Hello, I'm trying to determine the best/recommended AVAudioSession configuration (i.e category, mode, and options) for the following use-case. Essentially, I'd like to switch between periods of playing an audio file and then recognizing speech. The audio file is typically speech and I don't intend for playback and speech recognition to occur simultaneously. I'd like for the user to sill be able to interact with Siri and I'd like for it to work with CarPlay where navigation prompts can occur. I would assume the category to use is 'playAndRecord', but I'm not sure if it's better to just set that once for the entire lifecycle, or set to 'playback' for audio file playback and then switch to 'playAndRecord' for speech recognition . I'm also not sure on the best 'mode' and 'options' to set. Any suggestions would be appreciated. Thanks.

ApplicationMusicPlayer is not available on watchOS but all other platforms. Is there a technical reason for that like battery life? Same goes for SystemMusicPlayer and MPMusicPlayerController. I already filed feedbacks for that.

I created a virtual audio device to capture system audio with a sample rate of 44.1 kHz. After capturing the audio, I forward it to the hardware sound card using AVAudioEngine, also with a sample rate of 44.1 kHz. However, due to the clock sources being unsynchronized, problems occur after a period of playback. How can I retrieve the clock source of the hardware device and set it for the virtual device?

I am a graduate student conducting research in speech/audio signal processing and multimodal interaction. Apple Vision Pro is widely recognized as a multimodal interactive system supporting voice, eye, and gesture inputs. However, I could not find detailed specifications or documentation about the audio input sampling rate used by the device’s built-in microphone array when capturing user audio. Specifically, I would like to understand: What is the default audio input sampling rate (e.g., 16 kHz, 44.1 kHz, 48 kHz, etc.) for the Vision Pro’s microphones? When developing with visionOS / AVAudioSession / AVAudioEngine, is there a documented or recommended sampling rate for audio capture? Are there any best practices or settings for enabling high-quality voice capture on Vision Pro (especially for voice research tasks)? For context, my work involves voice processing, analysis, and possibly on-device real-time speech recognition. Any pointers to relevant APIs, documentation or examples (especially regarding audio capture buffer size or available formats on visionOS) would be very helpful. Thank you in advance! Best regards.

I have some tried-and-tested code that records and plays back audio via AUHAL which breaks on Tahoe on Intel. The same code works fine on Sequioa and also works on Tahoe on Apple Silicon. To start with something simple, the following code to request access to the Microphone doesn't work as it should: bool RequestMicrophoneAccess () { __block AVAuthorizationStatus status = [AVCaptureDevice authorizationStatusForMediaType: AVMediaTypeAudio]; if (status == AVAuthorizationStatusAuthorized) return true; __block bool done = false; [AVCaptureDevice requestAccessForMediaType: AVMediaTypeAudio completionHandler: ^ (BOOL granted) { status = (granted) ? AVAuthorizationStatusAuthorized : AVAuthorizationStatusDenied; done = true; }]; while (!done) CFRunLoopRunInMode (kCFRunLoopDefaultMode, 2.0, true); return status == AVAuthorizationStatusAuthorized; } On Tahoe on Intel, the code runs to completion but granted is always returned as NO. Tellingly, the popup to ask the user to grant microphone access is never displayed, even though the app is not present in the Privacy pane and never appears there. On Apple Silicon, everything works fine. There are some other problems, but I'm hoping they have a common underlying cause and that the Apple guys can figure out what's wrong from the information in this post. I'd be happy to test any potential fix. Thanks.

