Hi, I'm trying to setup a AVAudioEngine for USB Audio recording and monitoring playthrough. As soon as I try to setup playthough I get an error in the console: AVAEInternal.h:83 required condition is false: [AVAudioEngineGraph.mm:1361:Initialize: (IsFormatSampleRateAndChannelCountValid(outputHWFormat))] Any ideas how to fix it? // Input-Device setzen try? setupInputDevice(deviceID: inputDevice) let input = audioEngine.inputNode // Stereo-Format erzwingen let inputHWFormat = input.inputFormat(forBus: 0) let stereoFormat = AVAudioFormat(commonFormat: inputHWFormat.commonFormat, sampleRate: inputHWFormat.sampleRate, channels: 2, interleaved: inputHWFormat.isInterleaved) guard let format = stereoFormat else { throw AudioError.deviceSetupFailed(-1) } print("Input format: \(inputHWFormat)") print("Forced stereo format: \(format)") audioEngine.attach(monitorMixer) audioEngine.connect(input, to: monitorMixer, format: format) // MonitorMixer -> MainMixer (Output) // Problem here, format: format also breaks. audioEngine.connect(monitorMixer, to: audioEngine.mainMixerNode, format: nil)

I have a PCM audio buffer (AVAudioPCMFormatInt16). When I try to play it using AVPlayerNode / AVAudioEngine an exception is thrown: "[[busArray objectAtIndexedSubscript:(NSUInteger)element] setFormat:format error:&nsErr]: returned false, error Error Domain=NSOSStatusErrorDomain Code=-10868 (related thread https://forums.developer.apple.com/forums/thread/700497?answerId=780530022#780530022) If I convert the buffer to AVAudioPCMFormatFloat32 playback works. My questions are: Does AVAudioEngine / AVPlayerNode require AVAudioPCMBuffer to be in the Float32 format? Is there a way I can configure it to accept another format instead for my application? If 1 is YES is this documented anywhere? If 1 is YES is this required format subject to change at any point? Thanks! I was looking to watch the "AVAudioEngine in Practice" session video from WWDC 2014 but I can't find it anywhere (https://forums.developer.apple.com/forums/thread/747008).

I have an iPadOS M-processor application with two different running configurations. In config1, the shared AVAudioSession is configured for .videoChat mode using the built-in microphone. The input/output nodes of the AVAudioEngine are configured with voice processing enabled. The built-in mic is formatted for 1 channel at 48KHz. In config2, the shared AVAudioSession is configured for .measurement mode using an external USB microphone. The input/output nodes of the AVAudioEngine are configured with voice processing disabled. The external mic is formatted for 2 channels at 44.1KHz I've written a configuration manager designed to safely switch between these two configurations. It works by stopping AVAudioEngine and detaching all but the input and output nodes, updating the shared audio session for the desired mic and sample-rates, and setting the appropriate state for voice processing to either true or false as required by the configuration. Finally the new audio graph is constructed by attaching appropriate nodes, connecting them, and re-starting AVAudioEngine I'm experiencing what I believe is a race-condition between switching voice processing on or off and then trying to re-build and start the new audio graph. Even though notifications, which are dumped to the console indicate that my requested input and sample-rate settings are in place, I crash when trying to start the audio engine because the sample-rate is wrong. Investigating further it looks like the switch from remote I/O to voice-processing I/O or vice-versa has not yet actually completed. I introduced a 100ms second delay and that seems to help but is obviously not a reliable way to build software that must work consistently. How can I make sure that what are apparently asynchronous configuration changes to the shared audio session and the input/output nodes have completed before I go on? I tried using route change notifications from the shared AVAudioSession but these lie. They say my preferred mic input and sample-rate setting is in place but when I dump the AVAudioEngine graph to the debugger console, I still see the wrong sample rate assigned to the input/output nodes. Also these are the wrong AU nodes. That is, VPIO is still in place when RIO should be, or vice-versa. How can I make the switch reliable without arbitrary time delays? Is my configuration manager approach appropriate (question for Apple engineers)?

