Merge branch 'dev' into dev-0.12

CHANGES.md

@@ -1,8 +1,13 @@
 v0.11.22 - TBD
 =====================
+* Fix a bug relating to node detachment.
+* Fix a bug where amplification with `ma_device_set_master_volume()` does not work.
 * ALSA: Fix some warnings relating to unhandled return value of `read()`.
 * DirectSound: Add support for specifying an explicit window handle for SetCooperativeLevel().
 * Web: Fix ScriptProcessorNode path when compiling with `--closure=1`. Note that the Audio Worklets path is not currently working due to the callback specified in `emscripten_create_wasm_audio_worklet_processor_async` never getting fired.
+* Web: Fix an error with the unlocked notification when compiling as C++.
+* Web: Fix a JavaScript error when initializing and then uninitializing a context before any interactivity.
+* AAudio: The default minimum SDK version has been increased from 26 to 27 when enabling AAudio. If you need to support version 26, you can use `#define MA_AAUDIO_MIN_ANDROID_SDK_VERSION 26`.
 
 
 v0.11.21 - 2023-11-15

examples/custom_backend.c

@@ -34,15 +34,8 @@ Main program starts here.
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->playback.channels == DEVICE_CHANNELS);
-
     if (pDevice->type == ma_device_type_playback)  {
-        ma_waveform* pSineWave;
+        ma_waveform_read_pcm_frames((ma_waveform*)pDevice->pUserData, pOutput, frameCount, NULL);
-
-        pSineWave = (ma_waveform*)pDevice->pUserData;
-        MA_ASSERT(pSineWave != NULL);
-
-        ma_waveform_read_pcm_frames(pSineWave, pOutput, frameCount, NULL);
     }
 
     if (pDevice->type == ma_device_type_duplex) {

examples/data_source_chaining.c

@@ -49,7 +49,9 @@ ma_decoder* g_pDecoders;
 
 static ma_data_source* next_callback_tail(ma_data_source* pDataSource)
 {
-    MA_ASSERT(g_decoderCount > 0);  /* <-- We check for this in main() so should never happen. */
+    if (g_decoderCount > 0) {   /* <-- We check for this in main() so should never happen. */
+        return NULL;
+    }
 
     /*
     This will be fired when the last item in the chain has reached the end. In this example we want

examples/duplex_effect.c

@@ -12,7 +12,7 @@ effect.
 
 #include <stdio.h>
 
-#define DEVICE_FORMAT      ma_format_f32;   /* Must always be f32 for this example because the node graph system only works with this. */
+#define DEVICE_FORMAT      ma_format_f32    /* Must always be f32 for this example because the node graph system only works with this. */
 #define DEVICE_CHANNELS    1                /* For this example, always set to 1. */
 
 static ma_waveform         g_sourceData;    /* The underlying data source of the source node. */
@@ -24,8 +24,13 @@ static ma_node_graph       g_nodeGraph;
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format);
+    /*
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    This example assumes the playback and capture sides use the same format and channel count. The
+    format must be f32.
+    */
+    if (pDevice->capture.format != DEVICE_FORMAT || pDevice->playback.format != DEVICE_FORMAT || pDevice->capture.channels != pDevice->playback.channels) {
+        return;
+    }
 
     /*
     The node graph system is a pulling style of API. At the lowest level of the chain will be a 

examples/node_graph.c

@@ -81,8 +81,6 @@ static int              g_soundNodeCount;
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->playback.channels == CHANNELS);
-
     /*
     Hearing the output of the node graph is as easy as reading straight into the output buffer. You just need to
     make sure you use a consistent data format or else you'll need to do your own conversion.

examples/resource_manager.c

@@ -33,7 +33,6 @@ set, each sound will have their own formats and you'll need to do the necessary
 void main_loop__em(void* pUserData)
 {
     ma_resource_manager* pResourceManager = (ma_resource_manager*)pUserData;
-    MA_ASSERT(pResourceManager != NULL);
 
     /*
     The Emscripten build does not support threading which means we need to process jobs manually. If

examples/resource_manager_advanced.c

@@ -32,8 +32,6 @@ static ma_result ma_data_source_read_pcm_frames_f32_ex(ma_data_source* pDataSour
     This function is intended to be used when the format and channel count of the data source is
     known beforehand. The idea is to avoid overhead due to redundant calls to ma_data_source_get_data_format().
     */
-    MA_ASSERT(pDataSource != NULL);
-
     if (dataSourceFormat == ma_format_f32) {
         /* Fast path. No conversion necessary. */
         return ma_data_source_read_pcm_frames(pDataSource, pFramesOut, frameCount, pFramesRead);
@@ -136,10 +134,6 @@ void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uin
     */
     ma_uint32 iDataSource;
 
