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'm working on adding Spatial Audio support to a game on the Mac. I'm looking at the SpatialAudioRenderer sample but having some issues. It's unclear to me when a device is compatible with Spatial Audio and when I should attempt to render Spatial Audio. There is no property that I can find on the Mac that advertises Spatial Audio compatibility on a device. The sample crashes when the output device is a USB device. This includes the Apple Studio Display. The Apple Studio Display is supposed to be capable of rendering Spatial Audio. The device doesn't work with the sample - do I still need to render down the 7.1.4 source on this device? The sample always renders down to Stereo, but the Apple Studio Display is not a Stereo device. I'm a bit confused by the sample and when/how I should configure the mixing unit.

Using an iPhone Pro 12 running iOS 26.0.1, with AirPods Pro 3. Camera app does capture video with what seems to be "Studio Quality Recording". Am trying to replicate that SQR with my own Camera like app, and while I can pull audio in from the APP3 mic, and my video capture app is recording a 48,000Hz high-bitrate video, the audio still sounds non-SQR. I'm seeing bluetoothA2DP , bluetoothLE , bluetoothHFP as portType, and not sure if SQR depends on one of those? Is there sample code demonstrating a SQR capture? Nevermind video and camera, just audio even? Also, I don't understand what SQR is doing between the APP3 and the iPhone. What codec is that? What bitrate is that? If I capture video using Capture and inspect the audio stream I see mono 74.14 kbit/s MPEG-4 AAC, 48000 Hz. But I assume that's been recompressed and not really giving me any insight into the APP3 H2 transmission?

I prefer to use the album fetched from the library instead of the catalog since this is faster. If doing so, how can I check if all tracks of an album are added to the library. In this case I'd like to fetch the catalog version or throw an error (for example when offline). Using .with(.tracks) on the library album fetches the tracks added to the library. The trackCount property is referring to the tracks that can be fetched from the library. The isComplete property is always nil when fetching from the library. One possible way is checking the trackNumber and discCount properties. However this only detects that not all tracks of an album are added to the library if there is a song not added ahead of one that is. I'd like to be able to handle this edge case as well. Is there currently a way to do this? I'd prefer to not rely on the apple music catalog for this since this is supposed to work offline as well. Fetching and storing all trackIDs when connected and later comparing against these would work, but this would potentially mean storing tens of thousands of track ids. Thank you

Environment→ ・Device: iPad 10th generation ・OS：**iOS18.3.2 We're using AVAudioPlayer to play a sound when a button is tapped. In our use case, this button can be tapped very frequently — roughly every 0.1 to 0.2 seconds. Each tap triggers the following function: var audioPlayer: AVAudioPlayer? func soundPlay(resource: String, type: String){ guard let path = Bundle.main.path(forResource: resource, ofType: type) else { return } do { audioPlayer = try AVAudioPlayer(contentsOf: URL(fileURLWithPath: path)) audioPlayer!.delegate = self try audioSession.setCategory(.playback) } catch { return } self.audioPlayer!.play() } The issue is that under high-frequency tapping (especially around 0.1–0.15s intervals), the app occasionally crashes. The crash does not occur every time, but it happens randomly — sometimes within 30 seconds, within 1 minute, or even 3 minutes of continuous tapping. Interestingly, adding a delay of 0.2 seconds between button taps seems to prevent the crash entirely. Delays shorter than 0.2 seconds (e.g.,0.15s,0.18s) still result in occasional crashes. My questions are: **Is this expected behavior from AVAudioPlayer or AVAudioSession? Could this be a known issue or a limitation in AVFoundation? Is there any documentation or guidance on handling frequent sound playback safely?** Any insights or recommendations on how to handle rapid, repeated audio playback more reliably would be appreciated.

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 have a music app I'm developing and having a weird issue where I can see now playing info for every other platform than tvOS. As far as I can tell I have correctly configured the MPNowPlayingInfoCenter MPNowPlayingInfoCenter.default().nowPlayingInfo = nowPlayingInfo MPNowPlayingInfoCenter.default().playbackState = .playing Are there any extra requirements to get my app's now-playing info showing in control center on tvOS? Another strange issue that might be related is I can use the apple TV remote to pause audio but not resume playback, so I feel like there's something I'm missing about registering audio playback on tvOS specifically.

