I have a bunch of Audio Unit v3 plugins that are approaching release, and I was considering using subscription-model pricing, as I have done in a soon to be released iOS app. However, whether this is possible or not is not at all obvious. Specifically: The plugin can, depending on the host app, be loaded in-process or out-of-process - yes, I know, Logic Pro and Garage Band will not load a plug-in in-process anymore, but I am not going to rule that out for other audio apps and force on them the overhead of IPC (I spent two solid weeks deciphering the process to actually make it possible for an AUv3 to run in-process - see this - https://github.com/timboudreau/audio_unit_rust_demo - example with notes) Depending on how it is loaded, the value of Bundle.main.bundleIdentifier will vary. If I use the StoreKit API, will that return product results for my bundle identifier when being called as a library from a foreign application? I would expect it would be a major security hole if random apps could query about purchases of other random apps, so I assume not. Even if I restricted the plugins to running out-of-process, I have to set up the in-app purchases on the app store for the App container's ID, not the extension's ID, and the extension is what run - the outer app that is what you purchase is just a toy demo that exists solely to register the audio unit. I have similar questions with regard to MetricKit, which I would similarly like to use, but which may be running inside some random app. If there were some sort of signed token, or similar mechanism, that could be bundled or acquired by the running plugin extension that could be used to ensure both StoreKit and MetricKit operate under the assumption that purchases and metrics should be accessed as if called from the container app, that would be very helpful. This is the difference between having a one-and-done sales model and something that provides ongoing revenue to maintain these products - I am a one-person shop - if I price these products where they would need to be to pay the bills assuming a single sale per customer ever, the price will be too high for anyone to want to try products from a small vendor they've never heard of. So, being able to do a free trial period and then subscription is the difference between this being a viable business or not.

The AVB AVnu MILAN Convention has a groweing Population. Many big companies (Cisco, Meyer Sound, d&b Audio, l‘acoustics, Presonus, digico etc.) implements the AVB AVnu Milan Standards. Is there a plan on the Apple side to also implement AVnu Milan on top of the AVB Protocol? The advantage for Apple Sound would be a great Integration in the professionell Audio market and a more stable intergration on top of the AVB protocol. The atdecc work, but Not that stable.

Hi, I am getting into a trap. Please check stack-trace, howto fix this? regards, Joël stack-trace with ExtAudioFileWrite

I'm developing the VisionOS app. I want to know how to play spatial audio in addition to RealityKit? If it's iOS or macOS, how to play spatial audio in addition to RealityKit?

Since many users like me use Apple Music on Android, the app is almost as feature-rich as iOS. It would be fantastic if the developers could add the new iOS 26 features to the Android app, along with a minor UI change. I know it’s challenging to implement liquid glass on Android hardware or design, but features like auto-mix, pronunciation, and translation could be added. kindly consider this request !!!!

