Is anyone experiencing an issue with initial ‘AVAudioSession.sharedInstance().outputVolume’ returning 0 after updating to iOS 18.0? I’m observing the outputVolume of AVAudioSession, and when I adjust the device volume, the change value returns correctly. However, when I call AVAudioSession.sharedInstance().outputVolume to get the current volume before knowing the change value, it returns 0, even though the device volume is not actually 0.

After upgrading to iOS 18 CarPlay with 2023 Lexus and iPhone 15 Pro Max shows multiple issues: • speakers reduced to Mono sound (going back to normal after some minutes and then reducing again) • no speaker sound at all • touching / moving phone while driving resulting in “on and off” sound No Reboot / Shutdown helps No Cable connection works @Apple: do you test your software professionally or is this outsourced to the community? Doesn’t look at all like a professional approach? Please solve this dangerous (traffic!) and annoying topic ASAP! Thanks - Torsten

My Code private let synthesizer = AVSpeechSynthesizer() override func didReceive(_ request: UNNotificationRequest, withContentHandler contentHandler: @escaping (UNNotificationContent) -> Void) { self.contentHandler = contentHandler bestAttemptContent = (request.content.mutableCopy() as? UNMutableNotificationContent) do { try AVAudioSession.sharedInstance().setCategory(AVAudioSession.Category.playback) let utterance = AVSpeechUtterance(string: "Hello Sim") utterance.voice = AVSpeechSynthesisVoice(language: "th_TH") utterance.rate = 1.0 utterance.pitchMultiplier = 1.0 utterance.volume = 1.0 utterance.preUtteranceDelay = 0 self.synthesizer.usesApplicationAudioSession = false self.synthesizer.speak(utterance) } catch { print(error.localizedDescription) } if let bestAttemptContent = self.bestAttemptContent { contentHandler(bestAttemptContent) } } Info.plist enable UIBackgroundModes audio fetch processing remote-notification payloadNotification { "aps":{ "alert":{ "title":"title" "subtitle":"subtitle" "body":"body" } "mutable-content":1 "content-available":1 "sound":"tester.caf" } } This code can play in simulator but can't play in real device

I have an app that plays audio and the behaviour of it has changed in watchOS 11. I can no longer figure out how to play the audio through the headphones. To play audio I.. let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, policy: .longFormAudio, options: [] let activated = try await session.activate() if activated { // play audio } In previous versions, 'try await session.activate()` would bring up a route picker where the user could select their headphones. Now on watchOS 11 it just plays the audio out of the speaker. Maybe that's what some people want but if they do want it to play out of the headphones I can't see how I can give that option now? There's no AVRoutePickerView available on watchOS for selecting it. I've tried setting the category to .multiRoute instead of .playback and that does bring up the picker but selecting the speaker results in an error code and selecting the headphones results in it saying it cannot find my headphones (which shouldn't be the case since Apple Music on watchOS finds them). Tried overriding the output with try session.overrideOutputAudioPort(.speaker) but the compiler complains that speaker isn't available on watchOS, which is strange as if I understand correctly it's possible to play through the speaker now at least on some Apple Watches. So is this a bug or is there some way I've not found of playing audio through the headphones?

Hi, I am trying to detect if an audio stream is Dolby Atmos. I have existing code that determines if a stream is Dolby Atmos based on the following: Channel count is greater than equal to 8 Binaural is true Immersive is true Downmix is false I am trying to determine if these rules are correct and documentation that specifies these rules that I can reference in the future. Any help you can provide is greatly appreciated. Regards, John

Is it possible to play audio in the Background or when the app is Terminated? If yes, how can I play audio in the Background or when the app is Terminated in iOS using Swift? I am receiving an audio link in a Firebase notification. How can I play this audio link when the app is in the Background or Terminated?

