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 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 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.

I need to apply headphone-specific scenario only when headphones are the sole active playback device in my iOS audio app. Problem that there is no absolute way to definitively understand that headphones are the sole active playback device AVAudioSession.currentRoute.outputs portTypes don't guarantee headphones: let session = AVAudioSession.sharedInstance() let outputs = session.currentRoute.outputs let headphonesOnly = outputs.count == 1 && (outputs.first?.portType == .headphones || outputs.first?.portType == .bluetoothA2DP || outputs.first?.portType == .bluetoothHFP || outputs.first?.portType == .bluetoothLE) The issue in code above that listed bluetooth profiles (A2DP, HFP, LE) can be used by any audio device, not only headphones Is there any public API on iOS that can: Distinguish Bluetooth headphones vs Bluetooth speakers when both use A2DP/LE? Expose the user’s “Device Type” classification (headphones / speaker / car stereo, etc.) that is shown in Settings → Bluetooth → Device Type? Provide a more reliable way to know “this route is definitely headphones” for A2DP devices, beyond portType and portName string heuristics?

When using the [AVAudioSession setCategory:withOptions:error:] API, the call hangs for a long time and eventually returns an error.This issue occurs on iOS 16, and did not appear in earlier versions. Thread 135: 0 libsystem_kernel.dylib 0x00000002478e3cd4 _mach_msg2_trap :8 (in libsystem_kernel.dylib) 1 libsystem_kernel.dylib 0x00000002478e7214 _mach_msg_overwrite :428 (in libsystem_kernel.dylib) 2 libsystem_kernel.dylib 0x00000002478e705c _mach_msg :24 (in libsystem_kernel.dylib) 3 libdispatch.dylib 0x00000001d63ffe84 __dispatch_mach_send_and_wait_for_reply :548 (in libdispatch.dylib) 4 libdispatch.dylib 0x00000001d6400224 _dispatch_mach_send_with_result_and_wait_for_reply :60 (in libdispatch.dylib) 5 libxpc.dylib 0x00000001b2114e04 _xpc_connection_send_message_with_reply_sync :256 (in libxpc.dylib) 6 Foundation 0x000000019b6249f0 ___NSXPCCONNECTION_IS_WAITING_FOR_A_SYNCHRONOUS_REPLY__ :16 (in Foundation) 7 Foundation 0x000000019c06d1b4 -[NSXPCConnection _sendInvocation:orArguments:count:methodSignature:selector:withProxy:] :2100 (in Foundation) 8 CoreFoundation 0x000000019dfcb1cc ____forwarding___ :1072 (in CoreFoundation) 9 CoreFoundation 0x000000019dfd3200 ___forwarding_prep_0___ :96 (in CoreFoundation) 10 AudioSession 0x00000001c77498b0 __ZN4avas6client11SessionCore10HandlePingEv :192 (in AudioSession) 11 AudioSession 0x00000001c77497b0 ____ZN4avas6client11SessionCore12DispatchPingEv_block_invoke :52 (in AudioSession) 12 libdispatch.dylib 0x00000001d63e4adc __dispatch_call_block_and_release :32 (in libdispatch.dylib) 13 libdispatch.dylib 0x00000001d63fe7ec __dispatch_client_callout :16 (in libdispatch.dylib) 14 libdispatch.dylib 0x00000001d63ed468 __dispatch_lane_serial_drain :740 (in libdispatch.dylib) 15 libdispatch.dylib 0x00000001d63edf78 __dispatch_lane_invoke :440 (in libdispatch.dylib) 16 libdispatch.dylib 0x00000001d63f6f48 __dispatch_root_queue_drain :364 (in libdispatch.dylib) 17 libdispatch.dylib 0x00000001d63f6d08 __dispatch_worker_thread :268 (in libdispatch.dylib) 18 libsystem_pthread.dylib 0x00000001f9ff144c __pthread_start :136 (in libsystem_pthread.dylib) 19 libsystem_pthread.dylib 0x00000001f9fed8cc _thread_start :8 (in libsystem_pthread.dylib) Thread 132: 0 libsystem_kernel.dylib 0x00000002478e3cd4 _mach_msg2_trap :8 (in libsystem_kernel.dylib) 1 libsystem_kernel.dylib 0x00000002478e7214 _mach_msg_overwrite :428 (in libsystem_kernel.dylib) 2 libsystem_kernel.dylib 0x00000002478e705c _mach_msg :24 (in libsystem_kernel.dylib) 3 libdispatch.dylib 0x00000001d63ffe84 __dispatch_mach_send_and_wait_for_reply :548 (in libdispatch.dylib) 4 libdispatch.dylib 0x00000001d6400224 _dispatch_mach_send_with_result_and_wait_for_reply :60 (in libdispatch.dylib) 5 libxpc.dylib 0x00000001b2114e04 _xpc_connection_send_message_with_reply_sync :256 (in libxpc.dylib) 6 Foundation 0x000000019b6249f0 ___NSXPCCONNECTION_IS_WAITING_FOR_A_SYNCHRONOUS_REPLY__ :16 (in Foundation) 7 Foundation 0x000000019c06d1b4 -[NSXPCConnection _sendInvocation:orArguments:count:methodSignature:selector:withProxy:] :2100 (in Foundation) 8 CoreFoundation 0x000000019dfcb1cc ____forwarding___ :1072 (in CoreFoundation) 9 CoreFoundation 0x000000019dfd3200 ___forwarding_prep_0___ :96 (in CoreFoundation) 10 AudioSession 0x00000001c7754198 __ZNK4avas6client11SessionCore18SetBatchPropertiesEP12NSDictionaryIP8NSStringPU25objcproto14NSSecureCoding11objc_objectEPU15__autoreleasingP7NSArrayIPS2_IS4_P8NSNumberEENS_30AVAudioSessionBatchSetStrategyEbb :548 (in AudioSession) 11 AudioSession 0x00000001c7753e58 __ZNK4avas6client11SessionCore20SetBatchPropertiesMXEP12NSDictionaryIP8NSStringPU25objcproto14NSSecureCoding11objc_objectE :92 (in AudioSession) 12 AudioSession 0x00000001c775179c __ZN4avas6client11SessionCore11setCategoryEP8NSStringS3_32AVAudioSessionRouteSharingPolicym :472 (in AudioSession) 13 AudioSession 0x00000001c7768f88 -[AVAudioSession setCategory:withOptions:error:] :68 (in AudioSession) 14 AlipayWallet 0x000000010140580c -[AVAudioSession(APMHook) apmhook_setCategory:withOptions:error:] APMHookAudioSession.m:35 (in AlipayWallet) 15 AlipayWallet 0x00000001014001a4 -[APMAudioSessionManager resume] APMAudioSessionManager.m:718 (in AlipayWallet) 16 libdispatch.dylib 0x00000001d63e4adc __dispatch_call_block_and_release :32 (in libdispatch.dylib) 17 libdispatch.dylib 0x00000001d63fe7ec __dispatch_client_callout :16 (in libdispatch.dylib) 18 libdispatch.dylib 0x00000001d63ed468 __dispatch_lane_serial_drain :740 (in libdispatch.dylib) 19 libdispatch.dylib 0x00000001d63edf44 __dispatch_lane_invoke :388 (in libdispatch.dylib) 20 libdispatch.dylib 0x00000001d63f83ec __dispatch_root_queue_drain_deferred_wlh :292 (in libdispatch.dylib) 21 libdispatch.dylib 0x00000001d63f7ce4 __dispatch_workloop_worker_thread :692 (in libdispatch.dylib) 22 libsystem_pthread.dylib 0x00000001f9fee3b8 __pthread_wqthread :292 (in libsystem_pthread.dylib) 23 libsystem_pthread.dylib 0x00000001f9fed8c0 _start_wqthread :8 (in libsystem_pthread.dylib)

