feat(gx): add incomplete 'CGxDeviceGLSDL' (#2)

* chore(build): add vendored SDL 3.0.0 library * chore(build): add vendored glew-cmake-2.2.0 library * feat(console): in the presence of -opengl launch flag, change GxApi to OpenGl * feat(gx): add uncompleted CGxDeviceGLSDL targeting Windows and Linux * chore(build): change SDL3 linkage from shared (bad) to to static (good)
2025-12-14 12:02:30 +00:00 · 2023-11-18 10:50:16 -05:00 · 2023-11-18 10:50:16 -05:00 · 706c8903a1
commit 706c8903a1
parent 934e0fb600
2043 changed files with 663533 additions and 5 deletions
--- a/vendor/sdl-3.0.0/src/audio/aaudio/SDL_aaudio.c
+++ b/vendor/sdl-3.0.0/src/audio/aaudio/SDL_aaudio.c
@ -0,0 +1,536 @@
+/*
+  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.
+*/
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_AAUDIO
+
+#include "../SDL_sysaudio.h"
+#include "SDL_aaudio.h"
+
+#include "../../core/android/SDL_android.h"
+#include <stdbool.h>
+#include <aaudio/AAudio.h>
+
+#if __ANDROID_API__ < 31
+#define AAUDIO_FORMAT_PCM_I32 4
+#endif
+
+struct SDL_PrivateAudioData
+{
+    AAudioStream *stream;
+    int num_buffers;
+    Uint8 *mixbuf;          // Raw mixing buffer
+    size_t mixbuf_bytes;    // num_buffers * device->buffer_size
+    size_t callback_bytes;
+    size_t processed_bytes;
+    SDL_Semaphore *semaphore;
+    SDL_AtomicInt error_callback_triggered;
+    SDL_bool resume;  // Resume device if it was paused automatically
+};
+
+// Debug
+#if 0
+#define LOGI(...) SDL_Log(__VA_ARGS__);
+#else
+#define LOGI(...)
+#endif
+
+#define LIB_AAUDIO_SO "libaaudio.so"
+
+typedef struct AAUDIO_Data
+{
+    void *handle;
+#define SDL_PROC(ret, func, params) ret (*func) params;
+#include "SDL_aaudiofuncs.h"
+} AAUDIO_Data;
+static AAUDIO_Data ctx;
+
+static int AAUDIO_LoadFunctions(AAUDIO_Data *data)
+{
+#define SDL_PROC(ret, func, params)                                                             \
+    do {                                                                                        \
+        data->func = (ret (*) params)SDL_LoadFunction(data->handle, #func);                                     \
+        if (!data->func) {                                                                      \
+            return SDL_SetError("Couldn't load AAUDIO function %s: %s", #func, SDL_GetError()); \
+        }                                                                                       \
+    } while (0);
+#include "SDL_aaudiofuncs.h"
+    return 0;
+}
+
+
+static void AAUDIO_errorCallback(AAudioStream *stream, void *userData, aaudio_result_t error)
+{
+    LOGI("SDL AAUDIO_errorCallback: %d - %s", error, ctx.AAudio_convertResultToText(error));
+
+    // You MUST NOT close the audio stream from this callback, so we cannot call SDL_AudioDeviceDisconnected here.
+    // Just flag the device so we can kill it in PlayDevice instead.
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userData;
+    SDL_AtomicSet(&device->hidden->error_callback_triggered, (int) error);  // AAUDIO_OK is zero, so !triggered means no error.
+    SDL_PostSemaphore(device->hidden->semaphore);  // in case we're blocking in WaitDevice.
