Attempt to fix ubsan issue under macOS #295

miniaudio.h

@@ -1901,7 +1901,7 @@ typedef enum
     ma_standard_sample_rate_32000  = 32000,     /* Lows */
     ma_standard_sample_rate_24000  = 24000,
     ma_standard_sample_rate_22050  = 22050,
-    
+
     ma_standard_sample_rate_88200  = 88200,     /* Highs */
     ma_standard_sample_rate_96000  = 96000,
     ma_standard_sample_rate_176400 = 176400,
@@ -6860,7 +6860,7 @@ static MA_INLINE void ma_yield()
         #else
             #if defined(__DMC__)
                 /* Digital Mars does not recognize the PAUSE opcode. Fall back to NOP. */
-                __asm nop;  
+                __asm nop;
             #else
                 __asm pause;
             #endif
@@ -13611,7 +13611,7 @@ static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDeviceStateChanged(m
     ma_bool32 isThisDevice = MA_FALSE;
     ma_bool32 isCapture    = MA_FALSE;
     ma_bool32 isPlayback   = MA_FALSE;
-    
+
 
 #ifdef MA_DEBUG_OUTPUT
     printf("IMMNotificationClient_OnDeviceStateChanged(pDeviceID=%S, dwNewState=%u)\n", (pDeviceID != NULL) ? pDeviceID : L"(NULL)", (unsigned int)dwNewState);
@@ -13783,7 +13783,7 @@ static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDefaultDeviceChanged
 
                 if (pThis->pDevice->wasapi.isDetachedPlayback) {
                     pThis->pDevice->wasapi.isDetachedPlayback = MA_FALSE;
-                    
+
                     if (pThis->pDevice->type == ma_device_type_duplex && pThis->pDevice->wasapi.isDetachedCapture) {
                         restartDevice = MA_FALSE;   /* It's a duplex device and the capture side is detached. We cannot be restarting the device just yet. */
                     } else {
@@ -13795,7 +13795,7 @@ static HRESULT STDMETHODCALLTYPE ma_IMMNotificationClient_OnDefaultDeviceChanged
 
                 if (pThis->pDevice->wasapi.isDetachedCapture) {
                     pThis->pDevice->wasapi.isDetachedCapture = MA_FALSE;
-                    
+
                     if (pThis->pDevice->type == ma_device_type_duplex && pThis->pDevice->wasapi.isDetachedPlayback) {
                         restartDevice = MA_FALSE;   /* It's a duplex device and the playback side is detached. We cannot be restarting the device just yet. */
                     } else {
@@ -13851,7 +13851,7 @@ typedef ma_IUnknown ma_WASAPIDeviceInterface;
 static ma_context_command__wasapi ma_context_init_command__wasapi(int code)
 {
     ma_context_command__wasapi cmd;
-    
+
     MA_ZERO_OBJECT(&cmd);
     cmd.code = code;
 
@@ -13896,7 +13896,7 @@ static ma_result ma_context_post_command__wasapi(ma_context* pContext, const ma_
         /* Now that the command has been added, release the semaphore so ma_context_next_command__wasapi() can return. */
         ma_semaphore_release(&pContext->wasapi.commandSem);
     }
-    ma_mutex_unlock(&pContext->wasapi.commandLock);    
+    ma_mutex_unlock(&pContext->wasapi.commandLock);
 
     if (isUsingLocalEvent) {
         ma_event_wait(&localEvent);
@@ -13922,7 +13922,7 @@ static ma_result ma_context_next_command__wasapi(ma_context* pContext, ma_contex
             pContext->wasapi.commandCount -= 1;
         }
         ma_mutex_unlock(&pContext->wasapi.commandLock);
-    }    
+    }
 
     return result;
 }
@@ -13958,7 +13958,7 @@ static ma_thread_result MA_THREADCALL ma_context_command_thread__wasapi(void* pU
 
             case MA_CONTEXT_COMMAND_RELEASE_IAUDIOCLIENT__WASAPI:
             {
-                if (cmd.data.releaseAudioClient.deviceType == ma_device_type_playback) { 
+                if (cmd.data.releaseAudioClient.deviceType == ma_device_type_playback) {
                     if (cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback != NULL) {
                         ma_IAudioClient_Release((ma_IAudioClient*)cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback);
                         cmd.data.releaseAudioClient.pDevice->wasapi.pAudioClientPlayback = NULL;
@@ -14000,7 +14000,7 @@ static ma_result ma_device_create_IAudioClient_service__wasapi(ma_context* pCont
     cmd.data.createAudioClient.pAudioClient         = (void*)pAudioClient;
     cmd.data.createAudioClient.ppAudioClientService = ppAudioClientService;
     cmd.data.createAudioClient.result               = MA_SUCCESS;
-    
+
     result = ma_context_post_command__wasapi(pContext, &cmd);  /* This will not return until the command has actually been run. */
     if (result != MA_SUCCESS) {
         return result;
@@ -15047,7 +15047,7 @@ static ma_result ma_device_init_internal__wasapi(ma_context* pContext, ma_device
 
     pData->usingAudioClient3 = wasInitializedUsingIAudioClient3;
 
