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 have a custom USB Audio Class 2 (UAC2) compatible device. When I connect this custom device to a MacBook with a configuration of up to 10 channels (16-bit), everything seems to work fine. However, when I increase the channel count to 12, the MacBook does not recognize the 12 channels. It only shows the channel count as 0. TN2274 is the only source where I found some information about Apple's Audio Class Drivers, but it doesn't mention any limitations regarding channel counts. Could you let me know the current limitations of the Audio Class Drivers on the latest macOS versions? What configuration should I use to get 12 channels working? P.S. I also found that a 12-channel, 8-bit configuration is detected by the MacBook, bit I want it to work with 16bits. For more detail please check FB17098863

According to the documentation (https://developer.apple.com/documentation/avfoundation/avplayeritem/externalmetadata), AVPlayerItem should have an externalMetadata property. However it does not appear to be visible to my app. When I try, I get: Value of type 'AVPlayerItem' has no member 'externalMetadata' Documentation states iOS 12.2+; I am building with a minimum deployment target of iOS 18. Code snippet: import Foundation import AVFoundation /// ... in function ... // create metadata as described in https://developer.apple.com/videos/play/wwdc2022/110338 var title = AVMutableMetadataItem() title.identifier = .commonIdentifierAlbumName title.value = "My Title" as NSString? title.extendedLanguageTag = "und" var playerItem = await AVPlayerItem(asset: composition) playerItem.externalMetadata = [ title ]

I'm developing the VisionOS app. I want to know how to play spatial audio in addition to RealityKit? If it's iOS or macOS, how to play spatial audio in addition to RealityKit?

I'm experiencing issues with audio playback in my React video player component specifically on iOS mobile devices (iPhone/iPad). Even after implementing several recommended solutions, including Apple's own guidelines, the audio still isn't working properly on iOS Safari. It works completely fine on Android. On iOS, I ensured the video doesn't autoplay (it requires user interaction). Here are all the details: Environment iOS Safari (latest version) React 18 TypeScript Video files: MP4 with AAC audio codec Current Implementation const VideoPlayer: React.FC<VideoPlayerProps> = ({ src, autoplay = true, }) => { const videoRef = useRef<HTMLVideoElement>(null); const isIOSDevice = isIOS(); // Custom iOS detection const [touchStartY, setTouchStartY] = useState<number | null>(null); const [touchStartTime, setTouchStartTime] = useState<number | null>(null); // Handle touch start event for gesture detection const handleTouchStart = (e: React.TouchEvent) => { setTouchStartY(e.touches[0].clientY); setTouchStartTime(Date.now()); }; // Handle touch end event with gesture validation const handleTouchEnd = (e: React.TouchEvent) => { if (touchStartY === null || touchStartTime === null) return; const touchEndY = e.changedTouches[0].clientY; const touchEndTime = Date.now(); // Validate if it's a legitimate tap (not a scroll) const verticalDistance = Math.abs(touchEndY - touchStartY); const touchDuration = touchEndTime - touchStartTime; // Only trigger for quick taps (< 200ms) with minimal vertical movement if (touchDuration < 200 && verticalDistance < 10) { handleVideoInteraction(e); } setTouchStartY(null); setTouchStartTime(null); }; // Simplified video interaction handler following Apple's guidelines const handleVideoInteraction = (e: React.MouseEvent | React.TouchEvent) => { console.log('Video interaction detected:', { type: e.type, timestamp: new Date().toISOString() }); // Ensure keyboard is dismissed (iOS requirement) if (document.activeElement instanceof HTMLElement) { document.activeElement.blur(); } e.stopPropagation(); const video = videoRef.current; if (!video || !video.paused) return; // Attempt playback in response to user gesture video.play().catch(err => console.error('Error playing video:', err)); }; // Effect to handle video source and initial state useEffect(() => { console.log('VideoPlayer props:', { src, loadingState }); setError(null); setLoadingState('initial'); setShowPlayButton(false); // Never show custom play button on iOS if (videoRef.current) { // Set crossOrigin attribute for CORS videoRef.current.crossOrigin = "anonymous"; if (autoplay && !hasPlayed && !isIOSDevice) { // Only autoplay on non-iOS devices dismissKeyboard(); setHasPlayed(true); } } }, [src, autoplay, hasPlayed, isIOSDevice]); return ( <Paper shadow="sm" radius="md" withBorder onClick={handleVideoInteraction} onTouchStart={handleTouchStart} onTouchEnd={handleTouchEnd} > <video ref={videoRef} autoPlay={!isIOSDevice && autoplay} playsInline controls crossOrigin="anonymous" preload="auto" onLoadedData={handleLoadedData} onLoadedMetadata={handleMetadataLoaded} onEnded={handleVideoEnd} onError={handleError} onPlay={dismissKeyboard} onClick={handleVideoInteraction} onTouchStart={handleTouchStart} onTouchEnd={handleTouchEnd} {...(!isFirefoxBrowser && { "x-webkit-airplay": "allow", "x-webkit-playsinline": true, "webkit-playsinline": true })} > <source src={videoSrc} type="video/mp4" /> </video> </Paper> ); }; Apple's Guidelines Implementation Removed custom play controls on iOS Using native video controls for user interaction Ensuring audio playback is triggered by user gesture Following Apple's audio session guidelines Properly handling the canplaythrough event Current Behavior Video plays but without sound on iOS mobile Mute/unmute button in native video controls doesn't work Audio works fine on desktop browsers and Android devices Videos are confirmed to have AAC audio codec No console errors related to audio playback User interaction doesn't trigger audio as expected Questions Are there any additional iOS-specific requirements I'm missing? Could this be related to iOS audio session handling? Are there known issues with React's handling of video elements on iOS? Should I be implementing additional audio context initialization? Any insights or suggestions would be greatly appreciated!

