Hey y'all, I'm new here and in the process of building some audio software. I'm hitting a roadblock trying to incorporate VPIO into audio playback states without it ducking what's currently playing in my DAW. I have a few questions: Can VPIO other-audio ducking be completely disabled on macOS, beyond fixed/minimum? Minimum is not enough for professional audio environments. Is attenuation of an unrelated application routed through a different Core Audio output device expected? Is split-device VPIO—built-in microphone input with Apollo/Universal Audio output—supported? Why can AVAudioEngine.start() succeed while the engine remains stopped and immediately emits a configuration-change notification? Thanks for any insight into this!

Hi all,  Apple dropping on-going development for FireWire devices that were supported with the Core Audio driver standard is a catastrophe for a lot of struggling musicians who need to both keep up to date on security updates that come with new OS releases, and continue to utilise their hard earned investments in very expensive and still pristine audio devices that have been reduced to e-waste by Apple's seemingly tone-deaf ignorance in the cries for on-going support.  I have one of said audio devices, and I'd like to keep using it while keeping my 2019 Intel Mac Book Pro up to date with the latest security updates and OS features.  Probably not the first time you gurus have had someone make the logical leap leading to a request for something like this, but I was wondering if it might be somehow possible of shoe-horning the code used in previous versions of Mac OS that allowed the Mac to speak with the audio features of such devices to run inside the Ventura version of the OS.  Would it possible? Would it involve a lot of work? I don't think I'd be the only person willing to pay for a third party application or utility that restored this functionality. There has to be 100's of thousands of people who would be happy to spare some cash to stop their multi-thousand dollar investment in gear to be so thoughtlessly resigned to the scrap heap.  Any comments or layman-friendly explanations as to why this couldn’t happen would be gratefully received!  Thanks,  em

I would need to know which gender is used by the iOS system for speechSynthesis. I have tried let aVoice = AVSpeechSynthesisVoice() print(#function, #line, aVoice.gender.rawValue) But always get 0, for unspecified

Tested iPhone 16 Pro Max running DB2 Since using iOS 27 Developer Beta, Progressive Web Apps (PWAs) and applications using WKWebView that route live streaming audio (such as Icecast/SHOUTcast streams) through the Web Audio API (AudioContext / MediaElementAudioSourceNode for canvas visualizers) experience silent playback freezes during track changes. The JavaScript context remains active, and the HTMLMediaElement reports no errors or pauses, but the low-level system audio daemon (audiod) disconnects. The logs indicate a state desynchronization in MediaToolbox triggered by inline metadata frame updates, coupled with massive sandbox violations. Steps to reproduce: Run a PWA or WKWebView on iOS 27 Developer Beta that loads a live audio stream using an tag. Route the media element into the Web Audio API (e.g., audioContext.createMediaElementSource(audioElement)). Play the stream and lock the screen or let it run in the foreground. Wait for a track transition to occur on the live stream server (which pushes an inline ICY metadata update). Observed Result: Audio instantly goes silent at the track boundary. JavaScript rendering loops remain active, but the stream is dead. List of important errors: MediaToolbox parsing failure: <<<< FigStreamPlayer >>>> fpfs_CacheRenderChain: Caching unexpected mediatype metadata Reasoning: When the stream updates its inline metadata, MediaToolbox (FigStreamPlayer) fails to handle the frame descriptor correctly, halting the stream decoder render chain. Audio engine state desynchronization: <<<< FigStreamPlayer >>>> fpfs_SetRateOnTrack: [...] setting rate on track before it has reached playing state - 4 instead Reasoning: WebKit attempts to keep playing or setting the playback rate, but MediaToolbox fails because the underlying decoder track is stuck in an uninitialized/stalled state (4). System Audio Daemon Disconnection: Subsystem: com.apple.audioanalytics | Process: audiod Reporter disconnected. ( function=sendMessage, reporterID=... ) Reasoning: Because the rendering stream stalled, the system audio daemon (audiod) forcibly severs the audio session connection to the browser container. Massive Sandbox Log Flooding (WebKit.WebContent): Sandbox: com.apple.WebKit.WebContent(2719) deny(1) syscall-unix 179 Reasoning: WebKit's WebAudio thread tries to log trace markers to XNU's kdebug_trace64 (syscall 179) but is blocked by the WebContent sandbox profile, resulting in a loop of thousands of denials that clog the thread. What needs to be fixed: MediaToolbox (FigStreamPlayer): Fix the regression where incoming live stream metadata frames stall the render chain when connected to a Web Audio node. WebKit Sandbox Profiles: Update the iOS WebContent, Network, and GPU sandbox rules to allow necessary tracing/diagnostic calls (syscall-unix 179 and mach-lookup com.apple.diagnosticd) in developer beta builds to prevent thread-blocking loops.