+}
+
+static aaudio_data_callback_result_t AAUDIO_dataCallback(AAudioStream *stream, void *userData, void *audioData, int32_t numFrames)
+{
+    SDL_AudioDevice *device = (SDL_AudioDevice *) userData;
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    size_t framesize = SDL_AUDIO_FRAMESIZE(device->spec);
+    size_t callback_bytes = numFrames * framesize;
+    size_t old_buffer_index = hidden->callback_bytes / device->buffer_size;
+
+    if (device->iscapture) {
+        const Uint8 *input = (const Uint8 *)audioData;
+        size_t available_bytes = hidden->mixbuf_bytes - (hidden->callback_bytes - hidden->processed_bytes);
+        size_t size = SDL_min(available_bytes, callback_bytes);
+        size_t offset = hidden->callback_bytes % hidden->mixbuf_bytes;
+        size_t end = (offset + size) % hidden->mixbuf_bytes;
+        SDL_assert(size <= hidden->mixbuf_bytes);
+
+//LOGI("Recorded %zu frames, %zu available, %zu max (%zu written, %zu read)\n", callback_bytes / framesize, available_bytes / framesize, hidden->mixbuf_bytes / framesize, hidden->callback_bytes / framesize, hidden->processed_bytes / framesize);
+
+        if (offset <= end) {
+            SDL_memcpy(&hidden->mixbuf[offset], input, size);
+        } else {
+            size_t partial = (hidden->mixbuf_bytes - offset);
+            SDL_memcpy(&hidden->mixbuf[offset], &input[0], partial);
+            SDL_memcpy(&hidden->mixbuf[0], &input[partial], end);
+        }
+
+        SDL_MemoryBarrierRelease();
+        hidden->callback_bytes += size;
+
+        if (size < callback_bytes) {
+            LOGI("Audio recording overflow, dropped %zu frames\n", (callback_bytes - size) / framesize);
+        }
+    } else {
+        Uint8 *output = (Uint8 *)audioData;
+        size_t available_bytes = (hidden->processed_bytes - hidden->callback_bytes);
+        size_t size = SDL_min(available_bytes, callback_bytes);
+        size_t offset = hidden->callback_bytes % hidden->mixbuf_bytes;
+        size_t end = (offset + size) % hidden->mixbuf_bytes;
+        SDL_assert(size <= hidden->mixbuf_bytes);
+
+//LOGI("Playing %zu frames, %zu available, %zu max (%zu written, %zu read)\n", callback_bytes / framesize, available_bytes / framesize, hidden->mixbuf_bytes / framesize, hidden->processed_bytes / framesize, hidden->callback_bytes / framesize);
+
+        SDL_MemoryBarrierAcquire();
+        if (offset <= end) {
+            SDL_memcpy(output, &hidden->mixbuf[offset], size);
+        } else {
+            size_t partial = (hidden->mixbuf_bytes - offset);
+            SDL_memcpy(&output[0], &hidden->mixbuf[offset], partial);
+            SDL_memcpy(&output[partial], &hidden->mixbuf[0], end);
+        }
+        hidden->callback_bytes += size;
+
+        if (size < callback_bytes) {
+            LOGI("Audio playback underflow, missed %zu frames\n", (callback_bytes - size) / framesize);
+            SDL_memset(&output[size], device->silence_value, (callback_bytes - size));
+        }
+    }
+
+    size_t new_buffer_index = hidden->callback_bytes / device->buffer_size;
+    while (old_buffer_index < new_buffer_index) {
+        // Trigger audio processing
+        SDL_PostSemaphore(hidden->semaphore);
+        ++old_buffer_index;
+    }
+
+    return AAUDIO_CALLBACK_RESULT_CONTINUE;
+}
+
+static Uint8 *AAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *bufsize)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    size_t offset = (hidden->processed_bytes % hidden->mixbuf_bytes);
+    return &hidden->mixbuf[offset];
+}
+
+static int AAUDIO_WaitDevice(SDL_AudioDevice *device)
+{
+    SDL_WaitSemaphore(device->hidden->semaphore);
+    return 0;
+}
+
+static int BuildAAudioStream(SDL_AudioDevice *device);
+
+static int RecoverAAudioDevice(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+
+    // attempt to build a new stream, in case there's a new default device.
