Context: I am currently developing an app using the Push-to-Talk (PTT) framework. I have reviewed both the PTT framework documentation and the CallKit demo project to better understand how to properly manage audio session activation and AVAudioEngine setup. I am not activating the audio session manually. The audio session configuration is handled in the incomingPushResult or didBeginTransmitting callbacks from the PTChannelManagerDelegate. I am using a single AVAudioEngine instance for both input and playback. The engine is started in the didActivate callback from the PTChannelManagerDelegate. When I receive a push in full duplex mode, I set the active participant to the user who is speaking. Issue When I attempt to talk while the other participant is already speaking, my input tap on the input node takes a few seconds to return valid PCM audio data. Initially, it returns an empty PCM audio block. Details: The audio session is already active and configured with .playAndRecord. The input tap is already installed when the engine is started. When I talk from a neutral state (no one is speaking), the system plays the standard "microphone activation" tone, which covers this initial delay. However, this does not happen when I am already receiving audio. Assumptions / Current Setup Because the audio session is active in play and record, I assumed that microphone input would be available immediately, even while receiving audio. However, there seems to be a delay before valid input is delivered to the tap, only occurring when switching from a receive state to simultaneously talking. Questions Is this expected behavior when using the PTT framework in full duplex mode with a shared AVAudioEngine? Should I be restarting or reconfiguring the engine or audio session when beginning to talk while receiving audio? Is there a recommended pattern for managing microphone readiness in this scenario to avoid the initial empty PCM buffer? Would using separate engines for input and output improve responsiveness? I would like to confirm the correct approach to handling simultaneous talk and receive in full duplex mode using PTT framework and AVAudioEngine. Specifically, I need guidance on ensuring the microphone is ready to capture audio immediately without the delay seen in my current implementation. Relevant Code Snippets Engine Setup func setup() { let input = audioEngine.inputNode do { try input.setVoiceProcessingEnabled(true) } catch { print("Could not enable voice processing \(error)") return } input.isVoiceProcessingAGCEnabled = false let output = audioEngine.outputNode let mainMixer = audioEngine.mainMixerNode audioEngine.connect(pttPlayerNode, to: mainMixer, format: outputFormat) audioEngine.connect(beepNode, to: mainMixer, format: outputFormat) audioEngine.connect(mainMixer, to: output, format: outputFormat) // Initialize converters converter = AVAudioConverter(from: inputFormat, to: outputFormat)! f32ToInt16Converter = AVAudioConverter(from: outputFormat, to: inputFormat)! audioEngine.prepare() } Input Tap Installation func installTap() { guard AudioHandler.shared.checkMicrophonePermission() else { print("Microphone not granted for recording") return } guard !isInputTapped else { print("[AudioEngine] Input is already tapped!") return } let input = audioEngine.inputNode let microphoneFormat = input.inputFormat(forBus: 0) let microphoneDownsampler = AVAudioConverter(from: microphoneFormat, to: outputFormat)! let desiredFormat = outputFormat let inputFramesNeeded = AVAudioFrameCount((Double(OpusCodec.DECODED_PACKET_NUM_SAMPLES) * microphoneFormat.sampleRate) / desiredFormat.sampleRate) input.installTap(onBus: 0, bufferSize: inputFramesNeeded, format: input.inputFormat(forBus: 0)) { [weak self] buffer, when in guard let self = self else { return } // Output buffer: 1920 frames at 16kHz guard let outputBuffer = AVAudioPCMBuffer(pcmFormat: desiredFormat, frameCapacity: AVAudioFrameCount(OpusCodec.DECODED_PACKET_NUM_SAMPLES)) else { return } outputBuffer.frameLength = outputBuffer.frameCapacity let inputBlock: AVAudioConverterInputBlock = { inNumPackets, outStatus in outStatus.pointee = .haveData return buffer } var error: NSError? let converterResult = microphoneDownsampler.convert(to: outputBuffer, error: &error, withInputFrom: inputBlock) if converterResult != .haveData { DebugLogger.shared.print("Downsample error \(converterResult)") } else { self.handleDownsampledBuffer(outputBuffer) } } isInputTapped = true }

