2 files changed, 845 insertions, 0 deletions

diff --git a/contrib/SDL-3.2.8/src/audio/openslES/SDL_openslES.c b/contrib/SDL-3.2.8/src/audio/openslES/SDL_openslES.c
new file mode 100644
index 0000000..4d5b3bd
--- /dev/null
+++ b/contrib/SDL-3.2.8/src/audio/openslES/SDL_openslES.c
@@ -0,0 +1,807 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 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_OPENSLES
+
+// For more discussion of low latency audio on Android, see this:
+//   https://googlesamples.github.io/android-audio-high-performance/guides/opensl_es.html
+
+#include "../SDL_sysaudio.h"
+#include "SDL_openslES.h"
+
+#include "../../core/android/SDL_android.h"
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+#include <android/log.h>
+
+
+#define NUM_BUFFERS 2 // -- Don't lower this!
+
+struct SDL_PrivateAudioData
+{
+    Uint8 *mixbuff;
+    int next_buffer;
+    Uint8 *pmixbuff[NUM_BUFFERS];
+    SDL_Semaphore *playsem;
+};
+
+#if 0
+#define LOG_TAG   "SDL_openslES"
+#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
+#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
+//#define LOGV(...)  __android_log_print(ANDROID_LOG_VERBOSE,LOG_TAG,__VA_ARGS__)
+#define LOGV(...)
+#else
+#define LOGE(...)
+#define LOGI(...)
+#define LOGV(...)
+#endif
+
+/*
+#define SL_SPEAKER_FRONT_LEFT            ((SLuint32) 0x00000001)
+#define SL_SPEAKER_FRONT_RIGHT           ((SLuint32) 0x00000002)
+#define SL_SPEAKER_FRONT_CENTER          ((SLuint32) 0x00000004)
+#define SL_SPEAKER_LOW_FREQUENCY         ((SLuint32) 0x00000008)
+#define SL_SPEAKER_BACK_LEFT             ((SLuint32) 0x00000010)
+#define SL_SPEAKER_BACK_RIGHT            ((SLuint32) 0x00000020)
+#define SL_SPEAKER_FRONT_LEFT_OF_CENTER  ((SLuint32) 0x00000040)
+#define SL_SPEAKER_FRONT_RIGHT_OF_CENTER ((SLuint32) 0x00000080)
+#define SL_SPEAKER_BACK_CENTER           ((SLuint32) 0x00000100)
+#define SL_SPEAKER_SIDE_LEFT             ((SLuint32) 0x00000200)
+#define SL_SPEAKER_SIDE_RIGHT            ((SLuint32) 0x00000400)
+#define SL_SPEAKER_TOP_CENTER            ((SLuint32) 0x00000800)
+#define SL_SPEAKER_TOP_FRONT_LEFT        ((SLuint32) 0x00001000)
+#define SL_SPEAKER_TOP_FRONT_CENTER      ((SLuint32) 0x00002000)
+#define SL_SPEAKER_TOP_FRONT_RIGHT       ((SLuint32) 0x00004000)
+#define SL_SPEAKER_TOP_BACK_LEFT         ((SLuint32) 0x00008000)
+#define SL_SPEAKER_TOP_BACK_CENTER       ((SLuint32) 0x00010000)
+#define SL_SPEAKER_TOP_BACK_RIGHT        ((SLuint32) 0x00020000)
+*/
+#define SL_ANDROID_SPEAKER_STEREO (SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT)
+#define SL_ANDROID_SPEAKER_QUAD   (SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_BACK_LEFT | SL_SPEAKER_BACK_RIGHT)
+#define SL_ANDROID_SPEAKER_5DOT1  (SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER | SL_SPEAKER_LOW_FREQUENCY)
+#define SL_ANDROID_SPEAKER_7DOT1  (SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_SIDE_LEFT | SL_SPEAKER_SIDE_RIGHT)
+
+// engine interfaces
+static SLObjectItf engineObject = NULL;
+static SLEngineItf engineEngine = NULL;
+
+// output mix interfaces
+static SLObjectItf outputMixObject = NULL;
+
+// buffer queue player interfaces
+static SLObjectItf bqPlayerObject = NULL;
+static SLPlayItf bqPlayerPlay = NULL;
+static SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue = NULL;
+#if 0
+static SLVolumeItf bqPlayerVolume;
+#endif
+
+// recorder interfaces
+static SLObjectItf recorderObject = NULL;
+static SLRecordItf recorderRecord = NULL;
+static SLAndroidSimpleBufferQueueItf recorderBufferQueue = NULL;
+
+#if 0
+static const char *sldevaudiorecorderstr = "SLES Audio Recorder";
+static const char *sldevaudioplayerstr   = "SLES Audio Player";
+
+#define SLES_DEV_AUDIO_RECORDER sldevaudiorecorderstr
+#define SLES_DEV_AUDIO_PLAYER   sldevaudioplayerstr
+static void OPENSLES_DetectDevices( int recording )
+{
+    LOGI( "openSLES_DetectDevices()" );
