tomato/include/SDL3/SDL_audio.h
Green Sky 644725478f Squashed 'external/sdl/SDL/' changes from ec0042081..399bc709b
399bc709b build-scripts.pl: Added add-source-to-projects.pl
ac6827187 Visual-WinRT: dos2unix the project files to match other Visual Studio targets.
34719cba9 Fixed crash in hid_init() if the HIDDeviceManager isn't available
2e92e94eb Make sure we update device->sample_frames in SDL_AudioDeviceFormatChangedAlreadyLocked()
9964e5c5b wayland: Don't retrieve the drag offer strings with every pointer motion event
bac7eeaaa Added missing include
a541e2ac1 audio: Change a few SDL_memcpy calls to SDL_copyp.
54125c140 audio: Only update bound audiostreams' formats when necessary.
e0b0f9a36 testaudio: Fix mouseover testing.
2f3deec24 wayland: Don't process drag & drop events from surfaces not owned by SDL
42bdced05 events: Log file drop position events and print the pointer coordinates
c10d93d3a wayland: Replace magic constant with define
500852153 emscripten: Restore compatibility with existing emsdk releases.
953b55dd6 Use EM_ASM_PTR when the return value is a pointer
a4541a255 audio: SDL_GetAudioStreamQueued now returns bytes, not frames.
703aefbce Sync SDL3 wiki -> header
99421b64d linux: Add portal drag and drop
952c5059b Remove stray Â
eebd5d18a linux: Handle upower's UP_DEVICE_STATE_PENDING_CHARGE, PENDING_DISCHARGE
f8fdb20d8 audio: Destroy all existing SDL_AudioStreams on shutdown.
62d445997 audio: Removed declarations of functions that don't exist anymore.
b656720bc loopwave: Use SDL_GetAudioStreamQueued() for more accurate results.
34b931f7e audio: Added SDL_GetAudioStreamQueued
23206b9e3 audio: Added SDL_EVENT_AUDIO_DEVICE_FORMAT_CHANGED
c7e6d7a1f audio: Changed debug logging output.
87ec6acf2 audio: Added a FIXME
ac88ffb7e audio: don't allocate buffer in SDL_SetAudioPostmixCallback for NULL callback.
2a950f6ae audio: Replace some SDL_memcpy calls with SDL_copyp.
0dc0434a3 audio: Fixed race condition in subsystem shutdown.
23f60203a audio: precalculate if we can use simple copies instead of the full mixer.
36b0f1141 audio: Optimize setting device formats during audio thread iteration.
4c3e84897 testspriteminimal: make standalone by embedding icon.bmp
2a01f9dcb tests: plug leaks when running with --trackmem
f42bbeca2 SDL_test: track stack frames of allocations on Windows
12c0be028 SDL_test: clear text cache on exit event
b4bfb1831 SDL_test: free state before logging allocations
248b1edd3 SDL_test: destroy windows in SDL_CommonQuit
98da2dd30 SDL_test: don't warn about expected allocations when running with --trackmem
6a381567b Support audio rate conversion up to 384KHz
b2b548a1f Don't hang if IAudioRenderClient_GetBuffer() fails indefinitely
a3a5e1728 Fixed build warning '=': conversion from 'Uint32' to 'Uint16', possible loss of data
6d3e21c27 Fixed android build warnings
fca2f5318 Fixed warning: this function declaration is not a prototype
a72dfa6a5 Fixed sensor timestamp units for third-party PS5 controllers
f6756047a Fixed error: array subscript 2 is above array bounds of ‘const Uint8[2]’
7059a55cc Fixed sensor timestamp calculation for third-party PS5 controllers
c0443e5d1 Fixed crash in SDL_IMMDevice_FindByDevID()
fde8499f6 Use around 20ms for the audio buffer size
e5739d7d1 video: Remove SDL_GetFocusWindow()
39c2f9737 Fix NULL dereference in SDL_OpenAudio
9a23d0e3f Added new audio files to the Xcode project
a62e62f97 Refactored SDL_audiocvt.c
31229fd47 include: Added a note about SDL's iOS app delegate functions.
