If I call AudioDeviceStart on an AudioDevice in my application then "Hey Siri!" will not wake Siri up. Our users have complained that Siri does not get activated with my application is running. We found that calling AudioDeviceStart is causing the issue. How should we handle this?

Hi everyone, I wanted to bring up a question about Core Audio and its potential for future updates or improvements, specifically regarding latency optimization. As someone who relies on Core Audio for real-time audio processing, any enhancements in this area would be incredibly beneficial for professionals in the industry. Does anyone know if Apple has shared any plans or updates regarding Core Audio’s performance, particularly for low-latency applications? I’d appreciate any insights or advice from the community! Thanks so much! Best, Michael

We are using a VoiceProcessingIO audio unit in our VoIP application on Mac. In certain scenarios, the AudioComponentInstanceNew call blocks for up to five seconds (at least two). We are using the following code to initialize the audio unit: OSStatus status; AudioComponentDescription desc; AudioComponent inputComponent; desc.componentType = kAudioUnitType_Output; desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO; desc.componentFlags = 0; desc.componentFlagsMask = 0; desc.componentManufacturer = kAudioUnitManufacturer_Apple; inputComponent = AudioComponentFindNext(NULL, &desc); status = AudioComponentInstanceNew(inputComponent, &unit); We are having the issue with current MacOS versions on a host of different Macs (x86 and x64 alike). It takes two to three seconds until AudioComponentInstanceNew returns. We also see the following errors in the log multiple times: AUVPAggregate.cpp:2560 AggInpStreamsChanged wait failed and those right after (which I don't know if they matter to this issue): KeystrokeSuppressorCore.cpp:44 ERROR: KeystrokeSuppressor initialization was unsuccessful. Invalid or no plist was provided. AU will be bypassed. vpStrategyManager.mm:486 Error code 2003332927 reported at GetPropertyInfo

Hello, I'm developing a Command Line Tool in XCode, in order to capture system audio and save it to a file, which will then be used by a separate process. Everything works perfectly when running it from either XCode or the native terminal application (see image below), but as soon as I try to run it from any 3rd party application, it doesn't ask for permissions to record sound, and the resultant file ends up soundless. When archiving it and then running it from other 3rd party applications, e.g Warp (terminal) or spawning it as a child process from a bundled Electron application, it doesn't ask for permissions. Things of note: I've codesigned the application with "Developer ID Application" I've added NSAudioCaptureUsageDescriptionto Info.plist I've included Info.plist in the binary (see image below) I've added the com.apple.security.device.audio-input entitlement I've used the following resources as inspiration: https://github.com/insidegui/AudioCap https://developer.apple.com/documentation/coreaudio/capturing-system-audio-with-core-audio-taps As my use-case involves spawning the executable from Electron as a child process, I've tried to include the appropriate permissions to the parent application too, without success. I'm really at a loss here, it feels like I've tried everything. Any pointers are much appreciated! Thanks

Periodically when testing I am running into a situation where the app hangs and beach balls forever when using AVAudioEngine. This seems to log out when this affect happens: Now when this happens if I pause the debugger it's hanging at a call to: [engine connect:playerNode to:engine.mainMixerNode format:buffer.format]; #0 0x000000019391ca9c in __psynch_mutexwait () #1 0x0000000104d49100 in _pthread_mutex_firstfit_lock_wait () #2 0x0000000104d49014 in _pthread_mutex_firstfit_lock_slow () #3 0x00000001938928ec in std::__1::recursive_mutex::lock () #4 0x00000001ef80e988 in CADeprecated::RealtimeMessenger::_PerformPendingMessages () #5 0x00000001ef818868 in AVAudioNodeTap::Uninitialize () #6 0x00000001ef7fdc68 in AUGraphNodeBase::Uninitialize () #7 0x00000001ef884f38 in AVAudioEngineGraph::PerformCommand () #8 0x00000001ef88e780 in AVAudioEngineGraph::_Connect () #9 0x00000001ef8b7e70 in AVAudioEngineImpl::Connect () #10 0x00000001ef8bc05c in -[AVAudioEngine connect:to:format:] () Current all my audio engine related calls are on the main queue (though I am curious about this https://forums.developer.apple.com/forums/thread/123540?answerId=816827022#816827022). In any case, anyone know where I'm going wrong here?