Hi, I am looking for a good way to play sounds at a high frequency. At the moment I am using the AVAudioEngine, and create a couple AVAudioPlayerNode and for each sound I need to play I create a AVAudioPCMBuffer. When the app needs to play a sound, I get the correct AVAudioPCMBuffer for the sound and use the first available AVAudioPlayerNode and feed it to the buffer. The timing for a metronome app has to be very precise because if it's of by about 16ms the user can hear that it is not playing had the right interval. For low speeds this is working without any problems, but at high speeds it is getting worse. Maybe anyone has an idea on how I can improve my method. Its a Plugin for Flutter. import AVFoundation class FastSoundPlayer { private var audioPlayers: [SoundPlayer?] = [] private var sounds: [String: Sound] = [:] private var engine = AVAudioEngine() let session = AVAudioSession.sharedInstance() init() { do { try session.setCategory(AVAudioSession.Category.playback, mode: AVAudioSession.Mode.default, options: [AVAudioSession.CategoryOptions.mixWithOthers]) try session.setActive(true) createSoundPlayers(count: 20) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } // Selector method to handle applicationDidBecomeActiveNotification func applicationDidBecomeActive() { // Reinitialize AVAudioEngine and reattach all nodes do { engine.reset() objc_sync_enter(audioPlayers) audioPlayers.removeAll() createSoundPlayers(count: 20) objc_sync_exit(audioPlayers) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } func createSoundPlayers(count: Int) { for _ in 0..<count { let player = SoundPlayer() engine.attach(player.player) engine.connect(player.player, to: engine.mainMixerNode, format: nil) audioPlayers.append(player) } } func load(sound: Data, name: String) { let sound = Sound(soundData: sound) sounds[name] = sound } func play(name: String) { if !engine.isRunning { applicationDidBecomeActive() } guard let sound = sounds[name] else { print("Sound not found") return } if let player = getAvailablePlayer() { player.play(sound: sound) } } func getAvailablePlayer() -> SoundPlayer? { for player in audioPlayers { if !player!.isPlaying { return player } } return nil } } class SoundPlayer { let player = AVAudioPlayerNode() var isPlaying = false init() { player.volume = 1.0 } func play(sound: Sound) { player.scheduleBuffer(sound.sound!, at: nil, options: .interrupts, completionCallbackType: .dataPlayedBack) { _ in self.complete() } if (player.engine != nil && player.engine!.isRunning) { player.play() isPlaying = true } } func complete() { isPlaying = false } } class Sound { var sound: AVAudioPCMBuffer? init(soundData: Data) { do { let temporaryURL = FileManager.default.temporaryDirectory.appendingPathComponent("tempSound.wav") try soundData.write(to: temporaryURL) // Create AVAudioFile from the temporary file URL let audioFile = try AVAudioFile(forReading: temporaryURL) // Define the format for the PCM buffer (44100Hz, stereo) let format = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44100, channels: 2, interleaved: false) // Create AVAudioPCMBuffer guard let pcmBuffer = AVAudioPCMBuffer(pcmFormat: format!, frameCapacity: AVAudioFrameCount(audioFile.length)) else { // Failed to create PCM buffer self.sound = nil return } // Read audio file into PCM buffer try audioFile.read(into: pcmBuffer) // Assign the created AVAudioPCMBuffer to the sound property self.sound = pcmBuffer } catch { print("Error loading sound file: \(error.localizedDescription)") self.sound = nil } } } Thanks!

Sequoia 15.4.1 (24E263) XCode: 16.3 (16E140) Logic Pro: 11.2.1 I’ve been developing a complex audio unit for Mac OS that works perfectly well in its own bespoke host app and is now well into its beta testing stage. It did take some effort to get it to work well in Logic Pro however and all was fine and working well until: The AU part is an empty app extension with a framework containing its code. The framework contains Swift code for the UI and C code for the DSP parts. When the framework is compiled using the Swift 5 compiler the AU will run in Logic with no problems. (I should also mention that AU passes the most strict auval tests). But… when the framework is compiled with Swift 6 Logic Pro cannot load it. Logic displays a message saying the audio unit could not be loaded and to contact the developer. My own host app loads the AU perfectly well with the Swift 6 version, so I know there’s nothing wrong with the audio unit. I cannot find any differences in any of the built output files except, of course, the actual binary code in the framework. I’ve worked for hours on this and cannot find a solution other than to build the framework in Swift 5. (I worked hard to get all the async code updated and working with Swift 6! so I feel a little cheated!) What is happening? Is this a bug in Logic? Is this a bug in Swift 6 compiler/linker? I’m at the Duh! hands in the air, tearing out hair stage! ( once again!)

I'm using the new SpeechAnalyzer framework to detect certain commands and want to improve accuracy by giving context. Seems like AnalysisContext is the solution for this, but couldn't find any usage example. So I want to make sure that I'm doing it right or not. let context = AnalysisContext() context.contextualStrings = [ AnalysisContext.ContextualStringsTag("commands"): [ "set speed level", "set jump level", "increase speed", "decrease speed", ... ], AnalysisContext.ContextualStringsTag("vocabulary"): [ "speed", "jump", ... ] ] try await analyzer.setContext(context) With this implementation, it still gives outputs like "Set some speed level", "It's speed level", etc. Also, is it possible to make it expect number after those commands, in order to eliminate results like "set some speed level to" (instead of two).