I noticed that AVSampleBufferDisplayLayerContentLayer is not released when the AVSampleBufferDisplayLayer is removed and released. It is possible to reproduce the issue with the simple code: import AVFoundation import UIKit class ViewController: UIViewController { var displayBufferLayer: AVSampleBufferDisplayLayer? override func viewDidLoad() { super.viewDidLoad() let displayBufferLayer = AVSampleBufferDisplayLayer() displayBufferLayer.videoGravity = .resizeAspectFill displayBufferLayer.frame = view.bounds view.layer.insertSublayer(displayBufferLayer, at: 0) self.displayBufferLayer = displayBufferLayer DispatchQueue.main.asyncAfter(deadline: .now() + 1) { self.displayBufferLayer?.flush() self.displayBufferLayer?.removeFromSuperlayer() self.displayBufferLayer = nil } } } In my real project I have mutliple AVSampleBufferDisplayLayer created and removed in different view controllers, this is problematic because the amount of leaked AVSampleBufferDisplayLayerContentLayer keeps increasing. I wonder that maybe I should use a pool of AVSampleBufferDisplayLayer and reuse them, however I'm slightly afraid that this can also lead to strange bugs. Edit: It doesn't cause leaks on iOS 18 device but leaks on iPad Pro, iOS 17.5.1

This is an issue with the Insta360 Flow Pro 2. My iOS app uses DockKit to control the gimbal; in particular, my app disables tracking and sends angular velocity commands to control the gimbal's orientation. I only try to modify the yaw (rotation around the vertical axis); never the pitch or yaw. Note that I don't send the gimbal to a particular orientation directly; I modify the velocity. Everything works great for a long period of time: typically for a continuous run of 4-6 hours; in the most recent case, I managed about 36 hours of continous operation before the following problem occurred. I came back to check on the system, and because no visual activity had occurred in the camera's field of view for a while, the phone had commanded the gimbal to rotate back to a yaw angle of 0 degrees. So the phone in the gimbal should have been looking straight ahead (i.e. the 0 degree yaw position), but it was definitely looking off at an angle. I've seen this twice now. The first time, when it should have been looking straight ahead, it was in fact looking 60 degrees off center. This time (caught on video, see below), it was off by 22 degrees from center. Here's the weird part: the gimbal reports this way off center positioning as zero degrees (well close enough to zero, like 0.2 or something that's fine). But, mechanically, the gimbal still knows where zero degrees is: if we double click on the trigger of the Flow Pro 2, which is supposed to reset the gimbal to 0 degrees yaw and pitch, the gimbal responds correctly and reorients to a 0 degree position. However, the yaw values it reports are not zero, but as shown in my video, 22 degrees off axis or so. Power cycling the gimbal and restarting immediately fixes the problem. Also, I switched from my app to the Insta360 app, which caused the phone to flip from landscape to portrait, then when I returned to my app and switched back to landscape, the gimbal now started reporting correct yaw angles. Is there a possibility this is a bug in the DockKit framework? Has anyone seen this? I have a case open with Insta360, but although it's clearly a software issue, it's not clear if it's in Insta360's code or the DockKit layer. Any ideas for how I can get out of this mode? My concern is that the phone is in a tripod about 10' off the floor, and not very accessible. Also, if all goes well, we may have about 50 of these systems running, and having to fix them one by one after a few hours is not good. For a demonstration of this bug, see the following video: https://octoparry.com/offset.MOV Any help greatly appreciated.

Can i use iokit usb lib to disable build-in camera?

