Hi, hoping for guidance on what's a long-running bug for our app. The problem We have a transcription app on iPhone 17 Pro Max running iOS 26.5. Recording flow uses AVAudioEngine.installTap(onBus:) to capture PCM into a JS bridge for streaming to a remote transcription service. A parallel AVAudioRecorder writes the same audio to disk as backup. When the user starts a recording and locks the phone, iOS terminates our process with SIGKILL at approximately 50 seconds of continuous background time, despite: UIBackgroundModes includes audio (verified in shipping IPA's Info.plist) AVAudioSession.setCategory(.playAndRecord, mode: .default) is active AVAudioEngine is running with installTap producing PCM buffers right up to the moment of death UIApplication.backgroundTimeRemaining returns Double.greatestFiniteMagnitude at applicationDidEnterBackground (verified in our event log) No AVAudioSession.interruptionNotification is delivered before the kill. iOS terminates the process cleanly with no warning event to our observer. Evidence Our Swift observer module writes an event log to disk on every system event. On relaunch we ship it to our crash reporter. Excerpt from a recent kill on iOS 26.5 / build 2.1.32: T=0.000s session-start (engineRunning: true) T=57.199s app-will-resign-active (bufferCallbackCount: 22) T=58.913s app-did-enter-background (backgroundTimeRemaining: infinity, bufferCallbackCount: 39) [no further audio events captured] [Swift heartbeat written every 5s for next ~46 seconds] T~105s Process SIGKILLed (heartbeat last-alive: 09:31:01.597Z) Background time before kill: ~46 seconds. engineRunning: true and bufferCallbackCount was still incrementing at the moment the event log stops capturing - the audio engine was alive and feeding buffers when iOS terminated us. What we've tried (35 documented attempts) Hopefully not all relevant but listing for completeness: Various AVAudioSession category/mode/options combinations (Default, Measurement, VoiceChat, .mixWithOthers, .defaultToSpeaker, .allowBluetoothHFP) Parallel AVAudioRecorder writing a .caf file as a "real recording app" signal SFSpeechRecognizer with requiresOnDeviceRecognition = true consuming PCM in-process (50s request rotation) BGContinuedProcessingTask with Progress.completedUnitCount reporting monotonic progress every 5 seconds Live Activity (ActivityKit) with NSSupportsLiveActivitiesFrequentUpdates = true Live Activity update pushes via APNs (confirmed wake widget extension only, not host) Silent device-token APNs background pushes (confirmed iOS ~5/day rate limit) CallKit fake call (CXProvider + CXCallController) - works but creates the green pill UI which our product can't ship WebRTC peer connection with active media stream (via react-native-webrtc loopback) UIBackgroundModes: voip declaration (without CallKit) beginBackgroundTask + engine bounce (Apple's own guidance says don't, our test confirmed it's actively harmful) CLLocationManager background updates All die at ~50s background. None of them survive. What works on the same device Three App Store transcription apps survive indefinite background recording on our exact device + iOS version. We have inspected their IPAs (Mach-O LC_LOAD_DYLIB analysis + embedded entitlement extraction): Otter (com.aisense.otter) - UIBackgroundModes: audio + fetch + processing + remote-notification. Uses OneSignal-driven Live Activity push tokens + NotificationServiceExtension. No CallKit, PushKit, or WebRTC. Granola (com.granola.ios-prod) - has UIBackgroundModes: voip but the voip is for their separate outbound-phone-call feature (TwilioVoice + CallKit, lives in their PhoneCalls.framework). Recording-path uses ONLY AVAudioRecorder + PlayAndRecord + ModeDefault + Live Activity with frequentPushesEnabled. Zero PushKit anywhere in the bundle. Transcribe Speech to Text by DENIVIP (ru.denivip.transcribe) - the smallest API surface: UIBackgroundModes: audio + remote-notification only. AVAudioEngine + .playAndRecord + .default + SFSpeechRecognizer consuming PCM. No CallKit, PushKit, BGTask, Live Activity, WebRTC, or VoIP. Three apps, three different mechanisms, all working. We've implemented bits of all three approaches in our app and still die at 50s. Apple Voice Memos (system app, private entitlements) also survives indefinite recording on the same device. Questions What is the supported API path for indefinite background microphone-only recording on iOS 26.5? Voice Memos and competitor apps clearly accomplish this - what's the missing piece? Why does UIApplication.backgroundTimeRemaining return Double.greatestFiniteMagnitude at applicationDidEnterBackground but the process is terminated ~50 seconds later? Is the meaning of this property changing in iOS 26? What causes the iOS 26 process scheduler to revoke the audio-mode background runtime classification? No AVAudioSession.interruptionNotification is delivered before SIGKILL. Where can we observe the classification change? Does iOS 26 distinguish "audio recording with no audible output" from "audio recording with audible output (e.g. a media playback session)"? If so, what is the supported API to register as a recording-only background-audio app? Does BGContinuedProcessingTask (new in iOS 26) actually extend background CPU time for an app that is also using UIBackgroundModes: audio and an active AVAudioSession? Or is it for finish-what-you-started bursts only (per WWDC 2025 session 227)? Any guidance - even pointers to specific WWDC sessions, sample code, or technotes - would be hugely appreciated. We've spent ~40+ hours on this and want to know what the supported path looks like in iOS 26. Happy to share more event-log data, IPA inspection notes, or build a focused Xcode reproduction if helpful. Thanks!