Dear All, I am currently developing a high throughput audio system which operates via PCIe tunneled into a USB4 interface. This include a custom FPGA based hardware and custom Audio DriverKit driver. While performing read operation via the hw DMA (that is a Host to Device transfer), I am noticing sparse latency spikes into the read transfers. Specifically, 4KB operations (which I assume including MRd + CpID) take normally from 5us to 40us to be completed, perfectly fine for my case. However, in some rare occasions, they can end up to 400us, which causes me overruns. The measurements have been carried out from the FPGA and they include the overall request and transfer time. While trying to tackle the problem, I'm investigating the possible power saving options and performance constraint methods at my disposal. I currently use these methods to mitigate the problem. ChangePowerState(kIOServicePowerCapabilityOn); SetPowerOverride(true); RequireMaxBusStall(kIOMaxBusStall25usec); CreatePMAssertion(kIOServicePMAssertionCPUBit | kIOServicePMAssertionForceFullWakeupBit, &ivars->PMAssertionID, false); The buffers are currently about 16MB, single segment, 16KB aligned and, of course, "prepared" for DMA. The system run for 3 hours without any overrun, but I'm not still fully convinced about its reliability. May someone provide me some comments on this? Are there profiling tools that I can use? Feel free to request me any required detail. The testing system is a MacBook Pro M2 Pro. Many Thanks and Best Regards Francesco

https://streamable.com/y16s4q When I leave the Spotify Web Player open on its homepage and remain completely idle (without any physical user interaction or clicks), the page somehow self-triggers an active audio stream. Instantly, Safari's standard "This tab is playing audio" speaker icon appears normally on the right side of the URL/Address bar. Simultaneously, the browser emits a 1–2 second burst of extremely accelerated, fast-forwarded, and chaotic audio (resembling a time-stretch or "Alvin and the Chipmunks" effect) before going silent or stabilizing.

I'm building a virtual audio device using the HAL Plugin (AudioServerPlugin) API on macOS. The plugin loads correctly, WriteMix is called with non-zero audio data, but Plugin_StartIO is never called from my app's AUHAL input client. Environment: macOS 26.5 Xcode (latest) Bundle ID: com.private.SubON Plugin installed at: /Library/Audio/Plug-Ins/HAL/SubONHALPlugin.driver What works: Plugin loads and is visible as "SubON Virtual Device" Plugin_AddDeviceClient is called with bundleID=com.private.SubON WriteMix is called with non-zero maxSrc values (audio is reaching the plugin) AUHAL setup completes: EnableIO input=1 output=0, CurrentDevice set correctly, AudioUnitInitialize succeeds pre-start readback confirms: currentDevice matches, enableIOIn=1 What doesn't work: Plugin_StartIO is never called from the SubON.app input client WillDoIOOperation is never called ReadInput is never called maxAmplitude in the input tap is always 0.0 AUHAL setup code (Swift): AudioUnitUninitialize(auhal) AudioUnitSetProperty(auhal, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &one, size) AudioUnitSetProperty(auhal, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &zero, size) AudioUnitSetProperty(auhal, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &deviceID, size) AudioUnitInitialize(auhal) HAL Plugin — WillDoIOOperation: outWillDo = true; outWillDoInPlace = true; Returns true for both ReadInput and WriteMix. Observation: Even when selecting "SubON Virtual Device" as the system input device in System Settings, Plugin_StartIO is still never called. Any ideas what could prevent Plugin_StartIO from being called despite AUHAL configuration appearing correct?

