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https://github.com/mackron/miniaudio.git
synced 2026-04-22 08:14:04 +02:00
More tweaks to the Core Audio backend.
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@@ -13243,6 +13243,33 @@ mal_result mal_get_AudioObject_closest_buffer_size_in_frames(AudioObjectID devic
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} else {
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*pBufferSizeInFramesOut = bufferSizeInFramesIn;
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}
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return MAL_SUCCESS;
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}
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mal_result mal_set_AudioObject_buffer_size_in_frames(AudioObjectID deviceObjectID, mal_device_type deviceType, mal_uint32* pBufferSizeInOut)
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{
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mal_uint32 chosenBufferSizeInFrames;
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mal_result result = mal_get_AudioObject_closest_buffer_size_in_frames(deviceObjectID, deviceType, *pBufferSizeInOut, &chosenBufferSizeInFrames);
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if (result != MAL_SUCCESS) {
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return result;
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}
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// Try setting the size of the buffer... If this fails we just use whatever is currently set.
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AudioObjectPropertyAddress propAddress;
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propAddress.mSelector = kAudioDevicePropertyBufferFrameSize;
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propAddress.mScope = (deviceType == mal_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
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propAddress.mElement = kAudioObjectPropertyElementMaster;
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OSStatus status = AudioObjectSetPropertyData(deviceObjectID, &propAddress, 0, NULL, sizeof(chosenBufferSizeInFrames), &chosenBufferSizeInFrames);
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if (status != kAudioHardwareNoError) {
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// Getting here means we were unable to set the buffer size. In this case just use whatever is currently selected.
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UInt32 dataSize = sizeof(*pBufferSizeInOut);
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OSStatus status = AudioObjectGetPropertyData(deviceObjectID, &propAddress, 0, NULL, &dataSize, pBufferSizeInOut);
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if (status != kAudioHardwareNoError) {
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return mal_result_from_OSStatus(status);
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}
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}
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return MAL_SUCCESS;
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}
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@@ -13757,14 +13784,7 @@ mal_thread_result MAL_THREADCALL mal_AudioQueue_worker_thread__coreaudio(void* p
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if (status != kAudioHardwareNoError) {
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// An error occurred while stopping the audio queue... just continue on I suppose...
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}
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#if 0
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status = AudioQueueReset((AudioQueueRef)pDevice->coreaudio.audioQueue);
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if (status != kAudioHardwareNoError) {
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// Don't think we need to do anything if resetting fails.
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}
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#endif
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// Make sure the client is notified that the device has stopped, but make sure it's done after the status has been set.
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mal_stop_proc onStop = pDevice->onStop;
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if (onStop) {
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@@ -13830,47 +13850,50 @@ mal_result mal_device_init__coreaudio(mal_context* pContext, mal_device_type dev
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// Buffer size.
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mal_uint32 requestedBufferSizeInFrames = pDevice->bufferSizeInFrames / pDevice->periods;
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if (requestedBufferSizeInFrames == 0) {
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requestedBufferSizeInFrames = 1;
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mal_uint32 actualBufferSizeInFrames = pDevice->bufferSizeInFrames;
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if (actualBufferSizeInFrames < pDevice->periods) {
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actualBufferSizeInFrames = pDevice->periods;
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}
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if (pDevice->usingDefaultBufferSize) {
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// CPU speed is a factor to consider when determine how large of a buffer we need.
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float fCPUSpeed = mal_calculate_cpu_speed_factor();
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#if 0 // TODO: Test capture latency.
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// In my testing, capture seems to have worse latency than playback for some reason.
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float fType;
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if (type == mal_device_type_playback) {
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fType = 1.0f;
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// In my admittedly limited testing, capture latency seems to be about the same as playback with Core Audio, at least on my MacBook Pro. On other
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// backends, however, this is often different. I am therefore leaving the logic below in place just in case I need to do some capture/playback
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// specific tweaking.
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float fDeviceType;
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if (deviceType == mal_device_type_playback) {
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fDeviceType = 1.0f;
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} else {
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fType = 2.0f;
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fDeviceType = 1.0f;
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}
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#else
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float fType = 1.0f;
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#endif
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// Backend tax. Need to fiddle with this.
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float fBackend = 1.5f; // <-- mini_al's implementation is actually kind of bad at the moment. TODO: Revisit this after AudioUnit implementation.
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float fBackend = 0.5f;
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requestedBufferSizeInFrames = mal_calculate_default_buffer_size_in_frames(pConfig->performanceProfile, pConfig->sampleRate, fCPUSpeed*fType*fBackend);
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if (requestedBufferSizeInFrames == 0) {
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requestedBufferSizeInFrames = 1;
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actualBufferSizeInFrames = mal_calculate_default_buffer_size_in_frames(pConfig->performanceProfile, pConfig->sampleRate, fCPUSpeed*fDeviceType*fBackend);
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if (actualBufferSizeInFrames < pDevice->periods) {
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actualBufferSizeInFrames = pDevice->periods;
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}
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}
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mal_uint32 closestBufferSizeInFrames;
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result = mal_get_AudioObject_closest_buffer_size_in_frames(deviceObjectID, deviceType, requestedBufferSizeInFrames, &closestBufferSizeInFrames);
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actualBufferSizeInFrames = actualBufferSizeInFrames / pDevice->periods;
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result = mal_set_AudioObject_buffer_size_in_frames(deviceObjectID, deviceType, &actualBufferSizeInFrames);
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if (result != MAL_SUCCESS) {
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return result;
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}
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// The audio queue is initialized in the worker thread. We need a signal to know when the thread has been initialized.
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// The audio queue is initialized in the worker thread.
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mal_AudioQueue_worker_thread_data__coreaudio threadData;
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threadData.pDevice = pDevice;
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threadData.queueBufferSizeInBytes = closestBufferSizeInFrames * mal_get_bytes_per_frame(pDevice->internalFormat, pDevice->internalChannels);
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// In my testing, it appears that Core Audio likes queue buffers to be larger than device's IO buffer which was set above.
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threadData.queueBufferSizeInBytes = actualBufferSizeInFrames*2 * mal_get_bytes_per_frame(pDevice->internalFormat, pDevice->internalChannels);
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// We need a signal to know when the thread has been initialized.
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mal_event_init(pDevice->pContext, &threadData.initCompletionEvent);
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// The Core Audio backend is not using the main worker thread object so we can just reuse that one.
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