I have a visionOS app that plays audio using AVAudioEngine and presents both a window and an immersive space. If I close the window, the audio session gets interrupted and attempting to restart the session and audio engine has no effect. I need to dismiss the app, then reopen it, which reopens the main window, in order for audio to start playing again. This is in all visionOS 2 betas. Note that I have background audio enabled for my app.

I'm working on a podcast app and I'm running into a small quirk I'd like to fix. On Apple's Podcast app and on the Spotify app when I say, for example, "Hey Siri, skip" the audio pauses, the app performs the operation, and then immediately resumes playing the audio at the previous volume without waiting for the Siri overlay to dismiss. But my app doesn't do that. When I say "Hey Siri, skip" it pauses the audio, performs the operation, but then audio stays paused until the overlay dismisses or the audio resumes playing at a reduced volume until the overlay dismisses depending on which route I go. What I've tried: Stays paused until overlay dismisses: AVAudioSession.setCategory(.playback, mode: .spokenAudio), setActive(true) Register for AVAudioSession.interruptionNotification On .began interruption capture if audio is currently playing On .ended interruption: if it was playing before, call play() again Plays at reduced volume until the overlay dismisses: Same as above plus: Inside MPRemoteCommandCenter.shared().skipBackwardCommand, I call seek and then: AVAudioSession.sharedInstance().setActive(false, options: .notifyOthersOnDeactivation) AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, policy: .longFormAudio, options: []) AVAudioSession.sharedInstance().setActive(true) player.play() player.rate = playbackSpeed player.volume = 1.0 AVAudioSession.interruptionNotification finally arrives with .ended + .shouldResume, at which point volume snaps to normal. I tried that with and without setPrefersNoInterruptionsFromSystemAlerts(true) but there was no difference. Seems like .ended only arrives when the Siri overlay dismisses, and not during Siri's active state? While I was trying things XCode warned me that: Ignoring setPlaybackState because application does not contain entitlement com.apple.mediaremote.set-playback-state for platform Which, of course, I can't add b/c it's a private API. Do I need that to do what I want? Or am I missing something else? Thanks!

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!