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!

Is there any supported or recommended way to achieve user-configurable alarm volume while still using AlarmKit? Hi there! I’m currently building an alarm app on iOS using AlarmKit, and I’m running into a fundamental limitation around volume control. When using AlarmKit, alarm sounds are played at the system ringer volume. My goal is simple in concept: Allow users to set an alarm volume inside the app, and have the alarm sound play at that volume when triggered. However, AlarmKit does not provide any API to control or override the alarm volume. I’ve explored several approaches(AVSystemController, MPVolumeView...), but none achieved the desired result. Is there any supported or recommended way to achieve user-configurable alarm volume while still using AlarmKit? Any insights from developers who’ve shipped alarm apps, or from Apple engineers, would be greatly appreciated. Thanks in advance 🙏

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

I’m seeing what appears to be an iOS audio-session issue that occurs only when a phone call happens while the app is in the background. API: AVAudioSession, AVAudioRecorder Background Modes: Audio enabled (UIBackgroundModes = audio) Category: .playAndRecord Microphone permission: granted Expected Behavior If the app is recording audio in the background and a phone call interrupts it: AVAudioSession.interruptionNotification(.began) fires Call ends AVAudioSession.interruptionNotification(.ended) fires App should be able to re-activate its audio session and resume or restart recording Apple documentation suggests this should be supported for background audio apps. Actual Behavior When the app is in the background and phone call is ended: AVAudioSession.interruptionNotification(.ended) does fire Attempting to reactivate the audio session always fails: Error Domain=NSOSStatusErrorDomain Code=560557684 ("!int") "Session activation failed" The session appears to remain permanently “interrupted” Retrying activation (with delays) does not help Recreating AVAudioRecorder does not help Reactivation works only after the app is opened again