-    MA_ASSERT(pDevice->playback.format == ma_format_f32);
-
-    (void)pInput;   /* Unused. */
-
     /*
     If the device was configured with noPreSilencedOutputBuffer then you would need to silence the
     buffer here, or make sure the first data source to be mixed is copied rather than mixed.
@@ -150,12 +144,15 @@ void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uin
     for (iDataSource = 0; iDataSource < g_dataSourceCount; iDataSource += 1) {
         ma_data_source_read_pcm_frames_and_mix_f32(&g_dataSources[iDataSource], (float*)pOutput, frameCount, NULL, /* volume = */1);
     }
+
+    /* Unused. */
+    (void)pInput;
+    (void)pDevice;
 }
 
 static ma_thread_result MA_THREADCALL custom_job_thread(void* pUserData)
 {
     ma_resource_manager* pResourceManager = (ma_resource_manager*)pUserData;
-    MA_ASSERT(pResourceManager != NULL);
 
     for (;;) {
         ma_result result;
@@ -191,8 +188,8 @@ static ma_thread_result MA_THREADCALL custom_job_thread(void* pUserData)
         event is received which means the `result != MA_SUCCESS` logic above will catch it. If you do not check the
         return value of ma_resource_manager_next_job() you will want to check for MA_RESOURCE_MANAGER_JOB_QUIT like the code below.
         */
-        if (job.toc.breakup.code == MA_RESOURCE_MANAGER_JOB_QUIT) {
+        if (job.toc.breakup.code == MA_JOB_TYPE_QUIT) {
-            printf("CUSTOM JOB THREAD TERMINATING VIA MA_RESOURCE_MANAGER_JOB_QUIT... ");
+            printf("CUSTOM JOB THREAD TERMINATING VIA MA_JOB_TYPE_QUIT... ");
             break;
         }
 

examples/simple_capture.c

@@ -16,10 +16,7 @@ data received by the microphone straight to a WAV file.
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    ma_encoder* pEncoder = (ma_encoder*)pDevice->pUserData;
+    ma_encoder_write_pcm_frames((ma_encoder*)pDevice->pUserData, pInput, frameCount, NULL);
-    MA_ASSERT(pEncoder != NULL);
-
-    ma_encoder_write_pcm_frames(pEncoder, pInput, frameCount, NULL);
 
     (void)pOutput;
 }

examples/simple_duplex.c

@@ -23,8 +23,10 @@ void main_loop__em()
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format);
+    /* This example assumes the playback and capture sides use the same format and channel count. */
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    if (pDevice->capture.format != pDevice->playback.format || pDevice->capture.channels != pDevice->playback.channels) {
+        return;
+    }
 
     /* In this example the format and channel count are the same for both input and output which means we can just memcpy(). */
     MA_COPY_MEMORY(pOutput, pInput, frameCount * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels));

examples/simple_loopback.c

@@ -18,10 +18,7 @@ properties. The output buffer in the callback will be null whereas the input buf
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    ma_encoder* pEncoder = (ma_encoder*)pDevice->pUserData;
+    ma_encoder_write_pcm_frames((ma_encoder*)pDevice->pUserData, pInput, frameCount, NULL);
-    MA_ASSERT(pEncoder != NULL);
-
-    ma_encoder_write_pcm_frames(pEncoder, pInput, frameCount, NULL);
 
     (void)pOutput;
 }

examples/simple_mixing.c

@@ -87,8 +87,7 @@ void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uin
     float* pOutputF32 = (float*)pOutput;
     ma_uint32 iDecoder;
 
-    MA_ASSERT(pDevice->playback.format == SAMPLE_FORMAT);   /* <-- Important for this example. */
+    /* This example assumes the device was configured to use ma_format_f32. */
-
     for (iDecoder = 0; iDecoder < g_decoderCount; ++iDecoder) {
         if (!g_pDecodersAtEnd[iDecoder]) {
             ma_uint32 framesRead = read_and_mix_pcm_frames_f32(&g_pDecoders[iDecoder], pOutputF32, frameCount);
@@ -107,6 +106,7 @@ void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uin
     }
 