I added logging in an AudioServerPlugin to trace the arrival and removal of a particular client in the AddDeviceClient() and RemoveDeviceClient() methods. To my surprise the app was removed and added as a client often, even when it was in a steady state, for example when using the Audio MIDI Setup to change sample rates. Is this expected behavior, and if so, are there any explanations about this.

Have there been any changes in macOS 27 or iOS 27 that improve real-time audio level monitoring, WidgetKit updates, Live Activities, or audio route change handling for professional monitoring applications?

I have an AUv3 plug-in that works on both iOS and macOS. Some users have complained that after they create a track in Logic Pro (on either iPad or Mac version) and insert my plug-in as an instrument, the track always shows "Untitled" instead of the name of the active preset. Even if the user changes the factory preset, it still says "Untitled". I started debugging my AUv3 API usage for presets, specifically currentPreset and factoryPresets and tried pretty much everything with no success. For example, I tried overriding and not overriding currentPreset, and I tried issuing KVO for currentPreset. But Logic Pro still shows "Untitled" in all cases. In the following screenshot, I'm selecting a factory preset using Logic's interface. Here we see the factory preset is selected and diplayed using the UI of my plug-in, but Logic's interface is still showing Untitled. I noticed AUv2 plug-ins don't seem to suffer from this issue. For example, FabFilter Pro-Q 3 doesn't expose its factory parameters, but I can create a custom preset inside it or load a factory preset and Logic's interface will display the name of the plug-in. I'm assuming the v2 API kAudioUnitProperty_PresentPreset doesn't suffer fron this issue? Here's Logic Pro showing Pro-Q 3's custom preset name I created using the plug-in interface. Is there a solution to this? Or is this a bug? And then whose bug is it Logic's or Core Audio?

I'm trying to write 16-bit interleaved 2-channel data captured from a LiveSwitch audio source to a AVAudioFile. The buffer and file formats match but I get a bad parameter error from the API. Does this API not support the specified format or is there some other issue? Here is the debugger output. (lldb) po audioFile.url ▿ file:///private/var/mobile/Containers/Data/Application/1EB14379-0CF2-41B6-B742-4C9A80728DB3/tmp/Heart%20Sounds%201 - _url : file:///private/var/mobile/Containers/Data/Application/1EB14379-0CF2-41B6-B742-4C9A80728DB3/tmp/Heart%20Sounds%201 - _parseInfo : nil - _baseParseInfo : nil (lldb) po error Error Domain=com.apple.coreaudio.avfaudio Code=-50 "(null)" UserInfo={failed call=ExtAudioFileWrite(_impl->_extAudioFile, buffer.frameLength, buffer.audioBufferList)} (lldb) po buffer.format <AVAudioFormat 0x302a12b20: 2 ch, 44100 Hz, Int16, interleaved> (lldb) po audioFile.fileFormat <AVAudioFormat 0x302a515e0: 2 ch, 44100 Hz, Int16, interleaved> (lldb) po buffer.frameLength 882 (lldb) po buffer.audioBufferList ▿ 0x0000000300941e60 - pointerValue : 12894608992 This code