+    ctx.AAudioStream_requestStop(hidden->stream);
+    ctx.AAudioStream_close(hidden->stream);
+    hidden->stream = NULL;
+
+    SDL_aligned_free(hidden->mixbuf);
+    hidden->mixbuf = NULL;
+
+    SDL_DestroySemaphore(hidden->semaphore);
+    hidden->semaphore = NULL;
+
+    const int prev_sample_frames = device->sample_frames;
+    SDL_AudioSpec prevspec;
+    SDL_copyp(&prevspec, &device->spec);
+
+    if (BuildAAudioStream(device) < 0) {
+        return -1;  // oh well, we tried.
+    }
+
+    // we don't know the new device spec until we open the new device, so we saved off the old one and force it back
+    // so SDL_AudioDeviceFormatChanged can set up all the important state if necessary and then set it back to the new spec.
+    const int new_sample_frames = device->sample_frames;
+    SDL_AudioSpec newspec;
+    SDL_copyp(&newspec, &device->spec);
+
+    device->sample_frames = prev_sample_frames;
+    SDL_copyp(&device->spec, &prevspec);
+    if (SDL_AudioDeviceFormatChangedAlreadyLocked(device, &newspec, new_sample_frames) < 0) {
+        return -1;  // ugh
+    }
+    return 0;
+}
+
+
+static int AAUDIO_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+
+    // AAUDIO_dataCallback picks up our work and unblocks AAUDIO_WaitDevice. But make sure we didn't fail here.
+    const aaudio_result_t err = (aaudio_result_t) SDL_AtomicGet(&hidden->error_callback_triggered);
+    if (err) {
+        SDL_LogError(SDL_LOG_CATEGORY_AUDIO, "aaudio: Audio device triggered error %d (%s)", (int) err, ctx.AAudio_convertResultToText(err));
+
+        if (RecoverAAudioDevice(device) < 0) {
+            return -1;  // oh well, we went down hard.
+        }
+    } else {
+        SDL_MemoryBarrierRelease();
+        hidden->processed_bytes += buflen;
+    }
+    return 0;
+}
+
+static int AAUDIO_CaptureFromDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+
+    // AAUDIO_dataCallback picks up our work and unblocks AAUDIO_WaitDevice. But make sure we didn't fail here.
+    if (SDL_AtomicGet(&hidden->error_callback_triggered)) {
+        SDL_AtomicSet(&hidden->error_callback_triggered, 0);
+        return -1;
+    }
+
+    SDL_assert(buflen == device->buffer_size);  // If this isn't true, we need to change semaphore trigger logic and account for wrapping copies here
+    size_t offset = (hidden->processed_bytes % hidden->mixbuf_bytes);
+    SDL_MemoryBarrierAcquire();
+    SDL_memcpy(buffer, &hidden->mixbuf[offset], buflen);
+    hidden->processed_bytes += buflen;
+    return buflen;
+}
+
+static void AAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    LOGI(__func__);
+
+    if (hidden) {
+        if (hidden->stream) {
+            ctx.AAudioStream_requestStop(hidden->stream);
+            // !!! FIXME: do we have to wait for the state to change to make sure all buffered audio has played, or will close do this (or will the system do this after the close)?
+            // !!! FIXME: also, will this definitely wait for a running data callback to finish, and then stop the callback from firing again?