65aaf3a9a x11: Always update clipboard owner
f622f21e6 Fixed build
5774c9638 Prefer hidraw over libusb when libusb whitelisting is not enabled
9301f7ace hidapi/libusb: only enumerate each interface once
859dc14ad Replaced SDL_GetGamepadBindForAxis() and SDL_GetGamepadBindForButton() with SDL_GetGamepadBindings()
9e50048ab Revert "Removed SDL_GamepadBinding from the API"
9f17d1a9d Don't reference the same function in "see also"
86505ea63 fix SDL_AudioStreamCallback documentation
d885d5c31 Sync SDL3 wiki -> header
2f43f7bc5 audio: Allow querying of device buffer size.
cf9572113 audio: Added a hint to let apps force device buffer size.
47d8c77c6 audio: Choose better default sample frame counts.
8b26e95f9 audio: Change SDL_AudioStreamCallback
9da34e8fb docs: Updated README-emscripten.md.
fd1c54a00 detect fanatec steering wheels
cb4414608 docs: Whoops, this got added by the wiki bridge by accident!
cd633b9a8 Renamed SDL_IsAudioDevicePaused() to SDL_AudioDevicePaused()
c6cad07fa Sync SDL3 wiki -> header
a6e52f9e4 Sync SDL3 wiki -> header
2de2e9d03 Fix flickering of window when using desktop-fullscreen and borderless window on multiple monitors on Linux.  Closes #8186.
723835d16 Windows: fix for client rect resizing larger each time we came from exclusive fullscreen -> windowed on a monitor with HiDPI set.  The problem was we were using the monitor DPI rather than the window DPI so AdjustWindowRectExForDpi was giving us an incorrect size which would be too large for the client rect.  Closes #8237.
ce27363df wikiheaders: Sort undocumented functions.
e22282b09 Added README about transparent windows in Win32
1d1c6e630 Turn off COREAUDIO debug logging by default
52efefca0 wayland: Fix drag offer leak
3a992af44 audio: Added a postmix callback to logical devices.
7207bdce5 render: Enable clipping for zero-sized rectangles
22d81fb3e cmake: use MSVC_RUNTIME_LIBRARY to force MT
a2e17852d cmake: make sure SDL_GetPrefPath is run before testfilesystem
2fb266e0a ci: run tests in parallel
ad1313e75 testaudio: Patched to compile.
5747ddc01 testaudio: Clean up some messy memory management.
fafbea1ce audio: Move internal float32 mixing to a simplified function
116b0ec97 include: minor tweak to audio API documentation
fb1377035 include: Replaced old Bugzilla URL.
38c8fc05c audio: Remove ChooseMixStrategy.
b00cbd76a wikiheaders.pl: create Unsupported.md file with list of functions undocumented in either the headers or the wiki
37e1fc3b5 wayland: Ensure that the toplevel window is recreated when switching decoration modes
f2ca9a615 Added SDL_AUDIO_FRAMESIZE
53122593f Added SDL_AUDIO_BYTESIZE
544351c98 Sync SDL3 wiki -> header
2e7d2b94e Clarify that SDL_BlitSurface() ignores the width and height in dstrect
a2c1984d3 Detect Simagic wheel bases as wheels (#8198)
1d8dfbb22 avoid type redefinition errors after PR/8181
266b91d2f Detect Logitech G923 Playstation as wheel G923 have two different versions - Xbox version is already present in the wheel list, but not the PS version.
cde67ea49 Detect Logitech PRO Racing Wheel for Xbox (PC mode) as wheel Logitech PRO Racing Wheel have two different versions - for Playstation and Xbox. Vendor + Product ID for Playstation version already present in SDL sources, but not an Xbox version
3a932141e Restore audio format binary compatibility with SDL 2.0
e85206ffd wikiheaders.pl: add --rev= option to pass revision string
233789b0d Audio types have the same naming convention as other SDL endian types, e.g. [S|U][BITS][LE|BE]
36b5f3e35 Sync SDL3 wiki -> header
0e552761b Renamed AudioStreamSpeed to AudioStreamFrequencyRatio
47bcb078f Fixed some incorrect SDL_AUDIO_F32 uses
2833f2e7b Fixed OOB access in audio_convertAccuracy test
8387fae69 Sync SDL3 wiki -> header
832181345 docs: Add note about Wayland application icons
825d34475 Make sure that the same timestamp is used for all PS5 events from the same packet
9c1430324 Removed SDL_dataqueue
28b28bd8f Added audio_formatChange test
a59152688 Try and avoid overflow when handling very large audio streams
5394a805f Improved testaudiostreamdynamicresample
e55844274 Added SDL_(Get|Set)AudioStreamSpeed
43c3c5736 Track the formats of data in an SDL_AudioStream
337fed3df Tweaked ResampleFrame_SSE Use _mm_unpack(lo|hi)_ps instead of _mm_shuffle_ps
fd7cd91dc audio: Mix multiple streams in float32 to prevent clipping.