Hi everyone, I’m building a React Native iOS app where I’m integrating Wazo (native WebRTC) and Jitsi (WebView / WebRTC). Use case: Wazo is used to maintain a background call session (mainly signaling + audio keep-alive). Jitsi is used in the foreground for video calls. Problem: When Jitsi starts, it takes control of the microphone and camera. The Wazo call disconnects after ~5 minutes (likely due to media / audio session conflict). Even if Wazo audio/video is muted or tracks are disabled, the session still drops. My questions: Is it officially supported or recommended to run two WebRTC stacks (Wazo + Jitsi) simultaneously on iOS? Can Wazo stay connected without active audio/video tracks while Jitsi uses mic/camera? Is there a way to release Wazo media streams temporarily (but keep signaling alive) while Jitsi is loading or active? Are there any AVAudioSession / background mode limitations on iOS that make this impossible by design? If this is not supported, what is the recommended architecture (single WebRTC pipeline, switching media ownership, etc.)? Environment: iOS (React Native) Wazo SDK (native WebRTC) Jitsi Meet (WebView) CallKit + PushKit enabled Any guidance, documentation, or real-world experience would be greatly appreciated. Thanks in advance 🙏

I’m facing a strange audio routing issue that seems specific to iPhone 14 Pro / Pro Max. I’m using LiveKit (WebRTC) in a React Native app, which uses AVAudioSession internally for audio capture (VoIP / call-style usage). 🔍 What’s happening: I’m using an external USB microphone. On these devices: iPhone 11 → ✅ USB mic works iPhone 13 → ✅ USB mic works iPhone 17 Pro → ✅ USB mic works iPhone 14 Pro Max → ❌ USB mic does NOT work On iPhone 14 Pro Max: The same USB mic: ✅ Works in Voice Memos ✅ Works in Instagram Live ❌ Does NOT appear as an input option in my app ❌ Does NOT work in WhatsApp / Instagram calls Also: In my app on iPhone 14 Pro Max, iOS does not show the audio input selector UI On iPhone 17 Pro, the same app and same build does show the selector and the USB mic works ⚙️ My audio session config ( LiveKit ): await AudioSession.setAppleAudioConfiguration({ audioCategory: 'playAndRecord', audioMode: 'default', audioCategoryOptions: ['allowBluetooth', 'defaultToSpeaker'], }); await AudioSession.startAudioSession(); ❓ My questions: Is this a known limitation or behavior specific to iPhone 14 Pro / Pro Max? Does iPhone 14 Pro have different audio routing rules for call / VoIP mode compared to other devices? Why does the same USB mic work in recording apps (Voice Memos, Instagram Live) but not in call-style apps (LiveKit, WhatsApp, Instagram call)? Is there any documented difference in AVAudioSession behavior on iPhone 14 Pro regarding external USB audio inputs?

In my application, I use CallKit and have supportsHolding = true set. During my phone call, another call comes in (e.g., GSM). I accept the incoming call and put the current call on hold. If I end the active call myself, everything is fine, and CallKit calls the method provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession). However, if the other party ends the call, the second call remains on hold. In the application, the user clicks on unhold, and I notify CallKit that the hold has ended. But in this case, the didActivate method is not called at all. If I try to activate the audio myself after unhold, I receive the error: Domain=NSOSStatusErrorDomain Code=561017449 "Session activation failed" UserInfo={NSLocalizedDescription=Session activation failed} AVAudioSessionErrorInsufficientPriority == NSOSStatusErrorDomain Code: 561017449 What needs to be done for CallKit to activate my audio?

