Our app is a game written in Unity where we have most of our audio playback handled by Unity. However, one of our game experiences utilized microphone input for speech recognition, and so in order for us to perform echo cancellation (while the game has audio playback), we setup an audio stream from Unity to native Swift code that performs the mixing of the input/output nodes. We however found that by streaming the audio buffer to our AVAudioSession: The volume of the audio playback appears to output differently When capturing a screen recording of the app, the audio playback being played from AVAudioSession does not get captured at all. Looking to figure out what could be causing the discrepency in playback as well as capture behaviour during screen recordings. We setup the AVAudioSession with this configuration: AVAudioSession.sharedInstance().setCategory(AVAudioSession.Category.playAndRecord, options: .mixWithOthers) with inputNode.setVoiceProcessingEnabled(true) after connecting our IO and mixer nodes. Any suggestions or ideas on what to look out for would be appreciated!

Prior to iOS 17, I used AVAudioFile to open (for reading) the assetURL of MPMediaItem for songs that the user purchased through iTunes Store. With the iOS 17 Beta, this seems no longer possible as AVAudioFile throws this: ExtAudioFile.cpp:211 about to throw -54: open audio file AVAEInternal.h:109 [AVAudioFile.mm:135:AVAudioFileImpl: (ExtAudioFileOpenURL((CFURLRef)fileURL, &_extAudioFile)): error -54 Also can't copy the url to Documents directory because I get this: The file “item.m4a” couldn’t be opened because URL type ipod-library isn’t supported. This seems to be affecting other apps on the App Store besides mine, and it will reflect very badly on my app if this makes into the final iOS 17 because I have encouraged users to buy songs on iTunes Store to use with my app. Now there seems like there is no way to access them. Is this a known bug, or is there some type of workaround?

When recording audio over bluetooth from AirPods to iPhone using the AVAudioRecorder the Bluetooth audio codec used is always AAC-ELD independent of the codec to store which is selected in the AVAudioRecorder instance. As far as I know must every Bluetooth device support SBC, hence, it should be possible for the AirPods to transmit the recorded audio using the SBC codec instead of AAC-ELD. However, I could not find any resource on how the request this codec using the AVAudioRecorder or AVAudioEngine. Is it possible to request SBC at all and if yes how?

Hi! I am working on an audio application on iOS. This is how I retreive the workgroup from the remoteIO audiounit (ioUnit). The unit is initialized and is working fine (meaning that it is regularly called by the system). os_workgroup_t os_workgroup{nullptr}; uint32_t os_workgroup_index_size; if (status = AudioUnitGetProperty(ioUnit, kAudioOutputUnitProperty_OSWorkgroup, kAudioUnitScope_Global, 0, &os_workgroup, &os_workgroup_index_size); status != noErr) { throw runtime_error("AudioUnitSetProperty kAudioOutputUnitProperty_OSWorkgroup - Failed with OSStatus: " + to_string(status)); } However the resulting os_workgroup's value is 0x40. Which seems not correct. No wonder I cannot join any other realtime threads to the workgroup as well. The returned status however is a solid 0. Can anyone help?

From an app that reads audio from the built-in microphone, I'm receiving many crash logs where the AVAudioEngine fails to start again after the app was suspended. Basically, I'm calling these two methods in the app delegate's applicationDidBecomeActive and applicationDidEnterBackground methods respectively: let audioSession = AVAudioSession.sharedInstance() func startAudio() throws { self.audioEngine = AVAudioEngine() try self.audioSession.setCategory(.record, mode: .measurement)} try audioSession.setActive(true) self.audioEngine!.inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil, block: { ... }) self.audioEngine!.prepare() try self.audioEngine!.start() } func stopAudio() throws { self.audioEngine?.stop() self.audioEngine?.inputNode.removeTap(onBus: 0) self.audioEngine = nil try self.audioSession.setActive(false, options: [.notifyOthersOnDeactivation]) } In the crash logs (iOS 16.6) I'm seeing that this works fine several times as the app is opened and closed, but suddenly the audioEngine.start() call fails with the error Error Domain=com.apple.coreaudio.avfaudio Code=-10851 "(null)" UserInfo={failed call=err = AUGraphParser::InitializeActiveNodesInInputChain(ThisGraph, *GetInputNode())} and the audioEngine!.inputNode.outputFormat(forBus: 0) is something like <AVAudioFormat 0x282301c70: 2 ch, 0 Hz, Float32, deinterleaved> . Also, right before installing the tap, audioSession.availableInputs contains an entry of type MicrophoneBuiltIn but audioSession.currentRoute lists no inputs at all. I was not able to reproduce this situation on my own devices yet. Does anyone have an idea why this is happening?