Two issues: No matter what I set in try audioSession.setPreferredSampleRate(x) the sample rate on both iOS and macOS is always 48000 when the output goes through the speaker, and 24000 when my Airpods connect to an iPhone/iPad. Now, I'm checking the current output loudness to animate a 3D character, using mixerNode.installTap(onBus: 0, bufferSize: y, format: nil) { [weak self] buffer, time in Task { @MainActor in // calculate rms and animate character accordingly but any buffer size under 4800 is just ignored and the buffers I get are 4800 sized. This is ok, when the sampleRate is currently 48000, as 10 samples per second lead to decent visual results. But when AirPods connect, the samplerate is 24000, which means only 5 samples per second, so the character animation looks lame. My AVAudioEngine setup is the following: audioEngine.connect(playerNode, to: pitchShiftEffect, format: format) audioEngine.connect(pitchShiftEffect, to: mixerNode, format: format) audioEngine.connect(mixerNode, to: audioEngine.outputNode, format: nil) Now, I'd be fine if the outputNode runs at whatever if it needs, as long as my tap would get at least 10 samples per second. PS: Specifying my favorite format in the let format = AVAudioFormat(standardFormatWithSampleRate: 48_000, channels: 2)! mixerNode.installTap(onBus: 0, bufferSize: y, format: format) doesn't change anything either

Hi 👋! We have a SpriteKit-based app where we play AVAudio sounds in three different ways: Effects (incl. UI sounds) with AVAudioPlayer. Long looping tracks with AVAudioPlayer. Short animation effects on the timeline of SpriteKit's SKScene files (effectively SKAudioNode nodes). We've found that when you exit the app or otherwise interrupt audio plays, future audio plays often fail. For example, there's a WebKit-based video trailer inside the app, and if you play it, our looping background music track (2.) will stop playing, and won't resume as you close the trailer (return from WebKit). This is probably due to us not manually restarting the track (so may well be easily fixed). Periodically played AVAudioPlayer audio (1.) are not affected. However, the more concerning thing is that the audio tracks on SKScene file timelines (3.) will no longer play. My hypothesis is that AVAudioEngine gets interrupted, and needs to be restarted for those AVAudioNode elements to regain functionality. Thing is, we don't deal with AVAudioEngine at all currently in the app, meaning it is never initiated to begin with. Obviously things return to normal when you remove the app from short-term memory and restart it. However, it seems many of our users aren't doing this, and often report audio failing presumably due to some interruption in the past without the app ever being cleared from memory. Any idea why timeline-run SKAudioNodes would fail like this? Should the app react to app backgrounding/foregrounding regarding audio? Any help would be very much appreciated ✌️!

I'm using a 4 channel USB Audio interface, with 4 microphones, and want to process them through 4 independent effect chains. However the output from AVAudioInputNode is a single 4 channel bus. How can I split this into 4 mono busses? The following code splits the input into 4 copies, and routes them through the effects, but each bus contains all four channels. How can I remap the channels to remove the unwanted channels from the bus? I tried using channelMap on the mixer node but that had no effect. I'm currently using this code primarily on iOS but it should be portable between iOS and MacOS. It would be possible to do this through a Matrix Mixer Node, but that seems completely overkill, for such a basic operation. I'm already using a Matrix Mixer to combine the inputs, and it's not well supported in AVAudioEngine. AVAudioInputNode *inputNode=[engine inputNode]; [inputNode setVoiceProcessingEnabled:NO error:nil]; NSMutableArray *micDestinations=[NSMutableArray arrayWithCapacity:trackCount]; for(i=0;i<trackCount;i++) { fixMicFormat[i]=[AVAudioMixerNode new]; [engine attachNode:fixMicFormat[i]]; // And create reverb/compressor and eq the same way... [engine connect:reverb[i] to:matrixMixerNode fromBus:0 toBus:i format:nil]; [engine connect:eq[i] to:reverb[i] fromBus:0 toBus:0 format:nil]; [engine connect:compressor[i] to:eq[i] fromBus:0 toBus:0 format:nil]; [engine connect:fixMicFormat[i] to:compressor[i] fromBus:0 toBus:0 format:nil]; [micDestinations addObject:[[AVAudioConnectionPoint alloc] initWithNode:fixMicFormat[i] bus:0] ]; } AVAudioFormat *inputFormat = [inputNode outputFormatForBus: 1]; [engine connect:inputNode toConnectionPoints:micDestinations fromBus:1 format:inputFormat];

According to the header file the outputVolume properties supported range is 0.0-1.0: /*! @property outputVolume @abstract The mixer's output volume. @discussion This accesses the mixer's output volume (0.0-1.0, inclusive). @property (nonatomic) float outputVolume; However when setting the volume to 2.0 the audio does indeed play louder. Is the header file out of date and if so, what is the supported range for outputVolume? Thanks