+    if ( recording )
+            addfn( SLES_DEV_AUDIO_RECORDER );
+    else
+            addfn( SLES_DEV_AUDIO_PLAYER );
+}
+#endif
+
+static void OPENSLES_DestroyEngine(void)
+{
+    LOGI("OPENSLES_DestroyEngine()");
+
+    // destroy output mix object, and invalidate all associated interfaces
+    if (outputMixObject != NULL) {
+        (*outputMixObject)->Destroy(outputMixObject);
+        outputMixObject = NULL;
+    }
+
+    // destroy engine object, and invalidate all associated interfaces
+    if (engineObject != NULL) {
+        (*engineObject)->Destroy(engineObject);
+        engineObject = NULL;
+        engineEngine = NULL;
+    }
+}
+
+static bool OPENSLES_CreateEngine(void)
+{
+    const SLInterfaceID ids[1] = { SL_IID_VOLUME };
+    const SLboolean req[1] = { SL_BOOLEAN_FALSE };
+    SLresult result;
+
+    LOGI("openSLES_CreateEngine()");
+
+    // create engine
+    result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("slCreateEngine failed: %d", result);
+        goto error;
+    }
+    LOGI("slCreateEngine OK");
+
+    // realize the engine
+    result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeEngine failed: %d", result);
+        goto error;
+    }
+    LOGI("RealizeEngine OK");
+
+    // get the engine interface, which is needed in order to create other objects
+    result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("EngineGetInterface failed: %d", result);
+        goto error;
+    }
+    LOGI("EngineGetInterface OK");
+
+    // create output mix
+    result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 1, ids, req);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("CreateOutputMix failed: %d", result);
+        goto error;
+    }
+    LOGI("CreateOutputMix OK");
+
+    // realize the output mix
+    result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeOutputMix failed: %d", result);
+        goto error;
+    }
+    return true;
+
+error:
+    OPENSLES_DestroyEngine();
+    return false;
+}
+
+// this callback handler is called every time a buffer finishes recording
+static void bqRecorderCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
+{
+    struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *)context;
+
+    LOGV("SLES: Recording Callback");
+    SDL_SignalSemaphore(audiodata->playsem);
+}
+
+static void OPENSLES_DestroyPCMRecorder(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+    SLresult result;
+
+    // stop recording
+    if (recorderRecord != NULL) {
+        result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_STOPPED);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("SetRecordState stopped: %d", result);
+        }
+    }
+
+    // destroy audio recorder object, and invalidate all associated interfaces
+    if (recorderObject != NULL) {
+        (*recorderObject)->Destroy(recorderObject);
+        recorderObject = NULL;
+        recorderRecord = NULL;
+        recorderBufferQueue = NULL;
+    }
+
+    if (audiodata->playsem) {
+        SDL_DestroySemaphore(audiodata->playsem);
+        audiodata->playsem = NULL;
+    }
+
+    if (audiodata->mixbuff) {
+        SDL_free(audiodata->mixbuff);
+    }
+}
+
+// !!! FIXME: make this non-blocking!
+static void SDLCALL RequestAndroidPermissionBlockingCallback(void *userdata, const char *permission, bool granted)
+{
+    SDL_SetAtomicInt((SDL_AtomicInt *) userdata, granted ? 1 : -1);
+}
+
+static bool OPENSLES_CreatePCMRecorder(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+    SLDataFormat_PCM format_pcm;
+    SLDataLocator_AndroidSimpleBufferQueue loc_bufq;
+    SLDataSink audioSnk;
+    SLDataLocator_IODevice loc_dev;
+    SLDataSource audioSrc;
+    const SLInterfaceID ids[1] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
+    const SLboolean req[1] = { SL_BOOLEAN_TRUE };
+    SLresult result;
+    int i;
+
+    // !!! FIXME: make this non-blocking!