Hi all, I'm working on an audio visualizer app that plays files from the user's music library utilizing MediaPlayer and AVAudioEngine. I'm working on getting the music library functionality working before the visualizer aspect. After setting up the engine for file playback, my app inexplicably crashes with an EXC_BREAKPOINT with code = 1. Usually this means I'm unwrapping a nil value, but I think I'm handling the optionals correctly with guard statements. I'm not able to pinpoint where it's crashing. I think it's either in the play function or the setupAudioEngine function. I removed the processAudioBuffer function and my code still crashes the same way, so it's not that. The device that I'm testing this on is running iOS 26 beta 3, although my app is designed for iOS 18 and above. After commenting out code, it seems that the app crashes at the scheduleFile call in the play function, but I'm not fully sure. Here is the setupAudioEngine function: private func setupAudioEngine() { do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .default) try AVAudioSession.sharedInstance().setActive(true) } catch { print("Audio session error: \(error)") } engine.attach(playerNode) engine.attach(analyzer) engine.connect(playerNode, to: analyzer, format: nil) engine.connect(analyzer, to: engine.mainMixerNode, format: nil) analyzer.installTap(onBus: 0, bufferSize: 1024, format: nil) { [weak self] buffer, _ in self?.processAudioBuffer(buffer) } } Here is the play function: func play(_ mediaItem: MPMediaItem) { guard let assetURL = mediaItem.assetURL else { print("No asset URL for media item") return } stop() do { audioFile = try AVAudioFile(forReading: assetURL) guard let audioFile else { print("Failed to create audio file") return } duration = Double(audioFile.length) / audioFile.fileFormat.sampleRate if !engine.isRunning { try engine.start() } playerNode.scheduleFile(audioFile, at: nil) playerNode.play() DispatchQueue.main.async { [weak self] in self?.isPlaying = true self?.startDisplayLink() } } catch { print("Error playing audio: \(error)") DispatchQueue.main.async { [weak self] in self?.isPlaying = false self?.stopDisplayLink() } } } Here is a link to my test project if you want to try it out for yourself: https://github.com/aabagdi/VisualMan-example Thanks!

I'm trying to implement airplay into my app. I can successfully playback sound and trigger the airplay selector sheet. If the target device is a Bluetooth only device I can connect with no problem and stream the audio to the Bluetooth device, but if the audio device is a airplay specific device like a HomePod or an Apple TV when I select it, I get a spinning icon, indicating that it is trying to connect, and eventually it times out and stops without connecting. I don't believe it is an AirPlay audio issue because if I go to a different app, for example a podcast app and select my HomePods for output, and then switch back to my app. My audio will correctly stream to the HomePod. Not only that, I have it so that my icon will change color to indicate that it is connected via airplay and it is correctly indicating that it is connected via AirPlay. But I cannot then disconnect it using the Airplay selector. The issue appears to be in the AirPlay selection side, which I have spent several days attempting to troubleshoot mostly using ChatGPT to suggest code different than what I have to maybe work around the issue. Mostly it is focused on the audio player section, but it doesn't seem like that is really the route that is the problem.

I bought two "Apple USB-C to Headphone Jack Adapters". Upon closer inspection, they seems to be of different generations: The one with product ID 0x110a on top is working fine. The one with product ID 0x110b has two issues: There is a short but loud click noise on the headphone when I connect it to the iPad. When I play audio using AVAudioPlayer the first half of a second or so is cut off. Here's how I'm playing the audio: audioPlayer = try AVAudioPlayer(contentsOf: url) audioPlayer?.delegate = self audioPlayer?.prepareToPlay() audioPlayer?.play() Is this a known issue? Am I doing something wrong?

Hi there! We have a suite of AudioUnit v2 plugins that have been shipped for some time as aufx plugins, and we are looking into MIDI-related platform upgrades, so we need a way to update these plugins to request MIDI from Logic (and other AU hosts) but avoid changing our AU type and subtype so we don't break existing sessions. Any ideas on how we can do this?