I have a simple AVAudioEngine graph as follows: AVAudioPlayerNode -> AVAudioUnitEQ -> AVAudioUnitTimePitch -> AVAudioUnitReverb -> Main mixer node of AVAudioEngine. I noticed that whenever I have AVAudioUnitTimePitch or AVAudioUnitVarispeed in the graph, I noticed a very distinct crackling/popping sound in my Airpods Pro 2 when starting up the engine and playing the AVAudioPlayerNode and unable to find the reason why this is happening. When I remove the node, the crackling completely goes away. How do I fix this problem since i need the user to be able to control the pitch and rate of the audio during playback. import AVKit @Observable @MainActor class AudioEngineManager { nonisolated private let engine = AVAudioEngine() private let playerNode = AVAudioPlayerNode() private let reverb = AVAudioUnitReverb() private let pitch = AVAudioUnitTimePitch() private let eq = AVAudioUnitEQ(numberOfBands: 10) private var audioFile: AVAudioFile? private var fadePlayPauseTask: Task<Void, Error>? private var playPauseCurrentFadeTime: Double = 0 init() { setupAudioEngine() } private func setupAudioEngine() { guard let url = Bundle.main.url(forResource: "Song name goes here", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: url) } catch { print("Failed to load audio file: \(error)") return } reverb.loadFactoryPreset(.mediumHall) reverb.wetDryMix = 50 pitch.pitch = 0 // Increase pitch by 500 cents (5 semitones) engine.attach(playerNode) engine.attach(pitch) engine.attach(reverb) engine.attach(eq) // Connect: player -> pitch -> reverb -> output engine.connect(playerNode, to: eq, format: audioFile?.processingFormat) engine.connect(eq, to: pitch, format: audioFile?.processingFormat) engine.connect(pitch, to: reverb, format: audioFile?.processingFormat) engine.connect(reverb, to: engine.mainMixerNode, format: audioFile?.processingFormat) } func prepare() { guard let audioFile else { return } playerNode.scheduleFile(audioFile, at: nil) } func play() { DispatchQueue.global().async { [weak self] in guard let self else { return } engine.prepare() try? engine.start() DispatchQueue.main.async { [weak self] in guard let self else { return } playerNode.play() fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: true) // Ramp up volume until 1 is reached if volume >= 1 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 1 } } } } func pause() { fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: false) // Ramp down volume until 0 is reached if volume <= 0 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 0 playerNode.pause() // Shut down engine once ramp down completes DispatchQueue.global().async { [weak self] in guard let self else { return } engine.pause() } } } private func updateVolume(for x: Double, rising: Bool) -> Float { if rising { // Fade in return Float(pow(x, 2) * (3.0 - 2.0 * (x))) } else { // Fade out return Float(1 - (pow(x, 2) * (3.0 - 2.0 * (x)))) } } func setPitch(_ value: Float) { pitch.pitch = value } func setReverbMix(_ value: Float) { reverb.wetDryMix = value } } struct ContentView: View { @State private var audioManager = AudioEngineManager() @State private var pitch: Float = 0 @State private var reverb: Float = 0 var body: some View { VStack(spacing: 20) { Text("🎵 Audio Player with Reverb & Pitch") .font(.title2) HStack { Button("Prepare") { audioManager.prepare() } Button("Play") { audioManager.play() } .padding() .background(Color.green) .foregroundColor(.white) .cornerRadius(10) Button("Pause") { audioManager.pause() } .padding() .background(Color.red) .foregroundColor(.white) .cornerRadius(10) } VStack { Text("Pitch: \(Int(pitch)) cents") Slider(value: $pitch, in: -2400...2400, step: 100) { _ in audioManager.setPitch(pitch) } } VStack { Text("Reverb Mix: \(Int(reverb))%") Slider(value: $reverb, in: 0...100, step: 1) { _ in audioManager.setReverbMix(reverb) } } } .padding() } }