I'm getting Crashlytics crashes from some my users, deep in the Apple code: Crashed: AXSpeech EXC_BAD_ACCESS KERN_INVALID_ADDRESS 0x00000007ec54b360 0 libobjc.A.dylib 0x3c9c objc_retain_x8 + 16 1 AudioToolboxCore 0x99580 auoop::RenderPipeUser::~RenderPipeUser() + 112 2 AudioToolboxCore 0xe6090 -[AUAudioUnit_XPC internalDeallocateRenderResources] + 92 3 AVFAudio 0x90a0 AUInterfaceBaseV3::Uninitialize() + 60 4 AVFAudio 0x4cbe0 AVAudioEngineGraph::PerformCommand(AUGraphNodeBaseV3&, AVAudioEngineGraph::ENodeCommand, void*, unsigned int) const + 768 5 AVFAudio 0x56b0c AVAudioEngineGraph::_Uninitialize(NSError**) + 132 6 AVFAudio 0x7834 AVAudioEngineImpl::Stop(NSError**) + 388 7 AVFAudio 0x636c -[AVAudioEngine dealloc] + 52 8 TextToSpeech 0x30674 _TTSNameForVoiceInformation + 20864 9 libobjc.A.dylib 0x20a4 object_cxxDestructFromClass(objc_object*, objc_class*) + 116 10 libobjc.A.dylib 0x6e00 objc_destructInstance + 80 11 libobjc.A.dylib 0x104fc _objc_rootDealloc + 80 12 TextToSpeech 0x2d2f4 _TTSNameForVoiceInformation + 7680 13 TextToSpeech 0x496c TTSVocalizerCopyURLForFallbackResource + 8540 14 TextToSpeech 0x26094 TTSSpeechUnitTestingMode + 5548 15 libAXSpeechManager.dylib 0x108b0 -[AXSpeechManager .cxx_destruct] + 192 16 libobjc.A.dylib 0x20a4 object_cxxDestructFromClass(objc_object*, objc_class*) + 116 17 libobjc.A.dylib 0x6e00 objc_destructInstance + 80 18 libobjc.A.dylib 0x104fc _objc_rootDealloc + 80 19 libAXSpeechManager.dylib 0x5298 -[AXSpeechManager dealloc] + 268 20 Foundation 0x3b8a4 __NSThreadPerformPerform + 272 21 CoreFoundation 0xd3208 __CFRUNLOOP_IS_CALLING_OUT_TO_A_SOURCE0_PERFORM_FUNCTION__ + 28 22 CoreFoundation 0xdf864 __CFRunLoopDoSource0 + 176 23 CoreFoundation 0x646c8 __CFRunLoopDoSources0 + 244 24 CoreFoundation 0x7a1c4 __CFRunLoopRun + 828 25 CoreFoundation 0x7f4dc CFRunLoopRunSpecific + 612 26 Foundation 0x420c4 -[NSRunLoop(NSRunLoop) runMode:beforeDate:] + 212 27 libAXSpeechManager.dylib 0x13390 -[AXSpeechThread main] + 552 28 Foundation 0x5b634 __NSThread__start__ + 716 29 libsystem_pthread.dylib 0x16b8 _pthread_start + 148 30 libsystem_pthread.dylib 0xb88 thread_start + 8 It's most likely related to my use of AVSpeechSynthesizer. I do change some of the utterance fields, including the voice that's being used (which is set to a value from speechVoices()). UtilAudioIos_tts = AVSpeechSynthesizer() let utterance = AVSpeechUtterance utterance.voice = AVSpeechSynthesisVoice(identifier: voice.voiceCode) utterance.volume = volume utterance.pitchMultiplier = pitch utterance.rate = rate UtilAudioIos_tts!.speak(utterance) By coincidence or not, the following sometimes appears in the device log: 2023-05-30 20:35:29.948078+0100 <appname>[466:12882] [catalog] Unable to list voice folder and also, sometimes: 2023-05-30 20:37:35.345933+0100 <appname>[466:13298] [catalog] Query for com.apple.MobileAsset.VoiceServices.VoiceResources failed: 2 2023-05-30 20:37:35.360854+0100 rehearserfree[466:13433] [AXTTSCommon] MauiVocalizer: 11006 (Can't compile rule): regularExpression=\Oviedo(?=, (\x1b\\pause=\d+\\)?Florida)\b, message=unrecognized character follows \, characterPosition=1 2023-05-30 20:37:35.363163+0100 <appname>[466:13433] [AXTTSCommon] MauiVocalizer: 16038 (Resource load failed): component=ttt/re, uri=, contentType=application/x-vocalizer-rettt+text, lhError=88602000 2023-05-30 20:37:35.363182+0100 <appname>[466:13433] [AXTTSCommon] Error loading rules: 2147483648 All of these crashes have been on the various versions of iOS 16. Edit: I can't reproduce the crash myself - it's just some (not all) app users. The log entries above appear locally on my device (with no crash) but I can't see the logs of the users who have the crashes. Any idea what this might be caused by, or how to go about tracking the problem down?

