huginn/third-party/sokol/audio/audio.odin

// machine generated, do not edit

package sokol_audio

/*

    sokol_audio.h -- cross-platform audio-streaming API

    Project URL: https://github.com/floooh/sokol

    Do this:
        #define SOKOL_IMPL or
        #define SOKOL_AUDIO_IMPL
    before you include this file in *one* C or C++ file to create the
    implementation.

    Optionally provide the following defines with your own implementations:

    SOKOL_DUMMY_BACKEND - use a dummy backend
    SOKOL_ASSERT(c)     - your own assert macro (default: assert(c))
    SOKOL_AUDIO_API_DECL- public function declaration prefix (default: extern)
    SOKOL_API_DECL      - same as SOKOL_AUDIO_API_DECL
    SOKOL_API_IMPL      - public function implementation prefix (default: -)

    SAUDIO_RING_MAX_SLOTS           - max number of slots in the push-audio ring buffer (default 1024)
    SAUDIO_OSX_USE_SYSTEM_HEADERS   - define this to force inclusion of system headers on
                                      macOS instead of using embedded CoreAudio declarations

    If sokol_audio.h is compiled as a DLL, define the following before
    including the declaration or implementation:

    SOKOL_DLL

    On Windows, SOKOL_DLL will define SOKOL_AUDIO_API_DECL as __declspec(dllexport)
    or __declspec(dllimport) as needed.

    Link with the following libraries:

    - on macOS: AudioToolbox
    - on iOS: AudioToolbox, AVFoundation
    - on FreeBSD: asound
    - on Linux: asound
    - on Android: aaudio
    - on Windows with MSVC or Clang toolchain: no action needed, libs are defined in-source via pragma-comment-lib
    - on Windows with MINGW/MSYS2 gcc: compile with '-mwin32' and link with -lole32
    - on Vita: SceAudio
    - on 3DS: NDSP (libctru)

    FEATURE OVERVIEW
    ================
    You provide a mono- or stereo-stream of 32-bit float samples, which
    Sokol Audio feeds into platform-specific audio backends:

    - Windows: WASAPI
    - Linux: ALSA
    - FreeBSD: ALSA
    - macOS: CoreAudio
    - iOS: CoreAudio+AVAudioSession
    - emscripten: WebAudio with ScriptProcessorNode
    - Android: AAudio
    - Vita: SceAudio
    - 3DS: NDSP (libctru)

    Sokol Audio will not do any buffer mixing or volume control, if you have
    multiple independent input streams of sample data you need to perform the
    mixing yourself before forwarding the data to Sokol Audio.

    There are two mutually exclusive ways to provide the sample data:

    1. Callback model: You provide a callback function, which will be called
       when Sokol Audio needs new samples. On all platforms except emscripten,
       this function is called from a separate thread.
    2. Push model: Your code pushes small blocks of sample data from your
       main loop or a thread you created. The pushed data is stored in
       a ring buffer where it is pulled by the backend code when
       needed.

    The callback model is preferred because it is the most direct way to
    feed sample data into the audio backends and also has less moving parts
    (there is no ring buffer between your code and the audio backend).

    Sometimes it is not possible to generate the audio stream directly in a
    callback function running in a separate thread, for such cases Sokol Audio
    provides the push-model as a convenience.

    SOKOL AUDIO, SOLOUD AND MINIAUDIO
    =================================
    The WASAPI, ALSA and CoreAudio backend code has been taken from the
    SoLoud library (with some modifications, so any bugs in there are most
    likely my fault). If you need a more fully-featured audio solution, check
    out SoLoud, it's excellent:

        https://github.com/jarikomppa/soloud

    Another alternative which feature-wise is somewhere inbetween SoLoud and
    sokol-audio might be MiniAudio:

        https://github.com/mackron/miniaudio

    GLOSSARY
    ========
    - stream buffer:
        The internal audio data buffer, usually provided by the backend API. The
        size of the stream buffer defines the base latency, smaller buffers have
        lower latency but may cause audio glitches. Bigger buffers reduce or
        eliminate glitches, but have a higher base latency.

