Hi All, I am working on a DJ playout app (MACOS). The app has a few AVAudioPlayerNode's combined with the ApplicationMusicPlayer from Musickit. I can route the output of the AVaudioPlayer to a hardware device so that the audio files are directed to their own dedicated output on my Mac. The ApplicationMusicPlayer is following the default output and this is pretty annoying. Has anyone found a solution to chain the ApplicationMusicPlayer and get it set to a output device? Thanks Pancras

I am having issues deploying my iOS app, that uses ShazamKit, to get working on a Mac with Apple silicon. When uploading the archive to App Store Connect I do get ITMS-90863: Macs with Apple silicon support issue - The app links with libraries that aren’t present in macOS: /usr/lib/swift/libswiftShazamKit.dylib Is ShazamKit not supported for iOS apps that can run on Macs with Apple silicon? Or is there something I should fix in my setup / deployment?

I'm encountering errors while using AVAudioEngine with voice processing enabled (setVoiceProcessingEnabled(true)) in scenarios where the input and output audio devices are not the same. This issue arises specifically with mismatched devices, preventing the application from functioning as expected. Works: Paired devices (e.g., MacBook Pro mic → MacBook Pro speakers) Fails: Mismatched devices (e.g., AirPods mic → MacBook Pro speakers) When using paired input and output devices: The setup works as expected. Example: MacBook Pro microphone → MacBook Pro speakers. When using mismatched devices: AVAudioEngine setup fails during aggregate device construction. Example: AirPods microphone → MacBook Pro speakers. Error logs indicate a channel count mismatch. Here are the partial logs. Due to the content limit, I cannot post the entire logs. AUVPAggregate.cpp:1000 client-side input and output formats do not match (err=-10875) AUVPAggregate.cpp:1036 err=-10875 AVAEInternal.h:109 [AVAudioEngineGraph.mm:1344:Initialize: (err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)): error -10875 AggregateDevice.mm:329 Failed expectation of constructed aggregate (312): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (312): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (336): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (336): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AudioHardware-mac-imp.cpp:3484 AudioDeviceSetProperty: no device with given ID AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (348): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (348): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) Is it possible to use voice processing with different input/output devices? If yes, are there any specific configurations required to handle mismatched devices? How can we resolve channel count mismatch errors during aggregate device construction? Are there settings or API adjustments to enforce compatibility between input/output devices? Are there any workarounds or alternative approaches to achieve voice processing functionality with mismatched devices? For instance, can we force an intermediate channel configuration or downmix input/output formats?

Please include the line below in follow-up emails for this request. Case-ID: 11089799 When using AVSpeechUtterance and setting it to play in Mandarin, if Siri is set to Cantonese on iOS 18, it will be played in Cantonese. There is no such issue on iOS 17 and 16. 1.let utterance = AVSpeechUtterance(string: textView.text) let voice = AVSpeechSynthesisVoice(language: "zh-CN") utterance.voice = voice 2.In the phone settings, Siri is set to Cantonese

Please Update Andorid MusicKit，the version 1.1.2 will complied fail。the error msg：•SDKUriHandlerActivity>. Apps targeting Android 12 and higher are required to specify an explicit value for android:exported when the corres

Hi all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!

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

Hi, I would like to use macro-mode for the custom camera using AVCaptureDevice in my project. This feature might help to automatically adjust and switch between lenses to get a close up clear image. It looks like this feature is not available and there are no open apis to achieve macro mode from Apple. Is there a way to get this functionality in the custom camera without losing the image quality. Please let me know if this is possible. Thanks you, Adil Thamarasseri