USB cameras, microphones, HDMI capture cards, and audio interfaces are supposed to "just work" on macOS. In reality, it's often difficult to quickly access or monitor them without opening large and complicated software. Sometimes you simply want to see whether a USB camera is active. Sometimes you want to check an HDMI source connected through a capture card. And in other cases, you may want to use a Mac mini without a dedicated monitor by viewing its HDMI output through a USB capture device directly on another Mac. macOS supports many modern USB AV devices out of the box, but it surprisingly lacks a simple built-in utility for live monitoring and recording. Most users end up using oversized streaming or editing applications just to preview a video signal or monitor audio input. That becomes especially noticeable with: USB webcams HDMI capture adapters USB microphones audio interfaces secondary computers headless Mac mini setups A lightweight monitor utility is often much more practical when you only need real-time access to a device, want to record a stream, or quickly switch between multiple AV inputs. That's one of the reasons I built AV Monitor Pro  -  a native macOS app designed for monitoring and recording connected audio/video devices in real time. It can preview USB cameras, capture cards, microphones, and HDMI sources with minimal setup, and it's especially useful for workflows like running a Mac mini without a monitor, monitoring external devices, or recording live AV input directly on macOS.

Hello, I am developing Uniq Intercom, a voice-only group communication app for motorcyclists (always-on intercom over WebRTC, used continuously for multi-hour rides). I am seeking guidance on an iOS audio session and CarPlay HID interaction I have not been able to resolve through documented APIs. Problem: As soon as my app activates the microphone (yellow recording indicator visible), iOS appears to classify the app as a real-time communication participant and CarPlay re-routes the steering-wheel / handlebar HID transport buttons (left / right / ok) from the media-control role to the call-control role (answer/decline). Because I do not register a CallKit / LiveCommunicationKit call (the session is a continuous group voice channel, not a discrete telephony call), there is no call object for those buttons to act upon — they effectively become inert. Why this matters: Motorcyclists rely on the intercom for 4–6 hour rides. CarPlay is now built into a growing number of modern motorcycles and with aftermarket display units virtually any bike, and any rider who uses any voice communication platform alongside it — Uniq Intercom, WhatsApp Call currently runs into this same handlebar button remap. With the buttons inert, the rider's only remaining option is to reach for the motorcycle's touchscreen to skip a track or change navigation — this is unsafe. The exact same remap behavior occurs during a real WhatsApp or Phone call — but for those the remap is correct (answer/decline maps to a real call). For continuous voice apps without a CallKit-style discrete call, no equivalent path exists today. As both an iOS developer and a motorcyclist, I would very much like to see this resolved — solving it would meaningfully improve safety for every rider using an iPhone with CarPlay. Configurations I have tested (all reproduce the symptom on iOS 18+ / 26 with wireless CarPlay): AVAudioSession.Category.playAndRecord + .voiceChat mode + various option combinations (duckOthers, mixWithOthers, allowBluetoothHFP, allowBluetoothA2DP, defaultToSpeaker) Same category with .videoChat mode (which @livekit/react-native defaults to) Same category with .default mode (re-applied after setAudioModeAsync to defeat any framework override) — confirmed Mode = Default for ~2 s window in audiomxd log before WebRTC's setActive cycle returned mode to .voiceChat. Buttons remained remapped during the .default window. Disabling MPRemoteCommandCenter and clearing MPNowPlayingInfoCenter.default().nowPlayingInfo JS-side override of WebRTC's global RTCAudioSessionConfiguration via @livekit/react-native's AudioSession.setAppleAudioConfiguration({audioMode: 'default'}) bridge, applied both before connect and after setAudioModeAsync to defeat library overrides In every case the audiomxd system log confirms our session goes active (Mode = VoiceChat or Default, Recording = YES), and CarPlay HID buttons are immediately remapped to call-control. The middle "OK" button remains functional because it is not part of the call-control mapping — confirming the buttons are not blocked, only re-purposed. The remap occurs the instant the iOS recording indicator appears, regardless of audio session mode. This led me to conclude the trigger is not audio session mode but the combination of microphone permission + active session + (likely) the AUVoiceIO unit instantiated by WebRTC. I cannot find a public API path to suppress this classification while maintaining the always-on continuous voice channel. My questions: Is there an entitlement or API that allows an app with active microphone capture to declare itself as a non-call media participant, keeping CarPlay HID transport buttons in the media role? Is AVAudioSession.setPrefersEchoCancelledInput(_:) (iOS 18+) the intended path for retaining platform AEC under .default mode without the focus-engine "communication priority" marking? Documentation is sparse on its CarPlay arbitration implications. Does the PushToTalk framework affect HID arbitration differently from playAndRecord + voiceChat? Our continuous-channel UX does not fit the PTT transmit-on-press model, but understanding the contrast would help. If no current API exists, is this something the iOS Audio team would consider for future SDKs? Solving this would meaningfully improve safety for motorcycle and adventure-sport users on iOS. Thank you for your time and any guidance you can offer. — Emre Erkaya / Uniq Intercom