    - stream callback:
        Optional callback function which is called by Sokol Audio when it
        needs new samples. On Windows, macOS/iOS and Linux, this is called in
        a separate thread, on WebAudio, this is called per-frame in the
        browser thread.

    - channel:
        A discrete track of audio data, currently 1-channel (mono) and
        2-channel (stereo) is supported and tested.

    - sample:
        The magnitude of an audio signal on one channel at a given time. In
        Sokol Audio, samples are 32-bit float numbers in the range -1.0 to
        +1.0.

    - frame:
        The tightly packed set of samples for all channels at a given time.
        For mono 1 frame is 1 sample. For stereo, 1 frame is 2 samples.

    - packet:
        In Sokol Audio, a small chunk of audio data that is moved from the
        main thread to the audio streaming thread in order to decouple the
        rate at which the main thread provides new audio data, and the
        streaming thread consuming audio data.

    WORKING WITH SOKOL AUDIO
    ========================
    First call saudio_setup() with your preferred audio playback options.
    In most cases you can stick with the default values, these provide
    a good balance between low-latency and glitch-free playback
    on all audio backends.

    You should always provide a logging callback to be aware of any
    warnings and errors. The easiest way is to use sokol_log.h for this:

        #include "sokol_log.h"
        // ...
        saudio_setup(&(saudio_desc){
            .logger = {
                .func = slog_func,
            }
        });

    If you want to use the callback-model, you need to provide a stream
    callback function either in saudio_desc.stream_cb or saudio_desc.stream_userdata_cb,
    otherwise keep both function pointers zero-initialized.

    Use push model and default playback parameters:

        saudio_setup(&(saudio_desc){ .logger.func = slog_func });

    Use stream callback model and default playback parameters:

        saudio_setup(&(saudio_desc){
            .stream_cb = my_stream_callback
            .logger.func = slog_func,
        });

    The standard stream callback doesn't have a user data argument, if you want
    that, use the alternative stream_userdata_cb and also set the user_data pointer:

        saudio_setup(&(saudio_desc){
            .stream_userdata_cb = my_stream_callback,
            .user_data = &my_data
            .logger.func = slog_func,
        });

    The following playback parameters can be provided through the
    saudio_desc struct:

    General parameters (both for stream-callback and push-model):

        int sample_rate     -- the sample rate in Hz, default: 44100
        int num_channels    -- number of channels, default: 1 (mono)
        int buffer_frames   -- number of frames in streaming buffer, default: 2048

    The stream callback prototype (either with or without userdata):

        void (*stream_cb)(float* buffer, int num_frames, int num_channels)
        void (*stream_userdata_cb)(float* buffer, int num_frames, int num_channels, void* user_data)
            Function pointer to the user-provide stream callback.

    Push-model parameters:

        int packet_frames   -- number of frames in a packet, default: 128
        int num_packets     -- number of packets in ring buffer, default: 64

    The sample_rate and num_channels parameters are only hints for the audio
    backend, it isn't guaranteed that those are the values used for actual
    playback.

    To get the actual parameters, call the following functions after
    saudio_setup():

        int saudio_sample_rate(void)
        int saudio_channels(void);

    It's unlikely that the number of channels will be different than requested,
    but a different sample rate isn't uncommon.

    (NOTE: there's an yet unsolved issue when an audio backend might switch
    to a different sample rate when switching output devices, for instance
    plugging in a bluetooth headset, this case is currently not handled in
    Sokol Audio).

    You can check if audio initialization was successful with
    saudio_isvalid(). If backend initialization failed for some reason
    (for instance when there's no audio device in the machine), this
    will return false. Not checking for success won't do any harm, all
    Sokol Audio function will silently fail when called after initialization
    has failed, so apart from missing audio output, nothing bad will happen.

    Before your application exits, you should call

        saudio_shutdown();

    This stops the audio thread (on Linux, Windows and macOS/iOS) and
    properly shuts down the audio backend.