I have a memory leak, when using AVAudioPlayer. I managed to narrow down the issue into a very simple app, which code I paste in at the end. The memory leak start immediately when I start playing sound, but only in the emylator. On the real iPhone there is no memory leak. The memory leak on the Simulator looks like this: import SwiftUI import AVFoundation struct ContentView_Audio: View { var sound: AVAudioPlayer? init() { guard let path = Bundle.main.path(forResource: "cd201", ofType: "mp3") else { return } let url = URL(fileURLWithPath: path) do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .default, options: [.mixWithOthers]) } catch { return } do { try AVAudioSession.sharedInstance().setActive(true) } catch { return } do { sound = try AVAudioPlayer(contentsOf: url) } catch { return } } var body: some View { HStack { Button { playSound() } label: { ZStack { Circle() .fill(.mint.opacity(0.3)) .frame(width: 44, height: 44) .shadow(radius: 8) Image(systemName: "play.fill") .resizable() .frame(width: 20, height: 20) } } .padding() Button { stopSound() } label: { ZStack { Circle() .fill(.mint.opacity(0.3)) .frame(width: 44, height: 44) .shadow(radius: 8) Image(systemName: "stop.fill") .resizable() .frame(width: 20, height: 20) } } .padding() } } private func playSound() { guard sound != nil else { return } sound?.volume = 1 // sound?.numberOfLoops = -1 sound?.play() } func stopSound() { sound?.stop() } }

There appears to be no method of going forward or backwards in Get Info in the Music application,

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

I have an app that displays artwork via MPMediaItem.artwork, requesting an image with a specific size. How do I get a media item's MPMediaItemAnimatedArtwork, and how to get the preview image and video to display to the user?

Not able to record audio in AAC format with 96 kHz sample rate using AVAudioRecorder or Extended Audio File services with 96 kHz input audio from input device. The audio recording settings used are let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatMPEG4AAC), AVSampleRateKey: sampleRate AVNumberOfChannelsKey: 1 AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue ] When tried using AVAudioEngine using AVAudioFile, AVAudioFile(forWriting: fileURL, // file extension .m4a settings: fileSettings, commonFormat: AVAudioCommonFormat.pcmFormatFloat32, interleaved: interleaved) else { return } got error CodecConverterFactory.cpp:977 unable to select compatible encoder sample rate AudioConverter.cpp:1017 Failed to create a new in process converter -> from 1 ch, 96000 Hz, Float32 to 1 ch, 96000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame, with status 1718449215

Hi all, I've developed an audio DSP application in C++ using AudioToolbox and CoreAudio on MacOS 14.4.1 with Xcode 15. I use an AudioQueue for input and another for output. This works great. I'm now adding real-time audio analysis eg spectral analysis. I want this to run independently of my audio processing so it can not interfere with audio playback. Taps on AudioQueues seem to be a good way of doing this... Since the analytics won't modify the audio data, I am using a Siphon Tap by setting the AudioQueueProcessingTapFlags to kAudioQueueProcessingTap_PreEffects | kAudioQueueProcessingTap_Siphon; This works fine on my output queue. However, on my input queue the Tap callback is called once and then a EXC_BAD_ACCESS occurs - screen shot below. NB: I believe that a callback should only call AudioQueueProcessingTapGetSourceAudio when not using a Siphon, so I don't call it. Relevant code: AudioQueueProcessingTapCallback tap_callback) { // Makes an audio tap for a queue void * tap_data_ptr = NULL; AudioQueueProcessingTapFlags tap_flags = kAudioQueueProcessingTap_PostEffects | kAudioQueueProcessingTap_Siphon; uint32_t max_frames = 0; AudioStreamBasicDescription asbd; AudioQueueProcessingTapRef tap_ref; OSStatus status = AudioQueueProcessingTapNew(queue_ref, tap_callback, tap_data_ptr, tap_flags, &max_frames, &asbd, &tap_ref); if (status != noErr) printf("Error while making Tap\n"); else printf("Successfully made tap\n"); } void tapper(void * tap_data, AudioQueueProcessingTapRef tap_ref, uint32_t number_of_frames_in, AudioTimeStamp * ts_ptr, AudioQueueProcessingTapFlags * tap_flags_ptr, uint32_t * number_of_frames_out_ptr, AudioBufferList * buf_list) { // Callback function for audio queue tap printf("Tap callback"); }``` Image of exception stack provided by Xcode: ![]("https://developer.apple.com/forums/content/attachment/27479e8d-a118-459b-aa2d-7e30528910e3" "title=Screenshot 2025-06-14 at 1.29.14 PM.png;width=932;height=562") What have I missed? Appreciate any help you learned folks may be able to provide. Best, Geoff.