The problem I have at the moment is that if a phone call comes in during my recording, even if I don't answer, my recording will be interrupted The phenomenon of recording interruption is that the picture is stuck, and the recording can be resumed automatically after the call is over. But it will cause the recorded video sound and painting out of sync Through the AVCaptureSessionWasInterrupted listening, I can get to record the types of alerts and interrupt As far as I can tell, a ringing or vibrating phone can block the audio channel. I found the same scenario in other apps, you can turn off the ring tone or vibration, but I don't know how to do it, I tried a lot of ways, but it doesn't work BlackmagicCam or ProMovie App, when a call comes in during recording, there will only be a notification menu, and there will be no ringtone or vibration, which solves the problem of recording interruption I don't know if this requires some configuration or application, please let me know if it does

I am unable to access the Int32 error from the errors that CoreAudio throws in Swift type AudioHardwareError. This is critical. There is no way to access the errors or even create an AudioHardwareError to test for errors. do { _ = try AudioHardwareDevice(id: 0).streams // will throw } catch { if let error = error as? AudioHardwareError { // cast to AudioHardwareError print(error) // prints error code but not the errorDescription } } How can get reliably get the error.Int32? Or create a AudioHardwareError with an error constant? There is no way for me to handle these error with code or run tests without knowing what the error is. On top of that, by default the error localizedDescription does not contain the errorDescription unless I extend AudioHardwareError with CustomStringConvertible. extension AudioHardwareError: @retroactive CustomStringConvertible { public var description: String { return self.localizedDescription } }

Description As of iOS 18, AVAudioSession.setPreferredIOBufferDuration ignores the requested buffer size when Sound Recognition or Vocal Shortcuts is enabled. This results in 1) much larger buffer sizes and 2) mismatched buffer sizes between input and output buffers, which causes ‘glitchy’ audio and increased latency. Additionally, when this issue occurs AVAudioSession.setPreferredIOBufferDuration continues to return ‘true’ and no error is produced. Steps to Reproduce: Enable Vocal Shortcuts on a device running iOS 18. Enable at least one shortcut (e.g. Control Center). Open or clone the example project (https://github.com/cwalo/SoundRecognitionBug) Build and install the example project Attach a headset and launch the application Observe console logs showing a requested buffer size of 0.005805 (256 samples @ 48k) an actual buffer size of 0.023220 (1104 samples @48k - this is regularly the resulting buffer size in all of our tests) Quit the app and detach the headset. Enable mutesOutput in AudioSystem.mm (to avoid feedback) Launch the application Observe Same result from step 4 Mismatched hardware buffer size of 1104 and recorded frame count of 1024 Mismatched playbackCount and recordCount Quit the app and disable vocal shortcuts Launch the app Observe IOBufferDuration matching the requested duration and matched buffer sizes (expected behavior) Expected results: Requested IOBufferDuration is respected or AVAudioSession returns false or error is produced Input and output buffer sizes match Device(s): iPhone 11 Pro, iPad Pro OS: iOS 18.0.1 Environment: Xcode 16.1 FB: FB15715421 Related to: https://forums.developer.apple.com/forums/thread/765477

I’m experiencing an unusual audio issue with AirPods on macOS Sequoia while developing VoIP applications like Zoom and FaceTime. When AirPods are connected, the other party’s voice sometimes sounds unnaturally stretched (approximately twice as long). This problem can be temporarily fixed by switching the sound output settings from AirPods to speakers and then back to AirPods. From our analysis, the issue appears to be related to the sample rate provided by AudioObjectGetPropertyData. Here’s what we’ve observed: When the issue occurs, the AudioStreamBasicDescription.sampleRate for AirPods is reported as 48000. Under normal conditions, it’s reported as 24000. It seems like the system is mistakenly returning a sample rate that doesn’t match the AirPods’ actual settings, perhaps defaulting to a system speaker value. Once the output setting is toggled, the correct sampleRate (24000) is retrieved. This discrepancy causes our application to transmit the audio stream at 48000, leading to the distorted playback. Has anyone encountered a similar issue or knows how to resolve it?