My User Generated Content for my App is audio-based only and anonymous. All the content is deleted after 24 hours. Do I still need a report button, since I don't know the user and the content gets deleted anyway?

Hello, I have struggled to resolve issue above question. I could speak utterance when I turn on my iPhone, but when my iPhone goes to background mode(turn off iPhone), It doesn't speak any more. I think it is possible to play audio or speak utterance because I can play music on background status in youtube. Any help please??

Sound Pad throws an error on the audio device on iPad. Sound Pad Version 1.0.0 3 April 2023 Swift 5.8 Edition A fatal error was found in AudioPlayer.swift Line which causes this var engine: AVAudioEngine

Hi there, I'm having some trouble with AVAudioMixerNode only working when there is a single input, and outputting silence or very quiet buzzing when >1 input node is connected. My setup has voice processing enabled, input going to a sink, and N source nodes going to the main mixer node, going to the output node. In all cases I am connecting nodes in the graph with the same declared format: 48kHz 1 channel Float32 PCM. This is working great for 1 source node, but as soon as I add a second it breaks. I can reproduce this behaviour in the SignalGenerator sample, when the same format is used everywhere. Again, it'll work fine with 1 source node even in this configuration, but add another and there's silence. Am I doing something wrong with formats here? Is this expected? As I understood it with voice processing on and use of a mixer node I should be able to use my own format essentially everywhere in my graph? My SignalGenerator modified repro example follows: import Foundation import AVFoundation // True replicates my real app's behaviour, which is broken. // You can remove one source node connection // to make it work even when this is true. let showBrokenState: Bool = true // SignalGenerator constants. let frequency: Float = 440 let amplitude: Float = 0.5 let duration: Float = 5.0 let twoPi = 2 * Float.pi let sine = { (phase: Float) -> Float in return sin(phase) } let whiteNoise = { (phase: Float) -> Float in return ((Float(arc4random_uniform(UINT32_MAX)) / Float(UINT32_MAX)) * 2 - 1) } // My "application" format. let format: AVAudioFormat = .init(commonFormat: .pcmFormatFloat32, sampleRate: 48000, channels: 1, interleaved: true)! // Engine setup. let engine = AVAudioEngine() let mainMixer = engine.mainMixerNode let output = engine.outputNode try! output.setVoiceProcessingEnabled(true) let outputFormat = engine.outputNode.inputFormat(forBus: 0) let sampleRate = Float(format.sampleRate) let inputFormat = format var currentPhase: Float = 0 let phaseIncrement = (twoPi / sampleRate) * frequency let srcNodeOne = AVAudioSourceNode { _, _, frameCount, audioBufferList -> OSStatus in let ablPointer = UnsafeMutableAudioBufferListPointer(audioBufferList) for frame in 0..<Int(frameCount) { let value = sine(currentPhase) * amplitude currentPhase += phaseIncrement if currentPhase >= twoPi { currentPhase -= twoPi } if currentPhase < 0.0 { currentPhase += twoPi } for buffer in ablPointer { let buf: UnsafeMutableBufferPointer<Float> = UnsafeMutableBufferPointer(buffer) buf[frame] = value } } return noErr } let srcNodeTwo = AVAudioSourceNode { _, _, frameCount, audioBufferList -> OSStatus in let ablPointer = UnsafeMutableAudioBufferListPointer(audioBufferList) for frame in 0..<Int(frameCount) { let value = whiteNoise(currentPhase) * amplitude currentPhase += phaseIncrement if currentPhase >= twoPi { currentPhase -= twoPi } if currentPhase < 0.0 { currentPhase += twoPi } for buffer in ablPointer { let buf: UnsafeMutableBufferPointer<Float> = UnsafeMutableBufferPointer(buffer) buf[frame] = value } } return noErr } engine.attach(srcNodeOne) engine.attach(srcNodeTwo) engine.connect(srcNodeOne, to: mainMixer, format: inputFormat) engine.connect(srcNodeTwo, to: mainMixer, format: inputFormat) engine.connect(mainMixer, to: output, format: showBrokenState ? inputFormat : outputFormat) // Put the input node to a sink just to match the formats and make VP happy. let sink: AVAudioSinkNode = .init { timestamp, numFrames, data in .zero } engine.attach(sink) engine.connect(engine.inputNode, to: sink, format: showBrokenState ? inputFormat : outputFormat) mainMixer.outputVolume = 0.5 try! engine.start() CFRunLoopRunInMode(.defaultMode, CFTimeInterval(duration), false) engine.stop()