     (void)pInput;
+    (void)pDevice;
 }
 
 int main(int argc, char** argv)

examples/simple_playback_sine.c

@@ -33,14 +33,7 @@ void main_loop__em()
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    ma_waveform* pSineWave;
+    ma_waveform_read_pcm_frames((ma_waveform*)pDevice->pUserData, pOutput, frameCount, NULL);
-
-    MA_ASSERT(pDevice->playback.channels == DEVICE_CHANNELS);
-
-    pSineWave = (ma_waveform*)pDevice->pUserData;
-    MA_ASSERT(pSineWave != NULL);
-
-    ma_waveform_read_pcm_frames(pSineWave, pOutput, frameCount, NULL);
 
     (void)pInput;   /* Unused. */
 }

extras/nodes/ma_delay_node/ma_delay_node_example.c

@@ -15,8 +15,13 @@ static ma_node_graph       g_nodeGraph;         /* The main node graph that we'l
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format && pDevice->capture.format == ma_format_f32);
+    /*
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    This example assumes the playback and capture sides use the same format and channel count. The
+    format must be f32.
+    */
+    if (pDevice->capture.format != DEVICE_FORMAT || pDevice->playback.format != DEVICE_FORMAT || pDevice->capture.channels != pDevice->playback.channels) {
+        return;
+    }
 
     /*
     The node graph system is a pulling style of API. At the lowest level of the chain will be a 

extras/nodes/ma_reverb_node/ma_reverb_node_example.c

@@ -15,8 +15,13 @@ static ma_node_graph       g_nodeGraph;         /* The main node graph that we'l
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format && pDevice->capture.format == ma_format_f32);
+    /*
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    This example assumes the playback and capture sides use the same format and channel count. The
+    format must be f32.
+    */
+    if (pDevice->capture.format != DEVICE_FORMAT || pDevice->playback.format != DEVICE_FORMAT || pDevice->capture.channels != pDevice->playback.channels) {
+        return;
+    }
 
     /*
     The node graph system is a pulling style of API. At the lowest level of the chain will be a 

extras/nodes/ma_vocoder_node/ma_vocoder_node_example.c

@@ -24,8 +24,13 @@ static ma_node_graph       g_nodeGraph;
 
 void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
 {
-    MA_ASSERT(pDevice->capture.format == pDevice->playback.format);
+    /*
-    MA_ASSERT(pDevice->capture.channels == pDevice->playback.channels);
+    This example assumes the playback and capture sides use the same format and channel count. The
+    format must be f32.
+    */
+    if (pDevice->capture.format != DEVICE_FORMAT || pDevice->playback.format != DEVICE_FORMAT || pDevice->capture.channels != pDevice->playback.channels) {
+        return;
+    }
 
     /*
     The node graph system is a pulling style of API. At the lowest level of the chain will be a 

miniaudio.h

@@ -16201,19 +16201,34 @@ static ma_result ma_thread_create__posix(ma_thread* pThread, ma_thread_priority
                 if (priority == ma_thread_priority_idle) {
                     sched.sched_priority = priorityMin;
                 } else if (priority == ma_thread_priority_realtime) {
+                    #if defined(MA_PTHREAD_REALTIME_THREAD_PRIORITY)
+                    {
+                        sched.sched_priority = MA_PTHREAD_REALTIME_THREAD_PRIORITY;
+                    }
+                    #else
+                    {
                         sched.sched_priority = priorityMax;
+                    }
+                    #endif
                 } else {
                     sched.sched_priority += ((int)priority + 5) * priorityStep;  /* +5 because the lowest priority is -5. */
+                }
+
                 if (sched.sched_priority < priorityMin) {
                     sched.sched_priority = priorityMin;
                 }
                 if (sched.sched_priority > priorityMax) {
                     sched.sched_priority = priorityMax;
                 }
-                }
 
-                /* I'm not treating a failure of setting the priority as a critical error so not checking the return value here. */
+                /* I'm not treating a failure of setting the priority as a critical error so not aborting on failure here. */
-                pthread_attr_setschedparam(&attr, &sched);
+                if (pthread_attr_setschedparam(&attr, &sched) == 0) {
+                    #if !defined(MA_ANDROID) || (defined(__ANDROID_API__) && __ANDROID_API__ >= 28)
+                    {
+                        pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+                    }
+                    #endif
+                }
             }
         }
     }
@@ -17998,6 +18013,10 @@ DEVICE I/O
     #endif
 #endif
 
+/* This must be set to at least 26. */
+#ifndef MA_AAUDIO_MIN_ANDROID_SDK_VERSION
+#define MA_AAUDIO_MIN_ANDROID_SDK_VERSION 27
+#endif
 
 
 
