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

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 ran 5.1 audio tests in both YouTube and Apple Music, and I noticed that when sound is supposed to play from the rear or front surround speakers, it’s also duplicated in the front left and right channels. I’m absolutely sure the issue is with the Apple TV, because I played the same video directly through my TV’s native system, and the channel separation was correct. Everything used to work perfectly before, so this must be a software issue. I’m currently on tvOS 26 Developer Beta 5, but I’m certain the problem also existed on the stable tvOS 18.5. I’ve already reset and updated my Apple TV, and I also tried switching the audio format to forced Dolby Atmos 5.1. On the forums, I mostly see complaints about Dolby Atmos not working at all — in my case, everything technically works, but not the way it’s supposed to.

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

Using the official SwiftTranscriptionSampleApp from WWDC 2025, speech transcription takes 14+ seconds from audio input to first result, making it unusable for real-time applications. Environment iOS: 26.0 Beta Xcode: Beta 5 Device: iPhone 16 pro Sample App: Official Apple SwiftTranscriptionSampleApp from WWDC 2025 Configuration Tested Locale: en-US (properly allocated with AssetInventory.allocate(locale:)) and es-ES Setup: All optimizations applied (preheating, high priority, model retention) I started testing in my own app to replace SFSpeech API and include speech detection but after long fights with documentation (this part is quite terrible TBH) I tested the example (https://developer.apple.com/documentation/speech/bringing-advanced-speech-to-text-capabilities-to-your-app) and saw same results. I added some logs to check the specific time: 🎙️ [20:30:41.532] ✅ Analyzer started successfully - ready to receive audio! 🎙️ [20:30:41.532] Listening for transcription results... 🎙️ [20:30:56.342] 🚀 FIRST TRANSCRIPTION RESULT after 14.810s: 'Hello' (isFinal: false) Questions Is this expected performance for iOS 26 Beta, because old SFSpeech is far faster? Are there additional optimization steps for SpeechTranscriber? Should we expect significant performance improvements in later betas?

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?

After updating to iOS 18.5, we’ve observed that outgoing audio from our app intermittently stops being transmitted during VoIP calls using AVAudioSession configured with .playAndRecord and .voiceChat. The session is set active without errors, and interruptions are handled correctly, yet audio capture suddenly ceases mid-call. This was not observed in earlier iOS versions (≤ 18.4). We’d like to confirm if there have been any recent changes in AVAudioSession, CallKit, or related media handling that could affect audio input behavior during long-running calls. func configureForVoIPCall() throws { try setCategory( .playAndRecord, mode: .voiceChat, options: [.allowBluetooth, .allowBluetoothA2DP, .defaultToSpeaker]) try setActive(true) }

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?

Does anyone know how to pronounce the sound of a specific instrument when you tap a button on the screen on your iPhone or iPad? Now, in the middle of creating a music learning app, I'm thinking of assigning monotones or chords to the button-like frames on the keyboard and fingerboard on the screen. Can it be achieved with SwiftUI chords alone? Once upon a time, MIDI level 1 I remember that there was a pronunciation function of the instrument, but I don't think about implementing the same function in the current OS. Please lend me your wisdom.

Thread 5 Crashed: 0 libobjc.A.dylib 0x19af7b038 objc_msgSend + 56 1 CoreFoundation 0x19dfdb618 cow_cleanup + 135 2 CoreFoundation 0x19dfdb6fc -[__NSDictionaryM dealloc] + 147 3 MediaToolbox 0x1b167636c FigRemotePropertyCacheTeardown + 31 4 MediaToolbox 0x1b1c5b648 remoteXPCAsset_Finalize + 107 5 CoreMedia 0x1b1e9166c FigBaseObjectFinalize + 275 6 CoreFoundation 0x19dfcc5ec _CFRelease + 295 7 AVFCore 0x1b1054d64 -[AVFigAssetTrackInspector dealloc] + 151 8 AVFCore 0x1b0f818d8 -[AVAssetTrack dealloc] + 63 9 CoreFoundation 0x19dfdba28 RELEASE_OBJECTS_IN_THE_ARRAY + 115 10 CoreFoundation 0x19dfdb7e0 -[__NSArrayM dealloc] + 147 11 AVFCore 0x1b0f52e04 -[AVURLAsset dealloc] + 167 12 libobjc.A.dylib 0x19af821f8 object_cxxDestructFromClass(objc_object*, objc_class*) + 115 13 libobjc.A.dylib 0x19af7df20 objc_destructInstance_nonnull_realized(objc_object*) + 75 14 libobjc.A.dylib 0x19af7d4a4 _objc_rootDealloc + 71 15 AVFCore 0x1b0fef988 -[AVAssetReaderOutput dealloc] + 415 16 AVFCore 0x1b0ff11ec -[AVAssetReaderTrackOutput dealloc] + 127 17 CoreFoundation 0x19dfe20a4 -[__NSSingleObjectArrayI dealloc] + 63 18 libobjc.A.dylib 0x19af7d3f8 AutoreleasePoolPage::releaseUntil(objc_object**) + 203