    THE STREAM CALLBACK MODEL
    =========================
    To use Sokol Audio in stream-callback-mode, provide a callback function
    like this in the saudio_desc struct when calling saudio_setup():

    void stream_cb(float* buffer, int num_frames, int num_channels) {
        ...
    }

    Or the alternative version with a user-data argument:

    void stream_userdata_cb(float* buffer, int num_frames, int num_channels, void* user_data) {
        my_data_t* my_data = (my_data_t*) user_data;
        ...
    }

    The job of the callback function is to fill the *buffer* with 32-bit
    float sample values.

    To output silence, fill the buffer with zeros:

        void stream_cb(float* buffer, int num_frames, int num_channels) {
            const int num_samples = num_frames * num_channels;
            for (int i = 0; i < num_samples; i++) {
                buffer[i] = 0.0f;
            }
        }

    For stereo output (num_channels == 2), the samples for the left
    and right channel are interleaved:

        void stream_cb(float* buffer, int num_frames, int num_channels) {
            assert(2 == num_channels);
            for (int i = 0; i < num_frames; i++) {
                buffer[2*i + 0] = ...;  // left channel
                buffer[2*i + 1] = ...;  // right channel
            }
        }

    Please keep in mind that the stream callback function is running in a
    separate thread, if you need to share data with the main thread you need
    to take care yourself to make the access to the shared data thread-safe!

    THE PUSH MODEL
    ==============
    To use the push-model for providing audio data, simply don't set (keep
    zero-initialized) the stream_cb field in the saudio_desc struct when
    calling saudio_setup().

    To provide sample data with the push model, call the saudio_push()
    function at regular intervals (for instance once per frame). You can
    call the saudio_expect() function to ask Sokol Audio how much room is
    in the ring buffer, but if you provide a continuous stream of data
    at the right sample rate, saudio_expect() isn't required (it's a simple
    way to sync/throttle your sample generation code with the playback
    rate though).

    With saudio_push() you may need to maintain your own intermediate sample
    buffer, since pushing individual sample values isn't very efficient.
    The following example is from the MOD player sample in
    sokol-samples (https://github.com/floooh/sokol-samples):

        const int num_frames = saudio_expect();
        if (num_frames > 0) {
            const int num_samples = num_frames * saudio_channels();
            read_samples(flt_buf, num_samples);
            saudio_push(flt_buf, num_frames);
        }

    Another option is to ignore saudio_expect(), and just push samples as they
    are generated in small batches. In this case you *need* to generate the
    samples at the right sample rate:

    The following example is taken from the Tiny Emulators project
    (https://github.com/floooh/chips-test), this is for mono playback,
    so (num_samples == num_frames):

        // tick the sound generator
        if (ay38910_tick(&sys->psg)) {
            // new sample is ready
            sys->sample_buffer[sys->sample_pos++] = sys->psg.sample;
            if (sys->sample_pos == sys->num_samples) {
                // new sample packet is ready
                saudio_push(sys->sample_buffer, sys->num_samples);
                sys->sample_pos = 0;
            }
        }

    THE WEBAUDIO BACKEND
    ====================
    The WebAudio backend is currently using a ScriptProcessorNode callback to
    feed the sample data into WebAudio. ScriptProcessorNode has been
    deprecated for a while because it is running from the main thread, with
    the default initialization parameters it works 'pretty well' though.
    Ultimately Sokol Audio will use Audio Worklets, but this requires a few
    more things to fall into place (Audio Worklets implemented everywhere,
    SharedArrayBuffers enabled again, and I need to figure out a 'low-cost'
    solution in terms of implementation effort, since Audio Worklets are
    a lot more complex than ScriptProcessorNode if the audio data needs to come
    from the main thread).

    The WebAudio backend is automatically selected when compiling for
    emscripten (__EMSCRIPTEN__ define exists).

    https://developers.google.com/web/updates/2017/12/audio-worklet
    https://developers.google.com/web/updates/2018/06/audio-worklet-design-pattern

    "Blob URLs": https://www.html5rocks.com/en/tutorials/workers/basics/

    Also see: https://blog.paul.cx/post/a-wait-free-spsc-ringbuffer-for-the-web/

    THE COREAUDIO BACKEND
    =====================
    The CoreAudio backend is selected on macOS and iOS (__APPLE__ is defined).
    Since the CoreAudio API is implemented in C (not Objective-C) on macOS the
    implementation part of Sokol Audio can be included into a C source file.