+            ctx.AAudioStream_close(hidden->stream);
+        }
+
+        if (hidden->semaphore) {
+            SDL_DestroySemaphore(hidden->semaphore);
+        }
+
+        SDL_aligned_free(hidden->mixbuf);
+        SDL_free(hidden);
+        device->hidden = NULL;
+    }
+}
+
+static int BuildAAudioStream(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    const SDL_bool iscapture = device->iscapture;
+    aaudio_result_t res;
+
+    SDL_AtomicSet(&hidden->error_callback_triggered, 0);
+
+    AAudioStreamBuilder *builder = NULL;
+    res = ctx.AAudio_createStreamBuilder(&builder);
+    if (res != AAUDIO_OK) {
+        LOGI("SDL Failed AAudio_createStreamBuilder %d", res);
+        return SDL_SetError("SDL Failed AAudio_createStreamBuilder %d", res);
+    } else if (builder == NULL) {
+        LOGI("SDL Failed AAudio_createStreamBuilder - builder NULL");
+        return SDL_SetError("SDL Failed AAudio_createStreamBuilder - builder NULL");
+    }
+
+#if ALLOW_MULTIPLE_ANDROID_AUDIO_DEVICES
+    const int aaudio_device_id = (int) ((size_t) device->handle);
+    LOGI("Opening device id %d", aaudio_device_id);
+    ctx.AAudioStreamBuilder_setDeviceId(builder, aaudio_device_id);
+#endif
+
+    aaudio_format_t format;
+#ifdef SET_AUDIO_FORMAT
+    if ((device->spec.format == SDL_AUDIO_S32) && (SDL_GetAndroidSDKVersion() >= 31)) {
+        format = AAUDIO_FORMAT_PCM_I32;
+    } else if (device->spec.format == SDL_AUDIO_F32) {
+        format = AAUDIO_FORMAT_PCM_FLOAT;
+    } else {
+        format = AAUDIO_FORMAT_PCM_I16;  // sint16 is a safe bet for everything else.
+    }
+    ctx.AAudioStreamBuilder_setFormat(builder, format);
+    ctx.AAudioStreamBuilder_setSampleRate(builder, device->spec.freq);
+    ctx.AAudioStreamBuilder_setChannelCount(builder, device->spec.channels);
+#endif
+
+    const aaudio_direction_t direction = (iscapture ? AAUDIO_DIRECTION_INPUT : AAUDIO_DIRECTION_OUTPUT);
+    ctx.AAudioStreamBuilder_setDirection(builder, direction);
+    ctx.AAudioStreamBuilder_setErrorCallback(builder, AAUDIO_errorCallback, device);
+    ctx.AAudioStreamBuilder_setDataCallback(builder, AAUDIO_dataCallback, device);
+    // Some devices have flat sounding audio when low latency mode is enabled, but this is a better experience for most people
+    if (SDL_GetHintBoolean("SDL_ANDROID_LOW_LATENCY_AUDIO", SDL_TRUE)) {
+        ctx.AAudioStreamBuilder_setPerformanceMode(builder, AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+    }
+
+    LOGI("AAudio Try to open %u hz %u bit chan %u %s samples %u",
+         device->spec.freq, SDL_AUDIO_BITSIZE(device->spec.format),
+         device->spec.channels, SDL_AUDIO_ISBIGENDIAN(device->spec.format) ? "BE" : "LE", device->sample_frames);
+
+    res = ctx.AAudioStreamBuilder_openStream(builder, &hidden->stream);
+    if (res != AAUDIO_OK) {
+        LOGI("SDL Failed AAudioStreamBuilder_openStream %d", res);
+        ctx.AAudioStreamBuilder_delete(builder);
+        return SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+    }
+    ctx.AAudioStreamBuilder_delete(builder);
+
+    device->sample_frames = (int)ctx.AAudioStream_getFramesPerDataCallback(hidden->stream);
+    if (device->sample_frames == AAUDIO_UNSPECIFIED) {
+        // We'll get variable frames in the callback, make sure we have at least half a buffer available
+        device->sample_frames = (int)ctx.AAudioStream_getBufferCapacityInFrames(hidden->stream) / 2;
+    }
+
+    device->spec.freq = ctx.AAudioStream_getSampleRate(hidden->stream);
+    device->spec.channels = ctx.AAudioStream_getChannelCount(hidden->stream);