I have used AVQueuePlayer in my music app to play sequence of audios from a remote server, this how I have defined things my player in my ViewModel Variables private var cancellables = Set() private let audioSession = AVAudioSession.sharedInstance() private var avQueuePlayer: AVQueuePlayer? @Published var playbackSpeed: Float = 1.0 before starting playback, I am making sure that audio session is set properly, the code snippet used for that is do { try audioSession.setCategory(.playback, mode: .default, options: []) try audioSession.setActive(true, options: []) } catch { return } and this is the function I am using to update playback speed func updatePlaybackSpeed(_ newSpeed: Float){ if newSpeed > 0.0, newSpeed <= 2.0{ playbackSpeed = newSpeed avQueuePlayer?.rate = newSpeed print("requested speed is (newSpeed) and actual speed is (String(describing: avQueuePlayer?.rate))") } } sometimes whatever speed is set, player seems to play at the same speed as it was set, e.g. Once I got "requested speed is 1.5 and actual speed is 1.5", and player also seemed to play at the speed of 1.5 but another time I got "requested speed is 2.0 and actual speed is 2.0", but player still seemed to play at the speed of 1.0 to observe changes in rate, I used this **private func observeRateChanges() { guard let avQueuePlayer = self.avQueuePlayer else { return } NotificationCenter.default.publisher(for: AVQueuePlayer.rateDidChangeNotification, object: avQueuePlayer) .compactMap { $0.userInfo?[AVPlayer.rateDidChangeReasonKey] as? AVPlayer.RateDidChangeReason } .sink { reason in switch reason { case .appBackgrounded: print("The app transitioned to the background.") case .audioSessionInterrupted: print("The system interrupts the app’s audio session.") case .setRateCalled: print("The app set the player’s rate.") case .setRateFailed: print("An attempt to change the player’s rate failed.") default: break } } .store(in: &cancellables) }** when rate was set properly, I got this "The app set the player’s rate." from the above function, but when it wasn't, I got this "An attempt to change the player’s rate failed.," now I am not able to understand why rate is not being set, and if it gave "requested speed is 2.0 and actual speed is 2.0" from updatePlaybackSpeed function, why does the player seems to play with the speed of 1.0?

I am trying to use AVAudioEngine for recording and playing for a voice chat kind of app, but when the speaker plays any audio while recording, the recording take the speaker audio as input. I want to filter that out. Are there any suggestions for the swift code

In our Apple TV application, we use the native AVPlayer for live playback functionality. Until tvOS 17.6 and during the tvOS 18 beta, the Pause/Resume feature worked as expected, allowing us to pause live playback. However, after updating to tvOS 18.1, the pause functionality no longer works. The same app still works fine on tvOS 17, but on tvOS 18, attempting to pause live playback has no effect. We reviewed the tvOS 18 release notes but couldn't find any relevant changes or deprecations related to AVPlayer or live playback behavior. Has there been any change in the handling of live playback or the Pause/Resume functionality in tvOS 18.1? Any guidance or suggestions to address this issue would be greatly appreciated. Thank you!

Hi all, we try migrate project to Swift 6 Project use AVPlayer in MainActor Selection audio and subtitiles not work Task { @MainActor in let group = try await item.asset.loadMediaSelectionGroup(for: AVMediaCharacteristic.audible) get error: Non-sendable type 'AVMediaSelectionGroup?' returned by implicitly asynchronous call to nonisolated function cannot cross actor boundary and second example `if #available(iOS 15.0, *) { player?.currentItem?.asset.loadMediaSelectionGroup(for: AVMediaCharacteristic.audible, completionHandler: { group, error in if error != nil { return } if let groupWrp = group { DispatchQueue.main.async { self.setupAudio(groupWrp, audio: audioLang) } } }) }` get error: Sending 'groupWrp' risks causing data races