When using the AVSpeechSynthesizer() , I get an error after a couple of seconds :"IPCAUClient.cpp:139 IPCAUClient: can't connect to server (-66748) <0x104309130>", and then it speaks the text. The second time I call speak, there is no delay and error and it speaks immediately. Where does this error and delay come from and how can I resolve it? Intialization code: self.audioSession = AVAudioSession.sharedInstance() // 2) handle audio session first, before trying to read the text do { try audioSession.setCategory(.playback, mode: .voicePrompt, options: .duckOthers) try audioSession.setActive(false) } catch let error { Logger.model.debug("❓\(error.localizedDescription)") } speechSynthesizer = AVSpeechSynthesizer() speechSynthesizer.usesApplicationAudioSession = true Speak code: let utterance = AVSpeechUtterance(string: text) utterance.preUtteranceDelay = 0.1 utterance.rate = 0.5 utterance.pitchMultiplier = 0.75 utterance.prefersAssistiveTechnologySettings = false self.speechSynthesizer.speak(utterance) The last statement gives this error message!

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

Hi! I'm developing a music player app that interchanges between ApplicationMusicPlayer and AVAudioEngine. I'm facing an issue when switching from playback via ApplicationMusicPlayer to AVAudioEngine while the app is in background. Based on testing, it seems like the issue has to do with being unable to set audio focus in background, causing error AVAudioSessionErrorCodeCannotInterruptOthers. I would like to check if ApplicationMusicPlayer has its own audio focus separated from the app's own audio focus. If it is, is there anything that I can do to ensure that ApplicationMusicPlayer returns focus to the app? (I notice that the issue does not occur if we are moving playback from AVAudioEngine to ApplicationMusicPlayer. Not sure why the opposite does not work)

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 experiencing stuttering every time I record something with my iOS app on iOS 18 beta. The code ran fine on previous iOS versions. The stuttering occurs for the first 2 seconds. Here's an example: https://soundcloud.com/thomas-walther-219010679/ios-18-stuttering The way I set up AVAudioEngine and AVAudioSession was vetted quite thoroughly during sessions at WWDC '23. Here is how the engine and the tap is configured: let engine = AVAudioEngine() let recorderNode = AVAudioMixerNode() engine.attach(recorderNode) engine.connect(engine.mainMixerNode, to: engine.outputNode, format: engine.outputNode.inputFormat(forBus: 0)) engine.connect(recorderNode, to: engine.mainMixerNode, format: recordingOutputFormat) engine.connect(engine.inputNode, to: recorderNode, format: engine.inputNode.inputFormat(forBus: 0)) let bufferSize: AVAudioFrameCount = 4096 recorderNode.installTap(onBus: 0, bufferSize: bufferSize, format: nil) { [weak self] buffer, time in guard let self = self else { return } do { // Write recording to disk try audioFile.write(buffer) } catch { // ... } } I tried setting a different buffer size, but with no luck. I also can't see any hangs in Instruments. Do you have any pointers on how to debug this?

Hi, Team. We are currently creating a VoIP calling app using pjsip and want to be able to end a call using the headset button while the app is in the middle of a call (AVAudioSession.category == .playAndRecord), but MPRemoteCommand does not receive any events. After trying various things, We found that the button will respond if the audio output destination is set to the speaker or if .allowBluetoothA2DP is set as an option, but this is not suitable for this use case because audio input and output would be from the device rather than the headset. ================================================= Problem Headset button events cannot be received from MPRemoteCommand during a call. What is expected to happen? When the headset button is pressed during a call, a handler registered in some MPRemoteCommand is called back. What does actually happen? No MPRemoteCommand responds when the headset button is pressed during a call. Information Sample code Echoes back the audio input with a 5-second delay to simulate a phone call. https://github.com/ryu-akaike/HeadsetTalkTest-iOS/ Versions macOS: Sonoma 14.5 Xcode: 15.3 iPhone: 11 iOS: 17.5.1 ================================================= Thank you. Ryu Akaike