I'm working on adding CarPlay support to an audio app and I'd like to mimic the behavior of the Apple Music app on launch. Forgive me, but I think using Gherkin syntax here will help to best describe the desired behavior: GIVEN the Apple Music app is in a cold state (not launched or in memory) AND another audio app is actively playing audio WHEN I launch the Apple Music app from CarPlay THEN the Now Playing template is shown via a push AND the appropriate Now Playing info is shown AND the Now Playing button is shown on the tab bar AND the actively playing audio from another audio app is NOT interrupted The current behavior I see in my own app is that I can push on the Now Playing template and fill out the MPNowPlayingInfoCenter's info dictionary, but it won't render the info or show the Now Playing button on the tab bar until I start playing audio. Also, is there a way to hide the Now Playing button after the queue of content has finished playing? I'm able to pop the Now Playing template, but the Now Playing button is still present and tapping it will navigate the user to the now blank Now Playing template.

I often find when doing basic actions in MusicKit it is incredibly slow compared to Apple's Music App. I've tried different versions, devices, networks, Apple's sample code, it all throughout the last several years, and it is all the same. Does anyone else have this issue?

Hello Apple Developers, I am experiencing an issue where USB audio input (e.g., external USB microphone) is blocked when using AirPlay screen mirroring from my iPhone to a Mac or Apple TV. However, the built-in microphone continues to work without any problem. Issue Details: I am using an iPhone 15 (or latest device) running iOS [your iOS version]. I connect a USB audio interface/microphone via a USB-C adapter or Lightning adapter. The USB microphone works perfectly for audio input before starting AirPlay. The moment I enable AirPlay screen mirroring, the USB microphone stops working, and it disappears from available audio input sources. The built-in microphone continues to function, but I cannot use the USB microphone while mirroring. When I stop screen mirroring, the USB microphone immediately becomes available again. Expected Behavior: I would expect iOS to allow me to continue using an external USB microphone while mirroring my screen, just like it allows the built-in microphone to work. Questions: Is this an intentional restriction in iOS? Is there any workaround to enable USB audio input while using AirPlay screen mirroring? Is there a way to request a feature or configuration option to allow external USB microphones during AirPlay? I appreciate any insights or guidance from the Apple team or fellow developers. Thanks in advance! Best regards,

A month ago I was listening to music in Apple Music but then I saw a song, not anything weird, then I wanted to download it but it took a longer than usual and then an error apeared, I cant upload the picture of the text but it says, "The Song Name / Album / Artist 0% - Stopped (err = -12884)", I uninstall the app and then install it and it fixes, but when i want to enable Lossless Audio, the error appears, the error still aprearing until today and Im tired of uninstalling and installing the app so much times, can anyone help me?