Is there a way to destroy MIDIUMPMutableEndpoint again? In my app, the user has a setting to enable and disable MIDI 2.0. If MIDI 2.0 should not be supported (or if iOS version < 18), it creates a virtual destination and a virtual source. And if MIDI 2.0 should be enabled, it instead creates a MIDIUMPMutableEndpoint, which itself creates the virtual destination and source automatically. So here is my problem: I didn't find any way to destroy the MIDIUMPMutableEndpoint again. There is a method to disable it (setEnabled:NO), but that doesn't destroy or hide the virtual destination and source. So when the user turns MIDI 2.0 support off, I will have two virtual destinations and sources, and cannot get rid of the 2.0 ones. What is the correct way to get rid of the MIDIUMPMutableEndpoint once it is created?

Hi all, with my app ScreenFloat, you can record your screen, along with system- and microphone audio. Those two audio feeds are recorded into separate audio tracks in order to individually remove or edit them later on. Now, these recordings you create with ScreenFloat can be drag-and-dropped to other apps instantly. So far, so good, but some apps, like Slack, or VLC, or even websites like YouTube, do not play back multiple audio tracks, just one. So what I'm trying to do is, on dragging the video recording file out of ScreenFloat, instantly baking together the two individual audio tracks into one, and offering that new file as the drag and drop file, so that all audio is played in the target app. But it's slow. I mean, it's actually quite fast, but for drag and drop, it's slow. My approach is this: "Bake together" the two audio tracks into a one-track m4a audio file using AVMutableAudioMix and AVAssetExportSession Take the video track, add the new audio file as an audio track to it, and render that out using AVAssetExportSession For a quick benchmark, a 3'40'' movie, step 1 takes ~1.7 seconds, and step two adds another ~1.5 seconds, so we're at ~3.2 seconds. That's an eternity for a drag and drop, where the user might cancel if there's no immediate feedback. I could also do it in one step, but then I couldn't use the AV*Passthrough preset, and that makes it take around 32 seconds then, because I assume it touches the video data (which is unnecessary in this case, so I think the two-step approach here is the fastest). So, my question is, is there a faster way? The best idea I can come up with right now is, when initially recording the screen with system- and microphone audio as separate tracks, to also record both of them into a third, muted, "hidden" track I could use later on, basically eliminating the need for step one and just ripping the two single audio tracks out of the movie and only have the video and the "hidden" track (then unmuted), but I'd still have a ~1.5 second delay there. Also, there's the processing and data overhead (basically doubling the movie's audio data). All this would be great for an export operation (where one expects it to take a little time), but for a drag-and-drop operation, it's not ideal. I've discarded the idea of doing a promise file drag, because many apps do not accept those, and I want to keep wide compatibility with all sorts of apps. I'd appreciate any ideas or pointers. Thank you kindly, Matthias

In iOS 18, CarPlay shows an error: “There was a problem loading this content” after playback starts. Audio works fine, but the Now Playing screen doesn’t load. I’m using MPPlayableContentManager. This worked fine in iOS 17. Anyone else seeing this error in iOS 18?

Environment Device: iPhone 16e iOS Version: 18.4.1 - 18.7.1 Framework: AVFoundation (AVAudioEngine) Problem Summary On iPhone 16e (iOS 18.4.1-18.7.1), the installTap callback stops being invoked after resuming from a phone call interruption. This issue is specific to phone call interruptions and does not occur on iPhone 14, iPhone SE 3, or earlier devices. Expected Behavior After a phone call interruption ends and audioEngine.start() is called, the previously installed tap should continue receiving audio buffers. Actual Behavior After resuming from phone call interruption: Tap callback is no longer invoked No audio data is captured No errors are thrown Engine appears to be running normally Note: Normal pause/resume (without phone call interruption) works correctly. Steps to Reproduce Start audio recording on iPhone 16e Receive or make a phone call (triggers AVAudioSession interruption) End the phone call Resume recording with audioEngine.start() Result: Tap callback is not invoked Tested devices: iPhone 16e (iOS 18.4.1-18.7.1): Issue reproduces ✗ iPhone 14 (iOS 18.x): Works correctly ✓ iPhone SE 3 (iOS 18.x): Works correctly ✓ Code Initial Setup (Works) let inputNode = audioEngine.inputNode inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in self.processAudioBuffer(buffer, at: time) } audioEngine.prepare() try audioEngine.start() Interruption Handling NotificationCenter.default.addObserver( forName: AVAudioSession.interruptionNotification, object: AVAudioSession.sharedInstance(), queue: nil ) { notification in guard let userInfo = notification.userInfo, let typeValue = userInfo[AVAudioSessionInterruptionTypeKey] as? UInt, let type = AVAudioSession.InterruptionType(rawValue: typeValue) else { return } if type == .began { self.audioEngine.pause() } else if type == .ended { try? self.audioSession.setActive(true) try? self.audioEngine.start() // Tap callback doesn't work after this on iPhone 16e } } Workaround Full engine restart is required on iPhone 16e: func resumeAfterInterruption() { audioEngine.stop() inputNode.removeTap(onBus: 0) inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in self.processAudioBuffer(buffer, at: time) } audioEngine.prepare() try audioSession.setActive(true) try audioEngine.start() } This works but adds latency and complexity compared to simple resume. Questions Is this expected behavior on iPhone 16e? What is the recommended way to handle phone call interruptions? Why does this only affect iPhone 16e and not iPhone 14 or SE 3? Any guidance would be appreciated!