handles the details of converting the Live Switch frame into an AVAudioPCMBuffer. extension FMLiveSwitchAudioFrame { func convertedToPCMBuffer() -> AVAudioPCMBuffer { Self.convertToAVAudioPCMBuffer(from: self)! } static func convertToAVAudioPCMBuffer(from frame: FMLiveSwitchAudioFrame) -> AVAudioPCMBuffer? { // Retrieve the audio buffer and format details from the FMLiveSwitchAudioFrame guard let buffer = frame.buffer(), let format = buffer.format() as? FMLiveSwitchAudioFormat else { return nil } // Extract PCM format details from FMLiveSwitchAudioFormat let sampleRate = Double(format.clockRate()) let channelCount = AVAudioChannelCount(format.channelCount()) // Determine bytes per sample based on bit depth let bitsPerSample = 16 let bytesPerSample = bitsPerSample / 8 let bytesPerFrame = bytesPerSample * Int(channelCount) let frameLength = AVAudioFrameCount(Int(buffer.dataBuffer().length()) / bytesPerFrame) // Create an AVAudioFormat from the FMLiveSwitchAudioFormat guard let avAudioFormat = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: sampleRate, channels: channelCount, interleaved: true) else { return nil } // Create an AudioBufferList to wrap the existing buffer let audioBufferList = UnsafeMutablePointer<AudioBufferList>.allocate(capacity: 1) audioBufferList.pointee.mNumberBuffers = 1 audioBufferList.pointee.mBuffers.mNumberChannels = channelCount audioBufferList.pointee.mBuffers.mDataByteSize = UInt32(buffer.dataBuffer().length()) audioBufferList.pointee.mBuffers.mData = buffer.dataBuffer().data().mutableBytes // Directly use LiveSwitch buffer // Transfer ownership of the buffer to AVAudioPCMBuffer let pcmBuffer = AVAudioPCMBuffer(pcmFormat: avAudioFormat, bufferListNoCopy: audioBufferList) /* { buffer in // Ensure the buffer is freed when AVAudioPCMBuffer is deallocated buffer.deallocate() // Only call this if LiveSwitch allows manual deallocation } */ pcmBuffer?.frameLength = frameLength return pcmBuffer } } This is the handler that is invoked with every frame in order to convert it for use with AVAudioFile and optionally update a scrolling signal display on the screen. private func onRaisedFrame(obj: Any!) -> Void { // Bail out early if no one is interested in the data. guard isMonitoring else { return } // Convert LS frame to AVAudioPCMBuffer (no-copy) let frame = obj as! FMLiveSwitchAudioFrame let buffer = frame.convertedToPCMBuffer() // Hand subscribers a reference to the buffer for rendering to display. bufferPublisher?.send(buffer) // If we have and output file, store the data there, as well. guard let audioFile = self.audioFile else { return } do { try audioFile.write(from: buffer) // FIXME: This call is throwing error -50 } catch { FMLiveSwitchLog.error(withMessage: "Failed to write buffer to audio file at \(audioFile.url): \(error)") self.audioFile = nil } } This is how the audio file is being setup. static var recordingFormat: AVAudioFormat = { AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44_100, channels: 2, interleaved: true)! }() let audioFile = try AVAudioFile(forWriting: outputURL, settings: Self.recordingFormat.settings)