My app encountered problems when trying to open an x86 audioUnit v2 on a Silicon Mac (although Rosetta is installed). There seems to be a XPC connection issue with the AUHostingService that I don't know how to fix. I observed other host apps opening the same plugins without problem, so there is probably something wrong or incompatible in my codes. I noticed that: The issue occurs whether or not the app is sandboxed. The issue does no longer occur when the app itself runs under Rosetta. There is no error reported by CoreAudio during allocation and initialization of the audio unit. The first notified errors appears when the unit calls AudioUnitRender from the rendering callback. With most x86 plugins, the error is on first call: kAudioUnitErr_RenderTimeout and on any subsequent call: kAudioComponentErr_InstanceInvalidated On the UI side, when the Cocoa View is loaded, it appears shortly, then disappears immediately leaving its superview empty. With another x86 plugin, the Cocoa View is loaded normally, but CoreAudio still emits kAudioUnitErr_NoConnection from AudioUnitRender, whether the view has been loaded or not, and the plugin produces no sound. I also find these messages in the console (printed in that order): CLIENT ERROR: RemoteAUv2ViewController does not override - and thus cannot react to catastrophic errors beyond logging them AUAudioUnit_XPC.mm:641 Crashed AU possible component description: aumu/Helm/Tyte My app uses the AUv2 API and I suspect that working with the AUv3 API would spare me these problems. However, considering how my audio system is built (audio units are wrapped into C++ classes and most connections between units are managed on the fly from the rendering callback), it would be a lot of work to convert, and I’m even not sure that all I do with the AUv2 API would be possible with the AUv3 API. I could possibly find an intermediate solution, but in the immediate future I'm looking for the simplest and fastest possible fix. If I cannot find better, I see two fallback options: In this part of the doc: “Beginning with macOS 11, the system loads audio units into a separate process that depends on the architecture or host preference”, does “host preference” means that it would be possible to disable the “out of process” behavior, for example from the app entitlements or info.plist? Otherwise, as a last resort, I could completely disable the use of x86 audioUnits when my app runs under ARM64, for at least making things cleaner. But the Audio Component API doesn’t give any info about the plugin architecture, how could I found it? Any tip or idea about this issue will be much appreciated. Thanks in advance!

Hi, I'm working on a project that uses the AVSpeechSynthesizer and AVSpeechUtterance. I discovered by chance that the AVSpeechSynthesizer automatically completes some words instead of just outputting what it's supposed to. These are abbreviations for days of the week or months, but not all of them. I don't want either of them automatically completed, but only the specified text. The completion transcends languages. I have written a short example program for demonstration purposes. import SwiftUI import AVFoundation import Foundation let synthesizer: AVSpeechSynthesizer = AVSpeechSynthesizer() struct ContentView: View { var body: some View { VStack { Button { utter("mon") } label: { Text("mon") } .buttonStyle(.borderedProminent) Button { utter("tue") } label: { Text("tue") } .buttonStyle(.borderedProminent) Button { utter("thu") } label: { Text("thu") } .buttonStyle(.borderedProminent) Button { utter("feb") } label: { Text("feb") } .buttonStyle(.borderedProminent) Button { utter("feb", lang: "de-DE") } label: { Text("feb DE") } .buttonStyle(.borderedProminent) Button { utter("wed") } label: { Text("wed") } .buttonStyle(.borderedProminent) } .padding() } private func utter(_ text: String, lang: String = "en-US") { let utterance = AVSpeechUtterance(string: text) let voice = AVSpeechSynthesisVoice(language: lang) utterance.voice = voice synthesizer.speak(utterance) } } #Preview { ContentView() } Thank you Christian

