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miniaudio
Commit 0a853ba3 authored by David Reid's avatar David Reid Committed by GitHub
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Merge pull request #296 from chainblocks/master 

Fix ubsan issue on macOS.

Public issue https://github.com/mackron/miniaudio/issues/295
parents 2f56829b b0017520
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Showing with 37 additions and 37 deletions
miniaudio.h
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......@@ -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,
......@@ -6883,7 +6883,7 @@ 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
......@@ -13638,7 +13638,7 @@ 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);
......@@ -13810,7 +13810,7 @@ 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 {
......@@ -13822,7 +13822,7 @@ 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 {
......@@ -13878,7 +13878,7 @@ 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;
......@@ -13923,7 +13923,7 @@ 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);
......@@ -13949,7 +13949,7 @@ 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;
}
......@@ -13985,7 +13985,7 @@ 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;
......@@ -14028,7 +14028,7 @@ 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.pResult = &cmdResult; /* Declared locally, but won't be dereferenced after this function returns since execution of the command will wait here. */
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;
......@@ -15075,7 +15075,7 @@ 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 {
......@@ -16219,7 +16219,7 @@ 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);
......@@ -20103,7 +20103,7 @@ 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];
......@@ -22582,7 +22582,7 @@ 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;
......@@ -25168,7 +25168,7 @@ 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
......@@ -25178,19 +25178,20 @@ 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) {
......@@ -25217,21 +25218,20 @@ 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.
......@@ -25247,14 +25247,14 @@ 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. */
}
}
}
}
ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
if (pDeviceInfo->nativeDataFormatCount >= ma_countof(pDeviceInfo->nativeDataFormats)) {
......@@ -25348,12 +25348,12 @@ 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;
/*
......@@ -25511,7 +25511,7 @@ 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) {
......@@ -25616,7 +25616,7 @@ 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;
......@@ -26739,7 +26739,7 @@ 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
......@@ -27026,7 +27026,7 @@ 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;
......@@ -27663,7 +27663,7 @@ 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 {
......@@ -29070,7 +29070,7 @@ 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. */
......@@ -29658,7 +29658,7 @@ 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);
......@@ -32963,13 +32963,13 @@ 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) {
......@@ -43349,7 +43349,7 @@ 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) {
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