Hello. I am attempting to display the music inside of my app in Now Playing. I've tried a few different methods and keep running into unknown issues. I'm new to Objective-C and Apple development so I'm at a loss of how to continue. Currently, I have an external call to viewDidLoad upon initialization. Then, when I'm ready to play the music, I call playMusic. I have it hardcoded to play an mp3 called "1". I believe I have all the signing set up as the music plays after I exit the app. However, there is nothing in Now Playing. There are no errors or issues that I can see while the app is running. This is the only file I have in Xcode relating to this feature. Please let me know where I'm going wrong or if there is another object I need to use! #import <Foundation/Foundation.h> #import <UIKit/UIKit.h> #import <MediaPlayer/MediaPlayer.h> #import <AVFoundation/AVFoundation.h> @interface ViewController : UIViewController <AVAudioPlayerDelegate> @property (nonatomic, strong) AVPlayer *player; @property (nonatomic, strong) MPRemoteCommandCenter *commandCenter; @property (nonatomic, strong) MPMusicPlayerController *controller; @property (nonatomic, strong) MPNowPlayingSession *nowPlayingSession; @end @implementation ViewController - (void)viewDidLoad { [super viewDidLoad]; NSLog(@"viewDidLoad started."); [self setupAudioSession]; [self initializePlayer]; [self createNowPlayingSession]; [self configureNowPlayingInfo]; NSLog(@"viewDidLoad completed."); } - (void)setupAudioSession { AVAudioSession *audioSession = [AVAudioSession sharedInstance]; NSError *setCategoryError = nil; if (![audioSession setCategory:AVAudioSessionCategoryPlayback error:&setCategoryError]) { NSLog(@"Error setting category: %@", [setCategoryError localizedDescription]); } else { NSLog(@"Audio session category set."); } NSError *activationError = nil; if (![audioSession setActive:YES error:&activationError]) { NSLog(@"Error activating audio session: %@", [activationError localizedDescription]); } else { NSLog(@"Audio session activated."); } } - (void)initializePlayer { NSString *soundFilePath = [NSString stringWithFormat:@"%@/base/game/%@",[[NSBundle mainBundle] resourcePath], @"bgm/1.mp3"]; if (!soundFilePath) { NSLog(@"Audio file not found."); return; } NSURL *soundFileURL = [NSURL fileURLWithPath:soundFilePath]; self.player = [AVPlayer playerWithURL:soundFileURL]; NSLog(@"Player initialized with URL: %@", soundFileURL); } - (void)createNowPlayingSession { self.nowPlayingSession = [[MPNowPlayingSession alloc] initWithPlayers:@[self.player]]; NSLog(@"Now Playing Session created with players: %@", self.nowPlayingSession.players); } - (void)configureNowPlayingInfo { MPNowPlayingInfoCenter *infoCenter = [MPNowPlayingInfoCenter defaultCenter]; CMTime duration = self.player.currentItem.duration; Float64 durationSeconds = CMTimeGetSeconds(duration); CMTime currentTime = self.player.currentTime; Float64 currentTimeSeconds = CMTimeGetSeconds(currentTime); NSDictionary *nowPlayingInfo = @{ MPMediaItemPropertyTitle: @"Example Title", MPMediaItemPropertyArtist: @"Example Artist", MPMediaItemPropertyPlaybackDuration: @(durationSeconds), MPNowPlayingInfoPropertyElapsedPlaybackTime: @(currentTimeSeconds), MPNowPlayingInfoPropertyPlaybackRate: @(self.player.rate) }; infoCenter.nowPlayingInfo = nowPlayingInfo; NSLog(@"Now Playing info configured: %@", nowPlayingInfo); } - (void)playMusic { [self.player play]; [self createNowPlayingSession]; [self configureNowPlayingInfo]; } - (void)pauseMusic { [self.player pause]; [self configureNowPlayingInfo]; } @end

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! I get personal recommendations MusicItemCollection using this code: func getRecommendations() async throws -> MusicItemCollection<MusicPersonalRecommendation> { let request = MusicPersonalRecommendationsRequest() let response = try await request.response() let recommendations = response.recommendations return recommendations } However, all recommendations contain no more than 12 MusicItem's, while the Music.app application provides much more for some recommendations, for example, for the You recently listened recommendation, the Music.app application displays 40 items. Each recommendation has an items property that contains a collection of musical items MusicItemCollection<MusicPersonalRecommendation.Item>, the hasNextBatch property for these collections is always false. I expected that for some collections loading of new items would be available. Please tell me if I'm doing something wrong or is this a MusicKit bug? Thank you!

使用AVSpeechUtterance实现iOS语音播报，选择语言为简体中文“zh-CN”，读取中文“袆”（hui 第一声)错误，读成了“祎”（yi 第一声），希望能优化。