I’m working on a macOS app, written in Swift. My goal is to record audio from an external microphone, e.g., one connected via USB. For this, I’m using an AVCaptureSession and recording its output with an AVAssetWriter. This works perfectly in principle (and reliably with internal microphones, for example). The problem occurs after my app has successfully completed the first recording and I then want to make additional recordings (which makes me think it might be process-dependent, because it works again after restarting the app). The problem: Noisy or distorted-sounding audio files. In addition, the following error message appears in the Console from CoreAudio / its AudioConverter: Input data proc returned inconsistent 512 packets for 2048 bytes; at 3 bytes per packet, that is actually 682 packets It is easy to reproduce. This problem is reproducible even if I don’t configure the AVAssetWriter manually and instead let it receive its audioSettings using a preset from an AVOutputSettingsAssistant. I’m running on macOS 15.0 (24A335). I’ve filed a feedback including a demo project → FB15333298 🎟️ I would greatly appreciate any help! Have a great day, Martin

I am developing an iOS app that needs to play spoken audio on demand from a server, while ducking the audio of background music from another app (e.g., SoundtrackYourBrand or Apple Music). This must work even when the app is in the background, and the server dictates when and what audio is played. Ideally, the message should be played within a minute of the server requesting it. Current Attempt & Observations I initially tried using Firebase Cloud Messaging (FCM) silent notifications to send a URL to an audio file, which the app would then play using AVPlayer. This works consistently when the app is active, but in the background, it only works about 60% of the time. In cases where it fails, iOS ducks the background music (e.g., from SoundtrackYourBrand) but never plays the spoken audio. Interestingly, when I play the audio without enabling audio ducking, it seems to work 100% of the time from my limited testing, even in the background. The app has background modes enabled for Audio, Background Fetch, and Remote Notifications. Best Approach to Achieve This? I’d like guidance on the best Apple-compliant approach to reliably play audio on command from the server, even when the app is in the background. Some possible paths: Ensuring the app remains active in the background – Are there recommended ways to prevent the app from getting suspended, such as background tasks, a special background mode, or a persistent connection to the server? Alternative triggering mechanisms – Would something like VoIP, Push-to-Talk, or another background service be better suited for this use case? Built-in iOS speech synthesis (AVSpeechSynthesizer) – If playing external audio is unreliable, would generating speech dynamically from text be a more robust approach? Streaming audio instead of sending a URL – Could continuous streaming from the server keep the app active and allow playback at the right moment? I want to ensure the solution is reliable and works 100% of the time when needed. Any recommendations on the best approach for this would be greatly appreciated. Thank you for your time and guidance.