    However on iOS, Sokol Audio must be compiled as Objective-C due to it's
    reliance on the AVAudioSession object. The iOS code path support both
    being compiled with or without ARC (Automatic Reference Counting).

    For thread synchronisation, the CoreAudio backend will use the
    pthread_mutex_* functions.

    The incoming floating point samples will be directly forwarded to
    CoreAudio without further conversion.

    macOS and iOS applications that use Sokol Audio need to link with
    the AudioToolbox framework.

    THE WASAPI BACKEND
    ==================
    The WASAPI backend is automatically selected when compiling on Windows
    (_WIN32 is defined).

    For thread synchronisation a Win32 critical section is used.

    WASAPI may use a different size for its own streaming buffer then requested,
    so the base latency may be slightly bigger. The current backend implementation
    converts the incoming floating point sample values to signed 16-bit
    integers.

    The required Windows system DLLs are linked with #pragma comment(lib, ...),
    so you shouldn't need to add additional linker libs in the build process
    (otherwise this is a bug which should be fixed in sokol_audio.h).

    THE ALSA BACKEND
    ================
    The ALSA backend is automatically selected when compiling on Linux
    ('linux' is defined).

    For thread synchronisation, the pthread_mutex_* functions are used.

    Samples are directly forwarded to ALSA in 32-bit float format, no
    further conversion is taking place.

    You need to link with the 'asound' library, and the <alsa/asoundlib.h>
    header must be present (usually both are installed with some sort
    of ALSA development package).

    THE VITA BACKEND
    ================
    The VITA backend is automatically selected when compiling with vitasdk
    ('PSP2_SDK_VERSION' is defined).

    For thread synchronisation, the pthread_mutex_* functions are used.

    Samples are converted from float to short (uint16_t) to maintain
    all the same interface/api as other platforms.

    You may use any supported sample rate you wish, but all audio MUST
    match the same sample rate you choose.

    This uses the "BGM" port to allow selecting the sample rate ("Main"
    port is restricted to 48000 only).

    You need to link with the 'SceAudio' library, and the <psp2/audioout.h>
    header must be present (usually both are installed with the vitasdk).

    THE 3DS BACKEND
    ================
    The 3DS backend is automatically selected when compiling with libctru
    ('__3DS__' is defined).

    Running a separate thread on the older 3ds is not a good idea and I
    was not able to get it working without slowing down the main thread
    too much (it has a single core available with cooperative threads).

    The NDSP seems to work better by using its ndspSetCallback method.

    You may use any supported sample rate you wish, but all audio MUST
    match the same sample rate you choose or it will sound slowed down
    or sped up.

    The queue size and other NDSP specific parameters can be chosen by
    the provided 'saudio_n3ds_desc' type. Defaults will be used if
    nothing is provided.

    There is a known issue of a noticeable delay when starting a new
    sound on emulators. I was not able to improve this to my liking
    and ~300ms can be expected. This can be improved by using a lower
    buffer size than the 2048 default but I would not suggest under
    1536. It may crash under 1408, and they must be in multiples of 128.
    Note: I was NOT able to reproduce this issue on a real device and
    the audio worked perfectly.


    MEMORY ALLOCATION OVERRIDE
    ==========================
    You can override the memory allocation functions at initialization time
    like this:

        void* my_alloc(size_t size, void* user_data) {
            return malloc(size);
        }

        void my_free(void* ptr, void* user_data) {
            free(ptr);
        }

        ...
            saudio_setup(&(saudio_desc){
                // ...
                .allocator = {
                    .alloc_fn = my_alloc,
                    .free_fn = my_free,
                    .user_data = ...,
                }
            });
        ...

    If no overrides are provided, malloc and free will be used.