Our multimedia application Boinx FotoMagico displays media files of various kinds with a Metal rendering engine. At the moment we still use .bgra8Unorm pixel format and sRGB color space and only render in SDR, which is increasingly a problem, as much of the video content is HDR nowadays (e.g. videos shot on an iPhone). For that reason we would like to switch to EDR rendering with .rgba16Float pixel format and extendedLinearDisplayP3 color space. We have already worked out how to do this for HDR image files, but still have a technical problem when rendering HDR video files. We are using AVFoundation to get the video frames as CVPixelBuffers and convert them to MTLTexture using a CVMetalTextureCache. MTLTextures are then further processed in various compute shaders before being rendered to screen. However the pixel values in the texture are not what we expected. Video frames appear too bright/overexposed. In WWDC21 session "Explore HDR rendering with EDR" Ken Greenebaum mentioned: “AVFoundation does not presently decode HDR formats, such as HDR10, to EDR. Consequently, these need to be adapted for use with EDR rendering. This conversion is straightforward and involves two steps. First, converting to linear light by applying the inverse transfer function. And second, dividing by the medium's reference white.” https://developer.apple.com/videos/play/wwdc2021/10161?time=1498 However, the session does not explain, how to get or calculate the correct value for "reference white". We could not find any relevant info on the web. This is why we need DTS assistance. We need the code that calculates the correct value for reference white for any kind of video, whether it is SDR or HDR, and regardless of codec and encoding. I assume that Ken Greenebaum is the best Apple engineer to ask in this case, because he recorded most of the EDR related WWDC sessions in recent years? We have written a small test app that renders a short sample video (HLG encoding). The window contains two views. The upper view uses an AVPlayerLayer and renders the video natively just like QuickTime Player. The video content looks correct here. BTW, the window background is SDR white, so that bright EDR pixels can be clearly identified, e.g. the clouds just above the mountains in the upper left corner of the sample video. You may need to lower display brightness a bit if these clouds do not appear brighter than the white window background. The bottom view uses a CAMetalLayer and low-level Metal rendering. The CVPixelBuffers we receive from AVFoundation still need to be scaled down so that SDR reference white reaches pixel value 1.0. Entering a value of 9.0 to 10.0 for reference white in the text field makes it look about right on my Studio Display. But that is just experimental for this sample video file. We need code to calculate the correct value for reference white for any kind of video file! We have a couple of questions: SDR videos should probably use 1.0 as reference white, as their encoded pixel values can already be used as is? Is this assumption correct? Different video encoding of HDR video (HLG, PQ, etc) will probably lead to different values for reference white? Is the value for reference white constant throughout a video, or can it vary over time, either scene by scene, or even frame by frame? If it can vary, does the CVPixelBuffer of the current video frame contain all the necessary metadata to calculate the correct value? Does the NSScreen.maximumExtendedDynamicRangeColorComponentValue also influence the reference white value? The attached sample project is structured in a way that the only piece of code that needs to be modified is the ViewController.sdrReferenceWhiteValue() function. Please read the comments and the #warning in this function. This is where the code for calculating the reference white value should be inserted. Here is the download link for the sample project: https://www.dropbox.com/scl/fi/4w5gmftav5xhbixu9u6pb/HDRMetalTest.zip?rlkey=n8cm02soux3rx03vplgo6h1lm&dl=0

We have a Push To Talk application which allow user to record video and audio. When user is recording a video using AVCaptureSession and receive's an Push To Talk call, from moment the Push To Talk call is received the audio in the video which is being captured is stopped while the video capture is still in progress. Here after the PTT call is completed, we have tried restarting the audio session, there are no errors that are getting printed but we still don't see the audio getting restarted in video capture. We have also tried to add a new input for AVCaptureSession we are receiving error that is resulting in video capture stopping, error mentioned below: [OS-PLT] [CameraManager] Movie file finished with error: Error Domain=AVFoundationErrorDomain Code=-11818 "Recording Stopped" UserInfo={AVErrorRecordingSuccessfullyFinishedKey=true, NSLocalizedDescription=Recording Stopped, NSLocalizedRecoverySuggestion=Stop any other actions using the recording device and try again., AVErrorRecordingFailureDomainKey=1, NSUnderlyingError=0x3026bff60 {Error Domain=NSOSStatusErrorDomain Code=-16414 "(null)"}}, success We have also raised a Feedback Ticket on same: https://feedbackassistant.apple.com/feedback/16050598

Hello All, It seems that it's "very easy" (😬) to implement a little Swift code inside the prepared AU using Xcode 16.2 on Sequoia 15.1.1 and a Mac Studio M1 Ultra, but my issue is that I finally don't know... where. The documentation says that I've to find the AudioUnitViewController.swift file and then modify the render block : audioUnit.renderBlock = { (numFrames, ioData) in // Process audio here } in the Xcode project automatically generated, but I didn't find such a file... If somebody can help me in showing where is the file to be modified, I'll be very grateful ! Thank you very much. J

Hi I'm new to the forum, I'm planning an app just for Apple watch, I would like to use bluetooth audio in background, how can I do it? The messages I send via bluetooth stop as soon as the watch display turns off. Thank you! Nax