+
+    format = ctx.AAudioStream_getFormat(hidden->stream);
+    if (format == AAUDIO_FORMAT_PCM_I16) {
+        device->spec.format = SDL_AUDIO_S16;
+    } else if (format == AAUDIO_FORMAT_PCM_I32) {
+        device->spec.format = SDL_AUDIO_S32;
+    } else if (format == AAUDIO_FORMAT_PCM_FLOAT) {
+        device->spec.format = SDL_AUDIO_F32;
+    } else {
+        return SDL_SetError("Got unexpected audio format %d from AAudioStream_getFormat", (int) format);
+    }
+
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    // Allocate a double buffered mixing buffer
+    hidden->num_buffers = 2;
+    hidden->mixbuf_bytes = (hidden->num_buffers * device->buffer_size);
+    hidden->mixbuf = (Uint8 *)SDL_aligned_alloc(SDL_SIMDGetAlignment(), hidden->mixbuf_bytes);
+    if (hidden->mixbuf == NULL) {
+        return SDL_OutOfMemory();
+    }
+    hidden->processed_bytes = 0;
+    hidden->callback_bytes = 0;
+
+    hidden->semaphore = SDL_CreateSemaphore(iscapture ? 0 : hidden->num_buffers);
+    if (!hidden->semaphore) {
+        LOGI("SDL Failed SDL_CreateSemaphore %s iscapture:%d", SDL_GetError(), iscapture);
+        return -1;
+    }
+
+    LOGI("AAudio Actually opened %u hz %u bit chan %u %s samples %u, buffers %d",
+         device->spec.freq, SDL_AUDIO_BITSIZE(device->spec.format),
+         device->spec.channels, SDL_AUDIO_ISBIGENDIAN(device->spec.format) ? "BE" : "LE", device->sample_frames, hidden->num_buffers);
+
+    res = ctx.AAudioStream_requestStart(hidden->stream);
+    if (res != AAUDIO_OK) {
+        LOGI("SDL Failed AAudioStream_requestStart %d iscapture:%d", res, iscapture);
+        return SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+    }
+
+    LOGI("SDL AAudioStream_requestStart OK");
+
+    return 0;
+}
+
+static int AAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+#if ALLOW_MULTIPLE_ANDROID_AUDIO_DEVICES
+    SDL_assert(device->handle != NULL);  // AAUDIO_UNSPECIFIED is zero, so legit devices should all be non-zero.
+#endif
+
+    LOGI(__func__);
+
+    if (device->iscapture) {
+        if (!Android_JNI_RequestPermission("android.permission.RECORD_AUDIO")) {
+            LOGI("This app doesn't have RECORD_AUDIO permission");
+            return SDL_SetError("This app doesn't have RECORD_AUDIO permission");
+        }
+    }
+
+    device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
+    if (device->hidden == NULL) {
+        return SDL_OutOfMemory();
+    }
+
+    return BuildAAudioStream(device);
+}
+
+static SDL_bool PauseOneDevice(SDL_AudioDevice *device, void *userdata)
+{
+    struct SDL_PrivateAudioData *hidden = (struct SDL_PrivateAudioData *)device->hidden;
+    if (hidden != NULL) {
+        if (hidden->stream) {
+            aaudio_result_t res;
+
+            if (device->iscapture) {
+                // Pause() isn't implemented for 'capture', use Stop()
+                res = ctx.AAudioStream_requestStop(hidden->stream);
+            } else {
+                res = ctx.AAudioStream_requestPause(hidden->stream);
+            }
+
+            if (res != AAUDIO_OK) {
+                LOGI("SDL Failed AAudioStream_requestPause %d", res);
+                SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            }
+
+            SDL_LockMutex(device->lock);
+            hidden->resume = SDL_TRUE;
+        }
+    }
+    return SDL_FALSE;  // keep enumerating.
+}
+
+// Pause (block) all non already paused audio devices by taking their mixer lock
+void AAUDIO_PauseDevices(void)
+{
+    if (ctx.handle != NULL) {  // AAUDIO driver is used?