I am having an issue with the code that I posted below. I capture voice in my CarPlay app, then allow the user to have it read back to them using AVSpeechUtterance. This works fine on some cars, but many of my beta testers report no audio being played. I have also experienced this in a rental car where the audio was either too quiet or the audio didn't play. Does anyone see any issue with the code that I posted? This is for CarPlay specifically. class CarPlayTextToSpeechService: NSObject, ObservableObject, AVSpeechSynthesizerDelegate { private var speechSynthesizer = AVSpeechSynthesizer() static let shared = CarPlayTextToSpeechService() /// Completion callback private var completionCallback: (() -> Void)? override init() { super.init() speechSynthesizer.delegate = self } func configureAudioSession() { do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .voicePrompt, options: [.duckOthers, .interruptSpokenAudioAndMixWithOthers, .allowBluetoothHFP]) } catch { print("Failed to set audio session category: \(error.localizedDescription)") } } public func speak(_ text: String, completion: (() -> Void)? = nil) { self.configureAudioSession() // Store the completion callback self.completionCallback = completion Task(priority: .high) { let speechUtterance = AVSpeechUtterance(string: text) let langCode = Locale.preferredLocalLanguageCountryCode if langCode == "en-US" { speechUtterance.voice = AVSpeechSynthesisVoice(identifier: AVSpeechSynthesisVoiceIdentifierAlex) } else { speechUtterance.voice = AVSpeechSynthesisVoice(language: langCode) } try AVAudioSession.sharedInstance().setActive(true, options: .notifyOthersOnDeactivation) speechSynthesizer.speak(speechUtterance) } } func speechSynthesizer(_ synthesizer: AVSpeechSynthesizer, didFinish utterance: AVSpeechUtterance) { Task { stopSpeech() try AVAudioSession.sharedInstance().setActive(false) } // Call completion callback if available self.completionCallback?() self.completionCallback = nil } func stopSpeech() { speechSynthesizer.stopSpeaking(at: .immediate) } }

I have a keyboard in my iOS Morse Code app that has always been able to play audio via AVAudioEngine. Recently it has been failing to produce audio. I see that startAndReturnError: is now failing with this error: Error Domain=com.apple.coreaudio.avfaudio Code=268435459 "(null)" UserInfo={failed call=err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)} What's going on? Have keyboards lost the ability to play audio? Here's how I set things up: _engine = [AVAudioEngine new]; _prefs = [[NSUserDefaults alloc] initWithSuiteName:kSharedAppGroupID]; AVAudioMixerNode* mainMixerNode = _engine.mainMixerNode; AVAudioOutputNode* outputNode = _engine.outputNode; AVAudioFormat* format = [outputNode inputFormatForBus:0]; AVAudioFormat* inputFormat = [[AVAudioFormat alloc] initWithCommonFormat:AVAudioPCMFormatFloat32 sampleRate:44100 channels:1 interleaved:NO]; self.srcNode = [[AVAudioSourceNode alloc] initWithRenderBlock:^OSStatus(BOOL* _Nonnull isSilence, const AudioTimeStamp* _Nonnull timestamp, AVAudioFrameCount frameCount, AudioBufferList* _Nonnull outputData) { // This block builds the data, but is never called, so it is not the culprit. }]; [_engine attachNode:self.srcNode]; [_engine connect:self.srcNode to:mainMixerNode format:inputFormat]; [_engine connect:mainMixerNode to:_engine.outputNode format:nil]; [_engine prepare];