9097573e3 audio: Choose a mixing strategy on each iteration.
bbe2e012a Don't provide the SDL3 header path
c17a35f09 Fixed typo
4f72255eb Fixed README.md link
e0ab59754 Simplified SDL_main.h migration notes
d44bde61e Added SDL migration information to the top level README.md
6ff31e10c metal: Add hint to select low power device instead of the default one (#8182)
8a8aed477 Make sure that we process touch events that position the mouse
f84c87f20 Sync SDL3 wiki -> header
a7eea9997 macOS: Don't raise the parent top-level window when raising a child window, only raise the child window to the top of the parent
a5e721479 Add SDL_WINDOW_NOT_FOCUSABLE flag to set that the window should not be able to gain key focus
b385dc3b6 n3dsaudio: Patched to compile.
4e0c7c91f audio: PlayDevice() should return an error code.
a94d724f1 wayland: add SDL_VIDEO_DRIVER_WAYLAND_DYNAMIC_EGL
da5d93d3d wayland: don't define SDL_VIDEO_DRIVER_WAYLAND_DYNAMIC_* macro's
f002f7d12 ci: build emscripten with Debug buid type
3699b12ed audio: Fixed some "is_*" variables to be cleaner and/or more specific.
2471d8cc2 audio: Fixed logic error in SDL_OpenAudioDeviceStream.
1b03a2430 testsurround: fix order of arguments of callback
82db2b58f Renamed audio stream callback and moved the userdata parameter first
5bdad5210 Sync SDL3 wiki -> header
58c859f64 audio: Rename SDL_GetAudioStreamBinding to SDL_GetAudioStreamDevice.
efd2023a7 audio: Fixed documentation.
1e775e0ee audio: Replace SDL_CreateAndBindAudioStream with SDL_OpenAudioDeviceStream.
bd088c2f9 Revert "Clarify whether an audio function expects a physical or logical device ID"
82e481b52 Added --randmem test parameter
ea68bb802 Add some additional checks to audio_convertAudio
f8286df16 Fixed ResampleFrame_SSE doing unnecessary work
b1d63be53 Fixed audio_resampleLoss test
c191d6c30 Better Win32 transparent window support
923d612ca hidapi: sync macOS code with mainstream.
363f4fa9c avoid type redefinition errors after commit ee806597b9.
615824a80 Updated documentation now that SDL_GetAudioDevices() has been split into separate functions for output and capture devices
506a133d8 Clarify whether an audio function expects a physical or logical device ID
3b1d1e4e3 hidapi: sync the hidraw changes with mainstream
f617918e0 cmake: check linkage to libusb too, instead of libusb.h presence only.
041dbd6b5 Fixed GetResamplerAvailableOutputFrames Non-euclidean division is a pain
b49d0a607 x11: Avoid including full Vulkan headers.
4d2f9f3a3 yuv_rgb: Comment out unused code.
3c3486e2a wayland: Don't include full Vulkan headers when not necessary.
f066bbe98 x11: Don't include system headers twice.
d86d02bbb updated dynapi after SDL_GDKGetDefaultUser addition
4355f9cec Fixed warning C4389: '!=': signed/unsigned mismatch
5755de07a Fixed build warnings
0f80d47bb Fixed thread-safety warning
ee806597b Removed SDL_vulkan_internal.h from SDL_sysvideo.h
34860b932 Fixed testautomation --filter pixels_allocFreeFormat
6f8a6a31c gdk: GetBasePath should be a UTF8 version of Win32 GetBasePath
e30e5c77e Sync SDL3 wiki -> header
c0cd8c814 gdk: Add SDL_GDKGetDefaultUser, SDL_GetPrefPath implementation
106abce69 Refactored GetAudioStreamDataInternal buffer handling The final conversion step should now always go straight into the output buffer.