+    {
+        SDL_AtomicInt permission_response;
+        SDL_SetAtomicInt(&permission_response, 0);
+        if (!SDL_RequestAndroidPermission("android.permission.RECORD_AUDIO", RequestAndroidPermissionBlockingCallback, &permission_response)) {
+            return false;
+        }
+
+        while (SDL_GetAtomicInt(&permission_response) == 0) {
+            SDL_Delay(10);
+        }
+
+        if (SDL_GetAtomicInt(&permission_response) < 0) {
+            LOGE("This app doesn't have RECORD_AUDIO permission");
+            return SDL_SetError("This app doesn't have RECORD_AUDIO permission");
+        }
+    }
+
+    // Just go with signed 16-bit audio as it's the most compatible
+    device->spec.format = SDL_AUDIO_S16;
+    device->spec.channels = 1;
+    //device->spec.freq = SL_SAMPLINGRATE_16 / 1000;*/
+
+    // Update the fragment size as size in bytes
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    LOGI("Try to open %u hz %u bit %u channels %s samples %u",
+         device->spec.freq, SDL_AUDIO_BITSIZE(device->spec.format),
+         device->spec.channels, (device->spec.format & 0x1000) ? "BE" : "LE", device->sample_frames);
+
+    // configure audio source
+    loc_dev.locatorType = SL_DATALOCATOR_IODEVICE;
+    loc_dev.deviceType = SL_IODEVICE_AUDIOINPUT;
+    loc_dev.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT;
+    loc_dev.device = NULL;
+    audioSrc.pLocator = &loc_dev;
+    audioSrc.pFormat = NULL;
+
+    // configure audio sink
+    loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+    loc_bufq.numBuffers = NUM_BUFFERS;
+
+    format_pcm.formatType = SL_DATAFORMAT_PCM;
+    format_pcm.numChannels = device->spec.channels;
+    format_pcm.samplesPerSec = device->spec.freq * 1000; // / kilo Hz to milli Hz
+    format_pcm.bitsPerSample = SDL_AUDIO_BITSIZE(device->spec.format);
+    format_pcm.containerSize = SDL_AUDIO_BITSIZE(device->spec.format);
+    format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+    format_pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
+
+    audioSnk.pLocator = &loc_bufq;
+    audioSnk.pFormat = &format_pcm;
+
+    // create audio recorder
+    // (requires the RECORD_AUDIO permission)
+    result = (*engineEngine)->CreateAudioRecorder(engineEngine, &recorderObject, &audioSrc, &audioSnk, 1, ids, req);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("CreateAudioRecorder failed: %d", result);
+        goto failed;
+    }
+
+    // realize the recorder
+    result = (*recorderObject)->Realize(recorderObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeAudioPlayer failed: %d", result);
+        goto failed;
+    }
+
+    // get the record interface
+    result = (*recorderObject)->GetInterface(recorderObject, SL_IID_RECORD, &recorderRecord);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_RECORD interface get failed: %d", result);
+        goto failed;
+    }
+
+    // get the buffer queue interface
+    result = (*recorderObject)->GetInterface(recorderObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &recorderBufferQueue);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_BUFFERQUEUE interface get failed: %d", result);
+        goto failed;
+    }
+
+    // register callback on the buffer queue
+    // context is '(SDL_PrivateAudioData *)device->hidden'
+    result = (*recorderBufferQueue)->RegisterCallback(recorderBufferQueue, bqRecorderCallback, device->hidden);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RegisterCallback failed: %d", result);
+        goto failed;
+    }
+
+    // Create the audio buffer semaphore
+    audiodata->playsem = SDL_CreateSemaphore(0);
+    if (!audiodata->playsem) {
+        LOGE("cannot create Semaphore!");
+        goto failed;
+    }
+
+    // Create the sound buffers
+    audiodata->mixbuff = (Uint8 *)SDL_malloc(NUM_BUFFERS * device->buffer_size);
+    if (!audiodata->mixbuff) {