-    
+
     if (deviceType == ma_device_type_playback) {
         result = ma_device_create_IAudioClient_service__wasapi(pContext, deviceType, (ma_IAudioClient*)pData->pAudioClient, (void**)&pData->pRenderClient);
     } else {
@@ -16191,7 +16191,7 @@ static ma_result ma_context_uninit__wasapi(ma_context* pContext)
 {
     MA_ASSERT(pContext != NULL);
     MA_ASSERT(pContext->backend == ma_backend_wasapi);
-    
+
     if (pContext->wasapi.commandThread != NULL) {
         ma_context_command__wasapi cmd = ma_context_init_command__wasapi(MA_CONTEXT_COMMAND_QUIT__WASAPI);
         ma_context_post_command__wasapi(pContext, &cmd);
@@ -20073,7 +20073,7 @@ static ma_result ma_context_get_device_info__alsa(ma_context* pContext, ma_devic
     We want to ensure the the first data formats are the best. We have a list of favored sample
     formats and sample rates, so these will be the basis of our iteration.
     */
-    
+
     /* Formats. We just iterate over our standard formats and test them, making sure we reset the configuration space each iteration. */
     for (iFormat = 0; iFormat < ma_countof(g_maFormatPriorities); iFormat += 1) {
         ma_format format = g_maFormatPriorities[iFormat];
@@ -22552,7 +22552,7 @@ static ma_result ma_device_init__pulse(ma_device* pDevice, const ma_device_confi
         if (pDescriptorCapture->pDeviceID != NULL) {
             devCapture = pDescriptorCapture->pDeviceID->pulse;
         }
-        
+
         format     = pDescriptorCapture->format;
         channels   = pDescriptorCapture->channels;
         sampleRate = pDescriptorCapture->sampleRate;
@@ -25137,7 +25137,7 @@ static ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_
         if (deviceObjectID == defaultDeviceObjectID) {
             pDeviceInfo->isDefault = MA_TRUE;
         }
-        
+
         /*
         There could be a large number of permutations here. Fortunately there is only a single channel count
         being reported which reduces this quite a bit. For sample rates we're only reporting those that are
@@ -25147,19 +25147,20 @@ static ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_
         sample rate.
         */
         pDeviceInfo->nativeDataFormatCount = 0;
-        
+
         /* Formats. */
         {
             ma_format uniqueFormats[ma_format_count];
+            memset(uniqueFormats, 0xFF, sizeof(ma_format) * ma_format_count);
             ma_uint32 uniqueFormatCount = 0;
             ma_uint32 channels;
-            
+
             /* Channels. */
             result = ma_get_AudioObject_channel_count(pContext, deviceObjectID, deviceType, &channels);
             if (result != MA_SUCCESS) {
                 return result;
             }
-            
+
             /* Formats. */
             result = ma_get_AudioObject_stream_descriptions(pContext, deviceObjectID, deviceType, &streamDescriptionCount, &pStreamDescriptions);
             if (result != MA_SUCCESS) {
@@ -25186,21 +25187,20 @@ static ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_
                         break;
                     }
                 }
-                
+
                 /* If we've already handled this format just skip it. */
                 if (hasFormatBeenHandled) {
                     continue;
                 }
-                
-                uniqueFormatCount += 1;
-                
+
+                uniqueFormats[uniqueFormatCount++] = format;
 
                 /* Sample Rates */
                 result = ma_get_AudioObject_sample_rates(pContext, deviceObjectID, deviceType, &sampleRateRangeCount, &pSampleRateRanges);
                 if (result != MA_SUCCESS) {
                     return result;
                 }
-                
+
                 /*
                 Annoyingly Core Audio reports a sample rate range. We just get all the standard rates that are
                 between this range.
@@ -25216,14 +25216,14 @@ static ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_
                             pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].sampleRate = standardSampleRate;
                             pDeviceInfo->nativeDataFormats[pDeviceInfo->nativeDataFormatCount].flags      = 0;
                             pDeviceInfo->nativeDataFormatCount += 1;
-                            
+
                             if (pDeviceInfo->nativeDataFormatCount >= ma_countof(pDeviceInfo->nativeDataFormats)) {
                                 break;  /* No more room for any more formats. */
                             }
                         }
                     }
                 }
-                
+
                 if (pDeviceInfo->nativeDataFormatCount >= ma_countof(pDeviceInfo->nativeDataFormats)) {
                     break;  /* No more room for any more formats. */
                 }
@@ -25315,12 +25315,12 @@ static ma_result ma_context_get_device_info__coreaudio(ma_context* pContext, ma_
 