I'm trying to setup a listener for kAudioProcessPropertyIsRunningOutput but it's never triggered. I get calls for kAudioProcessPropertyIsRunning and kAudioProcessPropertyDevices but not for kAudioProcessPropertyIsRunningInput or kAudioProcessPropertyIsRunningOutput. class MyDelegate: PropertyListenerDelegate { func propertiesChanged(properties: [AudioObjectPropertyAddress]) { print(properties) } } var myDelegate = MyDelegate() var processes = try AudioHardwareSystem.shared.processes for process in processes { process.delegates += [myDelegate] try process.addListener(forProperties: [AudioObjectPropertyAddress(mSelector: kAudioPropertyWildcardPropertyID, mScope: kAudioObjectPropertyScopeWildcard, mElement: kAudioObjectPropertyElementWildcard)]) } Xcode 16.1 macOS 15.0.1

Hi, I have configured the stream as interleaved, but I am unsure if the function produces interleaved samples. So here my question: Does AudioDeviceCreateIOProcID produce interleaved samples with microphone input?

Our capture application records system audio via HAL plugin, however, with the latest macOS 15 Sequoia, all audio buffer values are zero. I am attaching sample code that replicates the problem. Compile as a Command Line Tool application with Xcode. STEPS TO REPRODUCE Install BlackHole 2ch audio driver: https://existential.audio/blackhole/download/?code=1579271348 Start some system audio, e.g. YouTube. Compile and run the sample application. On macOS up to Sonoma, you will hear audio via loopback and see audio values in the debug/console window. On macOS Sequoia, you will not hear audio and the audio values are 0. #import <AVFoundation/AVFoundation.h> #import <CoreAudio/CoreAudio.h> #define BLACKHOLE_UID @"BlackHole2ch_UID" #define DEFAULT_OUTPUT_UID @"BuiltInSpeakerDevice" @interface AudioCaptureDelegate : NSObject <AVCaptureAudioDataOutputSampleBufferDelegate> @end void setDefaultAudioDevice(NSString *deviceUID); @implementation AudioCaptureDelegate // receive samples from CoreAudio/HAL driver and print amplitute values for testing // this is where samples would normally be copied and passed downstream for further processing which // is not needed in this simple sample application - (void)captureOutput:(AVCaptureOutput *)captureOutput didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection *)connection { // Access the audio data in the sample buffer CMBlockBufferRef blockBuffer = CMSampleBufferGetDataBuffer(sampleBuffer); if (!blockBuffer) { NSLog(@"No audio data in the sample buffer."); return; } size_t length; char *data; CMBlockBufferGetDataPointer(blockBuffer, 0, NULL, &length, &data); // Process the audio samples to calculate the average amplitude int16_t *samples = (int16_t *)data; size_t sampleCount = length / sizeof(int16_t); int64_t sum = 0; for (size_t i = 0; i < sampleCount; i++) { sum += abs(samples[i]); } // Calculate and log the average amplitude float averageAmplitude = (float)sum / sampleCount; NSLog(@"Average Amplitude: %f", averageAmplitude); } @end // set the default audio device to Blackhole while testing or speakers when done // called by main void setDefaultAudioDevice(NSString *deviceUID) { AudioObjectPropertyAddress address; AudioDeviceID deviceID = kAudioObjectUnknown; UInt32 size; CFStringRef uidString = (__bridge CFStringRef)deviceUID; // Gets the device corresponding to the given UID. AudioValueTranslation translation; translation.mInputData = &uidString; translation.mInputDataSize = sizeof(uidString); translation.mOutputData = &deviceID; translation.mOutputDataSize = sizeof(deviceID); size = sizeof(translation); address.mSelector = kAudioHardwarePropertyDeviceForUID; address.mScope = kAudioObjectPropertyScopeGlobal; //???? address.mElement = kAudioObjectPropertyElementMain; OSStatus status = AudioObjectGetPropertyData(kAudioObjectSystemObject, &address, 0, NULL, &size, &translation); if (status != noErr) { NSLog(@"Error: Could not retrieve audio device ID for UID %@. Status code: %d", deviceUID, (int)status); return; } AudioObjectPropertyAddress propertyAddress; propertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; propertyAddress.mScope = kAudioObjectPropertyScopeGlobal; status = AudioObjectSetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, sizeof(AudioDeviceID), &deviceID); if (status == noErr) { NSLog(@"Default audio device set to %@", deviceUID); } else { NSLog(@"Failed to set default audio device: %d", status); } } // sets Blackhole device as default and configures it as AVCatureDeviceInput // sets the speakers as loopback so we can hear what is being captured // sets up queue to receive capture samples // runs session for 30 seconds, then restores speakers as default output int main(int argc, const char * argv[]) { @autoreleasepool { // Create the capture session AVCaptureSession *session = [[AVCaptureSession alloc] init]; // Select the audio device AVCaptureDevice *audioDevice = nil; NSString *audioDriverUID = nil; audioDriverUID = BLACKHOLE_UID; setDefaultAudioDevice(audioDriverUID); audioDevice = [AVCaptureDevice deviceWithUniqueID:audioDriverUID]; if (!audioDevice) { NSLog(@"Audio device %s not found!", [audioDriverUID UTF8String]); return -1; } else { NSLog(@"Using Audio device: %s", [audioDriverUID UTF8String]); } // Configure the audio input with the selected device (Blackhole) NSError *error = nil; AVCaptureDeviceInput *audioInput = [AVCaptureDeviceInput deviceInputWithDevice:audioDevice error:&error]; if (error || !audioInput) { NSLog(@"Failed to create audio input: %@", error); return -1; } [session addInput:audioInput]; // Configure the audio data output AVCaptureAudioDataOutput *audioOutput = [[AVCaptureAudioDataOutput alloc] init]; AudioCaptureDelegate *delegate = [[AudioCaptureDelegate alloc] init]; dispatch_queue_t queue = dispatch_queue_create("AudioCaptureQueue", NULL); [audioOutput setSampleBufferDelegate:delegate queue:queue]; [session addOutput:audioOutput]; // Set audio settings NSDictionary *audioSettings = @{ AVFormatIDKey: @(kAudioFormatLinearPCM), AVSampleRateKey: @48000, AVNumberOfChannelsKey: @2, AVLinearPCMBitDepthKey: @16, AVLinearPCMIsFloatKey: @NO, AVLinearPCMIsNonInterleaved: @NO }; [audioOutput setAudioSettings:audioSettings]; AVCaptureAudioPreviewOutput * loopback_output = nil; loopback_output = [[AVCaptureAudioPreviewOutput alloc] init]; loopback_output.volume = 1.0; loopback_output.outputDeviceUniqueID = DEFAULT_OUTPUT_UID; [session addOutput:loopback_output]; const char *deviceID = loopback_output.outputDeviceUniqueID ? [loopback_output.outputDeviceUniqueID UTF8String] : "nil"; NSLog(@"session addOutput for preview/loopback: %s", deviceID); // Start the session [session startRunning]; NSLog(@"Capturing audio data for 30 seconds..."); [[NSRunLoop currentRunLoop] runUntilDate:[NSDate dateWithTimeIntervalSinceNow:30.0]]; // Stop the session [session stopRunning]; NSLog(@"Capture session stopped."); setDefaultAudioDevice(DEFAULT_OUTPUT_UID); } return 0; }