+        LOGE("mixbuffer allocate - out of memory");
+        goto failed;
+    }
+
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        audiodata->pmixbuff[i] = audiodata->mixbuff + i * device->buffer_size;
+    }
+
+    // in case already recording, stop recording and clear buffer queue
+    result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_STOPPED);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Record set state failed: %d", result);
+        goto failed;
+    }
+
+    // enqueue empty buffers to be filled by the recorder
+    for (i = 0; i < NUM_BUFFERS; i++) {
+        result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue, audiodata->pmixbuff[i], device->buffer_size);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("Record enqueue buffers failed: %d", result);
+            goto failed;
+        }
+    }
+
+    // start recording
+    result = (*recorderRecord)->SetRecordState(recorderRecord, SL_RECORDSTATE_RECORDING);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Record set state failed: %d", result);
+        goto failed;
+    }
+
+    return true;
+
+failed:
+    return SDL_SetError("Open device failed!");
+}
+
+// this callback handler is called every time a buffer finishes playing
+static void bqPlayerCallback(SLAndroidSimpleBufferQueueItf bq, void *context)
+{
+    struct SDL_PrivateAudioData *audiodata = (struct SDL_PrivateAudioData *)context;
+
+    LOGV("SLES: Playback Callback");
+    SDL_SignalSemaphore(audiodata->playsem);
+}
+
+static void OPENSLES_DestroyPCMPlayer(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    // set the player's state to 'stopped'
+    if (bqPlayerPlay != NULL) {
+        const SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_STOPPED);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("SetPlayState stopped failed: %d", result);
+        }
+    }
+
+    // destroy buffer queue audio player object, and invalidate all associated interfaces
+    if (bqPlayerObject != NULL) {
+        (*bqPlayerObject)->Destroy(bqPlayerObject);
+
+        bqPlayerObject = NULL;
+        bqPlayerPlay = NULL;
+        bqPlayerBufferQueue = NULL;
+    }
+
+    if (audiodata->playsem) {
+        SDL_DestroySemaphore(audiodata->playsem);
+        audiodata->playsem = NULL;
+    }
+
+    if (audiodata->mixbuff) {
+        SDL_free(audiodata->mixbuff);
+    }
+}
+
+static bool OPENSLES_CreatePCMPlayer(SDL_AudioDevice *device)
+{
+    /* If we want to add floating point audio support (requires API level 21)
+       it can be done as described here:
+        https://developer.android.com/ndk/guides/audio/opensl/android-extensions.html#floating-point
+    */
+    if (SDL_GetAndroidSDKVersion() >= 21) {
+        const SDL_AudioFormat *closefmts = SDL_ClosestAudioFormats(device->spec.format);
+        SDL_AudioFormat test_format;
+        while ((test_format = *(closefmts++)) != 0) {
+            if (SDL_AUDIO_ISSIGNED(test_format)) {
+                break;
+            }
+        }
+
+        if (!test_format) {
+            // Didn't find a compatible format :
+            LOGI("No compatible audio format, using signed 16-bit audio");
+            test_format = SDL_AUDIO_S16;
+        }
+        device->spec.format = test_format;
+    } else {
+        // Just go with signed 16-bit audio as it's the most compatible
+        device->spec.format = SDL_AUDIO_S16;
+    }
+
+    // Update the fragment size as size in bytes
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    LOGI("Try to open %u hz %s %u bit %u channels %s samples %u",
+         device->spec.freq, SDL_AUDIO_ISFLOAT(device->spec.format) ? "float" : "pcm", SDL_AUDIO_BITSIZE(device->spec.format),
+         device->spec.channels, (device->spec.format & 0x1000) ? "BE" : "LE", device->sample_frames);
+
+    // configure audio source
+    SLDataLocator_AndroidSimpleBufferQueue loc_bufq;
+    loc_bufq.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
