We have the necessary background recording entitlements, and for many users... do not run into any issues. However, there is a subset of users that routinely get recordings ending.. we have narrowed this down and believe it to be the work of the watch dog. First we removed the entire view hierarchy when app is backgrounded. There is just 'Text("Recording")' This got the CPU usage in profiler down to 0%. We saw massive improvements to recording success rate. We walked away assuming that was enough. However we are still seeing the same sort of crashes. All in the background. We're using Observation to drive audio state changes to a Live Activity. Are those Observations causing the problem? Why doesn't apple provide a better API to background audio? The internet is full of weird issues https://stackoverflow.com/questions/76010213/why-is-my-react-native-app-sometimes-terminated-in-the-background-while-tracking https://stackoverflow.com/questions/71656047/why-is-my-react-native-app-terminating-in-the-background-while-recording-ios-r https://github.com/expo/expo/issues/16807 This is such a terrible user experience. And we have very little visibility into what is happening and why. No where in apple documentation states that in order for background recording to work, the app can only be 'Text("Recording")' It does not outline a CPU or memory threshold. It just kills us.

I have an app under development - demo here - https://youtu.be/VbAfUk_eYl0?si=s6EDBx-4G6P_QbZO - which is sort of an audio player for airdropped files - something useful to musicians who dump work in progress to their phone, make notes, revise and update. I've been testing my handling of audio session interruption notifications, but seems to be a lot of inconsistency in how, when and why iOS delivers them, and I'm wondering if there is some rhyme or reason to it that I'm just not detecting. For example, I am playing a song in my app. Switch to Apple Music and start playing a song there. My app gets an interruption began notification - this is consistent. Switch back to my app, and about half the time, I will get an interruption ended notification (coupled often with a blast of the tail of whatever audio buffer was partially played when the interruption started, even though the engine was stopped - and followed by call to my AVAudioPlayerNodeCompletionCallback - is there some way to avoid this?). Half the time I don't get an interruption ended notification; my app can (as expected) end the interruption by activating the AVAudioSession and playing something. I have not been able to determine any pattern to this behavior, other than that if my app started playing using AVAudioPlayerNode.scheduleSegment rather than scheduleFile I think the notification will be consistently delivered on app activation rather than when I activate the session programmatically. I would like my app to behave deterministically, and would appreciate any help in deciphering what causes the inconsistent behavior in notifications from iOS.

Let's consider the following code. I've created an actor that loads a list of .mp3 files from a Bundle and then makes it available for audio reproduction. Unfortunately, I'm experiencing a memory leak. At the play method. player.play() From Instruments I get _malloc_type_malloc_outlined libsystem_malloc.dylib start_wqthread libsystem_pthread.dylib private actor AudioActor { enum Failure: Error { case soundsNotLoaded([AudioPlayerClient.Sound: Error]) } enum Player { case music(AVAudioPlayer) } var players: [Sound: Player] = [:] let bundles: [Bundle] init(bundles: UncheckedSendable<[Bundle]>) { self.bundles = bundles.wrappedValue } func load(sounds: [Sound]) throws { try AVAudioSession.sharedInstance().setActive(true, options: []) var errors: [Sound: Error] = [:] for sound in sounds { guard let url = bundle.url(forResource: sound.name, withExtension: "mp3") else { continue } do { self.players[sound] = try .music(AVAudioPlayer(contentsOf: url)) } catch { errors[sound] = error } } guard errors.isEmpty else { throw Failure.soundsNotLoaded(errors) } } func play(sound: Sound, loops: Int?) throws { guard let player = self.players[sound] else { return } switch player { case let .music(player): player.numberOfLoops = loops ?? -1 player.play() } } func stop(sound: Sound) throws { guard let player = self.players[sound] else { throw Failure.soundsNotLoaded([:]) } switch player { case let .music(player): player.stop() } } }

Dear Apple Team, I am facing an issue with UDP networking in my watchOS app for duplex audio streaming using NWConnection. I have already added the necessary capabilities, including background mode for audio, to ensure smooth operation. Issue Details: The UDP connection works fine on the simulator since it uses macOS networking and allows low-level access. However, on a real Apple Watch (running watchOS 10), the connection remains in a "waiting" state and fails with Error 50. I am aware of Technical Note TN3135 regarding low-level networking on watchOS, but even after following these guidelines, the issue persists. Questions: Does watchOS impose additional restrictions on UDP networking compared to iOS/macOS? Are there any specific entitlements or configurations required to allow UDP connections on a real Apple Watch? Is there a workaround or debugging method to get more insights into why the connection fails? I would appreciate any guidance or recommendations on resolving this issue.