         /* Only a single format is being reported for iOS. */
         pDeviceInfo->nativeDataFormatCount = 1;
-        
+
         result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pDeviceInfo->nativeDataFormats[0].format);
         if (result != MA_SUCCESS) {
             return result;
         }
-        
+
         pDeviceInfo->nativeDataFormats[0].channels = bestFormat.mChannelsPerFrame;
 
         /*
@@ -25478,7 +25478,7 @@ static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFl
                     if (framesToRead > framesRemaining) {
                         framesToRead = framesRemaining;
                     }
-                    
+
                     ma_device_handle_backend_data_callback(pDevice, tempBuffer, NULL, framesToRead);
 
                     for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) {
@@ -25583,7 +25583,7 @@ static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFla
                     if (framesToSend > framesRemaining) {
                         framesToSend = framesRemaining;
                     }
-                    
+
                     ma_device_handle_backend_data_callback(pDevice, NULL, silentBuffer, framesToSend);
 
                     framesRemaining -= framesToSend;
@@ -26706,7 +26706,7 @@ static ma_result ma_device_init__coreaudio(ma_device* pDevice, const ma_device_c
     #endif
     }
 
-    
+
 
     /*
     When stopping the device, a callback is called on another thread. We need to wait for this callback
@@ -26993,7 +26993,7 @@ static ma_result ma_context_init__coreaudio(ma_context* pContext, const ma_conte
 #endif
 
     pContext->coreaudio.noAudioSessionDeactivate = pConfig->coreaudio.noAudioSessionDeactivate;
-    
+
     pCallbacks->onContextInit             = ma_context_init__coreaudio;
     pCallbacks->onContextUninit           = ma_context_uninit__coreaudio;
     pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__coreaudio;
@@ -27630,7 +27630,7 @@ static ma_result ma_device_init_handle__sndio(ma_device* pDevice, const ma_devic
             par.sig  = 1;
         } break;
     }
-    
+
     if (deviceType == ma_device_type_capture) {
         par.rchan = channels;
     } else {
@@ -29037,7 +29037,7 @@ static ma_result ma_device_init_fd__oss(ma_device* pDevice, const ma_device_conf
     MA_ASSERT(pDevice != NULL);
     MA_ASSERT(pConfig != NULL);
     MA_ASSERT(deviceType != ma_device_type_duplex);
-    
+
     pDeviceID     = pDescriptor->pDeviceID;
     shareMode     = pDescriptor->shareMode;
     ossFormat     = ma_format_to_oss((pDescriptor->format != ma_format_unknown) ? pDescriptor->format : ma_format_s16); /* Use s16 by default because OSS doesn't like floating point. */
@@ -29625,7 +29625,7 @@ static ma_result ma_create_and_configure_AAudioStreamBuilder__aaudio(ma_context*
 
         /* Not sure how this affects things, but since there's a mapping between miniaudio's performance profiles and AAudio's performance modes, let go ahead and set it. */
         ((MA_PFN_AAudioStreamBuilder_setPerformanceMode)pContext->aaudio.AAudioStreamBuilder_setPerformanceMode)(pBuilder, (pConfig->performanceProfile == ma_performance_profile_low_latency) ? MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY : MA_AAUDIO_PERFORMANCE_MODE_NONE);
-    
+
         /* We need to set an error callback to detect device changes. */
         if (pDevice != NULL) {  /* <-- pDevice should never be null if pDescriptor is not null, which is always the case if we hit this branch. Check anyway for safety. */
             ((MA_PFN_AAudioStreamBuilder_setErrorCallback)pContext->aaudio.AAudioStreamBuilder_setErrorCallback)(pBuilder, ma_stream_error_callback__aaudio, (void*)pDevice);
@@ -32930,13 +32930,13 @@ MA_API ma_result ma_device_init(ma_context* pContext, const ma_device_config* pC
     pDevice->capture.channels           = pConfig->capture.channels;
     ma_channel_map_copy(pDevice->capture.channelMap, pConfig->capture.channelMap, pConfig->capture.channels);
     pDevice->capture.channelMixMode     = pConfig->capture.channelMixMode;
-    
+
     pDevice->playback.shareMode         = pConfig->playback.shareMode;
     pDevice->playback.format            = pConfig->playback.format;
     pDevice->playback.channels          = pConfig->playback.channels;
     ma_channel_map_copy(pDevice->playback.channelMap, pConfig->playback.channelMap, pConfig->playback.channels);
     pDevice->playback.channelMixMode    = pConfig->playback.channelMixMode;
-    
+
 
     result = ma_mutex_init(&pDevice->startStopLock);
     if (result != MA_SUCCESS) {
@@ -43290,7 +43290,7 @@ static ma_result ma_audio_buffer_init_ex(const ma_audio_buffer_config* pConfig,
     if (result != MA_SUCCESS) {
         return result;
     }
-    
+
     ma_allocation_callbacks_init_copy(&pAudioBuffer->allocationCallbacks, &pConfig->allocationCallbacks);
 
     if (doCopy) {