e44f54ec5 Avoid using hex-floats
5b696996c Added ResampleFrame_SSE
958b3cfae Tweaked and enabled audio_convertAudio test
7dbb9b65b audio_convertAccuracy: Shuffle the data in case of a bad SIMD implementation
f6a4080ff audio_resampleLoss: Add support for multiple channels
4f894e748 audio_resampleLoss: SDL_GetAudioStreamData now returns the correct length
ab83f75bb Make sure GetAudioStreamDataInternal is called with a valid length
6a73f74b6 Rebuild full ResamplerFilter (left wing + right wing) at runtime
0c15ce006 Add a missing int cast
b74ee86b1 Optimized ResampleAudio, with special cases for 1 and 2 channels This would also benefit from some SIMD, since it's just a bunch of multiply-adds
fba6e1e3d Removed ResamplerFilterDifference It takes 1 extra multiply to calculate the correct interpolation, but I think the improvement in cache locality (and binary size) outweighs that.
9f7a22fa4 Removed 64-bit handling from AudioConvertByteswap
1f5327a9f Removed future_buffer, left_padding, and right_padding from SDL_AudioStream
71ad52d6d Lowered SDL_GetAudioStreamData to 32 KB No particular reason for this number, but 1 MB was a bit silly
69aec8c91 Fixed the report format for the Razer Wolverine V2 Pro
7c2669c9d Accept key events from any source
1e9d31448 Updated to Android minSdkVersion 19 and targetSdkVersion 34
8924d0d92 Added missing function prototype for SDL_WriteS64BE()
845f3c745 Fixed mismatch between stdlib calloc() and SDL free()
fb7921173 emscriptenaudio: Fire the capture silence_callback at an interval.
5191b2054 emscriptenaudio: Don't bother undefining things about to be unreachable.
fd75a4ca0 emscriptenaudio: Deal with blocked audio devices better.
981b8a337 emscriptenaudio: Remove unnecessary functions.
c7588e426 Transparent window for Win32 + OpenGL (#8143)
f9581178d cmake: fixed a typo.
e6c878824 Fixed ResampleAudio interpolation factor calculation
498363863 Misc audio tweaks/cleanup
72d9d53de Invert the inner ResampleAudio loops to avoid doing unnecessary work
88123a510 The history buffer should always have the maximum possible padding frames
96e47f165 Clamp results of GetResampler(AvailableOutput|NeededInput)Frames
d2b9c8b80 Fixed maths in testaudiostreamdynamicresample (and just show the actual scale)
14e38b17d Removed assertions from inner ResampleAudio loop
9d413dfdc The history buffer doesn't need to be so large
2788e848f Allow resampling less than 1 frame of input
383084e0a Pre-calculate resampling rate, and use it instead of .freq in most places
40a6a445c Update resample_offset inside ResampleAudio
47fea7f06 Used fixed-point arithmetic in ResampleAudio
7bb4e806e Clear resample_offset in SDL_ClearAudioStream, not SetAudioStreamFormat Not entirely sure if ClearAudioStream is the right place, but SetAudioStreamFormat was the wrong place
b9541b9ea Improved ResampleAudio * filterindex2 was off-by-one * Generate ResamplerFilter using doubles * Transpose ResamplerFilter to improve access patterns
cdaa19869 Track offset within the current sample when resampling
d60ebb06d mouse: Ensure that the dummy default cursor is removed from the cursor list
e58c2731f mouse: Free the default cursor when destroyed
789ce17e1 audio: Don't resample in chunks for now.
cbab33482 audio: Don't call SDL_AudioStream callbacks for empty data sets.
3e1ae0c86 Clearified the libusb whitelist default logic
f4520821e Removed some unnecessary integer casts
0989b7e86 Avoid using designated initializers
c6c1e673c Optimized SDL_Convert_*_to_*_Scalar
f97b920b3 Optimized SDL_Convert_*_to_*_SSE2 Some of the SDL_Convert_F32_to_*_SSE2 do not explicitly clamp the input, but instead rely on saturating casts. Inputs very far outside the valid [-1.0, 1.0] range may produce an incorrect result, but I believe that is an acceptable trade-off.