Hi, I am looking for a good way to play sounds at a high frequency. At the moment I am using the AVAudioEngine, and create a couple AVAudioPlayerNode and for each sound I need to play I create a AVAudioPCMBuffer. When the app needs to play a sound, I get the correct AVAudioPCMBuffer for the sound and use the first available AVAudioPlayerNode and feed it to the buffer. The timing for a metronome app has to be very precise because if it's of by about 16ms the user can hear that it is not playing had the right interval. For low speeds this is working without any problems, but at high speeds it is getting worse. Maybe anyone has an idea on how I can improve my method. Its a Plugin for Flutter. import AVFoundation class FastSoundPlayer { private var audioPlayers: [SoundPlayer?] = [] private var sounds: [String: Sound] = [:] private var engine = AVAudioEngine() let session = AVAudioSession.sharedInstance() init() { do { try session.setCategory(AVAudioSession.Category.playback, mode: AVAudioSession.Mode.default, options: [AVAudioSession.CategoryOptions.mixWithOthers]) try session.setActive(true) createSoundPlayers(count: 20) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } // Selector method to handle applicationDidBecomeActiveNotification func applicationDidBecomeActive() { // Reinitialize AVAudioEngine and reattach all nodes do { engine.reset() objc_sync_enter(audioPlayers) audioPlayers.removeAll() createSoundPlayers(count: 20) objc_sync_exit(audioPlayers) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } func createSoundPlayers(count: Int) { for _ in 0..<count { let player = SoundPlayer() engine.attach(player.player) engine.connect(player.player, to: engine.mainMixerNode, format: nil) audioPlayers.append(player) } } func load(sound: Data, name: String) { let sound = Sound(soundData: sound) sounds[name] = sound } func play(name: String) { if !engine.isRunning { applicationDidBecomeActive() } guard let sound = sounds[name] else { print("Sound not found") return } if let player = getAvailablePlayer() { player.play(sound: sound) } } func getAvailablePlayer() -> SoundPlayer? { for player in audioPlayers { if !player!.isPlaying { return player } } return nil } } class SoundPlayer { let player = AVAudioPlayerNode() var isPlaying = false init() { player.volume = 1.0 } func play(sound: Sound) { player.scheduleBuffer(sound.sound!, at: nil, options: .interrupts, completionCallbackType: .dataPlayedBack) { _ in self.complete() } if (player.engine != nil && player.engine!.isRunning) { player.play() isPlaying = true } } func complete() { isPlaying = false } } class Sound { var sound: AVAudioPCMBuffer? init(soundData: Data) { do { let temporaryURL = FileManager.default.temporaryDirectory.appendingPathComponent("tempSound.wav") try soundData.write(to: temporaryURL) // Create AVAudioFile from the temporary file URL let audioFile = try AVAudioFile(forReading: temporaryURL) // Define the format for the PCM buffer (44100Hz, stereo) let format = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44100, channels: 2, interleaved: false) // Create AVAudioPCMBuffer guard let pcmBuffer = AVAudioPCMBuffer(pcmFormat: format!, frameCapacity: AVAudioFrameCount(audioFile.length)) else { // Failed to create PCM buffer self.sound = nil return } // Read audio file into PCM buffer try audioFile.read(into: pcmBuffer) // Assign the created AVAudioPCMBuffer to the sound property self.sound = pcmBuffer } catch { print("Error loading sound file: \(error.localizedDescription)") self.sound = nil } } } Thanks!