I've got a setup using AVAudioEngine with several tone generator nodes, each with a chain of processing nodes, the chains then mixed into the main output. Generator ➡️ Effect ➡️... ➡️ .mainMixerNode ➡️ .outputNode). Generator ➡️ Effect ➡️... ⤴️ ... Generator ➡️ Effect ➡️... ⤴️ The user should be able to mute any chain individually. I've found several potential approaches to muting, but not terribly happy with any of them. Adjust the amplitudes directly in my tone generators. Issue: Consumes CPU even when completely muted. 4 generators adds ~15% cpu, even when all chains are muted. Detach/attach chains that are muted/unmuted. Issue: Causes loud clicking/popping sounds whenever muted/unmuted. Fade mixer output volume while detaching/attaching a chain (just cutting the volume immediately to 0 doesn't get rid of the clicking/popping). Issue: Causes all channels to fade during the transition, so not ideal. The rest of these ideas are variations on making volume control+detatch/attach work for individual chains, since approach #3 worked well. Add an AVAudioMixer to the end of each chain (just for volume control). Issue: Only the mixer on the final chain functions -- the others block all output. Not sure what's going on there. Use matrix mixer (for multi-input volume control). Plus detach/attach to reduce CPU if necessary. Not yet attempted, due to perceived complexity and reports of fragility in order of wiring in. A bunch of effort before I even know if it's going to work. Develop my own fader node to put on the end of each channel. Unlike the tone generator (simple AVSourceNode), developing an effect node seems complex and time consuming. Might not even fix CPU use. I'm not completely averse to the learning curve of either 5 or 6, but would rather get some guidance on best approach before diving in. They both seem likely to take more effort than I'd like for the simple behavior I'm trying to achieve.

Hi, In my project I am using AVFoundation for recording the audio. We are using AVAudioMixerNode class below method to record the audio packet. **func installTap( onBus bus: AVAudioNodeBus, bufferSize: AVAudioFrameCount, format: AVAudioFormat?, block tapBlock: @escaping AVAudioNodeTapBlock ) ** It works perfectly fine. But in production env some small percentage of the user we are facing issue like after recording few packets it stops automatically without stopping the audio engine. Can anyone help here that why this happens? I have also observed for mediaServicesWereResetNotification and added log on receiving this notification but when this issue happens I don't see any occurence of this log. Also is there any callback when the engine stops?

I prefer to use the album fetched from the library instead of the catalog since this is faster. If doing so, how can I check if all tracks of an album are added to the library. In this case I'd like to fetch the catalog version or throw an error (for example when offline). Using .with(.tracks) on the library album fetches the tracks added to the library. The trackCount property is referring to the tracks that can be fetched from the library. The isComplete property is always nil when fetching from the library. One possible way is checking the trackNumber and discCount properties. However this only detects that not all tracks of an album are added to the library if there is a song not added ahead of one that is. I'd like to be able to handle this edge case as well. Is there currently a way to do this? I'd prefer to not rely on the apple music catalog for this since this is supposed to work offline as well. Fetching and storing all trackIDs when connected and later comparing against these would work, but this would potentially mean storing tens of thousands of track ids. Thank you