I need to find a way to allow recording from the mic while outputting two different sound streams to two different devices (speaker and headphones). I've done a fair bit of reading around using AVAudioSession.Category.multiroute but haven't found any modern examples. @theanalogkid posted a nice example using obj-C nine years ago, but others have noted that the code isn't readily translatable to Swift. To make matters worse, this is one of the very few examples on how to properly use multirouting. The official documentation is lacking, to say the least, and the WWDC 2012 session is, well, old enough to attend middle school and be a Taylor Swift fan, but definitely not in Swift. The few relevant forum posts here are spread over this middle schooler's life span and likely outdated, with most having no responses other than the poster's own plightful echo. They don't paint a pretty picture of .multiroute's health, with a recent poster noting that volume buttons don't work in this mode, contacting DTS and finding that there's no fix; another finding that it just doesn't work for certain devices, etc. Audio is giving me enough of a headache so I'd like to avoid slogging through this if possible. .multiroute feels like the developer mode of AVAudioSession, but without documentation. tl;dr - Without using .multiroute, is there a way to allow an app to output two different devices while simultaneously recording audio? If .multiroute is the only way to achieve this, can someone give me a quick rundown of how this category works?

I am developing a visionOS app that captions speech in real environments. Currently, I am using Apple's built-in speech recognizer. However, when I was testing the app with a Vision Pro, the device seemed to only pick up the user's voice (in other words, the voices of the wearer of the Vision Pro device). For example, when the speech recognition task is running, and another person in front of me is talking, the system does not pick up the speech well. I tried to set the AVAudioSession to be equally sensitive to all directions: private func configureAudioSession() { do { try audioSession.setCategory(.record, mode: .measurement) try audioSession.setActive(true) if #available(visionOS 1.0, *) { let availableDataSources = audioSession.availableInputs?.first?.dataSources if let omniDirectionalSource = availableDataSources?.first(where: {$0.preferredPolarPattern == .omnidirectional}) { try audioSession.setInputDataSource(omniDirectionalSource) } } } catch { print("Failed to set up audio session: \(error)") } } And here is how I set up the speech recognition and configure the microphone inputs: private func startSpeechRecognition(completion: @escaping (String) -> Void) { do { // Cancel the previous task if it's running. if let recognitionTask = recognitionTask { recognitionTask.cancel() self.recognitionTask = nil } // The AudioSession is already active, creating input node. let inputNode = audioEngine.inputNode try inputNode.setVoiceProcessingEnabled(false) // Create and configure the speech recognition request recognitionRequest = SFSpeechAudioBufferRecognitionRequest() guard let recognitionRequest = recognitionRequest else { fatalError("Unable to create a recognition request") } recognitionRequest.shouldReportPartialResults = true // Keep speech recognition data on device if #available(iOS 13, *) { recognitionRequest.requiresOnDeviceRecognition = true } // Create a recognition task for speech recognition session. // Keep a reference to the task so that it can be canceled. recognitionTask = speechRecognizer?.recognitionTask(with: recognitionRequest) { result, error in // var isFinal = false if let result = result { // Update the recognizedText completion(result.bestTranscription.formattedString) } else if let error = error { completion("Recognition error: \(error.localizedDescription)") } if error != nil || result?.isFinal == true { // Stop recognizing speech if there is a problem self.audioEngine.stop() inputNode.removeTap(onBus: 0) self.recognitionRequest = nil self.recognitionTask = nil } } // Configure the microphone input let recordingFormat = inputNode.outputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { (buffer, when) in self.recognitionRequest?.append(buffer) } audioEngine.prepare() try audioEngine.start() } catch { completion("Audio engine could not start: \(error.localizedDescription)") } }