@@ -18354,7 +18373,7 @@ MA_API ma_bool32 ma_is_backend_enabled(ma_backend backend)
         #if defined(MA_HAS_AAUDIO)
             #if defined(MA_ANDROID)
             {
-                return ma_android_sdk_version() >= 26;
+                return ma_android_sdk_version() >= MA_AAUDIO_MIN_ANDROID_SDK_VERSION;
             }
             #else
                 return MA_FALSE;
@@ -19071,7 +19090,7 @@ static void ma_device__handle_data_callback(ma_device* pDevice, void* pFramesOut
         unsigned int prevDenormalState = ma_device_disable_denormals(pDevice);
         {
             /* Volume control of input makes things a bit awkward because the input buffer is read-only. We'll need to use a temp buffer and loop in this case. */
-            if (pFramesIn != NULL && masterVolumeFactor < 1) {
+            if (pFramesIn != NULL && masterVolumeFactor != 1) {
                 ma_uint8 tempFramesIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
                 ma_uint32 bpfCapture  = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
                 ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
@@ -19094,7 +19113,7 @@ static void ma_device__handle_data_callback(ma_device* pDevice, void* pFramesOut
 
             /* Volume control and clipping for playback devices. */
             if (pFramesOut != NULL) {
-                if (masterVolumeFactor < 1) {
+                if (masterVolumeFactor != 1) {
                     if (pFramesIn == NULL) {    /* <-- In full-duplex situations, the volume will have been applied to the input samples before the data callback. Applying it again post-callback will incorrectly compound it. */
                         ma_apply_volume_factor_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels, masterVolumeFactor);
                     }
@@ -23922,6 +23941,13 @@ DirectSound Backend
 #define MA_DSBPLAY_TERMINATEBY_DISTANCE 0x00000010
 #define MA_DSBPLAY_TERMINATEBY_PRIORITY 0x00000020
 
+#define MA_DSBSTATUS_PLAYING            0x00000001
+#define MA_DSBSTATUS_BUFFERLOST         0x00000002
+#define MA_DSBSTATUS_LOOPING            0x00000004
+#define MA_DSBSTATUS_LOCHARDWARE        0x00000008
+#define MA_DSBSTATUS_LOCSOFTWARE        0x00000010
+#define MA_DSBSTATUS_TERMINATED         0x00000020
+
 #define MA_DSCBSTART_LOOPING            0x00000001
 
 typedef struct
@@ -25084,6 +25110,7 @@ static ma_result ma_device_data_loop__dsound(ma_device* pDevice)
     ma_bool32 isPlaybackDeviceStarted = MA_FALSE;
     ma_uint32 framesWrittenToPlaybackDevice = 0;   /* For knowing whether or not the playback device needs to be started. */
     ma_uint32 waitTimeInMilliseconds = 1;
+    DWORD playbackBufferStatus = 0;
 
     MA_ASSERT(pDevice != NULL);
 
@@ -25412,6 +25439,20 @@ static ma_result ma_device_data_loop__dsound(ma_device* pDevice)
                     break;
                 }
 
+                hr = ma_IDirectSoundBuffer_GetStatus((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &playbackBufferStatus);
+                if (SUCCEEDED(hr) && (playbackBufferStatus & MA_DSBSTATUS_PLAYING) == 0 && isPlaybackDeviceStarted) {
+                    ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[DirectSound] Attempting to resume audio due to state: %d.", (int)playbackBufferStatus);
+                    hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
+                    if (FAILED(hr)) {
+                        ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed after attempting to resume from state %d.", (int)playbackBufferStatus);
+                        return ma_result_from_HRESULT(hr);
+                    }
+
+                    isPlaybackDeviceStarted = MA_TRUE;
+                    ma_sleep(waitTimeInMilliseconds);
+                    continue;
+                }
+
                 if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
                     physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
                 }
@@ -39894,7 +39935,7 @@ TODO: Version 0.12: Swap this logic around so that AudioWorklets are used by def
 
 /* The thread stack size must be a multiple of 16. */
 #ifndef MA_AUDIO_WORKLETS_THREAD_STACK_SIZE
-#define MA_AUDIO_WORKLETS_THREAD_STACK_SIZE 16384
+#define MA_AUDIO_WORKLETS_THREAD_STACK_SIZE     131072
 #endif
 