I’m facing a problem while trying to achieve spatial audio effects in my iOS 18 app. I have tried several approaches to get good 3D audio, but the effect never felt good enough or it didn’t work at all. Also what mostly troubles me is I noticed that AirPods I have doesn’t recognize my app as one having spatial audio (in audio settings it shows "Spatial Audio Not Playing"). So i guess my app doesn't use spatial audio potential. First approach uses AVAudioEnviromentNode with AVAudioEngine. Chaining position of player as well as changing listener’s doesn’t seem to change anything in how audio plays. Here's simple how i initialize AVAudioEngine import Foundation import AVFoundation class AudioManager: ObservableObject { // important class variables var audioEngine: AVAudioEngine! var environmentNode: AVAudioEnvironmentNode! var playerNode: AVAudioPlayerNode! var audioFile: AVAudioFile? ... //Sound set up func setupAudio() { do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } audioEngine = AVAudioEngine() environmentNode = AVAudioEnvironmentNode() playerNode = AVAudioPlayerNode() audioEngine.attach(environmentNode) audioEngine.attach(playerNode) audioEngine.connect(playerNode, to: environmentNode, format: nil) audioEngine.connect(environmentNode, to: audioEngine.mainMixerNode, format: nil) environmentNode.listenerPosition = AVAudio3DPoint(x: 0, y: 0, z: 0) environmentNode.listenerAngularOrientation = AVAudio3DAngularOrientation(yaw: 0, pitch: 0, roll: 0) environmentNode.distanceAttenuationParameters.referenceDistance = 1.0 environmentNode.distanceAttenuationParameters.maximumDistance = 100.0 environmentNode.distanceAttenuationParameters.rolloffFactor = 2.0 // example.mp3 is mono sound guard let audioURL = Bundle.main.url(forResource: "example", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: audioURL) } catch { print("Failed to load audio file: \(error)") } } ... //Playing sound func playSpatialAudio(pan: Float ) { guard let audioFile = audioFile else { return } // left side playerNode.position = AVAudio3DPoint(x: pan, y: 0, z: 0) playerNode.scheduleFile(audioFile, at: nil, completionHandler: nil) do { try audioEngine.start() playerNode.play() } catch { print("Failed to start audio engine: \(error)") } ... } Second more complex approach using PHASE did better. I’ve made an exemplary app that allows players to move audio player in 3D space. I have added reverb, and sliders changing audio position up to 10 meters each direction from listener but audio seems to only really change left to right (x axis) - again I think it might be trouble with the app not being recognized as spatial. //Crucial class Variables: class PHASEAudioController: ObservableObject{ private var soundSourcePosition: simd_float4x4 = matrix_identity_float4x4 private var audioAsset: PHASESoundAsset! private let phaseEngine: PHASEEngine private let params = PHASEMixerParameters() private var soundSource: PHASESource private var phaseListener: PHASEListener! private var soundEventAsset: PHASESoundEventNodeAsset? // Initialization of PHASE init{ do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } // Init PHASE Engine phaseEngine = PHASEEngine(updateMode: .automatic) phaseEngine.defaultReverbPreset = .mediumHall phaseEngine.outputSpatializationMode = .automatic //nothing helps // Set listener position to (0,0,0) in World space let origin: simd_float4x4 = matrix_identity_float4x4 phaseListener = PHASEListener(engine: phaseEngine) phaseListener.transform = origin phaseListener.automaticHeadTrackingFlags = .orientation try! self.phaseEngine.rootObject.addChild(self.phaseListener) do{ try self.phaseEngine.start(); } catch { print("Could not start PHASE engine") } audioAsset = loadAudioAsset() // Create sound Source // Sphere soundSourcePosition.translate(z:3.0) let sphere = MDLMesh.newEllipsoid(withRadii: vector_float3(0.1,0.1,0.1), radialSegments: 14, verticalSegments: 14, geometryType: MDLGeometryType.triangles, inwardNormals: false, hemisphere: false, allocator: nil) let shape = PHASEShape(engine: phaseEngine, mesh: sphere) soundSource = PHASESource(engine: phaseEngine, shapes: [shape]) soundSource.transform = soundSourcePosition print(soundSourcePosition) do { try phaseEngine.rootObject.addChild(soundSource) } catch { print ("Failed to add a child object to the scene.") } let simpleModel = PHASEGeometricSpreadingDistanceModelParameters() simpleModel.rolloffFactor = rolloffFactor soundPipeline.distanceModelParameters = simpleModel let samplerNode = PHASESamplerNodeDefinition( soundAssetIdentifier: audioAsset.identifier, mixerDefinition: soundPipeline, identifier: audioAsset.identifier + "_SamplerNode") samplerNode.playbackMode = .looping do {soundEventAsset = try phaseEngine.assetRegistry.registerSoundEventAsset( rootNode: samplerNode, identifier: audioAsset.identifier + "_SoundEventAsset") } catch { print("Failed to register a sound event asset.") soundEventAsset = nil } } //Playing sound func playSound(){ // Fire new sound event with currently set properties guard let soundEventAsset else { return } params.addSpatialMixerParameters( identifier: soundPipeline.identifier, source: soundSource, listener: phaseListener) let soundEvent = try! PHASESoundEvent(engine: phaseEngine, assetIdentifier: soundEventAsset.identifier, mixerParameters: params) soundEvent.start(completion: nil) } ... } Also worth mentioning might be that I only own personal team account

I’ve been researching how to achieve a recording playback effect in iOS similar to the hands-free calling effect in the system’s phone app. How can this be implemented? I tried using the voice chat recording method, but found that the volume of the speaker output is too low. How should this issue be addressed? I couldn’t find a suitable API. Could you provide me with some documentation or sample code? Thank you.

Hi all, I have spent a lot of time reading the tech note and watching the WDDC video that introduce the PTTFramework on iOS. I currently have a custom setup where I am using AVAudioEngine to schedule and play buffers that are being streamed through a call. I am looking to use the PTTFramework to allow a user to trigger this push to talk behavior from the lock screen and the various places with the system UI it provides. However I am unsure what the correct behavior is regarding the handling of the audio session. Right now I am using .playback when there is no active voice transmission so that devices such as AirPods can be in AD2P mode where applicable, and then transitioning to .playbackAndRecord category only when the mic input should become active. Following this change in my AVAudioEngine manager I am then manually activating and deactivating the audio session manually when the engine is either playing/recording or idle. In the documentation it states that you should not attempt to activate or deactivate your audio session directly, but allow the framework to handle it. Does that mean that I need to either call the request to transmit delegate function or set an active participant on the channel manager first, and then wait for the didBecomeActive delegate method to trigger before I actually attempt to play or record any audio? (I am using the fullDuplex mode currently.) I noticed that that delegate method will only trigger if the audio session wasn't active before doing one of the above (setting active participant, requesting transmit). Lastly, when using the PTTFramework it also mentions that we get support for PTT devices and I notice on the didBeginTransmittingFrom property we have a handsfreeButton case. Is there any documentation or resources for what is actually supported out of the box for this? I am currently working on handling a lot of the push to talk through bluetooth LE, and wanted to make sure there wasn't overlap with what the system provides. Thank you!