Hello, I'm facing an issue with Xcode 15 and iOS 17: it seems impossible to get AVAudioEngine's audio input node to work on simulator. inputNode has a 0ch, 0kHz input format, connecting input node to any node or installing a tap on it fails systematically. What we tested: Everything works fine on iOS simulators <= 16.4, even with Xcode 15. Nothing works on iOS simulator 17.0 on Xcode 15. Everything works fine on iOS 17.0 device with Xcode 15. More details on this here: https://github.com/Fesongs/InputNodeFormat Any idea on this? Something I'm missing? Thanks for your help 🙏 Tom PS: I filed a bug on Feedback Assistant, but it usually takes ages to get any answer so I'm also trying here 😉

In voip application , when the CallKit is enabled if we try playing a video through AVplayer the video content is updated frame by frame and the audio of the content is not audible . This issue is observed only in iOS 17, any idea how can we resolve this

Hello, I started to set audio stereo recording (both audio and video are recorded) and the audio quality seems to be lower than quality obtained with native camera application (configured for stereo). Using console to check the log, I found a difference between camera app and mine regarding MXSessionMode (of mediaserverd) in fact, camera application gives MXSessionMode = SpatialRecording and mine MXSessionMode = VideoRecording How can I configure capture session to finally have MXSessionMode = SpatialRecording? Any suggestion? Best regards

Basically for this iPhone app I want to be able to record from either the built in microphone or from a connected USB audio device while simultaneously playing back processed audio on connected AirPods. It's a pretty simple AVAudioEngine setup that includes a couple of effects units. The category is set to .playAndRecord with the .allowBluetooth and .allowBluetoothA2DP options added. With no attempts to set the preferred input and AirPods connected, the AirPods mic will be used and output also goes to the AirPods. If I call setPreferredInput to either built in mic or a USB audio device I will get input as desired but then output will always go to the speaker. I don't really see a good explanation for this and overrideOutputAudioPort does not really seem to have suitable options. Testing this on iPhone 14 Pro

How can I record audio in a keyboard extension? I've enabled microphone support by enabling "RequestsOpenAccess". When I try to record, I get the error below in the console. This doesn't make sense as Apple's docs seem to say that microphone access is allowed with Full Keyboard Access. What is the point of enabling the microphone if the app cannot access the data from the microphone? -CMSUtilities- CMSUtility_IsAllowedToStartRecording: Client sid:0x2205e, XXXXX(17965), 'prim' with PID 17965 was NOT allowed to start recording because it is an extension and doesn't have entitlements to record audio.

As the title suggests I am using AVAudioEngine for SpeechRecognition input & AVAudioPlayer for sound output. Apple says in this talk https://developer.apple.com/videos/play/wwdc2019/510 that the setVoiceProcessingEnabled function very usefully cancels the output from speaker to the mic. I set voiceProcessing on the Input and output nodes. It seems to work however the volume is low, even when the system volume is turned up. Any solution to this would be much appreciated.