300d1ec3e Added audio_convertAccuracy test
32cecc2ea Fixed assertion in audio_convertAudio
33f11e21e Removed assertions in AudioConvert(To|From)Float
c2f388fd8 cmake: add SDL_HIDAPI_LIBUSB_SHARED option + test on ci
371cc2d17 wayland: Remove unnecessary flag and state settings
fe85e6e75 cocoa: Send a maximized event instead of restored if a deminiaturized window is zoomed
ddddcb78c cocoa: Use the close method to hide a miniaturized window
be8c42cfd Clarify that a window being 'hidden' means that it is unmapped/ordered out
a44338cbc Fix typo in SDL_audiocvt.c
f464eb2c5 SDL_hidapi.c: change 'use_libusb_whitelist_default' into a macro.
6607a3cfa Disable cache in python http server
181d5d285 hidapi: Enable libusb support by default.
f0f15e365 hidapi: Use a whitelist for libusb when other backends are available
c3f7a7dc4 Convert audio using SDL_AUDIO_F32SYS format instead of SDL_AUDIO_F32
796713b9d xxd.py: always write \n line endings
723bcd0a8 SDL_TriggerBreakppoint for riscv arch (both 32/64) version.

git-subtree-dir: external/sdl/SDL
git-subtree-split: 399bc709b7485bab57880f8261f826f29dc0d7b2
2023-09-23 18:45:49 +02:00

1483 lines
58 KiB
C

/*
Simple DirectMedia Layer
Copyright (C) 1997-2023 Sam Lantinga <slouken@libsdl.org>
This software is provided 'as-is', without any express or implied
warranty. In no event will the authors be held liable for any damages
arising from the use of this software.
Permission is granted to anyone to use this software for any purpose,
including commercial applications, and to alter it and redistribute it
freely, subject to the following restrictions:
1. The origin of this software must not be misrepresented; you must not
claim that you wrote the original software. If you use this software
in a product, an acknowledgment in the product documentation would be
appreciated but is not required.
2. Altered source versions must be plainly marked as such, and must not be
misrepresented as being the original software.
3. This notice may not be removed or altered from any source distribution.
*/
/**
* \file SDL_audio.h
*
* \brief Audio functionality for the SDL library.
*/
#ifndef SDL_audio_h_
#define SDL_audio_h_
#include <SDL3/SDL_stdinc.h>
#include <SDL3/SDL_error.h>
#include <SDL3/SDL_endian.h>
#include <SDL3/SDL_mutex.h>
#include <SDL3/SDL_thread.h>
#include <SDL3/SDL_rwops.h>
#include <SDL3/SDL_begin_code.h>
/* Set up for C function definitions, even when using C++ */
#ifdef __cplusplus
extern "C" {
#endif
/*
* For multi-channel audio, the default SDL channel mapping is:
* 2: FL FR (stereo)
* 3: FL FR LFE (2.1 surround)
* 4: FL FR BL BR (quad)
* 5: FL FR LFE BL BR (4.1 surround)
* 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR)
* 7: FL FR FC LFE BC SL SR (6.1 surround)
* 8: FL FR FC LFE BL BR SL SR (7.1 surround)
*/
/**
* \brief Audio format flags.
*
* These are what the 16 bits in SDL_AudioFormat currently mean...
* (Unspecified bits are always zero).
*
* \verbatim
++-----------------------sample is signed if set
||
|| ++-----------sample is bigendian if set
|| ||
|| || ++---sample is float if set
|| || ||
|| || || +---sample bit size---+
|| || || | |
15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
\endverbatim
*
* There are macros in SDL 2.0 and later to query these bits.
*/
typedef Uint16 SDL_AudioFormat;
/**
* \name Audio flags
*/
/* @{ */
#define SDL_AUDIO_MASK_BITSIZE (0xFF)
#define SDL_AUDIO_MASK_FLOAT (1<<8)
#define SDL_AUDIO_MASK_BIG_ENDIAN (1<<12)
#define SDL_AUDIO_MASK_SIGNED (1<<15)
#define SDL_AUDIO_BITSIZE(x) ((x) & SDL_AUDIO_MASK_BITSIZE)
#define SDL_AUDIO_BYTESIZE(x) (SDL_AUDIO_BITSIZE(x) / 8)
#define SDL_AUDIO_ISFLOAT(x) ((x) & SDL_AUDIO_MASK_FLOAT)
#define SDL_AUDIO_ISBIGENDIAN(x) ((x) & SDL_AUDIO_MASK_BIG_ENDIAN)
#define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x))
#define SDL_AUDIO_ISSIGNED(x) ((x) & SDL_AUDIO_MASK_SIGNED)
#define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x))
#define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x))
/**
* \name Audio format flags
*
* Defaults to LSB byte order.