I'm trying to implement airplay into my app. I can successfully playback sound and trigger the airplay selector sheet. If the target device is a Bluetooth only device I can connect with no problem and stream the audio to the Bluetooth device, but if the audio device is a airplay specific device like a HomePod or an Apple TV when I select it, I get a spinning icon, indicating that it is trying to connect, and eventually it times out and stops without connecting. I don't believe it is an AirPlay audio issue because if I go to a different app, for example a podcast app and select my HomePods for output, and then switch back to my app. My audio will correctly stream to the HomePod. Not only that, I have it so that my icon will change color to indicate that it is connected via airplay and it is correctly indicating that it is connected via AirPlay. But I cannot then disconnect it using the Airplay selector. The issue appears to be in the AirPlay selection side, which I have spent several days attempting to troubleshoot mostly using ChatGPT to suggest code different than what I have to maybe work around the issue. Mostly it is focused on the audio player section, but it doesn't seem like that is really the route that is the problem.

I have a SwiftUI app - (https://youtu.be/VbAfUk_eYl0?si=JxUBh0Bpb-vc1E1U) - which I thought was almost ready for release - a manager for airdropped audio files from Logic Pro or other music creation applications. It uses AVAudioEngine and AVAudioPlayerNode to play audio, and the MediaPlayer API to integrate with car audio and similar, all of which works well. It does not currently have an explicit CarPlay integration (and I'm slightly horrified at the amount of work that is going to require). I had the good or bad luck of getting a loaner car with carplay while mine is being repaired yesterday, and lo and behold, when connected to the vehicle via CarPlay, there is no audio output in the vehicle at all. The now playing panel correctly shows the information my app provides about the currently playing song; the player node believes it is playing, the AVAudioSession is configured as it should be. But there is no sound. Obviously I cannot ship it in this state. I've tried fiddling with the parameters the AVAudioSession is configured with, in case there was some parameter that was preventing audio output, to no avail - currently: var options = AVAudioSession.CategoryOptions() options.insert(.allowAirPlay) options.insert(.allowBluetooth) options.insert(.allowBluetoothA2DP) try session.setCategory(.playback, mode: .default, options: options) try? session.setPreferredIOBufferDuration(0.002) // ~96 samples at 44.1kHz try? session.setPrefersNoInterruptionsFromSystemAlerts(true) try? session.setPrefersInterruptionOnRouteDisconnect(false) try session.setActive(true, options: [.notifyOthersOnDeactivation]) All diagnostics within the app show the player operating correctly - files are played and flushed; AVAudioPlayerNodeCompletionCallbacks are called when they should be. But the output is not audible in the vehicle. I would much prefer to ship this app without full-blown CarPlay integration, but with working audio when connected via CarPlay, and work on full CarPlay integration for the next release. Is there some secret handshake I am just missing to make this work?

I did watch WWDC 2019 Session 716 and understand that an active audio session is key to unlocking low‑level networking on watchOS. I’m configuring my audio session and engine as follows: private func configureAudioSession(completion: @escaping (Bool) -> Void) { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: []) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) // Retrieve sample rate and configure the audio format. let sampleRate = audioSession.sampleRate print("Active hardware sample rate: \(sampleRate)") audioFormat = AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1) // Configure the audio engine. audioInputNode = audioEngine.inputNode audioEngine.attach(audioPlayerNode) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: audioFormat) try audioEngine.start() completion(true) } catch { print("Error configuring audio session: \(error.localizedDescription)") completion(false) } } private func setupUDPConnection() { let parameters = NWParameters.udp parameters.includePeerToPeer = true connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupTCPConnection() { let parameters = NWParameters.tcp connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupWebSocketConnection() { guard let url = URL(string: "ws://***.***.xxxxx.***:0000") else { print("Invalid WebSocket URL") return } let session = URLSession(configuration: .default) webSocketTask = session.webSocketTask(with: url) webSocketTask?.resume() print("WebSocket connection initiated") sendAudioToServer() receiveDataFromServer() sendWebSocketPing(after: 0.6) } private func setupNWConnectionHandlers() { connection?.stateUpdateHandler = { [weak self] state in DispatchQueue.main.async { switch state { case .ready: print("Connected (NWConnection)") self?.isConnected = true self?.failToConnect = false self?.receiveDataFromServer() self?.sendAudioToServer() case .waiting(let error), .failed(let error): print("Connection error: \(error.localizedDescription)") DispatchQueue.main.asyncAfter(deadline: .now() + 2) { self?.setupNetwork() } case .cancelled: print("NWConnection cancelled") self?.isConnected = false default: break } } } connection?.start(queue: .main) } I am reaching out to seek further assistance regarding the challenges I've been experiencing with establishing a UDP, TCP & web socket connection on watchOS using NWConnection for duplex audio streaming. Despite implementing the recommendations provided earlier, I am still encountering difficulties. Or duplex audio streaming not possible on apple watch?