My app listens for verbal commands "Roll" & "Skip". It was working well until I used it while listening to a podcast in another app. I am getting a crash with the error: Thread 1: "required condition is false: IsFormatSampleRateAndChannelCountValid(format)" . It crashes when I am playing audio from the apps Snipd (a podcast app) or the Apple Podcast app. When I am playing audio from Youtube or the Apple Music it does not crash. This is the code for when I start listening for the commands: // MARK: - Speech Recognition func startListening() { do { try configureAudioSession() createRecognitionRequest() try prepareAudioEngine() } catch { print("Audio Engine error: \(error.localizedDescription)") } } private func configureAudioSession() throws { let audioSession = AVAudioSession.sharedInstance() try audioSession.setCategory(.playAndRecord, mode: .measurement, options: [.interruptSpokenAudioAndMixWithOthers, .duckOthers]) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) } private func createRecognitionRequest() { recognitionRequest = SFSpeechAudioBufferRecognitionRequest() guard let recognitionRequest = recognitionRequest else { return } recognitionRequest.shouldReportPartialResults = true recognitionTask = speechRecognizer?.recognitionTask(with: recognitionRequest, resultHandler: handleRecognitionResult) } private func prepareAudioEngine() throws { let inputNode = audioEngine.inputNode inputNode.removeTap(onBus: 0) let inputFormat = inputNode.inputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: inputFormat) { [weak self] (buffer, _) in self?.recognitionRequest?.append(buffer) } audioEngine.prepare() try audioEngine.start() isActuallyListening = true } Thanks

I have a custom USB device that includes a microphone. I can see the microphone on macOS when I plug in the device so I know that it is working with the kernel and AV subsystems. I can enumerate and reference the microphone using AVCaptureDevice but I have not been able to figure out how to use this device reference with AVAudioEngine. I'm trying to accomplish two things with this microphone. I want to stream audio from the microphone and have it rendered to the speakers on my MacBook Pro. I want to capture sound data from the microphone and forward it to a live streaming API. To my mind, from what I've read, I need AVAudioEngine to do this but I'm having trouble determining from the documentation just how to go about it on macOS. It seems that there is a lot more information for iOS or iPadOS but since USB-C support is sparsely documented on those operating systems, I'm focusing on the desktop (macOS) for now. Can I convert an AVCaptureDevice into and audio input for AVAudioEngine? If not, how can I accomplish what I'm trying to do using whatever is available on AVFoundation?