    This only affects memory allocation calls done by sokol_audio.h
    itself though, not any allocations in OS libraries.

    Memory allocation will only happen on the same thread where saudio_setup()
    was called, so you don't need to worry about thread-safety.


    ERROR REPORTING AND LOGGING
    ===========================
    To get any logging information at all you need to provide a logging callback in the setup call
    the easiest way is to use sokol_log.h:

        #include "sokol_log.h"

        saudio_setup(&(saudio_desc){ .logger.func = slog_func });

    To override logging with your own callback, first write a logging function like this:

        void my_log(const char* tag,                // e.g. 'saudio'
                    uint32_t log_level,             // 0=panic, 1=error, 2=warn, 3=info
                    uint32_t log_item_id,           // SAUDIO_LOGITEM_*
                    const char* message_or_null,    // a message string, may be nullptr in release mode
                    uint32_t line_nr,               // line number in sokol_audio.h
                    const char* filename_or_null,   // source filename, may be nullptr in release mode
                    void* user_data)
        {
            ...
        }

    ...and then setup sokol-audio like this:

        saudio_setup(&(saudio_desc){
            .logger = {
                .func = my_log,
                .user_data = my_user_data,
            }
        });

    The provided logging function must be reentrant (e.g. be callable from
    different threads).

    If you don't want to provide your own custom logger it is highly recommended to use
    the standard logger in sokol_log.h instead, otherwise you won't see any warnings or
    errors.


    LICENSE
    =======

    zlib/libpng license

    Copyright (c) 2018 Andre Weissflog

    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.

*/

import "core:c"

_ :: c

SOKOL_DEBUG :: #config(SOKOL_DEBUG, ODIN_DEBUG)

DEBUG :: #config(SOKOL_AUDIO_DEBUG, SOKOL_DEBUG)
USE_GL :: #config(SOKOL_USE_GL, false)
USE_DLL :: #config(SOKOL_DLL, false)