*/
/* @{ */
#define SDL_AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
#define SDL_AUDIO_S8 0x8008 /**< Signed 8-bit samples */
#define SDL_AUDIO_S16LE 0x8010 /**< Signed 16-bit samples */
#define SDL_AUDIO_S16BE 0x9010 /**< As above, but big-endian byte order */
/* @} */
/**
* \name int32 support
*/
/* @{ */
#define SDL_AUDIO_S32LE 0x8020 /**< 32-bit integer samples */
#define SDL_AUDIO_S32BE 0x9020 /**< As above, but big-endian byte order */
/* @} */
/**
* \name float32 support
*/
/* @{ */
#define SDL_AUDIO_F32LE 0x8120 /**< 32-bit floating point samples */
#define SDL_AUDIO_F32BE 0x9120 /**< As above, but big-endian byte order */
/* @} */
/**
* \name Native audio byte ordering
*/
/* @{ */
#if SDL_BYTEORDER == SDL_LIL_ENDIAN
#define SDL_AUDIO_S16 SDL_AUDIO_S16LE
#define SDL_AUDIO_S32 SDL_AUDIO_S32LE
#define SDL_AUDIO_F32 SDL_AUDIO_F32LE
#else
#define SDL_AUDIO_S16 SDL_AUDIO_S16BE
#define SDL_AUDIO_S32 SDL_AUDIO_S32BE
#define SDL_AUDIO_F32 SDL_AUDIO_F32BE
#endif
/* @} */
/* @} *//* Audio flags */
/**
* SDL Audio Device instance IDs.
*/
typedef Uint32 SDL_AudioDeviceID;
#define SDL_AUDIO_DEVICE_DEFAULT_OUTPUT ((SDL_AudioDeviceID) 0xFFFFFFFF)
#define SDL_AUDIO_DEVICE_DEFAULT_CAPTURE ((SDL_AudioDeviceID) 0xFFFFFFFE)
typedef struct SDL_AudioSpec
{
SDL_AudioFormat format; /**< Audio data format */
int channels; /**< Number of channels: 1 mono, 2 stereo, etc */
int freq; /**< sample rate: sample frames per second */
} SDL_AudioSpec;
/* Calculate the size of each audio frame (in bytes) */
#define SDL_AUDIO_FRAMESIZE(x) (SDL_AUDIO_BYTESIZE((x).format) * (x).channels)
/* SDL_AudioStream is an audio conversion interface.
- 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.
- It can handle input/output format changes on the fly.
- You push data as you have it, and pull it when you need it
- It can also function as a basic audio data queue even if you
just have sound that needs to pass from one place to another.
- You can hook callbacks up to them when more data is added or
requested, to manage data on-the-fly.
*/
struct SDL_AudioStream; /* this is opaque to the outside world. */
typedef struct SDL_AudioStream SDL_AudioStream;
/* Function prototypes */
/**
* \name Driver discovery functions
*
* These functions return the list of built in audio drivers, in the
* order that they are normally initialized by default.
*/
/* @{ */
/**
* Use this function to get the number of built-in audio drivers.
*
* This function returns a hardcoded number. This never returns a negative
* value; if there are no drivers compiled into this build of SDL, this
* function returns zero. The presence of a driver in this list does not mean
* it will function, it just means SDL is capable of interacting with that
* interface. For example, a build of SDL might have esound support, but if
* there's no esound server available, SDL's esound driver would fail if used.
*
* By default, SDL tries all drivers, in its preferred order, until one is
* found to be usable.
*
* \returns the number of built-in audio drivers.
*
* \threadsafety It is safe to call this function from any thread.
*
* \since This function is available since SDL 3.0.0.
*
* \sa SDL_GetAudioDriver
*/
extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
/**
* Use this function to get the name of a built in audio driver.
*
* The list of audio drivers is given in the order that they are normally
* initialized by default; the drivers that seem more reasonable to choose
* first (as far as the SDL developers believe) are earlier in the list.
*
* The names of drivers are all simple, low-ASCII identifiers, like "alsa",
* "coreaudio" or "xaudio2". These never have Unicode characters, and are not
* meant to be proper names.