Environment Device: iPhone 16e iOS Version: 18.4.1 - 18.7.1 Framework: AVFoundation (AVAudioEngine) Problem Summary On iPhone 16e (iOS 18.4.1-18.7.1), the installTap callback stops being invoked after resuming from a phone call interruption. This issue is specific to phone call interruptions and does not occur on iPhone 14, iPhone SE 3, or earlier devices. Expected Behavior After a phone call interruption ends and audioEngine.start() is called, the previously installed tap should continue receiving audio buffers. Actual Behavior After resuming from phone call interruption: Tap callback is no longer invoked No audio data is captured No errors are thrown Engine appears to be running normally Note: Normal pause/resume (without phone call interruption) works correctly. Steps to Reproduce Start audio recording on iPhone 16e Receive or make a phone call (triggers AVAudioSession interruption) End the phone call Resume recording with audioEngine.start() Result: Tap callback is not invoked Tested devices: iPhone 16e (iOS 18.4.1-18.7.1): Issue reproduces ✗ iPhone 14 (iOS 18.x): Works correctly ✓ iPhone SE 3 (iOS 18.x): Works correctly ✓ Code Initial Setup (Works) let inputNode = audioEngine.inputNode inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in self.processAudioBuffer(buffer, at: time) } audioEngine.prepare() try audioEngine.start() Interruption Handling NotificationCenter.default.addObserver( forName: AVAudioSession.interruptionNotification, object: AVAudioSession.sharedInstance(), queue: nil ) { notification in guard let userInfo = notification.userInfo, let typeValue = userInfo[AVAudioSessionInterruptionTypeKey] as? UInt, let type = AVAudioSession.InterruptionType(rawValue: typeValue) else { return } if type == .began { self.audioEngine.pause() } else if type == .ended { try? self.audioSession.setActive(true) try? self.audioEngine.start() // Tap callback doesn't work after this on iPhone 16e } } Workaround Full engine restart is required on iPhone 16e: func resumeAfterInterruption() { audioEngine.stop() inputNode.removeTap(onBus: 0) inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in self.processAudioBuffer(buffer, at: time) } audioEngine.prepare() try audioSession.setActive(true) try audioEngine.start() } This works but adds latency and complexity compared to simple resume. Questions Is this expected behavior on iPhone 16e? What is the recommended way to handle phone call interruptions? Why does this only affect iPhone 16e and not iPhone 14 or SE 3? Any guidance would be appreciated!

Is there a way to permanently disable PHASE SDK logging? It seems to be a lot chattier than Apple's other SDKs. While developing a RealityKit app that uses AudioPlaybackController, I must manually hide the PHASE SDK log output several times each day so I can see my app's log messages. Thank you.

Hi! I am writing a browser extension that allows you to control the playback of media content on a music service website. Unfortunately Safari does not support tracking changes to the audible property in an event tabs.onUpdated. Is there an alternative to this event? I'm looking for a way to track when the automatic inference engine interrupts playback on a music service website. That you.

I've filed this as FB21446798 but figured I'd post here too. In the first build of macOS 26.3, playback via ApplicationMusicPlayer is completely broken. When starting playback of anything at all, the console shows the following error: applicationController: xpc service connection interrupted Failed to obtain remoteObject: Error Domain=NSCocoaErrorDomain Code=4099 "The connection to service created from an endpoint was invalidated from this process." UserInfo={NSDebugDescription=The connection to service created from an endpoint was invalidated from this process.} Failed to prepareToPlay with error: Error Domain=MPMusicPlayerControllerErrorDomain Code=10 "(null)" UserInfo={NSUnderlyingError=0xc92910ff0 {Error Domain=NSCocoaErrorDomain Code=4099 "The connection to service created from an endpoint was invalidated from this process." UserInfo={NSDebugDescription=The connection to service created from an endpoint was invalidated from this process.}}} In addition, several crash logs for RemotePlayerService are generated, showing my app as the parent process. This issue is 100% repeatable. No matter how I load the queue, whether it’s catalog or library content, any variation I can think of all fails like this. I really hope this can be fixed before 26.3 comes out, otherwise my app will be totally unusable. 😅

I need to implement a solution through an API or custom driver to completely block out the built-in speakers and microphone of Mac, because I need other apps to use specified external devices as audio input and output. Is there a way to achieve this requirement? What I mean is that even in system preferences, it should not be possible to choose the built-in microphone and speakers; only my external device can be used.