Hi I developed a music app that plays offline audio and displays lyrics using Live Activities. According to ActivityKit documentation, Live Activities can be updated from the background. However, in my case, updates stop after ~30 seconds when the app goes to the background or the device is locked. Important points: The app continues running in the background (audio playback works fine using AVAudioSession with .playback) Background code execution is working as expected Only the Live Activity stops updating I am not using push updates since this is an offline app. Is there any limitation or requirement for updating Live Activities continuously in the background during audio playback? Audio Session Configuration let session = AVAudioSession.sharedInstance() try session.setCategory( .playback, mode: .default, options: [.mixWithOthers] // ✅ DO NOT interrupt other audio ) try session.setActive(true) print("✅ [AudioSession] Activated with mixWithOthers") } catch { print("❌ [AudioSession] Error: \(error)") } Live Activity Update Methods guard let activity = getLiveActivity(for: recordID) else{ print("⚠️ No Live Activity found for recordID: \(recordID)") return } guard activity.activityState == .active else { print("⚠️ Activity is not active") return } Task { let content = ActivityContent( state: state, staleDate: Date().addingTimeInterval(60 * 60 * 12), relevanceScore: 1.0 ) await activity.update(content) print("✅ Live Activity updated with ActivityContent") } }

Hi all, I’m trying to determine whether the loudness gap I’m seeing between AlarmKit alert playback and normal app-managed playback is expected behavior, a sound-asset issue, or something that should be reported as a bug. Observed behavior When an alarm fires through AlarmKit while the device is locked, the alarm sound is significantly quieter than playback of the same or very similar audio once the app is active and using its own audio session. The difference is large enough that it does not feel like a small mastering difference. It feels like the AlarmKit / system alerting path is using a meaningfully lower effective output level than normal app playback. Test scenario My repro is roughly: Schedule an alarm with AlarmKit. Lock the device. Let the alarm fire and listen during the system alerting phase. Enter the app / continue into the app-driven alarm experience. Play the same or equivalent alarm asset via app-managed playback. Result: AlarmKit / lock-screen alerting phase sounds much quieter. In-app playback sounds noticeably louder and fuller on the same device. Current implementation Alarm flow is currently split into two paths: 1) System alarm path Alarm scheduling and alert surfacing via AlarmKit Device may be locked No attempt to manipulate system volume No private APIs 2) In-app playback path After app activation, playback uses: AVAudioSession category .playback AVAudioPlayer Audio is routed as normal app playback This path sounds substantially louder than the AlarmKit path Important detail I am not asking how to override system volume. I understand that AlarmKit appears to follow the system ringer / alert volume model and does not expose a public API for custom alarm loudness. My question is narrower: Is it expected that the same asset or an equivalent asset will sound materially quieter during the AlarmKit alerting phase than during ordinary app playback with AVAudioSession(category: .playback)? Questions Is the lower perceived loudness during AlarmKit alerting an expected property of the framework / system alarm path? Does AlarmKit playback use a different output path, gain policy, processing chain, or speaker treatment than normal app playback with .playback? Are there recommended authoring constraints for AlarmKit alarm sounds to maximize perceived loudness on iPhone speakers? transient-heavy mix stronger mids reduced low-end different LUFS / peak strategy shorter attack, etc. Has anyone measured this directly with: the same WAV / CAF file same device same system volume locked AlarmKit playback vs unlocked in-app playback If this is not expected, would Apple want this reported as a bug with: sample project exact iOS version device model screen recording / audio recording What I’m trying to figure out For alarm-app UX, this matters a lot because: AlarmKit is the most reliable lock-screen/system path. But if AlarmKit playback is substantially quieter than normal app playback, the alarm experience is inconsistent depending on device/app state. That makes it hard to know whether to treat this as: expected system behavior, a framework limitation, an asset/mastering problem, or a bug. If anyone has tested this in a controlled way or received guidance from Apple/DTS, I’d appreciate any technical detail. Thanks.