Description: I am developing a recording-only application that supports background recording using AVAudioEngine. The app segments the recording into 60-second files for further processing. For example, a 10-minute recording results in ten 60-second files. Problem: The application functions as expected in the background. However, after the app receives an interruption (such as a phone call) and the interruption ends, I can successfully restart the recording. The problem arises when the app then transitions to the background; it fails to restart the recording. Specifically, after ending the call and transitioning the app to the background, the app encounters an error and is unable to restart AVAudioSession and AVAudioEngine. The only resolution is to close and restart the app, which is not ideal for user experience. Steps to Reproduce: 1. Start recording using AVAudioEngine. 2. The app records and saves 60-second segments. 3. Receive an interruption (e.g., an incoming phone call). 4. End the call. 5. Transition the app to the background. 6. Transition the app to the foreground and the session will be activated again. 7. Attempt to restart the recording. Expected Behavior: The app should resume recording seamlessly after the interruption and background transition. Actual Behavior: The app fails to restart AVAudioSession and AVAudioEngine, resulting in a continuous error. The recording cannot be resumed without closing and reopening the app. How I’m Starting the Recording: Configuration: internal func setAudioSessionCategory() { do { try audioSession.setCategory( .playAndRecord, mode: .default, options: [.defaultToSpeaker, .mixWithOthers, .allowBluetooth] ) } catch { debugPrint(error) } } internal func setAudioSessionActivation() { if UIApplication.shared.applicationState == .active { do { try audioSession.setPrefersNoInterruptionsFromSystemAlerts(true) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) if audioSession.isInputGainSettable { try audioSession.setInputGain(1.0) } try audioSession.setPreferredIOBufferDuration(0.01) try setBuiltInPreferredInput() } catch { debugPrint(error) } } } Starting AVAudioEngine: internal func setupEngine() { if callObserver.onCall() { return } inputNode = audioEngine.inputNode audioEngine.attach(audioMixer) audioEngine.connect(inputNode, to: audioMixer, format: AVAudioFormat.validInputAudioFormat(inputNode)) } internal func beginRecordingEngine() { audioMixer.removeTap(onBus: 0) audioMixer.installTap(onBus: 0, bufferSize: 1024, format: AVAudioFormat.validInputAudioFormat(inputNode)) { [weak self] buffer, _ in guard let self = self, let file = self.audioFile else { return } write(file, buffer: buffer) } audioEngine.prepare() do { try audioEngine.start() recordingTimer = Timer.scheduledTimer(withTimeInterval: recordingInterval, repeats: true) { [weak self] _ in self?.handleRecordingInterval() } } catch { debugPrint(error) } } On the try audioEngine.start() call, I receive error code 561145187 in the catch block. Logs/Error Messages: • Error code: 561145187 Request: I would appreciate any guidance or solutions to ensure the app can resume recording after interruptions and background transitions without requiring a restart. Thank you for your assistance.

I'm building an app that will allow users to record voice notes. The functionality of all that is working great; I'm trying to now implement changes to the AudioSession to manage possible audio streams from other apps. I want it so that if there is audio playing from a different app, and the user opens my app; the audio keep playing. When we start recording, any third party app audio should stop, and can then can resume again when we stop recording. This is my main audio setup code: private var audioEngine: AVAudioEngine! private var inputNode: AVAudioInputNode! func setupAudioEngine() { audioEngine = AVAudioEngine() inputNode = audioEngine.inputNode audioPlayerNode = AVAudioPlayerNode() audioEngine.attach(audioPlayerNode) let format = AVAudioFormat(standardFormatWithSampleRate: AUDIO_SESSION_SAMPLE_RATE, channels: 1) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: format) } private func setupAudioSession() { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .default, options: [.defaultToSpeaker, .allowBluetooth]) try audioSession.setPreferredSampleRate(AUDIO_SESSION_SAMPLE_RATE) try audioSession.setPreferredIOBufferDuration(0.005) // 5ms buffer for lower latency try audioSession.setActive(true) // Add observers setupInterruptionObserver() } catch { audioErrorMessage = "Failed to set up audio session: \(error)" } } This is all called upon app startup so we're ready to record whenever the user presses the record button. However, currently when this happens, any outside audio stops playing. I isolated the issue to this line: inputNode = audioEngine.inputNode When that's commented out, the audio will play -- but I obviously need this for recording functionality. Is this a bug? Expected behavior?