I have a music app I'm developing and having a weird issue where I can see now playing info for every other platform than tvOS. As far as I can tell I have correctly configured the MPNowPlayingInfoCenter MPNowPlayingInfoCenter.default().nowPlayingInfo = nowPlayingInfo MPNowPlayingInfoCenter.default().playbackState = .playing Are there any extra requirements to get my app's now-playing info showing in control center on tvOS? Another strange issue that might be related is I can use the apple TV remote to pause audio but not resume playback, so I feel like there's something I'm missing about registering audio playback on tvOS specifically.

I have a Catalyst app ('container') which hosts an embedded AUv3 Audio Unit extension ('plugin'). This used to work for years and has worked with this project until a few days ago. it still works on iOS as expected on MacOS the extension is never registered/installed and won't load extension won't show up with AUVal seems to have stopped working with the 26.1 XCode update I'm fairly certain the problem is not code related (i.e. likely build settings, project settings, entitlements, signing, etc.) I have compared all settings with another still-working project and can't find any meaningful difference (I can't request code-level support because even the minimal thing vastly exceeds the 250 lines of code limit.) How can I debug the issue? I literally don't know where to start to fix this problem, short of rebuilding the entire thing and hope that it magically starts working again.

My audio app shows a control bar at the bottom of the window. The controls show nicely, but there is a black "slab" appearing behind the inline controls, the same size as the playerView. Setting the player view background color does nothing: playerView.wantsLayer = true playerView.layer?.backgroundColor = NSColor.clear.cgColor How can I clear the background? If I use .floating controlsStyle, I don't get the background "slab".

Your draft looks great! Here's a refined version with the iOS 17 comparison emphasized and slightly better flow: Hi Apple Engineers and fellow developers, I'm experiencing a critical regression with ShazamKit's background operation on iOS 18. ShazamKit's SHManagedSession stops identifying songs in the background after approximately 20 seconds on iOS 18, while the exact same code works perfectly on iOS 17. The behavior is consistent: the app works perfectly in the foreground, but when backgrounded or device is locked, it initially works for about 20 seconds then stops identifying new songs. The microphone indicator remains active suggesting audio access is maintained, but ShazamKit doesn't send identified songs in the background until you open the app again. Detection immediately resumes when bringing the app to foreground. My technical setup uses SHManagedSession for continuous matching with background modes properly configured in Info.plist including audio mode, and Background App Refresh enabled. I've tested this on physical devices running iOS 18.0 through 18.5 with the same results across all versions. The exact same code running on iOS 17 devices works flawlessly in the background. To reproduce: initialize SHManagedSession and start matching, begin song identification in foreground, background the app or lock device, play different songs which are initially detected for about 20 seconds, then after the timeout period new songs are no longer identified until you bring the app to foreground. This regression has impacted my production app as users who rely on continuous background music identification are experiencing a broken feature. I submitted this as Feedback ID FB15255903 last September with no solution so far. I've created a minimal demo project that reproduces this issue: https://github.com/tfmart/ShazamKitBackground Has anyone else experienced this ShazamKit background regression on iOS 18? Are there any known workarounds or alternative approaches? Given the time this issue has persisted, could we please get acknowledgment of this regression, expected timeline for a fix, or any recommended workarounds? Testing environment is Xcode 16.0+ on iOS 18.0-18.5 across multiple physical device models. Any guidance would be greatly appreciated.