Here is some code I have to create an AVAudioFile instance based on Int16 samples. let format = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44100.0, channels: 2, interleaved: false)! let audioFile = try AVAudioFile(forWriting: outputURL, settings: format.settings) When writing to the file I get the following runtime error, presumably from CoreAudio. CABufferList.h:184 ASSERTION FAILURE [(nBytes <= buf->mDataByteSize) != 0 is false]: I read this as a size mismatch between what is specified in the format used to create the file and the file's own internal processingFormat property, which is read-only. Here is my debugger console output showing the input format I created, along with the resulting AVAudioFile fileFormat and processingFormat properties. (lldb) po format <AVAudioFormat 0x300e553b0: 2 ch, 44100 Hz, Int16, deinterleaved> (lldb) po format.settings ▿ 7 elements ▿ 0 : 2 elements - key : "AVNumberOfChannelsKey" - value : 2 ▿ 1 : 2 elements - key : "AVLinearPCMBitDepthKey" - value : 16 ▿ 2 : 2 elements - key : "AVFormatIDKey" - value : 1819304813 ▿ 3 : 2 elements - key : "AVLinearPCMIsNonInterleaved" - value : 1 ▿ 4 : 2 elements - key : "AVLinearPCMIsBigEndianKey" - value : 0 ▿ 5 : 2 elements - key : "AVLinearPCMIsFloatKey" - value : 0 ▿ 6 : 2 elements - key : "AVSampleRateKey" - value : 44100 (lldb) po audioFile.fileFormat <AVAudioFormat 0x300ea5400: 2 ch, 44100 Hz, Int16, interleaved> (lldb) po audioFile.processingFormat <AVAudioFormat 0x300ea5450: 2 ch, 44100 Hz, Float32, deinterleaved> Please note that the input format I'm using does not match either the audio file fileFormat or processingFormat properties. The file format is interleaved even though I specified de-interleaved. This makes sense to me as working with audio files that are growing is much easier and more efficient with interleaved data. The head-scratcher is the processingFormat. I specified Int16 samples and it is expecting Float32? According to the format settings dictionary, we are specifying the correct key/value pairs. Is this expected behavior? Does Apple always insist on Float32 internally or is this a bug?