Hi there community, First and foremost, a big thank you to everyone who takes the time to read this. TL;DR: How, if even possible, can I record multiple audio streams simultaneously on an iOS application (iPad/iPhone)? I'm working on a recorder for the iPad to gather data for a machine learning project focused on speech recognition. Our goal is to capture extensive speech data, which requires recording from multiple microphones. Specifically, I need to record from all mics connected to our Scarlett 4i4 audio interface and, most importantly, also record from the built-in mic on the iPad or iPhone at the same time. As a newcomer to Swift development, I initially explored AVAudioRecorder. However, I quickly realized that it only supports one active audio node at a time, making multi-channel recording impossible. (perhaps you can proof me wrong, would make my day) Next, I transitioned to using AVAudioEngine, but encountered the same limitation: I couldn't manage to get input nodes for both the built-in mic and the Scarlett interface channels simultaneously. The application started behaving oddly, often resulting in identical audio data being recorded across all files. Determined to find a solution, I delved deeper into the Core Audio framework, specifically using Audio Toolbox. My approach involved creating and configuring multiple Audio Units, each corresponding to a different audio input device. Here's a brief overview of my current implementation: Listing Available Input Devices: I used AVAudioSession to enumerate all available input devices. Creating Audio Units: For each device, I created an Audio Unit and attempted to configure it for recording. Setting Up Callbacks: I set up input and output callbacks to handle the audio processing. Despite my efforts over the last few days, I haven't had much success. The callbacks for the Audio Units don't seem to be invoked correctly, and I'm struggling to achieve simultaneous multi-channel recording. Below is a snippet of my latest attempt: let audioUnitCallback: AURenderCallback = { ( inRefCon: UnsafeMutableRawPointer, ioActionFlags: UnsafeMutablePointer<AudioUnitRenderActionFlags>, inTimeStamp: UnsafePointer<AudioTimeStamp>, inBusNumber: UInt32, inNumberFrames: UInt32, ioData: UnsafeMutablePointer<AudioBufferList>? ) -> OSStatus in guard let ioData = ioData else { return noErr } print("Input callback invoked") let audioUnit = inRefCon.assumingMemoryBound(to: AudioUnit.self).pointee var bufferList = AudioBufferList( mNumberBuffers: 1, mBuffers: AudioBuffer( mNumberChannels: 1, mDataByteSize: 0, mData: nil ) ) let status = AudioUnitRender(audioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList) if status != noErr { print("AudioUnitRender failed: \(status)") return status } // Copy rendered data to output buffer let buffer = UnsafeMutableAudioBufferListPointer(ioData)[0] buffer.mData?.copyMemory(from: bufferList.mBuffers.mData!, byteCount: Int(bufferList.mBuffers.mDataByteSize)) buffer.mDataByteSize = bufferList.mBuffers.mDataByteSize print("Rendered audio data") return noErr } let outputCallback: AURenderCallback = { ( inRefCon: UnsafeMutableRawPointer, ioActionFlags: UnsafeMutablePointer<AudioUnitRenderActionFlags>, inTimeStamp: UnsafePointer<AudioTimeStamp>, inBusNumber: UInt32, inNumberFrames: UInt32, ioData: UnsafeMutablePointer<AudioBufferList>? ) -> OSStatus in guard let ioData = ioData else { return noErr } print("Output callback invoked") // Process the output data if needed return noErr } In essence, I'm stuck and in need of guidance. Has anyone here successfully implemented multi-channel recording on iOS, especially involving both built-in microphones and external audio interfaces? Any shared experiences, insights, or suggestions on how to proceed would be immensely appreciated. Thank you once again for your time and assistance!