Problem: When the screen is locked, an incoming call does not initiate the launch of the Flutter application required for audio and video communication through Jitsi Meet. In the unlocked state, the application functions correctly. The current implementation does not have a mechanism for activating the Flutter engine when receiving a call via CallKit while the screen is locked. Although CallKit UI displays the call acceptance interface and the audio session is configured, the Flutter application remains in a suspended state, making it impossible to connect to the media server. Audio session activated using didActivateAudioSession method.

My code that streams buffers into AVAudioPlayerNode is stuttering when the buffer is finished and before the next one is played. while engine.isRunning { let framesToCopy = min(buffer.frameLength - framePosition, Self.BufferSize) let srcRaw = UnsafeRawPointer(srcPtr) let playbackBuffer = AVAudioPCMBuffer(pcmFormat: buffer.format, frameCapacity: Self.BufferSize)! let playbackPtr = playbackBuffer.floatChannelData![0] let destRaw = UnsafeMutableRawPointer(mutating: playbackPtr) memcpy(destRaw, srcRaw, Int(framesToCopy) * MemoryLayout<Float>.stride) srcPtr = srcPtr.advanced(by: Int(framesToCopy)) playbackBuffer.frameLength = framesToCopy await player.scheduleBuffer(playbackBuffer, at: nil, options: [], completionCallbackType: .dataRendered) } I've tried to schedule multiple buffers at once using a combination of both the synchronous and async versions of scheduleBuffer because I thought the delay might be but it still stutters and the data copied into the playbackBuffer matches the source buffer. I've tried all combinations of options and completionCallbackType but no luck. I've tried increasing the buffer size but that just spaces out the stutters because the buffer is larger. What am I missing about this API?

Hello. To determine wether "AVB/EAV Mode" of a AV-capable network interfaces is turned on or off I query the IO registry and evaluate the property "AVBControllerState". I was wondering if this is the "correct" approach and if there is anything known about the values for this property? Network interfaces without AV capability may also carry this property (e.g.: for my WiFi adapter the value of 1) whereas the value for interfaces with AV capability can be 0 and 3. At least as far as I could observe with my limited amount of test devices at hand. Is it safe to assume that a value of 3 means this feature is turned on, 0 that it is turned off and ignore values of 1? Is there another approach to get to know the status of the "AVB/EAV Mode"? Thanks for any insight. Best regards, Ingo