I can reproduce the bug that CallKit doesn't active audiosession after the outgoing call put on hold because of an incoming call. VoIP calling with CallKit Steps to reproduce: Download SpeakerBox example app from the link above and start it with XCode Start a new outgoing call Call your phone from other phone Hold and Accept the call After a few secs finish the call from the other phone The outgoing call will be still on hold Click on the call and click Toggle Hold The call won't be active again because the audiosession is activated. Logs: Received provider(_:didDeactivate:) Received provider(_:didDeactivate:) Received provider(_:didDeactivate:) Received provider(_:didDeactivate:) Received provider(_:didDeactivate:) Requested transaction successfully Starting audio Type: stdio AURemoteIO.cpp:1162 failed: 561017449 (enable 3, outf< 1 ch, 44100 Hz, Float32> inf< 1 ch, 44100 Hz, Float32>) Type: Error | Timestamp: 2024-08-15 12:20:29.949437+02:00 | Process: Speakerbox | Library: libEmbeddedSystemAUs.dylib | Subsystem: com.apple.coreaudio | Category: aurioc | TID: 0x19540d AVAEInternal.h:109 [AVAudioEngineGraph.mm:1344:Initialize: (err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)): error 561017449 Type: Error | Timestamp: 2024-08-15 12:20:29.949619+02:00 | Process: Speakerbox | Library: AVFAudio | Subsystem: com.apple.avfaudio | Category: avae | TID: 0x19540d Couldn't start Apple Voice Processing IO: Error Domain=com.apple.coreaudio.avfaudio Code=561017449 "(null)" UserInfo={failed call=err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)} Type: Notice | Timestamp: 2024-08-15 12:20:29.949730+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d Route change: Type: Notice | Timestamp: 2024-08-15 12:20:30.167498+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d ReasonUnknown Type: Notice | Timestamp: 2024-08-15 12:20:30.167549+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d Previous route: Type: Notice | Timestamp: 2024-08-15 12:20:30.167568+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d <AVAudioSessionRouteDescription: 0x302c00bc0, inputs = ( "<AVAudioSessionPortDescription: 0x302c01330, type = MicrophoneBuiltIn; name = iPhone Mikrofon; UID = Built-In Microphone; selectedDataSource = (null)>" ); outputs = ( "<AVAudioSessionPortDescription: 0x302c004d0, type = Receiver; name = Vev\U0151; UID = Built-In Receiver; selectedDataSource = (null)>" )> Type: Notice | Timestamp: 2024-08-15 12:20:30.167626+02:00 | Process: Speakerbox | Library: Speakerbox | TID: 0x19540d

Currently, I have successfully used ChannelMap to map hardware input channels and obtained audio data from the hardware device's MIC and OTG inputs. Additionally, I have used ChannelMap to map output channels to freely feed data for playback to each output channel. However, I now have a problem. I have a hardware device that only has output channels (no input channels), and the system has set this hardware device as the default playback device. In this case, how can I obtain the audio data being played to the output channels for modification?

Hi, We've noticed that this issue occurs more frequently after upgrading to iOS 18.4.1 and can result in one-way audio. Our app uses CallKit with WebRTC to establish VoIP connections. However, on iOS 18.4.1, CallKit no longer triggers: func provider(_ provider: CXProvider, didActivate audioSession: AVAudioSession) We're currently comparing the occurrence rate across different iOS versions to better understand the impact. Could you please help analyze the root cause of this issue?

I am facing a severe audio routing instability issue when using CallKit and the Zego Express SDK on iOS. The problem is that the audio route immediately reverts from the Speaker back to the Earpiece, effectively disabling the Speaker button functionality.📝 Observed BehaviorWhen the user taps the native CallKit Speaker Button, the audio route is correctly changed to the Speaker, but then instantly flips back to the Receiver (Earpiece), as shown in the system log captured via AVAudioSession.routeChangeNotification monitoring.🧾 Log Evidence (Flapping Occurs in 0.4 seconds)The following log snippet clearly illustrates the system overriding the user's action (reason: 4) with an unexpected CategoryChange (reason: 3) event: TimestampComponentReason CodeDescriptionRouteIs Speaker16:31:18.009[CallKitManager]4Override (CallKit/ControlCenter)Loa ngoài (Speaker)true16:31:18.411[CallKitManager]3CategoryChangeThiết bị nhận (Receiver)false

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'm implementing the PushToTalk framework and have encountered an issue where channelManager(_:didActivate:) is not called under specific circumstances. What works: App is in foreground, receives PTT push → didActivate is called ✅ App receives audio in foreground, then is backgrounded → subsequent pushes trigger didActivate ✅ What doesn't work: App is launched, user joins channel, then immediately backgrounds PTT push arrives while app is backgrounded incomingPushResult is called, I return .activeRemoteParticipant(participant) The system UI shows the speaker name correctly However, didActivate is never called Audio data arrives via WebSocket but cannot be played (no audio session) Setup: Channel joined successfully before backgrounding UIBackgroundModes includes push-to-talk No manual audio session activation (setActive) anywhere in my code AVAudioEngine setup only happens inside didActivate delegate method Issue persists even after channel restoration via channelDescriptor(restoredChannelUUID:) Question: Is this expected behavior or a bug? If expected, what's the correct approach to handle incoming PTT audio when the app is backgrounded and hasn't received audio while in the foreground yet?

Hi everyone, I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms. Problem: When the app is recording audio and an interruption occurs: I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began). On .ended, I check for .shouldResume and call audioRecorder?.record() again. The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder. Repro: Start a recording with AVAudioRecorder Simulate a system interruption (e.g., incoming call) Resume recording after the interruption Stop and inspect the output audio file Expected: Full audio (before and after interruption) should be saved. Actual: Only the audio after interruption is saved; the earlier part is missing Notes: According to the documentation, calling .record() after .pause() should resume recording into the same file. I confirmed that the file URL does not change, and I do not recreate the recorder instance. No error is thrown by the system during this process. This behavior happens consistently when the app is interrupted and resumed. Question: Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen? Thanks in advance!