My app encountered problems when trying to open an x86 audioUnit v2 on a Silicon Mac (although Rosetta is installed). There seems to be a XPC connection issue with the AUHostingService that I don't know how to fix. I observed other host apps opening the same plugins without problem, so there is probably something wrong or incompatible in my codes. I noticed that: The issue occurs whether or not the app is sandboxed. The issue does no longer occur when the app itself runs under Rosetta. There is no error reported by CoreAudio during allocation and initialization of the audio unit. The first notified errors appears when the unit calls AudioUnitRender from the rendering callback. With most x86 plugins, the error is on first call: kAudioUnitErr_RenderTimeout and on any subsequent call: kAudioComponentErr_InstanceInvalidated On the UI side, when the Cocoa View is loaded, it appears shortly, then disappears immediately leaving its superview empty. With another x86 plugin, the Cocoa View is loaded normally, but CoreAudio still emits kAudioUnitErr_NoConnection from AudioUnitRender, whether the view has been loaded or not, and the plugin produces no sound. I also find these messages in the console (printed in that order): CLIENT ERROR: RemoteAUv2ViewController does not override - and thus cannot react to catastrophic errors beyond logging them AUAudioUnit_XPC.mm:641 Crashed AU possible component description: aumu/Helm/Tyte My app uses the AUv2 API and I suspect that working with the AUv3 API would spare me these problems. However, considering how my audio system is built (audio units are wrapped into C++ classes and most connections between units are managed on the fly from the rendering callback), it would be a lot of work to convert, and I’m even not sure that all I do with the AUv2 API would be possible with the AUv3 API. I could possibly find an intermediate solution, but in the immediate future I'm looking for the simplest and fastest possible fix. If I cannot find better, I see two fallback options: In this part of the doc: “Beginning with macOS 11, the system loads audio units into a separate process that depends on the architecture or host preference”, does “host preference” means that it would be possible to disable the “out of process” behavior, for example from the app entitlements or info.plist? Otherwise, as a last resort, I could completely disable the use of x86 audioUnits when my app runs under ARM64, for at least making things cleaner. But the Audio Component API doesn’t give any info about the plugin architecture, how could I found it? Any tip or idea about this issue will be much appreciated. Thanks in advance!

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?

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?

I'm using MusicKit for DRM track playback in my iOS app and a third party library to play local user-owned music on the file system and from the music library. This app is also supporting accessory devices that offer Bluetooth remote media control. The wish is to achieve parity between how the remote interacts with user owned music and the DRM / cloud / Apple Music tracks in my application music player. Track navigation, app volume (rather than system volume), and scrubbing need to work consistently on a mix of tracks which could alternate DRM and cloud status within one album or playlist. Apple Music queue and track pickers are not useful tools in my app. How can I support playing DRM and Apple Music tracks while not surrendering the remote control features to the system?

Hello, i can successfully match music using shazamkit on Apple using SwiftUI, a simple app that let user to load an audio file and exctracts the relative match, while i am unable to match music using shamzamkit on Android. I am trying to make the same simple app but i cannot match music as i get MATCH_ATTEMPT_FAILED every time i try to. I don't know what i am doing wrong but the shazam part in the kotlin Android code is in this method : suspend fun processAudioFileInBackground( filePath: String, developerTokenProvider: DeveloperTokenProvider ) = withContext(Dispatchers.IO) { val bufferSize = 1024 * 1024 val audioFile = FileInputStream(filePath) val byteBuffer = ByteBuffer.allocate(bufferSize) byteBuffer.order(ByteOrder.LITTLE_ENDIAN) var bytesRead: Int while (audioFile.read(byteBuffer.array()).also { bytesRead = it } != -1) { val signatureGenerator = (ShazamKit.createSignatureGenerator(AudioSampleRateInHz.SAMPLE_RATE_44100) as ShazamKitResult.Success).data signatureGenerator.append(byteBuffer.array(), bytesRead, System.currentTimeMillis()) val signature = signatureGenerator.generateSignature() println("Signature: ${signature.durationInMs}") val catalog = ShazamKit.createShazamCatalog(developerTokenProvider, Locale.ENGLISH) val session = (ShazamKit.createSession(catalog) as ShazamKitResult.Success).data val matchResult = session.match(signature) println("MatchResult : $matchResult") setMatchResult(matchResult) byteBuffer.clear() } audioFile.close() } I noticed that changing Locale in catalog creation results in different result as i get NoMatch without exception. Can you please help me with this? Do i need to create a custom catalog?