I am working on an app which plays audio - https://youtu.be/VbAfUk_eYl0?si=nJg5ayy2faWE78-g - and one of the features is, on restart, if you had paused playback of a file at the time the app was previously shut down (or were playing one at the time of shutdown), the paused state and position in the file is restored exactly as it was, on restart. The functionality works. However, it seems impossible to get the "now playing" information in iOS into the right state to reflect that via the MediaPlayer API. On restart, handlers are attached to the play/pause/togglePlayPause actions on MPRemoteCommandCenter.shared(), and the map of media info is updated on MPNowPlayingInfoCenter.default().nowPlayingInfo. What happens is that iOS's media view shows the audio as playing and offers a pause button - even though the play action is enabled and the pause action is disabled. Once playback has been initiated (my workaround is to have the pause action toggle the play state, since otherwise you wouldn't be able to initiate playback from controls in a car without initiating it once from a device first). I've created a simplified white-noise-player demo to illustrate the problem - simply build and deploy it, and then start the app, lock your device and look at the playback controls on the lock screen. It will show a pause button - same behavior I've described. https://github.com/timboudreau/ios-play-pause-demo I've tried a few things to narrow down the source of the issue - for example, thinking that not MPNowPlayingInfoPropertyPlaybackProgress and MPMediaItemPropertyPlaybackDuration might be the culprit (since the system interpolates elapsed time and it's recommended to update those properties infrequently) on startup might do the trick, but the result is the same, just without a duration or progress shown. What governs this behavior, and is there some way to explicitly tell the media player API your current state is paused?

Environment→ ・Device: iPad 10th generation ・OS：**iOS18.3.2 I'm using AVAudioSession to record sound in my application. But I recently came to realize that when the app starts a recording session on a tablet, OS automatically sets the tablet volume to 50% and when after recording ends, it doesn't change back to the previous volume level before starting the recording. So I would like to know whether this is an OS default behavior or a bug? If it's a default behavior, I much appreciate if I can get a link to the documentation.