I did watch WWDC 2019 Session 716 and understand that an active audio session is key to unlocking low‑level networking on watchOS. I’m configuring my audio session and engine as follows: private func configureAudioSession(completion: @escaping (Bool) -> Void) { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: []) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) // Retrieve sample rate and configure the audio format. let sampleRate = audioSession.sampleRate print("Active hardware sample rate: \(sampleRate)") audioFormat = AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1) // Configure the audio engine. audioInputNode = audioEngine.inputNode audioEngine.attach(audioPlayerNode) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: audioFormat) try audioEngine.start() completion(true) } catch { print("Error configuring audio session: \(error.localizedDescription)") completion(false) } } private func setupUDPConnection() { let parameters = NWParameters.udp parameters.includePeerToPeer = true connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupTCPConnection() { let parameters = NWParameters.tcp connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupWebSocketConnection() { guard let url = URL(string: "ws://***.***.xxxxx.***:0000") else { print("Invalid WebSocket URL") return } let session = URLSession(configuration: .default) webSocketTask = session.webSocketTask(with: url) webSocketTask?.resume() print("WebSocket connection initiated") sendAudioToServer() receiveDataFromServer() sendWebSocketPing(after: 0.6) } private func setupNWConnectionHandlers() { connection?.stateUpdateHandler = { [weak self] state in DispatchQueue.main.async { switch state { case .ready: print("Connected (NWConnection)") self?.isConnected = true self?.failToConnect = false self?.receiveDataFromServer() self?.sendAudioToServer() case .waiting(let error), .failed(let error): print("Connection error: \(error.localizedDescription)") DispatchQueue.main.asyncAfter(deadline: .now() + 2) { self?.setupNetwork() } case .cancelled: print("NWConnection cancelled") self?.isConnected = false default: break } } } connection?.start(queue: .main) } Duplex in this context refers to two-way audio transmission simultaneously recording and sending audio while also receiving and playing back incoming audio, similar to a VoIP/SIP call. The setup works fine on the simulator, which suggests that the core logic is correct. However, since the simulator doesn’t fully replicate WatchOS hardware behavior especially for audio sessions and networking issues might arise when running on a real device. The problem likely lies in either the Watch’s actual hardware limitations, permission constraints, or specific audio session configurations. I am reaching out to seek further assistance regarding the challenges I've been experiencing with establishing a UDP, TCP & web socket connection on watchOS using NWConnection for duplex audio streaming. Despite implementing the recommendations provided earlier, I am still encountering difficulties From what I can see, your implementation is focused on streaming audio playback with the server. In my case, I'm looking for a slightly different approach: I want to capture audio and send buffers of a specific size to the server while playing audio simultaneously, essentially achieving full duplex streaming similar to a VOIP call. Additionally, I’d like to ensure that if no external audio route is connected, the Apple Watch speaker is used by default. Any thoughts or insights on adapting this setup for those requirements would be very welcome.

I'm trying to implement Ambisonic B-Format audio playback on Vision Pro with head tracking. So far audio plays, head tracking works, and the sound appears to be stereo. The problem is that it is not a proper binaural playback when compared to playing back the audiofile with a DAW. Has anyone successfully implemented B-Format playback on Vision Pro? Any suggestions on my current implementation: func playAmbiAudioForum() async { do { try AVAudioSession.sharedInstance().setCategory(.playback) try AVAudioSession.sharedInstance().setActive(true) // AudioFile laoding/preperation guard let testFileURL = Bundle.main.url(forResource: "audiofile", withExtension: "wav") else { print("Test file not found") return } let audioFile = try AVAudioFile(forReading: testFileURL) let audioFileFormat = audioFile.fileFormat // create AVAudioFormat with Ambisonics B Format guard let layout = AVAudioChannelLayout(layoutTag: kAudioChannelLayoutTag_Ambisonic_B_Format) else { print("layout failed") return } let format = AVAudioFormat( commonFormat: audioFile.processingFormat.commonFormat, sampleRate: audioFile.fileFormat.sampleRate, interleaved: false, channelLayout: layout ) // write audiofile to buffer guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: UInt32(audioFile.length)) else { print("buffer failed") return } try audioFile.read(into: buffer) playerNode.renderingAlgorithm = .HRTF // connecting nodes audioEngine.attach(playerNode) audioEngine.connect(playerNode, to: audioEngine.outputNode, format: format) audioEngine.prepare() playerNode.scheduleBuffer(buffer, at: nil) { print("File finished playing") } try audioEngine.start() playerNode.play() } catch { print("Setup error:", error) } }

I'm experiencing audio issues while developing for visionOS when playing PCM data through AVAudioPlayerNode. Issue Description: Occasionally, the speaker produces loud popping sounds or distorted noise This occurs during PCM audio playback using AVAudioPlayerNode The issue is intermittent and doesn't happen every time Technical Details: Platform: visionOS Device: vision pro / simulator Audio Framework: AVFoundation Audio Node: AVAudioPlayerNode Audio Format: PCM I would appreciate any insights on: Common causes of audio distortion with AVAudioPlayerNode Recommended best practices for handling PCM playback in visionOS Potential configuration issues that might cause this behavior Has anyone encountered similar issues or found solutions? Any guidance would be greatly helpful. Thank you in advance!