I want to connect a BluetoothHFP microphone to MFi hearing aids. Whenever I make the microphone the input it steals the output and whenever I make the MFi the output it steals the input. Is it possible to do this using AVAudioSession? Is it possible to redefine the MFi hearing aids as a speaker and use Live Listen?

AVCaptureSession.startRunning() triggers AVCaptureSessionRuntimeErrorNotification with AVError.unknown (-11800), underlying OSStatus 2003329396 → fourCC 'what', every cold launch, but only when an audio AVCaptureDeviceInput is attached. Removing only the audio input makes the error disappear. Same code in a fresh project records audio fine — bug only appears in this app's binary. AVAudioApplication.shared.recordPermission == .granted. Info.plist has NSMicrophoneUsageDescription. No interruption notifications fire. Test device: iPhone 16 Pro, iOS 26.4.2. iOS deployment target 17.1. Minimal reproducer import AVFoundation let session = AVCaptureSession() session.beginConfiguration() let camera = AVCaptureDevice.default(.builtInWideAngleCamera, for: .video, position: .back)! session.addInput(try AVCaptureDeviceInput(device: camera)) // Removing ONLY this line makes the error disappear: let mic = AVCaptureDevice.default(for: .audio)! session.addInput(try AVCaptureDeviceInput(device: mic)) session.addOutput(AVCaptureMovieFileOutput()) session.addOutput(AVCapturePhotoOutput()) session.commitConfiguration() NotificationCenter.default.addObserver( forName: .AVCaptureSessionRuntimeError, object: session, queue: nil ) { print($0.userInfo ?? [:]) } session.startRunning() // -11800 / 'what' fires within ~2 sec Observed state at error time AVError.unknown (-11800) underlyingError = NSError(NSOSStatusErrorDomain, 2003329396) userInfo[AVErrorFourCharCode] = 'what' captureSession.isRunning = false ← never came up captureSession.isInterrupted = false captureSession.preset = .high captureSession.inputs = [Back Triple Camera, iPhone Microphone] AVAudioSession.sharedInstance(): category = .playAndRecord mode = .videoRecording sampleRate = 48000.0 isInputAvailable = true isOtherAudioPlaying = false availableInputs = [MicrophoneBuiltIn] (no BT/Continuity/AirPods) currentRoute.inputs = [] ← EMPTY currentRoute.outputs = [Speaker|Speaker] 2003329396 = 0x77686174 = 'what'. From a few SO threads this maps to AURemoteIO::StartIO returning a HAL-bring-up failure. The smoking gun: currentRoute.inputs is empty even though availableInputs contains the built-in mic, isInputAvailable is true, the category is .playAndRecord, and isOtherAudioPlaying is false. The HAL never routes the mic into the session, then 'what' follows. Nothing observable from AVAudioSession indicates a competing client. Environment / SDKs linked Firebase (SPM: Crashlytics, Performance, Messaging, Analytics, AppCheck, RemoteConfig, DynamicLinks), FBSDK, Kingfisher, MetalPetal. Multiple Google ad mediation pods present, but their audio session takeover is already disabled (audioVideoManager.isAudioSessionApplicationManaged = true, IMSdk.shouldAutoManageAVAudioSession(false)). What I've ruled out (all still produce 'what') Audio session config: .playAndRecord/.videoRecording, .playAndRecord/.default, .record/.measurement, .record/.default. With/without .defaultToSpeaker, .allowBluetooth, .allowBluetoothA2DP, .mixWithOthers. setActive(true) before vs. after attaching audio input. setPreferredInput(builtInMic) (verified accepted). 200ms Thread.sleep between setActive(true) and startRunning(). Setting usesApplicationAudioSession = false swaps the fourCC to '!rec' but produces the same outcome. Topology: sessionPreset = .high / .hd1920x1080 / .hd1280x720 / .medium. Camera = .builtInTripleCamera / .builtInDualWideCamera / .builtInWideAngleCamera. AVCam-style always-attached graph. Setting sessionPreset before vs. after adding inputs. Threading: All session mutations on a single dedicated DispatchQueue (vs. Swift actor). 1× and 2× full stopRunning()+startRunning() recovery cycles ("do it twice" pattern) — both re-fail with 'what'. SDK takeover prevention: GoogleMobileAdsMediation pods (Vungle, Mintegral, Pangle, Unity, InMobi), Google-Mobile-Ads-SDK, MediaPipeTasksVision removed via full pod uninstall + clean build — 'what' persists. Notifications during the failure window: 3 × AVAudioSession.routeChangeNotification reason categoryChange before the error fires, even though category stays .playAndRecord/.videoRecording. Disabling automaticallyConfiguresApplicationAudioSession drops this to 1, but the runtime error still fires. No AVAudioSession.interruptionNotification. No AVCaptureSessionWasInterruptedNotification. Symbol audit otool -L and nm of the bundle confirm none of the linked frameworks reference AVAudioRecorder, AudioComponentInstanceNew, AURemoteIO, or AudioUnitInitialize in their symbol tables. Only the app's own files reference any audio API. Yet adding AVCaptureDeviceInput(.audio) reproduces 100% in this binary and 0% in a fresh project. My questions Who is most likely holding the audio HAL in a process where no linked framework references the AudioUnit / HAL APIs directly? Are there framework load-time audio initializations that don't show up in symbol tables (e.g., dynamic dlopen, CFBundleLoadExecutable) that could grab the HAL? Is there an os_log subsystem / category that surfaces the underlying AURemoteIO::StartIO failure reason at runtime? com.apple.coreaudio shows 'what' but not the originating cause. currentRoute.inputs is empty at error time even though availableInputs = [MicrophoneBuiltIn], isInputAvailable = true, and the category is .playAndRecord. What does an empty input route under those conditions imply, and what other system-level holders could be preventing the HAL from routing the mic in? Has anyone seen 'what' resolve with a device reboot, an iOS update, or by removing a specific framework? Happy to share a sysdiagnose. Thanks!