I did watch WWDC 2019 Session 716 and understand that an active audio session is key to unlocking low‑level networking on watchOS. I’m configuring my audio session and engine as follows: private func configureAudioSession(completion: @escaping (Bool) -> Void) { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: []) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) // Retrieve sample rate and configure the audio format. let sampleRate = audioSession.sampleRate print("Active hardware sample rate: \(sampleRate)") audioFormat = AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1) // Configure the audio engine. audioInputNode = audioEngine.inputNode audioEngine.attach(audioPlayerNode) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: audioFormat) try audioEngine.start() completion(true) } catch { print("Error configuring audio session: \(error.localizedDescription)") completion(false) } } private func setupUDPConnection() { let parameters = NWParameters.udp parameters.includePeerToPeer = true connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupTCPConnection() { let parameters = NWParameters.tcp connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupWebSocketConnection() { guard let url = URL(string: "ws://***.***.xxxxx.***:0000") else { print("Invalid WebSocket URL") return } let session = URLSession(configuration: .default) webSocketTask = session.webSocketTask(with: url) webSocketTask?.resume() print("WebSocket connection initiated") sendAudioToServer() receiveDataFromServer() sendWebSocketPing(after: 0.6) } private func setupNWConnectionHandlers() { connection?.stateUpdateHandler = { [weak self] state in DispatchQueue.main.async { switch state { case .ready: print("Connected (NWConnection)") self?.isConnected = true self?.failToConnect = false self?.receiveDataFromServer() self?.sendAudioToServer() case .waiting(let error), .failed(let error): print("Connection error: \(error.localizedDescription)") DispatchQueue.main.asyncAfter(deadline: .now() + 2) { self?.setupNetwork() } case .cancelled: print("NWConnection cancelled") self?.isConnected = false default: break } } } connection?.start(queue: .main) } Duplex in this context refers to two-way audio transmission simultaneously recording and sending audio while also receiving and playing back incoming audio, similar to a VoIP/SIP call. The setup works fine on the simulator, which suggests that the core logic is correct. However, since the simulator doesn’t fully replicate WatchOS hardware behavior especially for audio sessions and networking issues might arise when running on a real device. The problem likely lies in either the Watch’s actual hardware limitations, permission constraints, or specific audio session configurations. I am reaching out to seek further assistance regarding the challenges I've been experiencing with establishing a UDP, TCP & web socket connection on watchOS using NWConnection for duplex audio streaming. Despite implementing the recommendations provided earlier, I am still encountering difficulties From what I can see, your implementation is focused on streaming audio playback with the server. In my case, I'm looking for a slightly different approach: I want to capture audio and send buffers of a specific size to the server while playing audio simultaneously, essentially achieving full duplex streaming similar to a VOIP call. Additionally, I’d like to ensure that if no external audio route is connected, the Apple Watch speaker is used by default. Any thoughts or insights on adapting this setup for those requirements would be very welcome.

Is there a way to configure the APNS notification sound volume to be louder? I am implementing some custom sounds(narrative sentences) for APNS, it does play the custom sound, but the volume of the custom sound is not loud enough even though I had set the device's volume and "RingTone and Alerts" volume to max. I tried to amplify the custom sound file, it does play louder but the result is minimum if I want to maintain the quality of the sound without it been distorted. I tried to use Notification Service Extension, AVAudioPlayer and AVAudioSession to play the sound, it does play louder in max volume compare with relying on default sound payload in APNS, but the problem is AVAudioPlayer and AVAudioSession do not seems to be usable when the application is in background or killed state, is there any other alternative I could use?

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!

I'm working on adding CarPlay support to an audio app and am running into an issue. Occasionally, when a user opens the app from CarPlay while the main app scene is either not connected or is currently in the background, I will receive an error when attempting to activate the audio session. The code below mimics my setup: do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio) try AVAudioSession.sharedInstance().setActive(true) } catch { print(error) // NSOSStatusErrorDomain - 560557684: Session activation failed } That error code maps to AVAudioSession.ErrorCode.cannotInterruptOthers. Once in this state, all subsequent attempts to play different pieces of content will fail. However, things will start working normally if the user opens the app on their phone and tries again from CarPlay (while the app is in the foreground on their phone). I'm not sure why it would behave this way and want to note that I do have the audio background mode capability enabled. Has anyone else encountered this? Are there any workarounds or changes I could make to prevent this from happening?

Is there anyway that I could use AVAudioSession, AVAudioPlayer or anything similar in Notification Service Extension? I am trying to implement Audio Playback in the Notification Service Extension to play specific audio file when receiving Notification regardless the app state(foreground, background or killed), but I am not able to activate audio session in Notification Service Extension. NSError *sessionError = nil; BOOL success = [[AVAudioSession sharedInstance] setCategory:AVAudioSessionCategoryPlayback error:&sessionError]; success = [[AVAudioSession sharedInstance] setActive:YES error:&sessionError]; if (!success) { NSLog(@"Error activating audio session: %@", sessionError); } Below is the error that I got when I am trying to run the code above in Notification Service Extension. Error Domain=NSOSStatusErrorDomain Code=561015905 "Session activation failed" UserInfo={NSLocalizedDescription=Session activation failed}

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?