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) } } }

Hi all, The use of setVoiceProcessingEnabled increases the channel count of my microphone audio from 1 to 5. This has downstream effects, because when I use AVAudioConverter to convert between PCM buffer types the output buffer contains only silence. Here is a reproduction showing the channel growth from 1 to 5: let avAudioEngine: AVAudioEngine = AVAudioEngine() let inputNode = avAudioEngine.inputNode print(inputNode.inputFormat(forBus: 0)) // Prints <AVAudioFormat 0x600002f7ada0: 1 ch, 48000 Hz, Float32> do { try inputNode.setVoiceProcessingEnabled(true) } catch { print("Could not enable voice processing \(error)") return } print(inputNode.inputFormat(forBus: 0)) // Prints <AVAudioFormat 0x600002f7b020: 5 ch, 44100 Hz, Float32, deinterleaved> If it helps, the reason I'm using setVoiceProcessingEnabled because I don't want the mic to pick up output from the speakers. Per wwdc When enabled, extra signal processing is applied on the incoming audio, and any audio that is coming from the device is taken Here is my conversion logic from the input PCM format (which in the case above is 5ch, 44.1kHZ, Float 32, deinterleaved) to the target format PCM16 with a single channel: let outputFormat = AVAudioFormat( commonFormat: .pcmFormatInt16, sampleRate: inputPCMFormat.sampleRate, channels: 1, interleaved: false ) guard let converter = AVAudioConverter( from: inputPCMFormat, to: outputFormat) else { fatalError("Demonstration") } let newLength = AVAudioFrameCount(outputFormat.sampleRate * 2.0) guard let outputBuffer = AVAudioPCMBuffer( pcmFormat: outputFormat, frameCapacity: newLength) else { fatalError("Demonstration") } outputBuffer.frameLength = newLength try! converter.convert(to: outputBuffer, from: inputBuffer) // Use the PCM16 outputBuffer The outputBuffer contains only silence. But if I comment out inputNode.setVoiceProcessingEnabled(true) in the first snippet, the outputBuffer then plays exactly how I would expect it to. So I have two questions: Why does setVoiceProcessingEnabled increase the channel count to 5? How should I convert the resulting format to a single channel PCM16 format? Thank you, Lou

Hi! I am creating a aumi AUv3 extension and I am trying to achieve simultaneous connections to multiple other avaudionodes. I would like to know it is possible to route the midi to different outputs inside the render process in the AUv3. I am using connectMIDI(_:to:format:eventListBlock:) to connect the output of the AUv3 to multiple AvAudioNodes. However, when I send midi out of the AUv3, it gets sent to all the AudioNodes connected to it. I can't seem to find any documentation on how to route the midi only to one of the connected nodes. Is this possible?

I'm using AVAudioEngine to play AVAudioPCMBuffers. I'd like to synchronize some events with the playback. For example if the audio's frame position is >= some point && less than some point trigger some code. So I'm looking at - (void)installTapOnBus:(AVAudioNodeBus)bus bufferSize:(AVAudioFrameCount)bufferSize format:(AVAudioFormat * __nullable)format block:(AVAudioNodeTapBlock)tapBlock; Now I have frame positions calculated (predetermined before audio is scheduled I already made all necessary computations) . So I just need to fire code at certain points during playback: [playerNode installTapOnBus:bus bufferSize:bufferSize format:format block:^(AVAudioPCMBuffer * _Nonnull buffer, AVAudioTime * _Nonnull when) { //Inspect current audio here and fire... }]; [playerNode scheduleBuffer:fullbuffer atTime:startTime options:0 completionCallbackType:AVAudioPlayerNodeCompletionDataPlayedBack completionHandler:^(AVAudioPlayerNodeCompletionCallbackType callbackType) { // some code is here, not important to this question. }]; The problem I'm having is figuring out at what point in full buffer I'm at within the tap block. The tap block passes chunks (not the full audio buffer). I tried using the when parameter of the block to calculate the frame position relative to the entire audio but have be unsuccessful so far. I'm assuming the when parameter is relative to the buffer passed in the tap block (not my entire audio buffer I scheduled). Not installing a tap and just using a timer before scheduling my fullBuffer has given me good results but I'd rather avoid using a timer if possible and use sample time.