Summary Using AudioHardwareCreateProcessTap + AudioHardwareCreateAggregateDevice for system audio capture. During long sessions, the AudioDeviceIOProc callback continues firing normally but every PCM sample is exactly 0.0f — while the system is producing audible output. Environment Field Value macOS 26.5 Beta Hardware MacBook Air (M2) API AudioHardwareCreateProcessTap + AudioHardwareCreateAggregateDevice Tap CATapDescription, processes = [], .unmuted, private Format 48,000 Hz, Float32, interleaved stereo Aggregate anchor kAudioAggregateDeviceMainSubDeviceKey = current default output UID Observed behavior After running normally for several minutes, the stream transitions into an all-zero state: AudioDeviceIOProc continues to fire at expected cadence Frame count, timestamps (mHostTime, mSampleTime), and mDataByteSize all look normal AudioBufferList pointers are valid Every sample in every buffer is exactly 0.0f Other apps are still producing audible output through the same output device The condition may self-recover or persist until the session is stopped Confirmed via RMS logging both inside the IOProc and after the ring buffer consumer — data is zero on delivery, not introduced downstream. Example: 51-minute session on MacBook Air M2 Segment 1 (~7 min): Three all-zero periods: 60 s, 53 s, 141 s. Real PCM briefly returned between them. Segment 2 (~44 min): Two all-zero periods: 16 min 3 s, 3 min 8 s. IOProc cadence, timestamp deltas, default output UID, and kAudioDevicePropertyDeviceIsRunningSomewhere all remained normal throughout. What I have ruled out Actual silence: User was in an active video call and could hear participants through the output device. Default output device change: Monitored kAudioHardwarePropertyDefaultOutputDevice — no change during affected periods. IOProc stall: Heartbeat and kAudioDevicePropertyDeviceIsRunningSomewhere remained normal. Aggregate device destroyed: AudioObjectGetPropertyData on the aggregate UID continued returning the expected device. Tap descriptor misconfiguration: The same tap produces valid PCM earlier in the same session, so this is not a startup-time issue. Why detection is hard All-zero buffers from a broken tap are indistinguishable from legitimate silence (muted participant, waiting room, paused media). kAudioProcessPropertyIsRunningOutput reports whether a process has active output IO, not whether it is contributing non-zero samples — a muted Zoom call still reports true. Possible correlations Sample-rate renegotiation on the output device (44.1 kHz ↔ 48 kHz) when another app changes output Bluetooth device state changes (AirPods sleep/wake) where UID stays the same MacBook Air more frequently affected than MacBook Pro Always occurs after extended uptime — first few minutes are consistently clean Current workaround Full teardown and rebuild restores real PCM. Restarting the IOProc alone or recreating only the aggregate device is not reliable — both the Process Tap and Aggregate Device must be destroyed and recreated. 1. AudioDeviceStop 2. AudioDeviceDestroyIOProcID 3. AudioHardwareDestroyAggregateDevice 4. AudioHardwareDestroyProcessTap 5. AudioHardwareCreateProcessTap 6. AudioHardwareCreateAggregateDevice 7. Create + start new IOProc Applying this automatically is risky because it cannot be reliably distinguished from legitimate silence. Questions Expected failure mode? Can a Process Tap continue delivering zero-filled buffers while the system output is audible? Is this expected under certain device or routing conditions? Detection signal? Is there any HAL property, notification, or diagnostic counter that distinguishes "sources are genuinely silent" from "the tap data path has stopped receiving the real mix"? Targeted recovery? Is there a supported way to re-anchor or reset the tap data path without destroying and recreating both objects? Full rebuild as intended workaround? If so, it would help to confirm this so developers can converge on a consistent approach. Mixer activity signal? kAudioProcessPropertyIsRunningOutput reflects IO registration, not sample contribution. Is there any AudioProcess property that indicates a process is currently delivering non-zero audio to the system mixer?