Overview We are producing audio in real time from an editing application and are trying to put that on an HLS stream. We attempt to submit PCM samples through an audio writer but are getting a crash after a select number of samples have been appended. Depending on the number of audio frames in the PCM buffer, we might get more iterations before the crash but it always has the same traceback (see below). Code The setup is rather simple. We took inspiration from a few sources around the web. NSMutableDictionary *audio = [[NSMutableDictionary alloc] init]; [audio setObject:@(kAudioFormatMPEG4AAC) forKey:AVFormatIDKey]; [audio setObject:[NSNumber numberWithInt:config.audioSampleRate] // 48000 forKey:AVSampleRateKey]; [audio setObject:[NSNumber numberWithInt:config.audioChannels] // 2 forKey:AVNumberOfChannelsKey]; [audio setObject:@160000 forKey:AVEncoderBitRateKey]; m_audioConfig = [[NSDictionary alloc] initWithDictionary:audio]; m_audio = [[AVAssetWriterInput alloc] initWithMediaType:AVMediaTypeAudio outputSettings:m_audioConfig]; AVAudioFrameCount audioFrames = BUFFER_SAMPLES * bCount; AVAudioPCMBuffer *pcmBuffer = [[AVAudioPCMBuffer alloc] initWithPCMFormat:m_full.pcmFormat frameCapacity:audioFrames]; pcmBuffer.frameLength = pcmBuffer.frameCapacity; AudioChannelLayout layout; memset(&layout, 0, sizeof(layout)); layout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo; CMFormatDescriptionRef format; OSStatus stats = CMAudioFormatDescriptionCreate( kCFAllocatorDefault, pcmBuffer.format.streamDescription, sizeof(layout), &layout, 0, nil, nil, &format ); for (int i = 0; i < bCount; i++) { AudioPCM pcm; audioCallback->callback(pcm); memcpy(*(pcmBuffer.int16ChannelData) + (bufferSize * i), pcm.data, bufferSize); } size_t samplesConsumed = BUFFER_SAMPLES * bCount; CMSampleBufferRef sampleBuffer; CMSampleTimingInfo timing; timing.duration = CMTimeMake(1, config.audioSampleRate); timing.presentationTimeStamp = presentationTime; timing.decodeTimeStamp = kCMTimeInvalid; OSStatus ostatus = CMSampleBufferCreate( kCFAllocatorDefault, nil, false, nil, nil, format, (CMItemCount)pcmBuffer.frameLength, 1, &timing, 0, nil, &sampleBuffer ); //// ostatus = CMSampleBufferSetDataBufferFromAudioBufferList( sampleBuffer, kCFAllocatorDefault, kCFAllocatorDefault, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, pcmBuffer.audioBufferList ); if (ostatus != noErr) { NSLog(@"fill audio sample from buffer list failed: %s", logAudioError(ostatus)); return; } ostatus = CMSampleBufferSetDataReady(sampleBuffer); if (ostatus != noErr) { NSLog(@"set sample buffer ready failed: %s", logAudioError(ostatus)); return; } // Finally we can attach it, then shove the presentation time forward [m_audio appendSampleBuffer:sampleBuffer]; The Crash The crash points towards some level of deallocation when the conversion tooling is done or has enough samples to process an output packet? It's had to say. 0 caulk 0x1a1e9532c caulk::alloc::tiered_allocator<caulk::alloc::size_range_tier<0ul, 1008ul, caulk::alloc::tree_allocator<caulk::alloc::chunk_allocator<caulk::alloc::page_allocator, caulk::alloc::bitmap_allocator, caulk::alloc::embed_block_memory, 16384ul, 16ul, 6ul>>>, caulk::alloc::size_range_tier<1009ul, 256000ul, caulk::alloc::guarded_edges_allocator<caulk::alloc::consolidating_free_map<caulk::alloc::page_allocator, 10485760ul>, 4ul>>, caulk::alloc::tracking_allocator<caulk::alloc::page_allocator>>::deallocate(caulk::alloc::block, unsigned long) + 636 1 AudioToolboxCore 0x1993fbfe4 ExtendedAudioBufferList_Destroy + 112 2 AudioToolboxCore 0x1993d5fe0 std::__1::__optional_destruct_base<ACCodecOutputBuffer, false>::~__optional_destruct_base[abi:ne180100]() + 68 3 AudioToolboxCore 0x1993d5f48 acv2::CodecConverter::~CodecConverter() + 196 4 AudioToolboxCore 0x1993d5e5c acv2::CodecConverter::~CodecConverter() + 16 5 AudioToolboxCore 0x1992574d8 std::__1::vector<std::__1::unique_ptr<acv2::AudioConverterBase, std::__1::default_delete<acv2::AudioConverterBase>>, std::__1::allocator<std::__1::unique_ptr<acv2::AudioConverterBase, std::__1::default_delete<acv2::AudioConverterBase>>>>::__clear[abi:ne180100]() + 84 6 AudioToolboxCore 0x199259acc acv2::AudioConverterChain::RebuildConverterChain(acv2::ChainBuildSettings const&) + 116 7 AudioToolboxCore 0x1992596ec acv2::AudioConverterChain::SetProperty(unsigned int, unsigned int, void const*) + 1808 8 AudioToolboxCore 0x199324acc acv2::AudioConverterV2::setProperty(unsigned int, unsigned int, void const*) + 84 9 AudioToolboxCore 0x199327f08 with_resolved(OpaqueAudioConverter*, caulk::function_ref<int (AudioConverterAPI*)>) + 60 10 AudioToolboxCore 0x1993281e4 AudioConverterSetProperty + 72 11 MediaToolbox 0x1a7566c2c FigSampleBufferProcessorCreateWithAudioCompression + 2296 12 MediaToolbox 0x1a754db08 0x1a70b5000 + 4819720 13 MediaToolbox 0x1a754dab4 FigMediaProcessorCreateForAudioCompressionWithFormatWriter + 100 14 MediaToolbox 0x1a77ebb98 0x1a70b5000 + 7564184 15 MediaToolbox 0x1a7804158 0x1a70b5000 + 7663960 16 MediaToolbox 0x1a7801da0 0x1a70b5000 + 7654816 17 AVFCore 0x1ada530c4 -[AVFigAssetWriterTrack addSampleBuffer:error:] + 192 18 AVFCore 0x1ada55164 -[AVFigAssetWriterAudioTrack _flushPendingSampleBuffersReturningError:] + 500 19 AVFCore 0x1ada55354 -[AVFigAssetWriterAudioTrack addSampleBuffer:error:] + 472 20 AVFCore 0x1ada4ebf0 -[AVAssetWriterInputWritingHelper appendSampleBuffer:error:] + 128 21 AVFCore 0x1ada4c354 -[AVAssetWriterInput appendSampleBuffer:] + 168 22 lib_devapple_hls.dylib 0x115d2c7cc detail::AppleHLSImplementation::audioRuntime() + 1052 23 lib_devapple_hls.dylib 0x115d2d094 void* std::__1::__thread_proxy[abi:ne180100]<std::__1::tuple<std::__1::unique_ptr<std::__1::__thread_struct, std::__1::default_delete<std::__1::__thread_struct>>, void (detail::AppleHLSImplementation::*)(), detail::AppleHLSImplementation*>>(void*) + 72 24 libsystem_pthread.dylib 0x196e5b2e4 _pthread_start + 136 Any insight would be welcome!