Hi everyone, After updating my Apple TV HD (model A1625) to tvOS 26, I’ve noticed a significant spike in CPU usage—up to 3× higher than before the update. Go from around 40% to 120% Model: Apple TV HD (A1625) tvOS Version: 26 (stable release) and beta version of 26.1, App downgrade stream due to lack of cpu power If anyone else is experiencing this, please share your findings or workarounds. Would love to hear from Apple engineers or other developers if this is a known regression or if there’s a recommended fix. Thanks!

Does anyone have a template of an Apple Projected Media Profile Format Description or a File of a Stereo wideFOV video? Use case I have 2 compatible cameras that I stereo sync and I want to move the projection information from the compatible video to the Spatial video that combines them. Every version I can come up with crashes the AVP and when viewing as Spatial in Tahoe I just get a black screen.

Our streaming app uses FairPlay-protected video streams, which previously worked fine when using AVAssetResourceLoaderDelegate to provide CKCs. Recently, we migrated to AVContentKeySession, and while everything works as expected during regular playback, we encountered an issue with AirPlay. Our CKC has a 120-second expiry, so we renew it by calling renewExpiringResponseData.. This trigger the didProvideRenewingContentKeyRequest delegate and we respond with updated CKC. However, when streaming via AirPlay, both video and audio freeze exactly after 120 seconds. To validate the issue, I tested with AVAssetResourceLoaderDelegate and found that I can reproduce the same freeze if I do not renew the key. This suggests that AirPlay is not accepting the renewed CKC when using AVContentKeySession. Additional Details: This issue occurs across different iOS versions and various AirPlay devices. The same content plays without issues when played directly on the device. The renewal process is successful, and segments continue to load, but playback remains frozen. Tried renewing the CKC bit early (100s). I also tried setting player.usesExternalPlaybackWhileExternalScreenIsActive = true, but the issue persists. We don't use persistentKey. Is there anything else that needs to be considered for proper key renewal when AirPlaying? Any help on how to fix this or confirmation if this is a known issue would be greatly appreciated.

I donate INPlayMediaIntent to systerm（donate success）, but not show in control center My code is as follows let mediaItems = mediaItems.map { $0.inMediaItem } let intent = if #available(iOS 13.0, *) { INPlayMediaIntent(mediaItems: mediaItems, mediaContainer: nil, playShuffled: false, playbackRepeatMode: .none, resumePlayback: true, playbackQueueLocation: .now, playbackSpeed: nil, mediaSearch: nil) } else { INPlayMediaIntent(mediaItems: mediaItems, mediaContainer: nil, playShuffled: false, playbackRepeatMode: .none, resumePlayback: true) } intent.suggestedInvocationPhrase = "播放音乐" let interaction = INInteraction(intent: intent, response: nil) interaction.donate { error in if let error = error { print("Intent 捐赠失败: \(error.localizedDescription)") } else { print("Intent 捐赠成功 ✅") } }

Hello, Basically, I am reading and writing an asset. To simplify, I am just reading the asset and rewriting it into an output video without any modifications. However, I want to add a fade-out effect to the last three seconds of the output video. I don’t know how to do this. So far, before adding the CMSampleBuffer to the output video, I tried reducing its volume using an extension on CMSampleBuffer. In the extension, I passed 0.4 for testing, aiming to reduce the video's overall volume by 60%. My question is: How can I directly adjust the volume of a CMSampleBuffer? Here is the extension: extension CMSampleBuffer { func adjustVolume(by factor: Float) -> CMSampleBuffer? { guard let blockBuffer = CMSampleBufferGetDataBuffer(self) else { return nil } var length = 0 var dataPointer: UnsafeMutablePointer<Int8>? guard CMBlockBufferGetDataPointer(blockBuffer, atOffset: 0, lengthAtOffsetOut: nil, totalLengthOut: &length, dataPointerOut: &dataPointer) == kCMBlockBufferNoErr else { return nil } guard let dataPointer = dataPointer else { return nil } let sampleCount = length / MemoryLayout<Int16>.size dataPointer.withMemoryRebound(to: Int16.self, capacity: sampleCount) { pointer in for i in 0..<sampleCount { let sample = Float(pointer[i]) pointer[i] = Int16(sample * factor) } } return self } }