when ODIN_OS == .Windows {
    when USE_DLL {
        when USE_GL {
            when DEBUG { foreign import sokol_audio_clib { "../sokol_dll_windows_x64_gl_debug.lib" } }
            else       { foreign import sokol_audio_clib { "../sokol_dll_windows_x64_gl_release.lib" } }
        } else {
            when DEBUG { foreign import sokol_audio_clib { "../sokol_dll_windows_x64_d3d11_debug.lib" } }
            else       { foreign import sokol_audio_clib { "../sokol_dll_windows_x64_d3d11_release.lib" } }
        }
    } else {
        when USE_GL {
            when DEBUG { foreign import sokol_audio_clib { "sokol_audio_windows_x64_gl_debug.lib" } }
            else       { foreign import sokol_audio_clib { "sokol_audio_windows_x64_gl_release.lib" } }
        } else {
            when DEBUG { foreign import sokol_audio_clib { "sokol_audio_windows_x64_d3d11_debug.lib" } }
            else       { foreign import sokol_audio_clib { "sokol_audio_windows_x64_d3d11_release.lib" } }
        }
    }
} else when ODIN_OS == .Darwin {
    when USE_DLL {
             when  USE_GL && ODIN_ARCH == .arm64 &&  DEBUG { foreign import sokol_audio_clib { "../dylib/sokol_dylib_macos_arm64_gl_debug.dylib" } }
        else when  USE_GL && ODIN_ARCH == .arm64 && !DEBUG { foreign import sokol_audio_clib { "../dylib/sokol_dylib_macos_arm64_gl_release.dylib" } }
        else when  USE_GL && ODIN_ARCH == .amd64 &&  DEBUG { foreign import sokol_audio_clib { "../dylib/sokol_dylib_macos_x64_gl_debug.dylib" } }
        else when  USE_GL && ODIN_ARCH == .amd64 && !DEBUG { foreign import sokol_audio_clib { "../dylib/sokol_dylib_macos_x64_gl_release.dylib" } }
        else when !USE_GL && ODIN_ARCH == .arm64 &&  DEBUG { foreign import sokol_audio_clib { "../dylib/sokol_dylib_macos_arm64_metal_debug.dylib" } }
        else when !USE_GL && ODIN_ARCH == .arm64 && !DEBUG { foreign import sokol_audio_clib { "../dylib/sokol_dylib_macos_arm64_metal_release.dylib" } }
        else when !USE_GL && ODIN_ARCH == .amd64 &&  DEBUG { foreign import sokol_audio_clib { "../dylib/sokol_dylib_macos_x64_metal_debug.dylib" } }
        else when !USE_GL && ODIN_ARCH == .amd64 && !DEBUG { foreign import sokol_audio_clib { "../dylib/sokol_dylib_macos_x64_metal_release.dylib" } }
    } else {
        when USE_GL {
            when ODIN_ARCH == .arm64 {
                when DEBUG { foreign import sokol_audio_clib { "sokol_audio_macos_arm64_gl_debug.a", "system:AudioToolbox.framework" } }
                else       { foreign import sokol_audio_clib { "sokol_audio_macos_arm64_gl_release.a", "system:AudioToolbox.framework" } }
            } else {
                when DEBUG { foreign import sokol_audio_clib { "sokol_audio_macos_x64_gl_debug.a", "system:AudioToolbox.framework" } }
                else       { foreign import sokol_audio_clib { "sokol_audio_macos_x64_gl_release.a", "system:AudioToolbox.framework" } }
            }
        } else {
            when ODIN_ARCH == .arm64 {
                when DEBUG { foreign import sokol_audio_clib { "sokol_audio_macos_arm64_metal_debug.a", "system:AudioToolbox.framework" } }
                else       { foreign import sokol_audio_clib { "sokol_audio_macos_arm64_metal_release.a", "system:AudioToolbox.framework" } }
            } else {
                when DEBUG { foreign import sokol_audio_clib { "sokol_audio_macos_x64_metal_debug.a", "system:AudioToolbox.framework" } }
                else       { foreign import sokol_audio_clib { "sokol_audio_macos_x64_metal_release.a", "system:AudioToolbox.framework" } }
            }
        }
    }
} else when ODIN_OS == .Linux {
    when USE_DLL {
        when DEBUG { foreign import sokol_audio_clib { "sokol_audio_linux_x64_gl_debug.so", "system:dl", "system:pthread" } }
        else       { foreign import sokol_audio_clib { "sokol_audio_linux_x64_gl_release.so", "system:dl", "system:pthread" } }
    } else {
        when DEBUG { foreign import sokol_audio_clib { "sokol_audio_linux_x64_gl_debug.a", "system:asound", "system:dl", "system:pthread" } }
        else       { foreign import sokol_audio_clib { "sokol_audio_linux_x64_gl_release.a", "system:asound", "system:dl", "system:pthread" } }
    }
} else when ODIN_ARCH == .wasm32 || ODIN_ARCH == .wasm64p32 {
    // Feed sokol_audio_wasm_gl_debug.a or sokol_audio_wasm_gl_release.a into emscripten compiler.
    foreign import sokol_audio_clib { "env.o" }
} else {
    #panic("This OS is currently not supported")
}

@(default_calling_convention="c", link_prefix="saudio_")
foreign sokol_audio_clib {
    // setup sokol-audio
    setup :: proc(#by_ptr desc: Desc)  ---
    // shutdown sokol-audio
    shutdown :: proc()  ---
    // true after setup if audio backend was successfully initialized
    isvalid :: proc() -> bool ---
    // return the saudio_desc.user_data pointer
    userdata :: proc() -> rawptr ---
    // return a copy of the original saudio_desc struct
    query_desc :: proc() -> Desc ---
    // actual sample rate
    sample_rate :: proc() -> c.int ---
    // return actual backend buffer size in number of frames
    buffer_frames :: proc() -> c.int ---
    // actual number of channels
    channels :: proc() -> c.int ---
    // return true if audio context is currently suspended (only in WebAudio backend, all other backends return false)
    suspended :: proc() -> bool ---
    // get current number of frames to fill packet queue
    expect :: proc() -> c.int ---
    // push sample frames from main thread, returns number of frames actually pushed
    push :: proc(frames: ^f32, #any_int num_frames: c.int) -> c.int ---
}

