* Convert some audio data of one format to another format.
|
|
*
|
|
* Please note that this function is for convenience, but should not be used
|
|
* to resample audio in blocks, as it will introduce audio artifacts on the
|
|
* boundaries. You should only use this function if you are converting audio
|
|
* data in its entirety in one call. If you want to convert audio in smaller
|
|
* chunks, use an SDL_AudioStream, which is designed for this situation.
|
|
*
|
|
* Internally, this function creates and destroys an SDL_AudioStream on each
|
|
* use, so it's also less efficient than using one directly, if you need to
|
|
* convert multiple times.
|
|
*
|
|
* \param src_spec The format details of the input audio
|
|
* \param src_data The audio data to be converted
|
|
* \param src_len The len of src_data
|
|
* \param dst_spec The format details of the output audio
|
|
* \param dst_data Will be filled with a pointer to converted audio data,
|
|
* which should be freed with SDL_free(). On error, it will be
|
|
* NULL.
|
|
* \param dst_len Will be filled with the len of dst_data
|
|
* \returns 0 on success or a negative error code on failure; call
|
|
* SDL_GetError() for more information.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*
|
|
* \sa SDL_CreateAudioStream
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_ConvertAudioSamples(const SDL_AudioSpec *src_spec,
|
|
const Uint8 *src_data,
|
|
int src_len,
|
|
const SDL_AudioSpec *dst_spec,
|
|
Uint8 **dst_data,
|
|
int *dst_len);
|
|
|
|
|
|
/**
|
|
* Get the appropriate memset value for silencing an audio format.
|
|
*
|
|
* The value returned by this function can be used as the second argument to
|
|
* memset (or SDL_memset) to set an audio buffer in a specific format to
|
|
* silence.
|
|
*
|
|
* \param format the audio data format to query.
|
|
* \returns A byte value that can be passed to memset.
|
|
*
|
|
* \threadsafety It is safe to call this function from any thread.
|
|
*
|
|
* \since This function is available since SDL 3.0.0.
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_GetSilenceValueForFormat(SDL_AudioFormat format);
|
|
|
|
|
|
/* Ends C function definitions when using C++ */
|
|
#ifdef __cplusplus
|
|
}
|
|
#endif
|
|
#include <SDL3/SDL_close_code.h>
|
|
|
|
#endif /* SDL_audio_h_ */
|