Hi all, I am working on an app where I have live prompts playing, in addition to a voice channel that sometimes becomes active. Right now I am using two different AVAudioSession Configurations so what we only switch to a mic enabled mode when we actually need input from the mic. These are defined below. When just using the device hardware, everything works as expected and the modes change and the playback continues as needed. However when using bluetooth devices such as AirPods where the switch from AD2P to HFP is needed, I am getting a AVAudioEngineConfigurationChange notification. In response I am tearing down the engine and creating a new one with the same 2 player nodes. This does work fine and there are no crashes, except all the audio I have scheduled on a player node has now been cleared. All the completion blocks marked with ".dataPlayedBack" return the second this event happens, and leaves me in a state where I now have a valid engine setup again but have no idea what actually played, or was errantly marked as such. Is this the expected behavior when getting a configuration change notification? Adding some information below to my audio graph for context: All my parts of the graph, I disconnect when getting this event and do the same to the new engine private var inputEngine: AVAudioEngine private var audioEngine: AVAudioEngine private let voicePlayerNode: AVAudioPlayerNode private let promptPlayerNode: AVAudioPlayerNode audioEngine.attach(voicePlayerNode) audioEngine.attach(promptPlayerNode) audioEngine.connect( voicePlayerNode, to: audioEngine.mainMixerNode, format: voiceNodeFormat ) audioEngine.connect( promptPlayerNode, to: audioEngine.mainMixerNode, format: nil ) An example of how I am scheduling playback, and where that completion is firing even if it didn't actually play. private func scheduleVoicePlayback(_ id: AudioPlaybackSample.Id, buffer: AVAudioPCMBuffer) async throws { guard !voicePlayerQueue.samples.contains(where: { $0 == id }) else { return } seprateQueue.append(buffer) if !isVoicePlaying { activateAudioSession() } voicePlayerQueue.samples.append(id) if !voicePlayerNode.isPlaying { voicePlayerNode.play() } if let convertedBuffer = buffer.convert(to: voiceNodeFormat) { await voicePlayerNode.scheduleBuffer(convertedBuffer, completionCallbackType: .dataPlayedBack) } else { throw AudioPlaybackError.failedToConvert } voiceSampleHasBeenPlayed(id) } And lastly my audio session configuration if its useful. extension AVAudioSession { static func setDefaultCategory() { do { try sharedInstance().setCategory( .playback, options: [ .duckOthers, .interruptSpokenAudioAndMixWithOthers ] ) } catch { print("Failed to set default category? \(error.localizedDescription)") } } static func setVoiceChatCategory() { do { try sharedInstance().setCategory( .playAndRecord, options: [ .defaultToSpeaker, .allowBluetooth, .allowBluetoothA2DP, .duckOthers, .interruptSpokenAudioAndMixWithOthers ] ) } catch { print("Failed to set category? \(error.localizedDescription)") } } }

I am experiencing an issue while recording audio using AVAudioEngine with the installTap method. I convert the AVAudioPCMBuffer to Data and send it to a UDP server. However, when I receive the Data and play it back, there is continuous crackling noise during playback. I am sending audio data using this library "https://github.com/mindAndroid/swift-rtp" by creating packet and send it. Please help me resolve this issue. I have attached the code reference that I am currently using. Thank you. ViewController.swift