Hi everyone, I was working on some code that involves recording audio with AVAudioEngine and got an issue that just crashes the app: EXC_BREAKPOINT Exception 6, Code 1, Subcode 4304279688 +0x009888 AudioRecordModule.setupAudioEngine +0x009788 AudioRecordModule.setupAudioEngine +0x00c5bc AudioRecordModule.handleConfigurationChange Below is the relevant code in the Recorder class. public class AudioRecordModule: Module { private var audioEngine: AVAudioEngine? private func startRecording(options recordingOptions: RecordingOptions) { try AVAudioSession.sharedInstance().setCategory(.playAndRecord, options: .mixWithOthers) try AVAudioSession.sharedInstance().setActive(true) outputFormat = AVAudioFormat( commonFormat: recordingOptions.bitDepth == 32 ? .pcmFormatInt32 : .pcmFormatInt16, sampleRate: Double(recordingOptions.sampleRate), channels: AVAudioChannelCount(recordingOptions.channels), interleaved: true )! let fileUri = URL(string: recordingOptions.fileUri)! let formatSettings: [String: Any] = [ AVFormatIDKey: kAudioFormatMPEG4AAC, AVSampleRateKey: recordingOptions.sampleRate, AVNumberOfChannelsKey: recordingOptions.channels, AVEncoderBitRateStrategyKey: AVAudioBitRateStrategy_Constant, AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue, ] self.recordedFile = try AVAudioFile( forWriting: fileUri, settings: formatSettings, commonFormat: outputFormat.commonFormat, interleaved: outputFormat.isInterleaved ) if !hadSetupNotification { setupNotifications() } } func handleConfigurationChange() { DispatchQueue.main.async { self.releaseAudioEngine() self.setupAudioEngine() if self.state == "recording" { // we could attempt to keep recording do { try self.audioEngine?.start() } catch { self.internalPauseRecording() self.sendInterruptEvent() } } } } func setupNotifications() { nc.addObserver( forName: Notification.Name.AVAudioEngineConfigurationChange, object: nil, queue: nil ) { [weak self] _ in guard let weakself = self else { return } if weakself.state != "inactive" { weakself.handleConfigurationChange() } } } private func setupAudioEngine() { self.audioEngine = nil let audioEngine = AVAudioEngine() self.audioEngine = audioEngine let inputNode = audioEngine.inputNode let inputFormat = inputNode.inputFormat(forBus: 0) let converter = AVAudioConverter(from: inputFormat, to: outputFormat)! inputNode.installTap(onBus: 0, bufferSize: 1024, format: inputFormat) { (buffer: AVAudioPCMBuffer!, time: AVAudioTime!) -> Void in do { let inputBlock: AVAudioConverterInputBlock = { _, outStatus in outStatus.pointee = AVAudioConverterInputStatus.haveData return buffer } let frameCapacity = AVAudioFrameCount(self.outputFormat.sampleRate) * buffer.frameLength / AVAudioFrameCount(buffer.format.sampleRate) let outputBuffer = AVAudioPCMBuffer( pcmFormat: self.outputFormat, frameCapacity: frameCapacity )! var error: NSError? converter.convert(to: outputBuffer, error: &error, withInputFrom: inputBlock) if let error = error { throw error } else { try self.recordedFile?.write(from: outputBuffer) } } catch { print(error) } } } private func releaseAudioEngine() { if let audioEngine = self.audioEngine { audioEngine.inputNode.removeTap(onBus: 0) audioEngine.stop() } audioEngine = nil } } Beside that, the record module works normally. It is just the configuration change that it does not handle well. I understand that when configuration changes, I need to reinit the audio engine to have the correct input format (since the new config/audio device can have different sample rate and such). If I don't do that, the app also crashes perhaps due to the mismatch. AVAudioRecorder is not an option for me. Thank you for your help.

I’m exploring enabling speech-to-commands processing for a game, but would like to try and do a baseline of voice recognition within that to allow two people in close proximity to interact , but not interfere with each others voice commands to this system. (it’s for an accessible game idea)

My project has uses an AVAudioEngine with a very simple setup: A Speech recognizer running on a tap on the engine's input with separate AVAudioPlayerNodes handling playback. try session.setCategory(.playAndRecord, mode: .default, options: []) try session.setActive(true, options: .notifyOthersOnDeactivation) try session.setAllowHapticsAndSystemSoundsDuringRecording(true) filePlayerNode ---> engine.mainMixerNode bufferPlayerNode --> engine.mainMixerNode engine.mainMixerNode --> engine.outputNode //bufferPlayer.scheduleBuffer() is called on its own queue The input works fine since the buffers can be collected into a file and plays back correctly, and also because the recognizer works fine; but when I try to play the live audio by sending the buffer to the bufferPlayer on this or another device, the buffer audio plays at a very low volume, sometimes with severe distortions. If I lower the sample rate via AVAudioConverter, the distortions get worse. I've tried experimenting with the AVAudioSession category options, having separate AVAudioEngines, and much, much more, yet I still haven't figured this out. It's gotten to the point where I've fixed almost all the arcane and minor issues in my audio system, yet I still can't play back my voice properly. The ability to both play and record simultaneously is a basic feature of phones--when on speaker mode, a phone doesn't need to behave like a walkie-talkie. In my mind, it's inconceivable that the relatively new AVAudioEngine doesn't have a implementation for this, since the main issue (feedback loops) can be dealt with via a simple primitive circuit. Live video chat apps like FaceTime wouldn't be possible without this, yet to my surprise I found no answers online (what I did find were articles explaining how to write a file while playback is occurring). Is there truly no way to do this on AVAudioEngine? Am I missing something fundamental? Any pointers would be greatly appreciated