We are a music app, encountered a scene, there is no way to resume playing music, so I would like to ask about the technical plan, how to achieve it. For example, when playing a video in another app, we pause the music playing and turn off the video, we should resume the music playing. Our code is implemented, so listen AVAudioSessionInterruptionNotification, when we received the notice and judge AVAudioSessionInterruptionOptionShouldResume, we play music came again, Error 560557684（AVAudioSessionErrorCodeCannotInterruptOthers） was reported. We were very confused NSError *error = nil; AVAudioSession *audioSession = [AVAudioSession sharedInstance]; [audioSession setCategory:AVAudioSessionCategoryPlayback withOptions:0 error:&error]; [audioSession setActive:YES error:&error]; We compared the apple music app and found that apple music can resume playing. Here is a video of the effects of our app: https://drive.google.com/file/d/1J94S2kxkEpNvG536yzCnKmE7IN3cGzIJ/view?usp=sharing Here's the apple music effect video: https://drive.google.com/file/d/1c1Kdgkn2nhy8SdDvRJAFF2sPvqJ8fL48/view?usp=sharing We want to improve our user experience. How can we do that?

Music app stops playing when switching to the background In apps that play music or music files, if you move to the home screen or run another app while the app is running, the music playback stops. Our app does not have the code to stop playing when switching to the background. We are guessing that some people experience this and others do not. We usually guide users to reboot their devices and try again. How can this phenomenon be improved in the code? Or is this a bug or error in the OS?

Hello, I am building a new iOS app which uses AVSpeechSynthesizer and should be able to mix audio nicely with audio from other apps. AVSpeechSynthesizer seems to handle setting the AVAudioSession to active on it's own, but does not deactivate the audio session. This leads to issues, namely that other audio sources remain "ducked" after AVSpeechSynthesizer is done speaking. I have implemented deactivating the audio session myself, which "works", in that it allows other audio sources to become "un-ducked", but it throws this exception each time even though it appears successful. Error Domain=NSOSStatusErrorDomain Code=560030580 "Session deactivation failed" UserInfo={NSLocalizedDescription=Session deactivation failed} It appears to be a bug with how AVSpeechSynthesizer handles activating/deactivating the audio session. Below is a minimal example which illustrates the problem. It has two buttons, one which manually deactivates the audio sessions, which throws the exception, but otherwise works, and another button which leaves audio session management to the AVSpeechSynthesizer but does not "un-duck" other audio. If you play some audio from another app (ex: Music), you'll see the button which throws/catches an exception successfully ducks/un-ducks the audio, while the one without attempting to deactivate the session ducks but does not un-duck the audio. import AVFoundation struct ContentView: View { let workingSynthesizer = UnduckingSpeechSynthesizer() let brokenSynthesizer = BrokenSpeechSynthesizer() init() { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playback, mode: .voicePrompt, options: [.duckOthers]) } catch { print("Setup error info: \(error)") } } var body: some View { VStack { Button("Works Correctly"){ workingSynthesizer.speak(text: "Hello planet") } Text("-------") Button("Does not work"){ brokenSynthesizer.speak(text: "Hello planet") } } .padding() } } class UnduckingSpeechSynthesizer: NSObject { var synth = AVSpeechSynthesizer() let audioSession = AVAudioSession.sharedInstance() override init(){ super.init() synth.delegate = self } func speak(text: String) { let utterance = AVSpeechUtterance(string: text) synth.speak(utterance) } } extension UnduckingSpeechSynthesizer: AVSpeechSynthesizerDelegate { func speechSynthesizer(_ synthesizer: AVSpeechSynthesizer, didFinish utterance: AVSpeechUtterance) { do { try audioSession.setActive(false, options: .notifyOthersOnDeactivation) } catch { // always throws an error // Error Domain=NSOSStatusErrorDomain Code=560030580 "Session deactivation failed" UserInfo={NSLocalizedDescription=Session deactivation failed} print("Deactivate error info: \(error)") } } } class BrokenSpeechSynthesizer { var synth = AVSpeechSynthesizer() let audioSession = AVAudioSession.sharedInstance() func speak(text: String) { let utterance = AVSpeechUtterance(string: text) synth.speak(utterance) } } (I have a separate issue where the first speech attempt takes a few seconds but I don't think it's related)