Hello, I'm fairly new to AVAudioEngine and I'm trying to connect 2 mono nodes as left/right input to a stereo node. I was successful in splitting the input audio to 2 mono nodes using AVAudioConnectionPoint and channelMap. But I can't figure out how to connect them back to a stereo node. I'll post the code I have so far. The use case for this is that I'm trying to process the left/right channels with separate audio units. Any ideas? let monoFormat = AVAudioFormat(standardFormatWithSampleRate: nativeFormat.sampleRate, channels: 1)! let leftInputMixer = AVAudioMixerNode() let rightInputMixer = AVAudioMixerNode() let leftOutputMixer = AVAudioMixerNode() let rightOutputMixer = AVAudioMixerNode() let channelMixer = AVAudioMixerNode() [leftInputMixer, rightInputMixer, leftOutputMixer, rightOutputMixer, channelMixer].forEach { engine.attach($0) } let leftConnectionR = AVAudioConnectionPoint(node: leftInputMixer, bus: 0) let rightConnectionR = AVAudioConnectionPoint(node: rightInputMixer, bus: 0) plugin.leftInputMixer = leftInputMixer plugin.rightInputMixer = rightInputMixer plugin.leftOutputMixer = leftOutputMixer plugin.rightOutputMixer = rightOutputMixer plugin.channelMixer = channelMixer leftInputMixer.auAudioUnit.channelMap = [0] rightInputMixer.auAudioUnit.channelMap = [1] engine.connect(previousNode, to: [leftConnectionR, rightConnectionR], fromBus: 0, format: monoFormat) // Process right channel, pass through left channel engine.connect(rightInputMixer, to: plugin.audioUnit, format: monoFormat) engine.connect(plugin.audioUnit, to: rightOutputMixer, format: monoFormat) engine.connect(leftInputMixer, to: leftOutputMixer, format: monoFormat) // Mix back to stereo? engine.connect(leftOutputMixer, to: channelMixer, format: stereoFormat) engine.connect(rightOutputMixer, to: channelMixer, format: stereoFormat)

I’m working on a memo app that records audio from the iPhone’s microphone (and other devices like MacBook or iPad) and processes it in 10-second chunks at a target sample rate of 16 kHz. However, I’ve encountered limitations with installTap in AVAudioEngine, which doesn’t natively support configuring a target sample rate on the mic input (the default being 44.1 kHz). To address this, I tried using AVAudioMixerNode to downsample the mic input directly. Although everything seems correctly configured, no audio is recorded—just a flat signal with zero levels. There are no errors, and all permissions are granted, so it seems like an issue with downsampling rather than the mic setup itself. To make progress, I implemented a workaround by tapping and resampling each chunk tapped using installTap (every 50ms in my case) with AVAudioConverter. While this works, it can introduce artifacts at the beginning and end of each chunk, likely due to separate processing instead of continuous downsampling. Here are the key issues and questions I have: 1. Can we change the mic input sample rate directly using AVAudioSession or another native API in AVAudio? Setting up the desired sample rate initially would be ideal for my use case. 2. Are there alternatives to installTap for recording audio at a different sample rate or for continuously downsampling the live input without chunk-based artifacts? This issue seems longstanding, as noted in a 2018 forum post: https://forums.developer.apple.com/forums/thread/111726 Any guidance on configuring or processing mic input at a lower sample rate in real-time would be greatly appreciated. Thank you!

I'm building a streaming app on visionOS that can play sound from audio buffers each frame. The audio format has a bitrate of 48000, and each buffer has 480 samples. I noticed when calling audioPlayerNode.scheduleBuffer(audioBuffer) The memory keeps increasing at the speed of 0.1MB per second And at around 4 minutes, the node seems to be full of buffers and had a hard reset, at which point, the audio is stopped temporary with a memory change. see attached screenshot. However, if I call audioPlayerNode.scheduleBuffer(audioBuffer, at: nil, options: .interrupts) The memory leak issue is gone, but the audio is broken (sounds like been shortened). Below is the full code snippet, anyone knows how to fix it? @Observable final class MyAudioPlayer { private var audioEngine: AVAudioEngine = .init() private var audioPlayerNode: AVAudioPlayerNode = .init() private var audioFormat: AVAudioFormat? init() { audioEngine.attach(audioPlayerNode) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: nil) try? AVAudioSession.sharedInstance().setCategory(.playback, mode: .default) try? AVAudioSession.sharedInstance().setActive(true) audioEngine.prepare() try? audioEngine.start() audioPlayerNode.play() } // more code... /// callback every frame private func audioFrameCallback_Non_Interleaved(buf: UnsafeMutablePointer<Float>?, samples: Int) { guard let buf, let format = AVAudioFormat(commonFormat: .pcmFormatFloat32, sampleRate: 48000, channels: 2, interleaved: false), let audioBuffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: AVAudioFrameCount(samples)) else { return } audioBuffer.frameLength = AVAudioFrameCount(samples) if let data = audioBuffer.floatChannelData { for channel in 0 ..< Int(format.channelCount) { for frame in 0 ..< Int(audioBuffer.frameLength) { data[channel][frame] = buf[frame * Int(format.channelCount) + channel] } } } // memory leak here audioPlayerNode.scheduleBuffer(audioBuffer) } }