In Instruments, I'm seeing "Zero Time Stamp" events in the "Audio Server" lane. What does that mean?

A bit of a novice to app development here but I have a paid developer account, I have registered the identifier for MusicKit on the developer website (using the bundle identifier I've selected in Xcode) but the option to add MusicKit as a capability is not available in Xcode? I've manually updated the certificates, closed the app and reopened it, started a new project and tried with a different demo project? Apologies if I am missing something obvious but could someone help me get this capability added?

I've filed this as FB21446798 but figured I'd post here too. In the first build of macOS 26.3, playback via ApplicationMusicPlayer is completely broken. When starting playback of anything at all, the console shows the following error: applicationController: xpc service connection interrupted Failed to obtain remoteObject: Error Domain=NSCocoaErrorDomain Code=4099 "The connection to service created from an endpoint was invalidated from this process." UserInfo={NSDebugDescription=The connection to service created from an endpoint was invalidated from this process.} Failed to prepareToPlay with error: Error Domain=MPMusicPlayerControllerErrorDomain Code=10 "(null)" UserInfo={NSUnderlyingError=0xc92910ff0 {Error Domain=NSCocoaErrorDomain Code=4099 "The connection to service created from an endpoint was invalidated from this process." UserInfo={NSDebugDescription=The connection to service created from an endpoint was invalidated from this process.}}} In addition, several crash logs for RemotePlayerService are generated, showing my app as the parent process. This issue is 100% repeatable. No matter how I load the queue, whether it’s catalog or library content, any variation I can think of all fails like this. I really hope this can be fixed before 26.3 comes out, otherwise my app will be totally unusable. 😅