Hello, I've discovered a buffer initialization bug in AVAudioUnitSampler that happens when loading presets with multiple zones referencing different regions in the same audio file (monolith/concatenated samples approach). Almost all zones output silence (i.e. zeros) at the beginning of playback instead of starting with actual audio data. The Problem Setup: Single audio file (monolith) containing multiple concatenated samples Multiple zones in an .aupreset, each with different sample start and sample end values pointing to different regions of the same file All zones load successfully without errors Expected Behavior: All zones should play their respective audio regions immediately from the first sample. Actual Behavior: Last zone in the zone list: Works perfectly - plays audio immediately All other zones: Output [0, 0, 0, 0, ..., _audio_data] instead of [real_audio_data] The number of zeros varies from event to event for each zone. It can be a couple of samples (<30) up to several buffers. After the initial zeros, the correct audio plays normally, so there is no shift in audio playback, just missing samples at the beginning. Minimal Reproduction 1. Create Test Monolith Audio File Create a single Wav file with 3 concatenated 1-second samples (44.1kHz): Sample 1: frames 0-44099 (constant amplitude 0.3) Sample 2: frames 44100-88199 (constant amplitude 0.6) Sample 3: frames 88200-132299 (constant amplitude 0.9) 2. Create Test Preset Create an .aupreset with 3 zones all referencing the same file: Pseudo code <Zone array> <zone 1> start : 0, end: 44099, note: 60, waveform: ref_to_monolith.wav; <zone 2> start sample: 44100, note: 62, end sample: 88199, waveform: ref_to_monolith.wav; <zone 3> start sample: 88200, note: 64, end sample: 132299, waveform: ref_to_monolith.wav; </Zone array> 3. Load and Test // Load preset into AVAudioUnitSampler let sampler = AVAudioUnitSampler() try sampler.loadAudioFiles(from: presetURL) // Play each zone (MIDI notes C4=60, D4=62, E4=64) sampler.startNote(60, withVelocity: 64, onChannel: 0) // Zone 1 sampler.startNote(62, withVelocity: 64, onChannel: 0) // Zone 2 sampler.startNote(64, withVelocity: 64, onChannel: 0) // Zone 3 4. Observed Result Zone 1 (C4): [0, 0, 0, ..., 0.3, 0.3, 0.3] ❌ Zeros at beginning Zone 2 (D4): [0, 0, 0, ..., 0.6, 0.6, 0.6] ❌ Zeros at beginning Zone 3 (E4): [0.9, 0.9, 0.9, ...] ✅ Works correctly (last zone) What I've Extensively Tested What DOES Work Separate files per zone: Each zone references its own individual audio file All zones play correctly without zeros Problem: Not viable for iOS apps with 500+ sample libraries due to file handle limitations What DOESN'T Work (All Tested) 1. Different Audio Formats: CAF (Float32 PCM, Int16 PCM, both interleaved and non-interleaved) M4A (AAC compressed) WAV (uncompressed) SF2 (SoundFont2) Bug persists across all formats 2. CAF Region Chunks: Created CAF files with embedded region chunks defining zone boundaries Set zones with no sampleStart/sampleEnd in preset (nil values) AVAudioUnitSampler completely ignores CAF region metadata Bug persists 3. Unique Waveform IDs: Gave each zone a unique waveform ID (268435456, 268435457, 268435458) Each ID has its own file reference entry (all pointing to same physical file) Hypothesized this might trigger separate buffer initialization Bug persists - no improvement 4. Different Sample Rates: Tested: 44.1kHz, 48kHz, 96kHz Bug occurs at all sample rates 5. Mono vs Stereo: Bug occurs with both mono and stereo files Environment macOS: Sonoma 14.x (tested across multiple minor versions) iOS: Tested on iOS 17.x with same results Xcode: 16.x Frameworks: AVFoundation, AudioToolbox Reproducibility: 100% reproducible with setup described above Impact & Use Case This bug severely impacts professional music applications that need: Small file sizes: Monolith files allow sharing compressed audio data (AAC/M4A) iOS file handle limits: Opening 400+ individual sample files is not viable on iOS Performance: Single file loading is much faster than hundreds of individual files Standard industry practice: Monolith/concatenated samples are used by EXS24, Kontakt, and most professional samplers Current Impact: Cannot use monolith files with AVAudioUnitSampler on iOS Forced to choose between: unusable audio (zeros at start) OR hitting iOS file limits No viable workaround exists Root Cause Hypothesis The bug appears to be in AVAudioUnitSampler's internal buffer initialization when: Multiple zones share the same source audio file Each zone specifies different sampleStart/sampleEnd offsets Key observation: The last zone in the zone array always works correctly. This is NOT related to: File permissions or security-scoped resources (separate files work fine) Audio codec issues (happens with uncompressed PCM too) Preset parsing (preset loads correctly, all zones are valid) Questions Is this a known issue? I couldn't find any documentation, bug reports, or discussions about this. Is there ANY workaround that allows monolith files to work with AVAudioUnitSampler? Alternative APIs? Is there a different API or approach for iOS that properly supports monolith sample files?

Dear Sirs, I'd like to add an icon to my audio driver based on AudioDriverKit. This icon should show up left of my audio device in the audio devices dialog. For an Audio Server Plugin I managed to do this using the property kAudioDevicePropertyIcon and CFBundleCopyResourceURL(...) but how would you do this with AudioDriverKit? Should I use IOUserAudioCustomProperty or IOUserAudioControl and how would I refer to the Bundle? Is there an example available somewhere? Thanks and best regards, Johannes

Hello, I have an existing AUv3 instrument plugin. In the plug in, users can access files (audio files, song projects) via a UIDocumentPickerViewController In Logic Pro, (and some other hosts, but not all), the document picker is unable to receive touches, while a keyboard case is attached to the iPad. Removing the case (this is an Apple brand iPad case) allows the interactions to resume and allows me to pick files in the usual way. One of my users reports this non-responsive behavior occurs even after disconnecting their keyboard. I have fiddled with entitlements all day, and have determined that is not the issue, since the keyboard disconnection appears to fix it every time for me. Here is my, very boilerplate, presentation code : guard let type = UTType("com.my.type") else { return } let fileBrowser = UIDocumentPickerViewController(forOpeningContentTypes: [type]) fileBrowser.overrideUserInterfaceStyle = .dark fileBrowser.delegate = self fileBrowser.directoryURL = myFileFolderURL() self.present(fileBrowser, animated: true) {