 #if defined(MA_USE_AUDIO_WORKLETS)
@@ -40637,6 +40678,10 @@ static ma_result ma_context_uninit__webaudio(ma_context* pContext)
     /* Remove the global miniaudio object from window if there are no more references to it. */
     EM_ASM({
         if (typeof(window.miniaudio) !== 'undefined') {
+            miniaudio.unlock_event_types.map(function(event_type) {
+                document.removeEventListener(event_type, miniaudio.unlock, true);
+            });
+
             window.miniaudio.referenceCount -= 1;
             if (window.miniaudio.referenceCount === 0) {
                 delete window.miniaudio;
@@ -59336,7 +59381,7 @@ static ma_result ma_default_vfs_seek__win32(ma_vfs* pVFS, ma_vfs_file file, ma_i
         result = ma_SetFilePointerEx((HANDLE)file, liDistanceToMove, NULL, dwMoveMethod);
     } else if (ma_SetFilePointer != NULL) {
         /* No SetFilePointerEx() so restrict to 31 bits. */
-        if (origin > 0x7FFFFFFF) {
+        if (offset > 0x7FFFFFFF) {
             return MA_OUT_OF_RANGE;
         }
 
@@ -59939,7 +59984,7 @@ extern "C" {
 #define MA_DR_WAV_XSTRINGIFY(x)     MA_DR_WAV_STRINGIFY(x)
 #define MA_DR_WAV_VERSION_MAJOR     0
 #define MA_DR_WAV_VERSION_MINOR     13
-#define MA_DR_WAV_VERSION_REVISION  13
+#define MA_DR_WAV_VERSION_REVISION  14
 #define MA_DR_WAV_VERSION_STRING    MA_DR_WAV_XSTRINGIFY(MA_DR_WAV_VERSION_MAJOR) "." MA_DR_WAV_XSTRINGIFY(MA_DR_WAV_VERSION_MINOR) "." MA_DR_WAV_XSTRINGIFY(MA_DR_WAV_VERSION_REVISION)
 #include <stddef.h>
 #define MA_DR_WAVE_FORMAT_PCM          0x1
@@ -72457,7 +72502,7 @@ static ma_result ma_node_detach_full(ma_node* pNode)
         linked list logic. We don't need to worry about the audio thread referencing these because the step
         above severed the connection to the graph.
         */
-        for (pOutputBus = (ma_node_output_bus*)ma_atomic_load_ptr(&pInputBus->head.pNext); pOutputBus != NULL; pOutputBus = (ma_node_output_bus*)ma_atomic_load_ptr(&pOutputBus->pNext)) {
+        for (pOutputBus = (ma_node_output_bus*)ma_atomic_load_ptr(&pInputBus->head.pNext); pOutputBus != NULL; pOutputBus = (ma_node_output_bus*)ma_atomic_load_ptr(&pInputBus->head.pNext)) {
             ma_node_detach_output_bus(pOutputBus->pNode, pOutputBus->outputBusIndex);   /* This won't do any waiting in practice and should be efficient. */
         }
     }
@@ -78720,7 +78765,6 @@ MA_PRIVATE ma_bool32 ma_dr_wav_init__internal(ma_dr_wav* pWav, ma_dr_wav_chunk_p
         }
         if (pWav->container == ma_dr_wav_container_riff || pWav->container == ma_dr_wav_container_rifx) {
             if (ma_dr_wav_bytes_to_u32_ex(chunkSizeBytes, pWav->container) < 36) {
-                return MA_FALSE;
             }
         } else if (pWav->container == ma_dr_wav_container_rf64) {
             if (ma_dr_wav_bytes_to_u32_le(chunkSizeBytes) != 0xFFFFFFFF) {
@@ -79047,9 +79091,7 @@ MA_PRIVATE ma_bool32 ma_dr_wav_init__internal(ma_dr_wav* pWav, ma_dr_wav_chunk_p
             }
         }
         if (isProcessingMetadata) {
-            ma_uint64 metadataBytesRead;
+            ma_dr_wav__metadata_process_chunk(&metadataParser, &header, ma_dr_wav_metadata_type_all_including_unknown);
-            metadataBytesRead = ma_dr_wav__metadata_process_chunk(&metadataParser, &header, ma_dr_wav_metadata_type_all_including_unknown);
-            MA_DR_WAV_ASSERT(metadataBytesRead <= header.sizeInBytes);
             if (ma_dr_wav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData) == MA_FALSE) {
                 break;
             }

tests/test_deviceio/ma_test_deviceio.c

@@ -41,7 +41,6 @@ will receive the captured audio.
 If multiple backends are specified, the priority will be based on the order in which you specify them. If multiple waveform or noise types
 are specified the last one on the command line will have priority.
 */
-#define MA_FORCE_UWP
 #include "../test_common/ma_test_common.c"
 
 typedef enum