We have an VoiP application, our application can be configured to amplify the PCM samples before feeding it to the Player to achieve volume gain at the receiver. In order to support this, We follow as below. If User has configured this Gain Settings within application, Application applies the amplification for the samples to introduce the gain. Application will also set the AVAudioSessionCategory to AVAudioSessionCategoryPlayback Provided the User has chosen the output to Speaker. This settings was working for us but we see there is a difference in behaviour w.r.t Volume Level System Settings between OS 26.3.1 and OS 26.4 When user has chosen earpiece as Output, then we will set the AVAudioSessionCategory to AVAudioSessionCategoryPlayAndRecord. User would have set the volume level to minimum. When user will change the output to Speaker, then we will set the AVAudioSessionCategory to AVAudioSessionCategoryPlayback. The expectation is, the volume level should be of AVAudioSessionCategoryPlayback what was set earlier instead we are seeing the volume level stays as minimum which was set to AVAudioSessionCategoryPlayAndRecord Could you please explain about this inconsistency w.r.t Volume level.

Hello, I am wondering if it is possible to have audio from my AirPods be sent to my speech to text service and at the same time have the built in mic audio input be sent to recording a video? I ask because I want my users to be able to say "CAPTURE" and I start recording a video (with audio from the built in mic) and then when the user says "STOP" I stop the recording.

I’m building a standalone Apple Watch metronome app for running. My goal is for these 3 apps to work at the same time: Runna owns the workout session Spotify plays music my app plays a metronome click in the background So far this is what I've found: Using HKWorkout​Session in my metronome app works well with Spotify, but conflicts with Runna and other workout apps, so I removed that. Using watchOS background audio with longFormAudio allows my app run in the background, and it can coexist with Runna. However, it seems to conflict with Spotify playback, and one app tends to stop the other. Is there any supported watchOS audio/background configuration that allows all 3 at once? More specifically this is what I need: another app owns HKWorkout​Session Spotify keeps playing my app keeps generating metronome clicks in the background Or is this simply not supported by current watchOS session/background rules? My metronome uses AVAudio​Engine / AVAudio​Player​Node with generated click audio. Thank you!

I'm seeing crashes in _MPRemoteCommandEventDispatch on iOS 26.x devices in 3 apps. According to Bugsnag logs they are: NSInternalInconsistencyException: event dispatch <_MPRemoteCommandEventDispatch: <MPRemoteCommandEvent: 0x11c049500 commandID=THV0 command=<MPRemoteCommand: 0x109ad1ea0 type=Play (0) enabled=YES handlers=[0x109b6a310]> sourceID=(null) ([HostedRoutingSessionDataSource] handleControlSendingCommand<2W5E>)> state:201> deallocated without calling continuation I attached a log from Xcode organizer matching Bugsnag crash. mpr_remote_command_event.crash When I set the brakpoint on the -[_MPRemoteCommandEventDispatch dealloc] I can see it it's hit every time I tap play or pause on locked screen play button. Thread 0 Crashed: 0 libsystem_kernel.dylib 0x00000002370420cc __pthread_kill + 8 (:-1) 1 libsystem_pthread.dylib 0x00000001e975c810 pthread_kill + 268 (pthread.c:1721) 2 libsystem_c.dylib 0x0000000198f8ff64 abort + 124 (abort.c:122) 3 libc++abi.dylib 0x000000018a7cf808 __abort_message + 132 (abort_message.cpp:66) 4 libc++abi.dylib 0x000000018a7be484 demangling_terminate_handler() + 304 (cxa_default_handlers.cpp:76) 5 libobjc.A.dylib 0x000000018a6cff78 _objc_terminate() + 156 (objc-exception.mm:496) 6 xxxxxxxxxxxxxx 0x00000001003a7db8 CPPExceptionTerminate() + 416 (BSG_KSCrashSentry_CPPException.mm:156) 7 libc++abi.dylib 0x000000018a7cebdc std::__terminate(void (*)()) + 16 (cxa_handlers.cpp:59) 8 libc++abi.dylib 0x000000018a7ceb80 std::terminate() + 108 (cxa_handlers.cpp:88) 9 CoreFoundation 0x000000018d7341c4 __CFRunLoopPerCalloutARPEnd + 256 (CFRunLoop.c:769) 10 CoreFoundation 0x000000018d70bb5c __CFRunLoopRun + 1976 (CFRunLoop.c:3179) 11 CoreFoundation 0x000000018d70aa6c _CFRunLoopRunSpecificWithOptions + 532 (CFRunLoop.c:3462) 12 GraphicsServices 0x000000022e31c498 GSEventRunModal + 120 (GSEvent.c:2049) 13 UIKitCore 0x00000001930ceba4 -[UIApplication _run] + 792 (UIApplication.m:3902) 14 UIKitCore 0x0000000193077a78 UIApplicationMain + 336 (UIApplication.m:5577) 15 xxxxxxxxxxxxxx 0x00000001000c0134 main + 308 (main.swift:15) 16 dyld 0x000000018a722e28 start + 7116 (dyldMain.cpp:1477) Is the crash happening when the app is being terminated? Thank you!