I have a device which is using linux gadget audio to receive audio input via USB, exposing 24 capture channels. This device works well with Mac, Windows, and Android phones. However, when sending audio from an iPhone (both USB-C iPhones and lightning iPhones using an official Apple lightning -> usb adaptor) I am seeing strange behaviour. Audio which is sent from the iPhone to any one of inputs 12, 19, 20, 21, or 22 appears in all of those channels, rather than only the channel to which audio is routed. I have confirmed on my linux device that these channels are not being erroneously populated by the software running on that device; the issue is visible in audio recorded directly from the gadget using arecord, meaning it is present in the audio being sent from the iPhone. I have confirmed that the gadget channel mask is correct for 24 channel audio (0xFFFFFF). As said above, audio routed to this device from any non-iPhone device (Mac, Windows, Android) works fine. The only sensible conclusion seems to be that the iPhone is populating the additional channels erroneously due to some bug in CoreAudio's handling of gadget audio devices. I would appreciate any insight on this from Apple developers, or from anyone else who has come across this issue and found a workaround.

If a Tap or AggregateDevice with the Private property set is created, does it automatically disappear when the process ends? If not, how can I remove the Tap or AggregateDevice before the main process terminates?

I just bought a monitor S2725QC from Dell Technologies and isn't fully-integrated with MacOS even though it says on the website it is compatible with MacOS. https://www.dell.com/en-us/shop/all-monitors/sac/monitors/all-monitors/macos-compatible?appliedRefinements=51765 The screen brightness and volume control buttons don't work with the monitors (I have two). What can I do in terms of writing code with Dell Monitor SDK and MacOS Frameworks/Technologies?

I am developing a standard UAC 2.0 device and encountered an issue where the channel names do not update according to the iChannelNames field in the Class Specific AS Interface Descriptor when switching between different channel counts. For example: AS1 (6 channels) is configured with the following channel names: ADAT 1, ADAT 2, ADAT 3, ADAT 4, HP L, HP R AS2 (4 channels) is configured with: ADAT 1, ADAT 2, HP L, HP R However, when switching from AS1 (6 channels) to AS2 (4 channels), the channel names displayed in Audio MIDI Setup do not reflect the change as expected. The actual result is: ADAT 1, ADAT 2, ADAT 3, ADAT 4 The system simply hides the last two channels; the names of the remaining channels are not updated. Initial Topology My original topology was as follows: Later, I discovered that macOS uses the iChannelNames field from the Input Terminal to display channel names. Therefore, I modified the USB device descriptors and updated the topology to the following: To distinguish the channel names for different channel counts, each Input Terminal is assigned a unique iChannelNames value. This method worked perfectly on macOS 15. However, after updating to macOS 26, this topology no longer displays the correct channel names. Question On macOS 26, what is the correct method to ensure that the channel names update dynamically when switching between different audio channel configurations?

I am developing a standard UAC 2.0 device and encountered an issue where the channel names do not update according to the iChannelNames field in the Class Specific AS Interface Descriptor when switching between different channel counts. For example: AS1 (6 channels) is configured with the following channel names: ADAT 1, ADAT 2, ADAT 3, ADAT 4, HP L, HP R AS2 (4 channels) is configured with: ADAT 1, ADAT 2, HP L, HP R However, when switching from AS1 (6 channels) to AS2 (4 channels), the channel names displayed in Audio MIDI Setup do not reflect the change as expected. The actual result is: ADAT 1, ADAT 2, ADAT 3, ADAT 4 The system simply hides the last two channels; the names of the remaining channels are not updated. Initial Topology My original topology was as follows: Later, I discovered that macOS uses the iChannelNames field from the Input Terminal to display channel names. Therefore, I modified the USB device descriptors and updated the topology to the following: To distinguish the channel names for different channel counts, each Input Terminal is assigned a unique iChannelNames value. This method worked perfectly on macOS 15. However, after updating to macOS 26, this topology no longer displays the correct channel names. Question On macOS 26, what is the correct method to ensure that the channel names update dynamically when switching between different audio channel configurations?

I am developing a macOS virtual audio device using an Audio Server Plug-In (HAL). I want the virtual device to be visible to all applications only when my main app is running, and completely hidden from all apps when the app is closed. The goal is to dynamically control device visibility based on app state without reinstalling the driver.What is the recommended way for the app to notify the HAL plug-in about its running or closed state ? Any guidance on best-practice architecture for this scenario would be appreciated.