*
* \param index the index of the audio driver; the value ranges from 0 to
* SDL_GetNumAudioDrivers() - 1
* \returns the name of the audio driver at the requested index, or NULL if an
* invalid index was specified.
*
* \threadsafety It is safe to call this function from any thread.
*
* \since This function is available since SDL 3.0.0.
*
* \sa SDL_GetNumAudioDrivers
*/
extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
/* @} */
/**
* Get the name of the current audio driver.
*
* The returned string points to internal static memory and thus never becomes
* invalid, even if you quit the audio subsystem and initialize a new driver
* (although such a case would return a different static string from another
* call to this function, of course). As such, you should not modify or free
* the returned string.
*
* \returns the name of the current audio driver or NULL if no driver has been
* initialized.
*
* \threadsafety It is safe to call this function from any thread.
*
* \since This function is available since SDL 3.0.0.
*/
extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
/**
* Get a list of currently-connected audio output devices.
*
* This returns of list of available devices that play sound, perhaps to
* speakers or headphones ("output" devices). If you want devices that record
* audio, like a microphone ("capture" devices), use
* SDL_GetAudioCaptureDevices() instead.
*
* This only returns a list of physical devices; it will not have any device
* IDs returned by SDL_OpenAudioDevice().
*
* \param count a pointer filled in with the number of devices returned
* \returns a 0 terminated array of device instance IDs which should be freed
* with SDL_free(), or NULL on error; call SDL_GetError() for more
* details.
*
* \threadsafety It is safe to call this function from any thread.
*
* \since This function is available since SDL 3.0.0.
*
* \sa SDL_OpenAudioDevice
* \sa SDL_GetAudioCaptureDevices
*/
extern DECLSPEC SDL_AudioDeviceID *SDLCALL SDL_GetAudioOutputDevices(int *count);
/**
* Get a list of currently-connected audio capture devices.
*
* This returns of list of available devices that record audio, like a
* microphone ("capture" devices). If you want devices that play sound,
* perhaps to speakers or headphones ("output" devices), use
* SDL_GetAudioOutputDevices() instead.
*
* This only returns a list of physical devices; it will not have any device
* IDs returned by SDL_OpenAudioDevice().
*
* \param count a pointer filled in with the number of devices returned
* \returns a 0 terminated array of device instance IDs which should be freed
* with SDL_free(), or NULL on error; call SDL_GetError() for more
* details.
*
* \threadsafety It is safe to call this function from any thread.
*
* \since This function is available since SDL 3.0.0.
*
* \sa SDL_OpenAudioDevice
* \sa SDL_GetAudioOutputDevices
*/
extern DECLSPEC SDL_AudioDeviceID *SDLCALL SDL_GetAudioCaptureDevices(int *count);
/**
* Get the human-readable name of a specific audio device.
*
* The string returned by this function is UTF-8 encoded. The caller should
* call SDL_free on the return value when done with it.
*
* \param devid the instance ID of the device to query.
* \returns the name of the audio device, or NULL on error.
*
* \threadsafety It is safe to call this function from any thread.
*
* \since This function is available since SDL 3.0.0.
*
* \sa SDL_GetAudioOutputDevices
* \sa SDL_GetAudioCaptureDevices
* \sa SDL_GetDefaultAudioInfo
*/
extern DECLSPEC char *SDLCALL SDL_GetAudioDeviceName(SDL_AudioDeviceID devid);
/**
* Get the current audio format of a specific audio device.
*
* For an opened device, this will report the format the device is currently
* using. If the device isn't yet opened, this will report the device's
* preferred format (or a reasonable default if this can't be determined).
*
* You may also specify SDL_AUDIO_DEVICE_DEFAULT_OUTPUT or
* SDL_AUDIO_DEVICE_DEFAULT_CAPTURE here, which is useful for getting a
* reasonable recommendation before opening the system-recommended default
* device.
*
* You can also use this to request the current device buffer size. This is
* specified in sample frames and represents the amount of data SDL will feed
* to the physical hardware in each chunk. This can be converted to
* milliseconds of audio with the following equation:
*
* `ms = (int) ((((Sint64) frames) * 1000) / spec.freq);`
*
* Buffer size is only important if you need low-level control over the audio
* playback timing. Most apps do not need this.