+    loc_bufq.numBuffers = NUM_BUFFERS;
+
+    SLDataFormat_PCM format_pcm;
+    format_pcm.formatType = SL_DATAFORMAT_PCM;
+    format_pcm.numChannels = device->spec.channels;
+    format_pcm.samplesPerSec = device->spec.freq * 1000; // / kilo Hz to milli Hz
+    format_pcm.bitsPerSample = SDL_AUDIO_BITSIZE(device->spec.format);
+    format_pcm.containerSize = SDL_AUDIO_BITSIZE(device->spec.format);
+
+    if (SDL_AUDIO_ISBIGENDIAN(device->spec.format)) {
+        format_pcm.endianness = SL_BYTEORDER_BIGENDIAN;
+    } else {
+        format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
+    }
+
+    switch (device->spec.channels) {
+    case 1:
+        format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT;
+        break;
+    case 2:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_STEREO;
+        break;
+    case 3:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_STEREO | SL_SPEAKER_FRONT_CENTER;
+        break;
+    case 4:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_QUAD;
+        break;
+    case 5:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_QUAD | SL_SPEAKER_FRONT_CENTER;
+        break;
+    case 6:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_5DOT1;
+        break;
+    case 7:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_5DOT1 | SL_SPEAKER_BACK_CENTER;
+        break;
+    case 8:
+        format_pcm.channelMask = SL_ANDROID_SPEAKER_7DOT1;
+        break;
+    default:
+        // Unknown number of channels, fall back to stereo
+        device->spec.channels = 2;
+        format_pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+        break;
+    }
+
+    SLDataSink audioSnk;
+    SLDataSource audioSrc;
+    audioSrc.pFormat = (void *)&format_pcm;
+
+    SLAndroidDataFormat_PCM_EX format_pcm_ex;
+    if (SDL_AUDIO_ISFLOAT(device->spec.format)) {
+        // Copy all setup into PCM EX structure
+        format_pcm_ex.formatType = SL_ANDROID_DATAFORMAT_PCM_EX;
+        format_pcm_ex.endianness = format_pcm.endianness;
+        format_pcm_ex.channelMask = format_pcm.channelMask;
+        format_pcm_ex.numChannels = format_pcm.numChannels;
+        format_pcm_ex.sampleRate = format_pcm.samplesPerSec;
+        format_pcm_ex.bitsPerSample = format_pcm.bitsPerSample;
+        format_pcm_ex.containerSize = format_pcm.containerSize;
+        format_pcm_ex.representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
+        audioSrc.pFormat = (void *)&format_pcm_ex;
+    }
+
+    audioSrc.pLocator = &loc_bufq;
+
+    // configure audio sink
+    SLDataLocator_OutputMix loc_outmix;
+    loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
+    loc_outmix.outputMix = outputMixObject;
+    audioSnk.pLocator = &loc_outmix;
+    audioSnk.pFormat = NULL;
+
+    // create audio player
+    const SLInterfaceID ids[2] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE, SL_IID_VOLUME };
+    const SLboolean req[2] = { SL_BOOLEAN_TRUE, SL_BOOLEAN_FALSE };
+    SLresult result;
+    result = (*engineEngine)->CreateAudioPlayer(engineEngine, &bqPlayerObject, &audioSrc, &audioSnk, 2, ids, req);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("CreateAudioPlayer failed: %d", result);
+        goto failed;
+    }
+
+    // realize the player
+    result = (*bqPlayerObject)->Realize(bqPlayerObject, SL_BOOLEAN_FALSE);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RealizeAudioPlayer failed: %d", result);
+        goto failed;
+    }
+
+    // get the play interface
+    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_PLAY, &bqPlayerPlay);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_PLAY interface get failed: %d", result);
+        goto failed;
+    }
+
+    // get the buffer queue interface
+    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &bqPlayerBufferQueue);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_BUFFERQUEUE interface get failed: %d", result);