Hello, I have been running into issues with setting nowPlayingInfo information, specifically updating information for CarPlay and the CPNowPlayingTemplate. When I start playback for an item, I see lock screen information update as expected, along with the CarPlay now playing information. However, the playing items are books with collections of tracks. When I select a new track(chapter) within the book, I set the MPMediaItemPropertyTitle to the new chapter name. This change is reflected correctly on the lock screen, but almost never appears correctly on the CarPlay CPNowPlayingTemplate. The previous chapter title remains set and never updates. I see "Application exceeded audio metadata throttle limit." in the debug console fairly frequently. From that a I figured that I need to minimize updates to the nowPlayingInfo dictionary. What I did: I store the metadata dictionary in a local dictionary and only set values in the main nowPlayingInfo dictionary when they are different from the current value. I kick off the nowPlayingInfo update via a task that initially sleeps for around 2 seconds (not a final value, just for my current testing). If a previous Task is active, it gets cancelled, so that only one update can happen within that time window. Neither of these things have been sufficient. I can switch between different titles entirely and the information updates (including cover art). But when I switch chapters within a title, the MPMediaItemPropertyTitle continues to get dropped. I know the value is getting set, because it updates on the lock screen correctly. In total, I have 12 keys I update for info, though with the above changes, usually 2-4 of them actually get updated with high frequency. I am running out of ideas to satisfy the throttling thresholds to accurately display metadata. I could use some advice. Thanks.

So I'm using AVAudioEngine. When playing audio I become the 'now playing' app using MPNowPlayingInfoCenter/MPRemoteCommandCenter APIs. When configuring MPRemoteCommandCenter I add a play/pause command target via -addTargetWithHandler on the togglePlayPauseCommand property. Now I also have a play/pause button in my app's UI. When I pause playback from my app's UI (which means I'm the active app, I'm in the foreground), what I do is this: -I pause the AVAudioPlayerNode I'm using with AVAudioEngine. I do not, stop, reset, etc. the AVAudioEngine. I only pause the player node. My thought process here is that the user just pressed pause and it is very likely that he will hit 'play' to resume playback in the near future because My app is in the foreground and the user just hit the pause button. Now if my app moves to the background and if I receive a memory warning I presume it'd make sense to tear down the engine or pause it. Perhaps I'm wrong about this? So when I initially hit the play button from my app's UI I also activate my AVAudioSession. I do this in high priority NSOperation since the documentation warns that "we recommend that applications not activate their session from a thread where a long blocking operation will be problematic." So now I'm playing and I hit pause from my app's UI. Then I quickly bring up the "Now Playing" center and I see I'm the "Now Playing" app but the play-pause button is showing the pause icon instead of the play icon but I'm in the pause state. I do set MPNowPlayingInfoCenter's playbackState to MPNowPlayingPlaybackStatePaused when I pause. Not surprisingly this doesn't work. The documentation states this is for macOS only. So the only way to get MPRemoteCommandCenter to show the "play" image for the play-pause button is to deactivate my AVAudioSession when I pause playback? Since I change the active state of my audio session in a NSOperation because documentation recommends "we recommend that applications not activate their session from a thread where a long blocking operation will be problematic." the play-pause toggle in the remote command center won't immediately update since I'm doing it on another thread. IMO it feels kind of inappropriate for a play-pause button to wait on a NSOperation activating the audio session before updating its UI when I already know my play/paused state, it should update right away like the button in my app does. Wouldn't it be nicer to just use MPNowPlayingInfoCenter's playbackState property on iOS too? If I'm no the longer the now playing app/active audio session it doesn't matter since I'm not in the now playing UI, just ignore it? Also is it recommended that I deactivate my audio session explicitly every time the user pauses audio in my app (when I'm in the foreground)? Also when I do deactivate the audio session I get an error: AVAudioSessionErrorCodeIsBusy (but the button in the now playing center updates to the proper image). I do this : -(void)pause { [self.playerNode pause]; [self runOperationToDeactivateAudioSession]; // This does nothing on iOS: MPNowPlayingInfoCenter *nowPlayingCenter = [MPNowPlayingInfoCenter defaultCenter]; nowPlayingCenter.playbackState = MPNowPlayingPlaybackStatePaused; } So in -runOperationToDeactivateAudioSession I get the AVAudioSessionErrorCodeIsBusy. According to the documentation Starting in iOS 8, if the session has running I/Os at the time that deactivation is requested, the session will be deactivated, but the method will return NO and populate the NSError with the code property set to AVAudioSessionErrorCodeIsBusy to indicate the misuse of the API. So pausing the player node when pausing isn't enough to meet the deactivation criteria. I guess I have to pause or stop the audio engine. I could probably wait until I receive a scene went to background notification or something before deactivating my audio session (which is async, so the button may not update to the correct image in time). This seems like a lot of code to have to write to get a play-pause toggle to update, especially in iPad-multi window scene environment. What's the recommended approach? Should I pause the AudioEngine instead of the player node always? Should I always explicitly deactivate my audio session when the user pauses playback from my app's UI even if I'm in the foreground? I personally like the idea of just being able to set [MPNowPlayingInfoCenter defaultCenter].playbackState = MPNowPlayingPlaybackStatePaused; But maybe that's because that would just make things easier on me. This does feels overcomplicated though. If anyone can share some tips on how I should handle this, I'd appreciate it.