AddInstanceForFactory: No factory registered for id <CFUUID 0x6000002e76c0> F8BB1C28-BAE8-11D6-9C31-00039315CD46 AudioQueueObject.cpp:1580 BuildConverter: AudioConverterNew returned -50 from: 0 ch, 16000 Hz, .... (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame to: 2 ch, 16000 Hz, Int16, interleaved HALSystem.cpp:2216 AudioObjectPropertiesChanged: no such object AQMEIO_HAL.cpp:2552 timeout AudioHardware-mac-imp.cpp:2706 AudioDeviceStop: no device with given ID AudioQueueObject.cpp:1580 BuildConverter: AudioConverterNew returned -50 from: 0 ch, 16000 Hz, .... (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame to: 2 ch, 16000 Hz, Int16, interleaved AudioQueueObject.cpp:6707 ConvertInput: aq@0x109994200: AudioConverterFillComplexBuffer returned -50, packetCount 5328 Xcode version 15.2(15C500b) iPhone 15Pro Version 17.2 (Simulator) Language : Swift In version 17.0 or above, there are no recording issues in the object-c project, iPhone and simulators can't start recording, Why?

Hello, I used kAudioDevicePropertyDeviceIsRunningSomewhere to check if an internal or external microphone is being used. My code works well for the internal microphone, and for microphones which are connected using a cable. External microphones which are connected using bluetooth are not reporting their status. The status is always requested successfully, but it is always reported as inactive. Main relevant parts in my code : static inline AudioObjectPropertyAddress makeGlobalPropertyAddress(AudioObjectPropertySelector selector) { AudioObjectPropertyAddress address = { selector, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster, }; return address; } static BOOL getBoolProperty(AudioDeviceID deviceID, AudioObjectPropertySelector selector) { AudioObjectPropertyAddress const address = makeGlobalPropertyAddress(selector); UInt32 prop; UInt32 propSize = sizeof(prop); OSStatus const status = AudioObjectGetPropertyData(deviceID, &address, 0, NULL, &propSize, &prop); if (status != noErr) { return 0; //this line never gets executed in my tests. The call above always succeeds, but it always gives back "false" status. } return static_cast<BOOL>(prop == 1); } ... __block BOOL microphoneActive = NO; iterateThroughAllInputDevices(^(AudioObjectID object, BOOL *stop) { if (getBoolProperty(object, kAudioDevicePropertyDeviceIsRunningSomewhere) != 0) { microphoneActive = YES; *stop = YES; } }); What could cause this and how could it be fixed? Thank you for your help in advance!