+        goto failed;
+    }
+
+    // register callback on the buffer queue
+    // context is '(SDL_PrivateAudioData *)device->hidden'
+    result = (*bqPlayerBufferQueue)->RegisterCallback(bqPlayerBufferQueue, bqPlayerCallback, device->hidden);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("RegisterCallback failed: %d", result);
+        goto failed;
+    }
+
+#if 0
+    // get the volume interface
+    result = (*bqPlayerObject)->GetInterface(bqPlayerObject, SL_IID_VOLUME, &bqPlayerVolume);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("SL_IID_VOLUME interface get failed: %d", result);
+        // goto failed;
+    }
+#endif
+
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    // Create the audio buffer semaphore
+    audiodata->playsem = SDL_CreateSemaphore(NUM_BUFFERS - 1);
+    if (!audiodata->playsem) {
+        LOGE("cannot create Semaphore!");
+        goto failed;
+    }
+
+    // Create the sound buffers
+    audiodata->mixbuff = (Uint8 *)SDL_malloc(NUM_BUFFERS * device->buffer_size);
+    if (!audiodata->mixbuff) {
+        LOGE("mixbuffer allocate - out of memory");
+        goto failed;
+    }
+
+    for (int i = 0; i < NUM_BUFFERS; i++) {
+        audiodata->pmixbuff[i] = audiodata->mixbuff + i * device->buffer_size;
+    }
+
+    // set the player's state to playing
+    result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Play set state failed: %d", result);
+        goto failed;
+    }
+
+    return true;
+
+failed:
+    return false;
+}
+
+static bool OPENSLES_OpenDevice(SDL_AudioDevice *device)
+{
+    device->hidden = (struct SDL_PrivateAudioData *)SDL_calloc(1, sizeof(*device->hidden));
+    if (!device->hidden) {
+        return false;
+    }
+
+    if (device->recording) {
+        LOGI("OPENSLES_OpenDevice() for recording");
+        return OPENSLES_CreatePCMRecorder(device);
+    } else {
+        bool ret;
+        LOGI("OPENSLES_OpenDevice() for playback");
+        ret = OPENSLES_CreatePCMPlayer(device);
+        if (!ret) {
+            // Another attempt to open the device with a lower frequency
+            if (device->spec.freq > 48000) {
+                OPENSLES_DestroyPCMPlayer(device);
+                device->spec.freq = 48000;
+                ret = OPENSLES_CreatePCMPlayer(device);
+            }
+        }
+
+        if (!ret) {
+            return SDL_SetError("Open device failed!");
+        }
+    }
+
+    return true;
+}
+
+static bool OPENSLES_WaitDevice(SDL_AudioDevice *device)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    LOGV("OPENSLES_WaitDevice()");
+
+    while (!SDL_GetAtomicInt(&device->shutdown)) {
+        // this semaphore won't fire when the app is in the background (OPENSLES_PauseDevices was called).
+        if (SDL_WaitSemaphoreTimeout(audiodata->playsem, 100)) {
+            return true;  // semaphore was signaled, let's go!
+        }
+        // Still waiting on the semaphore (or the system), check other things then wait again.
+    }
+    return true;
+}
+
+static bool OPENSLES_PlayDevice(SDL_AudioDevice *device, const Uint8 *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    LOGV("======OPENSLES_PlayDevice()======");
+
+    // Queue it up
+    const SLresult result = (*bqPlayerBufferQueue)->Enqueue(bqPlayerBufferQueue, buffer, buflen);
+
+    audiodata->next_buffer++;
+    if (audiodata->next_buffer >= NUM_BUFFERS) {
+        audiodata->next_buffer = 0;
+    }
+
+    // If Enqueue fails, callback won't be called.
+    // Post the semaphore, not to run out of buffer
+    if (SL_RESULT_SUCCESS != result) {
+        SDL_SignalSemaphore(audiodata->playsem);
+    }
+
+    return true;
+}
+
+///           n   playn sem
+// getbuf     0   -     1
+// fill buff  0   -     1
+// play       0 - 0     1
+// wait       1   0     0
+// getbuf     1   0     0
+// fill buff  1   0     0
+// play       0   0     0
+// wait
+//
+// okay..