I have a question regarding the behavior of AVAudioSession.sharedInstance().outputVolume. Observed behavior: When the app is in the foreground, I read audioSession.outputVolume (for example, 0.1). The app is then moved to the background. While the app is in the background, the user changes the system volume using the hardware buttons (for example, to 0.5). When the app returns to the foreground, audioSession.outputVolume still reports the previous value (0.1). From my testing, outputVolume only seems to update when the system volume is changed while the app is in the foreground. Volume changes made while the app is in the background are not reflected when the app returns to the foreground. Questions: According to Apple’s documentation for AVAudioSession.outputVolume: “The systemwide output volume set by the user.” https://developer.apple.com/documentation/avfaudio/avaudiosession/outputvolume However, based on our testing on iOS 18.6.2 and iOS 18.1, the observed behavior seems to differ from this description. Questions: The documentation states that outputVolume represents the system-wide volume set by the user. In our testing, the value does not reflect volume changes made while the app is in the background and only updates when the app is in the foreground.Is this the expected behavior of AVAudioSession.outputVolume? Is there any other recommended way in Swift to retrieve the current system volume that reflects user changes made both while the app is in the foreground and while it is in the background? Any clarification on the intended behavior or recommended handling would be greatly appreciated.

I’m implementing a custom playback pipeline using AVSampleBufferAudioRenderer together with AVSampleBufferRenderSynchronizer. hasSufficientMediaDataForReliablePlaybackStart appears to be the intended signal for determining when enough media has been queued to start playback. For local playback, this works well in practice — the property becomes true after a reasonable amount of media is enqueued. However, when the output route is AirPlay, using this property becomes difficult: AirPlay requires significantly more buffered media before the renderer reports sufficient data. The required preroll amount is much larger than for local playback. For short assets, it is possible to enqueue the entire audio track and still never observe hasSufficientMediaDataForReliablePlaybackStart == true. In that situation there is no more media data to enqueue, but the renderer still reports that playback is not ready. Given this behavior, what is the recommended way to determine playback readiness when using AVSampleBufferAudioRenderer with AirPlay?