*
* \param devid the instance ID of the device to query.
* \param spec On return, will be filled with device details.
* \param sample_frames Pointer to store device buffer size, in sample frames.
* Can be NULL.
* \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_GetAudioDeviceFormat(SDL_AudioDeviceID devid, SDL_AudioSpec *spec, int *sample_frames);
/**
* Open a specific audio device.
*
* You can open both output and capture devices through this function. Output
* devices will take data from bound audio streams, mix it, and send it to the
* hardware. Capture devices will feed any bound audio streams with a copy of
* any incoming data.
*
* An opened audio device starts out with no audio streams bound. To start
* audio playing, bind a stream and supply audio data to it. Unlike SDL2,
* there is no audio callback; you only bind audio streams and make sure they
* have data flowing into them (however, you can simulate SDL2's semantics
* fairly closely by using SDL_OpenAudioDeviceStream instead of this
* function).
*
* If you don't care about opening a specific device, pass a `devid` of either
* `SDL_AUDIO_DEVICE_DEFAULT_OUTPUT` or `SDL_AUDIO_DEVICE_DEFAULT_CAPTURE`. In
* this case, SDL will try to pick the most reasonable default, and may also
* switch between physical devices seamlessly later, if the most reasonable
* default changes during the lifetime of this opened device (user changed the
* default in the OS's system preferences, the default got unplugged so the
* system jumped to a new default, the user plugged in headphones on a mobile
* device, etc). Unless you have a good reason to choose a specific device,
* this is probably what you want.
*
* You may request a specific format for the audio device, but there is no
* promise the device will honor that request for several reasons. As such,
* it's only meant to be a hint as to what data your app will provide. Audio
* streams will accept data in whatever format you specify and manage
* conversion for you as appropriate. SDL_GetAudioDeviceFormat can tell you
* the preferred format for the device before opening and the actual format
* the device is using after opening.
*
* It's legal to open the same device ID more than once; each successful open
* will generate a new logical SDL_AudioDeviceID that is managed separately
* from others on the same physical device. This allows libraries to open a
* device separately from the main app and bind its own streams without
* conflicting.
*
* It is also legal to open a device ID returned by a previous call to this
* function; doing so just creates another logical device on the same physical
* device. This may be useful for making logical groupings of audio streams.
*
* This function returns the opened device ID on success. This is a new,
* unique SDL_AudioDeviceID that represents a logical device.
*
* Some backends might offer arbitrary devices (for example, a networked audio
* protocol that can connect to an arbitrary server). For these, as a change
* from SDL2, you should open a default device ID and use an SDL hint to
* specify the target if you care, or otherwise let the backend figure out a
* reasonable default. Most backends don't offer anything like this, and often
* this would be an end user setting an environment variable for their custom
* need, and not something an application should specifically manage.
*
* When done with an audio device, possibly at the end of the app's life, one
* should call SDL_CloseAudioDevice() on the returned device id.
*
* \param devid the device instance id to open, or
* SDL_AUDIO_DEVICE_DEFAULT_OUTPUT or
* SDL_AUDIO_DEVICE_DEFAULT_CAPTURE for the most reasonable
* default device.
* \param spec the requested device configuration. Can be NULL to use
* reasonable defaults.
* \returns The device ID on success, 0 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_CloseAudioDevice
* \sa SDL_GetAudioDeviceFormat
*/
extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(SDL_AudioDeviceID devid, const SDL_AudioSpec *spec);
/**
* Use this function to pause audio playback on a specified device.
*
* This function pauses audio processing for a given device. Any bound audio
* streams will not progress, and no audio will be generated. Pausing one
* device does not prevent other unpaused devices from running.
*
* Unlike in SDL2, audio devices start in an _unpaused_ state, since an app
* has to bind a stream before any audio will flow. Pausing a paused device is
* a legal no-op.
*
* Pausing a device can be useful to halt all audio without unbinding all the
* audio streams. This might be useful while a game is paused, or a level is
* loading, etc.
*
* 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_ResumeAudioDevice
* \sa SDL_AudioDevicePaused
*/
extern DECLSPEC int SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev);
/**
* Use this function to unpause audio playback on a specified device.
*
* 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 0 on success, or -1 on error.
*
* \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);
/**
* 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. The `samples` member is set to a sane default and all others are
* set to zero.
*
* 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_ */