I have a USB audio interface that is causing kernel traps and the audio output to "skip" or dropout every few seconds. This behavior occurs with a completely fresh install of Catalina as well as Big Sur with the stock Music app on a 2019 MacBook Pro 16 (full specs below). The Console logs show coreaudiod got an error from a kernel trap, a "USB Sound assertion" in AppleUSBAudio/AppleUSBAudio-401.4/KEXT/AppleUSBAudioDevice.cpp at line 6644, and the Music app "skipping cycle due to overload." I've added a short snippet from Console logs around the time of the audio skip/drop out. The more complete logs are at this gist: https://gist.github.com/djflux/08d9007e2146884e6df1741770de5105 I've also opened a Feedback Assistant ticket (FB9037528): https://feedbackassistant.apple.com/feedback/9037528 Does anyone know what could be causing this issue? Thanks for any help. Cheers, Flux aka Andy. Hardware Overview:  Model Name: MacBook Pro  Model Identifier: MacBookPro16,1  Processor Name: 8-Core Intel Core i9  Processor Speed: 2.4 GHz  Number of Processors: 1  Total Number of Cores: 8  L2 Cache (per Core): 256 KB  L3 Cache: 16 MB  Hyper-Threading Technology: Enabled  Memory: 64 GB  System Firmware Version: 1554.80.3.0.0 (iBridge: 18.16.14347.0.0,0) System Software Overview: System Version: macOS 11.2.3 (20D91)  Kernel Version: Darwin 20.3.0  Boot Volume: Macintosh HD  Boot Mode: Normal  Computer Name: mycomputername  User Name: myusername  Secure Virtual Memory: Enabled  System Integrity Protection: Enabled USB interface: Denon DJ DS1 Snippet of Console logs error 21:07:04.848721-0500 coreaudiod HALS_IOA1Engine::EndWriting: got an error from the kernel trap, Error: 0xE00002D7 default 21:07:04.848855-0500 Music HALC_ProxyIOContext::IOWorkLoop: skipping cycle due to overload default 21:07:04.857903-0500 kernel USB Sound assertion (Resetting engine due to error returned in Read Handler) in /AppleInternal/BuildRoot/Library/Caches/com.apple.xbs/Sources/AppleUSBAudio/AppleUSBAudio-401.4/KEXT/AppleUSBAudioDevice.cpp at line 6644 ... default 21:07:05.102746-0500 coreaudiod Audio IO Overload inputs: 'private' outputs: 'private' cause: 'Unknown' prewarming: no recovering: no default 21:07:05.102926-0500 coreaudiod   CAReportingClient.mm:508  message {   HostApplicationDisplayID = "com.apple.Music";   cause = Unknown;   deadline = 2615019;   "input_device_source_list" = Unknown;   "input_device_transport_list" = USB;   "input_device_uid_list" = "AppleUSBAudioEngine:Denon DJ:DS1:000:1,2";   "io_buffer_size" = 512;   "io_cycle" = 1;   "is_prewarming" = 0;   "is_recovering" = 0;   "issue_type" = overload;   lateness = "-535";   "output_device_source_list" = Unknown;   "output_device_transport_list" = USB;   "output_device_uid_list" = "AppleUSBAudioEngine:Denon DJ:DS1:000:1,2"; }: (null)

Hello, We have noticed an issue with SoundRecognition that causes glitching with our AudioUnit setup in Smule. Input and output frame sizes are inconsistent. Input frame size does not match [AVAudioSession sharedInstance].IOBufferDuration My best guess is that SoundRecognition influences the input frame size and not the output frame size. To reproduce use the example app here: https://github.com/MarkoGill/SoundRecognitionBug Hardware/OS iPhone 14 Pro on iOS 18 -> Experiences the problem iPhone 11 on iOS 18 -> Experiences the problem iPhone 15 on iOS 18 -> Not experiencing the problem Reproduction Steps Enable Sound Recognition (Settings > Accessibility > Sound Recognition > On) Enable a Sound for detection (Sounds > Dog > On) Open the example app with headset (it routes input to output) Notice glitching occurs Check the logs. Record and Playback buffer sizes vary Example Log: AU input sample rate: 48000.000000 AU output sample rate: 48000.000000 hardware sample rate: 48000.000000 hardware buffer size: 1104.000000 updated record frame counts: 1024 updated playback frame counts: 1104 Notes: You can disable Sound Recognition, restart the app, and playback behaves correctly.

Hello, I run into an issue on Monterey (12.7.5). I have a bundled XPC service in my application which is displaying some stuff and playin sounds via NSSound. I had a problem with playback due to service priority, so I use the trick with a reply block where I send a reply block to the service and basically just retain it and never call it. This worked fine so far, but we have users, predominantly on Monterey, who are having a problem with sound playback. It's choppy and distorted when their machine is under load (where "load" often just means playing a video on YouTube in Chrome). Is there anything else I can do to get the proper priority for my xpc service so I can avoid distorted sound? Additionally the service type is Application and RunLoopType is NSRunLoop with JoinExistingSession set to true. The QoS level of main queue is 0x21 (user interactive) and I'm calling all the NSSound APIs on main queue.