Hello! I'm experiencing an issue with iOS's audio routing system when trying to use Bluetooth headphones for audio output while also recording environmental audio from the built-in microphone. Desired behavior: Play audio through Bluetooth headset (AirPods) Record unprocessed environmental audio from the iPhone's built-in microphone Actual behavior: When explicitly selecting the built-in microphone, iOS reports it's using it (in currentRoute.inputs) However, the actual audio data received is clearly still coming from the AirPods microphone The audio is heavily processed with voice isolation/noise cancellation, removing environmental sounds Environment Details Device: iPhone 12 Pro Max iOS Version: 18.4.1 Hardware: AirPods Audio Framework: AVAudioEngine (also tried AudioQueue) Code Attempted I've tried multiple approaches to force the correct routing: func configureAudioSession() { let session = AVAudioSession.sharedInstance() // Configure to allow Bluetooth output but use built-in mic try? session.setCategory(.playAndRecord, options: [.allowBluetoothA2DP, .defaultToSpeaker]) try? session.setActive(true) // Explicitly select built-in microphone if let inputs = session.availableInputs, let builtInMic = inputs.first(where: { $0.portType == .builtInMic }) { try? session.setPreferredInput(builtInMic) print("Selected input: \(builtInMic.portName)") } // Log the current route let route = session.currentRoute print("Current input: \(route.inputs.first?.portName ?? "None")") // Configure audio engine with native format let inputNode = audioEngine.inputNode let nativeFormat = inputNode.inputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: nativeFormat) { buffer, time in // Process audio buffer // Despite showing "Built-in Microphone" in route, audio appears to be // coming from AirPods with voice isolation applied - welp! } try? audioEngine.start() } I've also tried various combinations of: Different audio session modes (.default, .measurement, .voiceChat) Different option combinations (with/without .allowBluetooth, .allowBluetoothA2DP) Setting session.setPreferredInput() both before and after activation Diagnostic Observations When AirPods are connected: AVAudioSession.currentRoute.inputs correctly shows "Built-in Microphone" after setPreferredInput() The actual audio data received shows clear signs of AirPods' voice isolation processing Background/environmental sounds are actively filtered out... When recording a test audio played near the phone (not through the app), the recording is nearly silent. Only headset voice goes through. Questions Is there a workaround to force iOS to actually use the built-in microphone while maintaining Bluetooth output? Are there any lower-level configurations that might resolve this issue? Any insights, workarounds, or suggestions would be greatly appreciated. This is blocking a critical feature in my application that requires environmental audio recording while providing audio feedback through headphones 😅

Hi 👋! We have a SpriteKit-based app where we play AVAudio sounds in three different ways: Effects (incl. UI sounds) with AVAudioPlayer. Long looping tracks with AVAudioPlayer. Short animation effects on the timeline of SpriteKit's SKScene files (effectively SKAudioNode nodes). We've found that when you exit the app or otherwise interrupt audio plays, future audio plays often fail. For example, there's a WebKit-based video trailer inside the app, and if you play it, our looping background music track (2.) will stop playing, and won't resume as you close the trailer (return from WebKit). This is probably due to us not manually restarting the track (so may well be easily fixed). Periodically played AVAudioPlayer audio (1.) are not affected. However, the more concerning thing is that the audio tracks on SKScene file timelines (3.) will no longer play. My hypothesis is that AVAudioEngine gets interrupted, and needs to be restarted for those AVAudioNode elements to regain functionality. Thing is, we don't deal with AVAudioEngine at all currently in the app, meaning it is never initiated to begin with. Obviously things return to normal when you remove the app from short-term memory and restart it. However, it seems many of our users aren't doing this, and often report audio failing presumably due to some interruption in the past without the app ever being cleared from memory. Any idea why timeline-run SKAudioNodes would fail like this? Should the app react to app backgrounding/foregrounding regarding audio? Any help would be very much appreciated ✌️!

My app requirement is to check that User is on call while doing transaction. If user on call then we need to show Caution alert. For this requirement we used CallKit to detect Call status and it's working fine but recently Apple has rejected the application because of Callkit that is banned in China. Could you please provide any solution to check the Call Status only.

I am work an app development on an app which request an audio function in background as an alert sound. during debug testing , the function work fine, but once I testing standalone without debugging , The function not work , it will play out the sound when I back to app. does any way to trace the issues ?