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 tried SpeechTranscriber with a lot of my devices (from iPhone 12 series ~ iPhone 17 series) without issues. However, SpeechTranscriber.isAvailable value is false for my iPhone 11 Pro. https://developer.apple.com/documentation/speech/speechtranscriber/isavailable I'am curious why the iPhone 11 Pro device is not supported. Are all iPhone 11 series not supported intentionally? Or is there any problem with my specific device? I've also checked the supportedLocales, and the value is an empty array. https://developer.apple.com/documentation/speech/speechtranscriber/supportedlocales

In the AVP project, a selector pops up, only wanting to filter spatial videos. When selecting the material of one of the spatial videos, the selection result returns empty. How can we obtain the video selected by the user and get the path and the URL of the file The code is as follows： PhotosPicker(selection: $selectedItem, matching: .videos) { Text("Choose a spatial photo or video") } func loadTransferable(from imageSelection: PhotosPickerItem) -> Progress { return imageSelection.loadTransferable(type: URL.self) { result in DispatchQueue.main.async { // guard imageSelection == self.imageSelection else { return } print("加载成功的图片集合:(result)") switch result { case .success(let url?): self.selectSpatialVideoURL = url print("获取视频链接:(url)") case .success(nil): break // Handle the success case with an empty value. case .failure(let error): print("spatial错误：(error)") // Handle the failure case with the provided error. } } } }

My app is properly configured with MusicKit. I've generated a JWT using my valid credentials (Team ID, Key ID, private key), and I’ve ensured the time settings are correct via NTP. When I call: https://api.music.apple.com/v1/catalog/jp/search?term=ado&amp;types=songs I consistently receive a 500 Internal Server Error. The JWT is generated using ES256 with valid iat and exp values. I’ve confirmed the token decodes properly using jwt.io, and it's passed via the Authorization: Bearer header. Things I’ve confirmed: Key ID, Team ID, private key are correct App ID is configured with MusicKit capability JWT is generated and signed correctly macOS time is synced via NTP Used both curl and Python to test — same result Is there anything else I should check on the Apple Developer Console (like App ID, Certificates, or provisioning profile)? Or could this be a backend issue on Apple’s side? Any guidance would be appreciated.