When a tab goes to sleep, all its resources gets killed. When the user has any interaction it gets active again. However, if there is an audio in the tab, it does not play again even after user interaction. One has to reload/reopen the tab. Is this how it should work? Is there a fix for this?

I like to suggest a different microphone dot icon for Voice Control. I had customized Voice Control to turn on a Flashlight. This caused confusion with the orange dot being switched on constantly. I made an error in sending a security vulnerability to Apple Security about the orange dot microphone in always ON mode when iPhone is unlocked.

Looking for help on getting "On Tap" to work inside RCP for my AVP project. I can get it to work when using "on added to scene" but if I switch to "on tap", the audio will not play when attaching the audio to an entity in my scene. I'm using the same entity for the tap gesture that the audio is using for the emitter. Here is my work flow for the "on added to scene" that works correctly to help troubleshoot my non working "on tap". Behaviors: "on added to scene". action - timeline Input target: check mark enabled, allowed all Collision set to default Audio library: source mp3 file Chanel Audio: resource mp3 file above Timeline: Play Audio with mp3 file added This set up in RCP allows my AVP project to launch correctly with audio "on added to scene". But when switching behaviors to "on tap", the audio will no longer play and I can not figure out why. I've tried several different options and nothing works. Please help!

I followed this guide, and added com.apple.developer.spatial-audio.profile-access as an entitlement to the app (via the + Capability button – Spatial Audio Profile). I have a audio graph that outputs to AVAudioEngine. However, the Xcode Cloud build ended up with this error: Invalid Code Signing Entitlements. Your application bundle's signature contains code signing entitlements that are not supported on iOS. Specifically, key 'com.apple.developer.spatial-audio.profile-access' in 'Payload/…' is not supported. This guide says it's available on iOS. Does it mean not on iOS 17? In which case how can I provide fallback for iOS 17?

Hello! I used the Apple CA Playthrough example code that pipes audio between devices. It uses AudioUnit callbacks to pipe the input to an output device, and I created a system equalizer with it - however users reported it stopped working in macOS 15. I am getting the error HALPlugIn.cpp:552 HALPlugIn::DeviceGetCurrentTime: got an error from the plug-in routine, Error: 1937010544 (stop) for the output device and no sound coming out of the speakers. The error only occurs when using a virtual device as an input, not using the microphone. First I thought the problem was in the loopback driver, but it also does not work with other loopback drivers like Blackhole. STEPS TO REPRODUCE Install a virtual device, for example "brew install blackhole-2ch" and run the CAPlayThrough example code (you need to add Mic Permission in the info.plist). Then set your system audio output to the virtual device, select the device as input in CAPlayThrough and hit start. You should see the error in console. My question: What did change in macOS 15 that could cause this? Is it something with the new permission handling maybe?

private var audioEngine = AVAudioEngine() private var inputNode: AVAudioInputNode! func startAnalyzing() { inputNode = audioEngine.inputNode let recordingFormat = inputNode.outputFormat(forBus: 0) let hardwareSampleRate = recordingSession.sampleRate inputNode.removeTap(onBus: 0) if recordingFormat.sampleRate != hardwareSampleRate { print("。") let newFormat = AVAudioFormat(commonFormat: recordingFormat.commonFormat, sampleRate: hardwareSampleRate, channels: recordingFormat.channelCount, interleaved: recordingFormat.isInterleaved) inputNode.installTap(onBus: 0, bufferSize: 1024, format: newFormat) { buffer, time in self.processAudioBuffer(buffer, time: time) } } else { inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { buffer, time in self.processAudioBuffer(buffer, time: time) } } do { audioEngine.prepare() try audioEngine.start() } catch { print(": \(error)") } } I back the app to the background and then call startAnalyzing(), which reports an error and the background recording permissions are configured。 error： [10429:570139] [aurioc] AURemoteIO.cpp:1668 AUIOClient_StartIO failed (561145187) [10429:570139] [avae] AVAEInternal.h:109 [AVAudioEngineGraph.mm:1545:Start: (err = PerformCommand(*ioNode, kAUStartIO, NULL, 0)): error 561145187 Audio engine couldn't start. Is background boot not allowed？