Log_Item :: enum i32 {
    OK,
    MALLOC_FAILED,
    ALSA_SND_PCM_OPEN_FAILED,
    ALSA_FLOAT_SAMPLES_NOT_SUPPORTED,
    ALSA_REQUESTED_BUFFER_SIZE_NOT_SUPPORTED,
    ALSA_REQUESTED_CHANNEL_COUNT_NOT_SUPPORTED,
    ALSA_SND_PCM_HW_PARAMS_SET_RATE_NEAR_FAILED,
    ALSA_SND_PCM_HW_PARAMS_FAILED,
    ALSA_PTHREAD_CREATE_FAILED,
    WASAPI_CREATE_EVENT_FAILED,
    WASAPI_CREATE_DEVICE_ENUMERATOR_FAILED,
    WASAPI_GET_DEFAULT_AUDIO_ENDPOINT_FAILED,
    WASAPI_DEVICE_ACTIVATE_FAILED,
    WASAPI_AUDIO_CLIENT_INITIALIZE_FAILED,
    WASAPI_AUDIO_CLIENT_GET_BUFFER_SIZE_FAILED,
    WASAPI_AUDIO_CLIENT_GET_SERVICE_FAILED,
    WASAPI_AUDIO_CLIENT_SET_EVENT_HANDLE_FAILED,
    WASAPI_CREATE_THREAD_FAILED,
    AAUDIO_STREAMBUILDER_OPEN_STREAM_FAILED,
    AAUDIO_PTHREAD_CREATE_FAILED,
    AAUDIO_RESTARTING_STREAM_AFTER_ERROR,
    USING_AAUDIO_BACKEND,
    AAUDIO_CREATE_STREAMBUILDER_FAILED,
    COREAUDIO_NEW_OUTPUT_FAILED,
    COREAUDIO_ALLOCATE_BUFFER_FAILED,
    COREAUDIO_START_FAILED,
    BACKEND_BUFFER_SIZE_ISNT_MULTIPLE_OF_PACKET_SIZE,
    VITA_SCEAUDIO_OPEN_FAILED,
    VITA_PTHREAD_CREATE_FAILED,
    N3DS_NDSP_OPEN_FAILED,
}

/*
    saudio_logger

    Used in saudio_desc to provide a custom logging and error reporting
    callback to sokol-audio.
*/
Logger :: struct {
    func : proc "c" (a0: cstring, a1: u32, a2: u32, a3: cstring, a4: u32, a5: cstring, a6: rawptr),
    user_data : rawptr,
}

/*
    saudio_allocator

    Used in saudio_desc to provide custom memory-alloc and -free functions
    to sokol_audio.h. If memory management should be overridden, both the
    alloc_fn and free_fn function must be provided (e.g. it's not valid to
    override one function but not the other).
*/
Allocator :: struct {
    alloc_fn : proc "c" (a0: c.size_t, a1: rawptr) -> rawptr,
    free_fn : proc "c" (a0: rawptr, a1: rawptr),
    user_data : rawptr,
}

N3ds_Ndspinterptype :: enum i32 {
    DSP_INTERP_POLYPHASE = 0,
    DSP_INTERP_LINEAR = 1,
    DSP_INTERP_NONE = 2,
}

N3ds_Desc :: struct {
    queue_count : c.int,
    interpolation_type : N3ds_Ndspinterptype,
    channel_id : c.int,
}

Desc :: struct {
    sample_rate : c.int,
    num_channels : c.int,
    buffer_frames : c.int,
    packet_frames : c.int,
    num_packets : c.int,
    stream_cb : proc "c" (a0: ^f32, a1: c.int, a2: c.int),
    stream_userdata_cb : proc "c" (a0: ^f32, a1: c.int, a2: c.int, a3: rawptr),
    user_data : rawptr,
    n3ds : N3ds_Desc,
    allocator : Allocator,
    logger : Logger,
}