Hi everyone, I am currently on MacOS Tahoe (26.1), and for some weird reason my mac is not connecting via HDMI. To be accurate: it is connecting and the LG TV shows up in the Displays settings, but no image shows up in it, I have no idea why. This used to work as I've tried this cable before with the same exact tv. The cable is a basic Amazon Basics HDMI one. Allow me just to advanced this question a little: usually terminal commands are more advanced recommendations, whereas basic questions like "have you connected it right" are just a waste of time

I am trying to get access to raw audio samples from mic. I've written a simple example application that writes the values to a text file. Below is my sample application. All the input samples from the buffers connected to the input tap is zero. What am I doing wrong? I did add the Privacy - Microphone Usage Description key to my application target properties and I am allowing microphone access when the application launches. I do find it strange that I have to provide permission every time even though in Settings > Privacy, my application is listed as one of the applications allowed to access the microphone. class AudioRecorder { private let audioEngine = AVAudioEngine() private var fileHandle: FileHandle? func startRecording() { let inputNode = audioEngine.inputNode let audioFormat: AVAudioFormat #if os(iOS) let hardwareSampleRate = AVAudioSession.sharedInstance().sampleRate audioFormat = AVAudioFormat(standardFormatWithSampleRate: hardwareSampleRate, channels: 1)! #elseif os(macOS) audioFormat = inputNode.inputFormat(forBus: 0) // Use input node's current format #endif setupTextFile() inputNode.installTap(onBus: 0, bufferSize: 1024, format: audioFormat) { [weak self] buffer, _ in self!.processAudioBuffer(buffer: buffer) } do { try audioEngine.start() print("Recording started with format: \(audioFormat)") } catch { print("Failed to start audio engine: \(error.localizedDescription)") } } func stopRecording() { audioEngine.stop() audioEngine.inputNode.removeTap(onBus: 0) print("Recording stopped.") } private func setupTextFile() { let tempDir = FileManager.default.temporaryDirectory let textFileURL = tempDir.appendingPathComponent("audioData.txt") FileManager.default.createFile(atPath: textFileURL.path, contents: nil, attributes: nil) fileHandle = try? FileHandle(forWritingTo: textFileURL) } private func processAudioBuffer(buffer: AVAudioPCMBuffer) { guard let channelData = buffer.floatChannelData else { return } let channelSamples = channelData[0] let frameLength = Int(buffer.frameLength) var textData = "" var allZero = true for i in 0..<frameLength { let sample = channelSamples[i] if sample != 0 { allZero = false } textData += "\(sample)\n" } if allZero { print("Got \(frameLength) worth of audio data on \(buffer.stride) channels. All data is zero.") } else { print("Got \(frameLength) worth of audio data on \(buffer.stride) channels.") } // Write to file if let data = textData.data(using: .utf8) { fileHandle!.write(data) } } }

I was trying to set custom audio output device for a generated audio on macCatalyst. While using let status = AudioUnitSetProperty(outputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &outputDeviceID, UInt32(MemoryLayout.size)) kAudioOutputUnitProperty_CurrentDevice is invalid, and status = -10879, indicating an error. STEPS TO REPRODUCE Set Run Destination to MacOS and run the program. "AudioUnitSetProperty: 0" should be printed, indicating it works fine. Set Run Destination to Mac Catalyst and run the program. "Error setting output device: -10879" should be printed, indicating an error.

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!

Hey there, I just upgraded to Mac OS Tahoe ,son an apple MacBook Pro 2019 16inch. am using IntellijIDEA and Flutter to develop a mobile app which I test on the simulator app running iOS 18.4 . the issue: when I start the simulator app. ( while in the loading phase and in the operation phase as well ), the audio from an already open YouTube tab on safari (this happens on chrome browser as well). the sound glitches and becomes Noise. a fix I found online is to kill the audio deamon on Mac OS, This works using the command: "sudo killall coreaudiod" this kills the audio process, (while the emulator is operational), then the macOS restarts the audio deamon then the audio works fine alongside with the simulator being open. I just want to ask is there a permanent fix for this? is Apple working on a fix for this in the upcoming update?