+
+static Uint8 *OPENSLES_GetDeviceBuf(SDL_AudioDevice *device, int *bufsize)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    LOGV("OPENSLES_GetDeviceBuf()");
+    return audiodata->pmixbuff[audiodata->next_buffer];
+}
+
+static int OPENSLES_RecordDevice(SDL_AudioDevice *device, void *buffer, int buflen)
+{
+    struct SDL_PrivateAudioData *audiodata = device->hidden;
+
+    // Copy it to the output buffer
+    SDL_assert(buflen == device->buffer_size);
+    SDL_memcpy(buffer, audiodata->pmixbuff[audiodata->next_buffer], device->buffer_size);
+
+    // Re-enqueue the buffer
+    const SLresult result = (*recorderBufferQueue)->Enqueue(recorderBufferQueue, audiodata->pmixbuff[audiodata->next_buffer], device->buffer_size);
+    if (SL_RESULT_SUCCESS != result) {
+        LOGE("Record enqueue buffers failed: %d", result);
+        return -1;
+    }
+
+    audiodata->next_buffer++;
+    if (audiodata->next_buffer >= NUM_BUFFERS) {
+        audiodata->next_buffer = 0;
+    }
+
+    return device->buffer_size;
+}
+
+static void OPENSLES_CloseDevice(SDL_AudioDevice *device)
+{
+    // struct SDL_PrivateAudioData *audiodata = device->hidden;
+    if (device->hidden) {
+        if (device->recording) {
+            LOGI("OPENSLES_CloseDevice() for recording");
+            OPENSLES_DestroyPCMRecorder(device);
+        } else {
+            LOGI("OPENSLES_CloseDevice() for playing");
+            OPENSLES_DestroyPCMPlayer(device);
+        }
+
+        SDL_free(device->hidden);
+        device->hidden = NULL;
+    }
+}
+
+static bool OPENSLES_Init(SDL_AudioDriverImpl *impl)
+{
+    LOGI("OPENSLES_Init() called");
+
+    if (!OPENSLES_CreateEngine()) {
+        return false;
+    }
+
+    LOGI("OPENSLES_Init() - set pointers");
+
+    // Set the function pointers
+    // impl->DetectDevices = OPENSLES_DetectDevices;
+    impl->ThreadInit = Android_AudioThreadInit;
+    impl->OpenDevice = OPENSLES_OpenDevice;
+    impl->WaitDevice = OPENSLES_WaitDevice;
+    impl->PlayDevice = OPENSLES_PlayDevice;
+    impl->GetDeviceBuf = OPENSLES_GetDeviceBuf;
+    impl->WaitRecordingDevice = OPENSLES_WaitDevice;
+    impl->RecordDevice = OPENSLES_RecordDevice;
+    impl->CloseDevice = OPENSLES_CloseDevice;
+    impl->Deinitialize = OPENSLES_DestroyEngine;
+
+    // and the capabilities
+    impl->HasRecordingSupport = true;
+    impl->OnlyHasDefaultPlaybackDevice = true;
+    impl->OnlyHasDefaultRecordingDevice = true;
+
+    LOGI("OPENSLES_Init() - success");
+
+    // this audio target is available.
+    return true;
+}
+
+AudioBootStrap OPENSLES_bootstrap = {
+    "openslES", "OpenSL ES audio driver", OPENSLES_Init, false, false
+};
+
+void OPENSLES_ResumeDevices(void)
+{
+    if (bqPlayerPlay != NULL) {
+        // set the player's state to 'playing'
+        SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PLAYING);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("OPENSLES_ResumeDevices failed: %d", result);
+        }
+    }
+}
+
+void OPENSLES_PauseDevices(void)
+{
+    if (bqPlayerPlay != NULL) {
+        // set the player's state to 'paused'
+        SLresult result = (*bqPlayerPlay)->SetPlayState(bqPlayerPlay, SL_PLAYSTATE_PAUSED);
+        if (SL_RESULT_SUCCESS != result) {
+            LOGE("OPENSLES_PauseDevices failed: %d", result);
+        }
+    }
+}
+
+#endif // SDL_AUDIO_DRIVER_OPENSLES
diff --git a/contrib/SDL-3.2.8/src/audio/openslES/SDL_openslES.h b/contrib/SDL-3.2.8/src/audio/openslES/SDL_openslES.h
new file mode 100644
index 0000000..0ae2664
--- /dev/null
+++ b/contrib/SDL-3.2.8/src/audio/openslES/SDL_openslES.h
@@ -0,0 +1,38 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 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"
+
+#ifndef SDL_openslesaudio_h_
+#define SDL_openslesaudio_h_
+
+#ifdef SDL_AUDIO_DRIVER_OPENSLES
+
+extern void OPENSLES_ResumeDevices(void);
+extern void OPENSLES_PauseDevices(void);
+
+#else
+
+#define OPENSLES_ResumeDevices()
+#define OPENSLES_PauseDevices()
+
+#endif
+
+#endif // SDL_openslesaudio_h_