I am trying to get MIDI output from the AU Host demo app using the recent MIDI processor example. The processor works correctly in Logic Pro, but I cannot send MIDI from the AUv3 extension in standalone mode using the default host app to another program (e.g., Ableton). The MIDI manager, which is part of the standalone host app, works fine, and I can send MIDI using it directly—Ableton receives it without issues. I have already set the midiOutputNames in the extension, and the midiOutBlock is mapped. However, the MIDI data from the AUv3 extension does not reach Ableton in standalone mode. I suspect the issue is that midiOutBlock might never be called in the plugin, or perhaps an input to the plugin is missing, which prevents it from sending MIDI. I am currently using the default routing. I have modified the MIDI manager such that it works well as described above. Here is a part of my code for SimplePlayEngine.swift and my MIDIManager.swift for reference: @MainActor @Observable public class SimplePlayEngine { private let midiOutBlock: AUMIDIOutputEventBlock = { sampleTime, cable, length, data in return noErr } var scheduleMIDIEventListBlock: AUMIDIEventListBlock? = nil public init() { engine.attach(player) engine.prepare() setupMIDI() } private func setupMIDI() { if !MIDIManager.shared.setupPort(midiProtocol: MIDIProtocolID._2_0, receiveBlock: { [weak self] eventList, _ in if let scheduleMIDIEventListBlock = self?.scheduleMIDIEventListBlock { _ = scheduleMIDIEventListBlock(AUEventSampleTimeImmediate, 0, eventList) } }) { fatalError("Failed to setup Core MIDI") } } func initComponent(type: String, subType: String, manufacturer: String) async -> ViewController? { reset() guard let component = AVAudioUnit.findComponent(type: type, subType: subType, manufacturer: manufacturer) else { fatalError("Failed to find component with type: \(type), subtype: \(subType), manufacturer: \(manufacturer))" ) } do { let audioUnit = try await AVAudioUnit.instantiate( with: component.audioComponentDescription, options: AudioComponentInstantiationOptions.loadOutOfProcess) self.avAudioUnit = audioUnit self.connect(avAudioUnit: audioUnit) return await audioUnit.loadAudioUnitViewController() } catch { return nil } } private func startPlayingInternal() { guard let avAudioUnit = self.avAudioUnit else { return } setSessionActive(true) if avAudioUnit.wantsAudioInput { scheduleEffectLoop() } let hardwareFormat = engine.outputNode.outputFormat(forBus: 0) engine.connect(engine.mainMixerNode, to: engine.outputNode, format: hardwareFormat) do { try engine.start() } catch { isPlaying = false fatalError("Could not start engine. error: \(error).") } if avAudioUnit.wantsAudioInput { player.play() } isPlaying = true } private func resetAudioLoop() { guard let avAudioUnit = self.avAudioUnit else { return } if avAudioUnit.wantsAudioInput { guard let format = file?.processingFormat else { fatalError("No AVAudioFile defined.") } engine.connect(player, to: engine.mainMixerNode, format: format) } } public func connect(avAudioUnit: AVAudioUnit?, completion: @escaping (() -> Void) = {}) { guard let avAudioUnit = self.avAudioUnit else { return } engine.disconnectNodeInput(engine.mainMixerNode) resetAudioLoop() engine.detach(avAudioUnit) func rewiringComplete() { scheduleMIDIEventListBlock = auAudioUnit.scheduleMIDIEventListBlock if isPlaying { player.play() } completion() } let hardwareFormat = engine.outputNode.outputFormat(forBus: 0) engine.connect(engine.mainMixerNode, to: engine.outputNode, format: hardwareFormat) if isPlaying { player.pause() } let auAudioUnit = avAudioUnit.auAudioUnit if !auAudioUnit.midiOutputNames.isEmpty { auAudioUnit.midiOutputEventBlock = midiOutBlock } engine.attach(avAudioUnit) if avAudioUnit.wantsAudioInput { engine.disconnectNodeInput(engine.mainMixerNode) if let format = file?.processingFormat { engine.connect(player, to: avAudioUnit, format: format) engine.connect(avAudioUnit, to: engine.mainMixerNode, format: format) } } else { let stereoFormat = AVAudioFormat(standardFormatWithSampleRate: hardwareFormat.sampleRate, channels: 2) engine.connect(avAudioUnit, to: engine.mainMixerNode, format: stereoFormat) } rewiringComplete() } } and my MIDI Manager @MainActor class MIDIManager: Identifiable, ObservableObject { func setupPort(midiProtocol: MIDIProtocolID, receiveBlock: @escaping @Sendable MIDIReceiveBlock) -> Bool { guard setupClient() else { return false } if MIDIInputPortCreateWithProtocol(client, portName, midiProtocol, &port, receiveBlock) != noErr { return false } for source in self.sources { if MIDIPortConnectSource(port, source, nil) != noErr { print("Failed to connect to source \(source)") return false } } setupVirtualMIDIOutput() return true } private func setupVirtualMIDIOutput() { let virtualStatus = MIDISourceCreate(client, virtualSourceName, &virtualSource) if virtualStatus != noErr { print("❌ Failed to create virtual MIDI source: \(virtualStatus)") } else { print("✅ Created virtual MIDI source: \(virtualSourceName)") } } func sendMIDIData(_ data: [UInt8]) { print("hey") var packetList = MIDIPacketList() withUnsafeMutablePointer(to: &packetList) { ptr in let pkt = MIDIPacketListInit(ptr) _ = MIDIPacketListAdd(ptr, 1024, pkt, 0, data.count, data) if virtualSource != 0 { let status = MIDIReceived(virtualSource, ptr) if status != noErr { print("❌ Failed to send MIDI data: \(status)") } else { print("✅ Sent MIDI data: \(data)") } } } } }

I'm trying to use the new Speech framework for streaming transcription on macOS 26.3, and I can reproduce a failure with SpeechAnalyzer.start(inputSequence:). What is working: SpeechAnalyzer + SpeechTranscriber offline path using start(inputAudioFile:finishAfterFile:) same Spanish WAV file transcribes successfully and returns a coherent final result What is not working: SpeechAnalyzer + SpeechTranscriber stream path using start(inputSequence:) same WAV, replayed as AnalyzerInput(buffer:bufferStartTime:) fails once replay starts with: _GenericObjCError domain=Foundation._GenericObjCError code=0 detail=nilError I also tried: DictationTranscriber instead of SpeechTranscriber no realtime pacing during replay Both still fail in stream mode with the same error. So this does not currently look like a ScreenCaptureKit issue or a Python integration issue. I reduced it to a pure Swift CLI repro. Environment: macOS 26.3 (25D122) Xcode 26.3 Swift 6.2.4 Apple Silicon Mac Has anyone here gotten SpeechAnalyzer.start(inputSequence:) working reliably on macOS 26.x? If so, I'd be interested in any workaround or any detail that differs from the obvious setup: prepareToAnalyze(in:) bestAvailableAudioFormat(...) AnalyzerInput(buffer:bufferStartTime:) replaying a known-good WAV in chunks I already filed Feedback Assistant: FB22149971