Hi everyone, We’re encountering an issue where AudioQueueNewOutput blocks indefinitely and never returns, and we’re hoping to get some insight or confirmation if this is a known behavior/regression on newer iOS versions. Issue Description When triggering audio playback, we create an output AudioQueue using AudioQueueNewOutput. On some devices, the call hangs inside AudioQueueNewOutput and never returns, with no OSStatus error and no subsequent logs. This behavior is reproducible mainly on iOS 18.3. Earlier iOS versions do not show this issue under the same code path. if (audioDes) { mAudioDes.mSampleRate = audioDes->mSampleRate; mAudioDes.mBitsPerChannel = audioDes->mBitsPerChannel; mAudioDes.mChannelsPerFrame = audioDes->mChannelsPerFrame; mAudioDes.mFormatID = audioDes->mFormatID; mAudioDes.mFormatFlags = audioDes->mFormatFlags; mAudioDes.mFramesPerPacket = audioDes->mFramesPerPacket; mAudioDes.mBytesPerFrame = audioDes->mBytesPerFrame; mAudioDes.mBytesPerPacket = audioDes->mBytesPerFrame; mAudioDes.mReserved = 0; } // Create AudioQueue for output OSStatus status = AudioQueueNewOutput( &mAudioDes, AQOutputCallback, this, NULL, NULL, 0, &audioQueue ); code-block The thread blocks inside AudioQueueNewOutput, and execution never reaches the next line. Additional Notes / Observations ASBD is confirmed to be valid Standard PCM output Sample rate, channels, bytes per frame/packet all consistent Same ASBD works correctly on earlier iOS versions AudioQueue is created on a background thread Not on the main thread Not inside the AudioQueue callback On first creation, AVAudioSession may not yet be active setCategory and setActive:YES may be called shortly before creating the AudioQueue There may be a timing window where the session is still activating Issue is reported mainly on iOS 18.3 Multiple user reports point to iOS 18.3 devices Same code path works on iOS 17.x and earlier No OSStatus error is returned — the call simply never returns. Questions Is it expected that AudioQueueNewOutput can block indefinitely while waiting for AVAudioSession / audio route / HAL readiness? Have there been any behavior changes in iOS 18.3 regarding AudioQueue creation or AudioSession synchronization? Is it unsafe to call AudioQueueNewOutput before AVAudioSession is fully active on recent iOS versions? Are there recommended patterns (or delays / callbacks) to ensure AudioQueue creation does not hang? Any insight or confirmation would be greatly appreciated. Thanks in advance!

I've got a streaming Radio app that loads an HLS stream into an AVAudioPlayer. I've set up an Intents extension that notifies SiriKit that my app must handle the INPlayMediaIntent in app, and, I'm able to successfully initiate the stream playing from my phone using the string "Play ". My intent handler in app looks like this: completionHandler(INPlayMediaIntentResponse(code: .success, userActivity: nil)) DispatchQueue.main.async { AudioPlayerService.shared.play() } The Audio Player service, in its init, does the following: try AVAudioSession.sharedInstance().setCategory( .playback, mode: .default, policy: .longFormAudio ) Additionally, in my Info.plist, I have the AirPlay optimization policy set to Long Form Audio. Having said all that, when I try to route my app to play "on a given HomePod speaker" ("play on ") the speaker routing instructions are never followed. I've looked and not been able to find where I might be able to instruct my app to follow the correct path here. I was assuming I could not trigger this behavior manually, as I believe I don't really have any control over AirPlay routing. Is there any guidance for working with SiriKit to do the right thing with regards to audio routing?

I’m using the shared instance of AVAudioSession. After activating it with .setActive(true), I observe the outputVolume, and it correctly reports the device’s volume. However, after deactivating the session using .setActive(false), changing the volume, and then reactivating it again, the outputVolume returns the previous volume (before deactivation), not the current device volume. The correct volume is only reported after the user manually changes it again using physical buttons or Control Center, which triggers the observer. What I need is a way to retrieve the actual current device volume immediately after reactivating the audio session, even on the second and subsequent activations. Disabling and re-enabling the audio session is essential to how my application functions. I’ve tested this behavior with my colleagues, and the issue is consistently reproducible on iOS 18.0.1, iOS 18.1, iOS 18.3, iOS 18.5 and iOS 18.6.2. On devices running iOS 17.6.1 and iOS 16.0.3, outputVolume correctly reflects the current volume immediately after calling .setActive(true) multiple times.

I’m building a teleprompter-style app that relies on Picture in Picture. PiP starts correctly on device. Everything works — until another app (e.g. TikTok / Instagram) starts active video recording. When camera capture begins in the foreground app, iOS terminates my PiP session. Some teleprompter apps appear to keep PiP active while recording in other apps, so I’m trying to understand the recommended architectural pattern for this scenario. Is there a documented approach or best practice to keep PiP stable during third-party camera capture? Looking specifically for guidance on the correct AVKit / AVAudioSession configuration for this use case.