+        (void) SDL_FindPhysicalAudioDeviceByCallback(PauseOneDevice, NULL);
+    }
+}
+
+// Resume (unblock) all non already paused audio devices by releasing their mixer lock
+static SDL_bool ResumeOneDevice(SDL_AudioDevice *device, void *userdata)
+{
+    struct SDL_PrivateAudioData *hidden = device->hidden;
+    if (hidden != NULL) {
+        if (hidden->resume) {
+            hidden->resume = SDL_FALSE;
+            SDL_UnlockMutex(device->lock);
+        }
+
+        if (hidden->stream) {
+            aaudio_result_t res = ctx.AAudioStream_requestStart(hidden->stream);
+            if (res != AAUDIO_OK) {
+                LOGI("SDL Failed AAudioStream_requestStart %d", res);
+                SDL_SetError("%s : %s", __func__, ctx.AAudio_convertResultToText(res));
+            }
+        }
+    }
+    return SDL_FALSE;  // keep enumerating.
+}
+
+void AAUDIO_ResumeDevices(void)
+{
+    if (ctx.handle != NULL) {  // AAUDIO driver is used?
+        (void) SDL_FindPhysicalAudioDeviceByCallback(ResumeOneDevice, NULL);
+    }
+}
+
+static void AAUDIO_Deinitialize(void)
+{
+    Android_StopAudioHotplug();
+
+    LOGI(__func__);
+    if (ctx.handle) {
+        SDL_UnloadObject(ctx.handle);
+    }
+    SDL_zero(ctx);
+    LOGI("End AAUDIO %s", SDL_GetError());
+}
+
+
+static SDL_bool AAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    LOGI(__func__);
+
+    /* AAudio was introduced in Android 8.0, but has reference counting crash issues in that release,
+     * so don't use it until 8.1.
+     *
+     * See https://github.com/google/oboe/issues/40 for more information.
+     */
+    if (SDL_GetAndroidSDKVersion() < 27) {
+        return SDL_FALSE;
+    }
+
+    SDL_zero(ctx);
+
+    ctx.handle = SDL_LoadObject(LIB_AAUDIO_SO);
+    if (ctx.handle == NULL) {
+        LOGI("SDL couldn't find " LIB_AAUDIO_SO);
+        return SDL_FALSE;
+    }
+
+    if (AAUDIO_LoadFunctions(&ctx) < 0) {
+        SDL_UnloadObject(ctx.handle);
+        SDL_zero(ctx);
+        return SDL_FALSE;
+    }
+
+    impl->ThreadInit = Android_AudioThreadInit;
+    impl->Deinitialize = AAUDIO_Deinitialize;
+    impl->OpenDevice = AAUDIO_OpenDevice;
+    impl->CloseDevice = AAUDIO_CloseDevice;
+    impl->WaitDevice = AAUDIO_WaitDevice;
+    impl->PlayDevice = AAUDIO_PlayDevice;
+    impl->GetDeviceBuf = AAUDIO_GetDeviceBuf;
+    impl->WaitCaptureDevice = AAUDIO_WaitDevice;
+    impl->CaptureFromDevice = AAUDIO_CaptureFromDevice;
+
+    impl->HasCaptureSupport = SDL_TRUE;
+
+#if ALLOW_MULTIPLE_ANDROID_AUDIO_DEVICES
+    impl->DetectDevices = Android_StartAudioHotplug;
+#else
+    impl->OnlyHasDefaultOutputDevice = SDL_TRUE;
+    impl->OnlyHasDefaultCaptureDevice = SDL_TRUE;
+#endif
+
+    LOGI("SDL AAUDIO_Init OK");
+    return SDL_TRUE;
+}
+
+AudioBootStrap AAUDIO_bootstrap = {
+    "AAudio", "AAudio audio driver", AAUDIO_Init, SDL_FALSE
+};
+
+#endif // SDL_AUDIO_DRIVER_AAUDIO