I developed an educational app that implements audio-video communication through RTC, while using WebView to display course materials during classes. However, some users are experiencing an issue where the audio playback from WebView is very quiet. I've checked that the AVAudioSessionCategory is set by RTC to AVAudioSessionCategoryPlayAndRecord, and the AVAudioSessionCategoryOption also includes AVAudioSessionCategoryOptionMixWithOthers. What could be causing the WebView audio to be suppressed, and how can this be resolved?

I’m developing a macOS audio monitoring app using AVAudioEngine, and I’ve run into a critical issue on macOS 26 beta where AVFoundation fails to detect any input devices, and AVAudioEngine.start() throws the familiar error 10877. FB#: FB19024508 Strange Behavior: AVAudioEngine.inputNode shows no channels or input format on bus 0. AVAudioEngine.start() fails with -10877 (AudioUnit connection error). AVCaptureDevice.DiscoverySession returns zero audio devices. Microphone permission is granted (authorized), and the app is properly signed and sandboxed with com.apple.security.device.audio-input. However, CoreAudio HAL does detect all input/output devices: Using AudioObjectGetPropertyDataSize and AudioObjectGetPropertyData with kAudioHardwarePropertyDevices, I can enumerate 14+ devices, including AirPods, USB DACs, and BlackHole. This suggests the lower-level audio stack is functional. I have tried: Resetting CoreAudio with sudo killall coreaudiod Rebuilding and re-signing the app Clearing TCC with tccutil reset Microphone Running on Apple Silicon and testing Rosetta/native detection via sysctl.proc_translated Using a fallback mechanism that logs device info from HAL and rotates logs for submission via Feedback Assistant I have submitted logs and a reproducible test case via Feedback Assitant : FB#: FB19024508]

Hello, I'm trying to receive parquet files using the example that provided in documentation. I've done all required steps but receive constantly error 500 with "Upstream Service Error". By looking into the issues list, seems this error exists for months. Is it possible to get it working?

Dear Sirs, I’ve written a virtual audio driver based on AudioDriverKit and running as dext in my MacOS app. Sometimes when waking up from a sleep state the recording side of my driver extension seems to hang and I don’t see any calls to my io_operation callback. Then the recording app like a DAW seems to hang when trying to start a recording. This doesn’t happen after short sleep states or after a complete new start of my MacBook. I already opened a case in Feedback-Assistant on 5th of May (FB17503622) which also includes a sysdiagnose and a ktrace but I didn't get any feedback so far. Meanwhile some of our customers are getting angry and I'd like to know if there's anything I could do to fix this problem on my side. We’re not sure whether this worked in previous MacOS versions, we think we didn’t observe this before 15.3.1 but at least since 15.3.1. we’ve seen this problem. Best regards, Johannes