AVAudioEngine

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Use a group of connected audio node objects to generate and process audio signals and perform audio input and output.

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Configuring Apple Vision Pro's microphones to effectively pick up other speaker's voice
I am developing a visionOS app that captions speech in real environments. Currently, I am using Apple's built-in speech recognizer. However, when I was testing the app with a Vision Pro, the device seemed to only pick up the user's voice (in other words, the voices of the wearer of the Vision Pro device). For example, when the speech recognition task is running, and another person in front of me is talking, the system does not pick up the speech well. I tried to set the AVAudioSession to be equally sensitive to all directions: private func configureAudioSession() { do { try audioSession.setCategory(.record, mode: .measurement) try audioSession.setActive(true) if #available(visionOS 1.0, *) { let availableDataSources = audioSession.availableInputs?.first?.dataSources if let omniDirectionalSource = availableDataSources?.first(where: {$0.preferredPolarPattern == .omnidirectional}) { try audioSession.setInputDataSource(omniDirectionalSource) } } } catch { print("Failed to set up audio session: \(error)") } } And here is how I set up the speech recognition and configure the microphone inputs: private func startSpeechRecognition(completion: @escaping (String) -> Void) { do { // Cancel the previous task if it's running. if let recognitionTask = recognitionTask { recognitionTask.cancel() self.recognitionTask = nil } // The AudioSession is already active, creating input node. let inputNode = audioEngine.inputNode try inputNode.setVoiceProcessingEnabled(false) // Create and configure the speech recognition request recognitionRequest = SFSpeechAudioBufferRecognitionRequest() guard let recognitionRequest = recognitionRequest else { fatalError("Unable to create a recognition request") } recognitionRequest.shouldReportPartialResults = true // Keep speech recognition data on device if #available(iOS 13, *) { recognitionRequest.requiresOnDeviceRecognition = true } // Create a recognition task for speech recognition session. // Keep a reference to the task so that it can be canceled. recognitionTask = speechRecognizer?.recognitionTask(with: recognitionRequest) { result, error in // var isFinal = false if let result = result { // Update the recognizedText completion(result.bestTranscription.formattedString) } else if let error = error { completion("Recognition error: \(error.localizedDescription)") } if error != nil || result?.isFinal == true { // Stop recognizing speech if there is a problem self.audioEngine.stop() inputNode.removeTap(onBus: 0) self.recognitionRequest = nil self.recognitionTask = nil } } // Configure the microphone input let recordingFormat = inputNode.outputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { (buffer, when) in self.recognitionRequest?.append(buffer) } audioEngine.prepare() try audioEngine.start() } catch { completion("Audio engine could not start: \(error.localizedDescription)") } }
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1.3k
Jul ’24
Setting Audio Input node for AVAudioEngine causes outside audio to stop
I'm building an app that will allow users to record voice notes. The functionality of all that is working great; I'm trying to now implement changes to the AudioSession to manage possible audio streams from other apps. I want it so that if there is audio playing from a different app, and the user opens my app; the audio keep playing. When we start recording, any third party app audio should stop, and can then can resume again when we stop recording. This is my main audio setup code: private var audioEngine: AVAudioEngine! private var inputNode: AVAudioInputNode! func setupAudioEngine() { audioEngine = AVAudioEngine() inputNode = audioEngine.inputNode audioPlayerNode = AVAudioPlayerNode() audioEngine.attach(audioPlayerNode) let format = AVAudioFormat(standardFormatWithSampleRate: AUDIO_SESSION_SAMPLE_RATE, channels: 1) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: format) } private func setupAudioSession() { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .default, options: [.defaultToSpeaker, .allowBluetooth]) try audioSession.setPreferredSampleRate(AUDIO_SESSION_SAMPLE_RATE) try audioSession.setPreferredIOBufferDuration(0.005) // 5ms buffer for lower latency try audioSession.setActive(true) // Add observers setupInterruptionObserver() } catch { audioErrorMessage = "Failed to set up audio session: \(error)" } } This is all called upon app startup so we're ready to record whenever the user presses the record button. However, currently when this happens, any outside audio stops playing. I isolated the issue to this line: inputNode = audioEngine.inputNode When that's commented out, the audio will play -- but I obviously need this for recording functionality. Is this a bug? Expected behavior?
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Jul ’24
Implementing Multi-Channel Audio Recording on iOS with Built-In and External Mics
Hi there community, First and foremost, a big thank you to everyone who takes the time to read this. TL;DR: How, if even possible, can I record multiple audio streams simultaneously on an iOS application (iPad/iPhone)? I'm working on a recorder for the iPad to gather data for a machine learning project focused on speech recognition. Our goal is to capture extensive speech data, which requires recording from multiple microphones. Specifically, I need to record from all mics connected to our Scarlett 4i4 audio interface and, most importantly, also record from the built-in mic on the iPad or iPhone at the same time. As a newcomer to Swift development, I initially explored AVAudioRecorder. However, I quickly realized that it only supports one active audio node at a time, making multi-channel recording impossible. (perhaps you can proof me wrong, would make my day) Next, I transitioned to using AVAudioEngine, but encountered the same limitation: I couldn't manage to get input nodes for both the built-in mic and the Scarlett interface channels simultaneously. The application started behaving oddly, often resulting in identical audio data being recorded across all files. Determined to find a solution, I delved deeper into the Core Audio framework, specifically using Audio Toolbox. My approach involved creating and configuring multiple Audio Units, each corresponding to a different audio input device. Here's a brief overview of my current implementation: Listing Available Input Devices: I used AVAudioSession to enumerate all available input devices. Creating Audio Units: For each device, I created an Audio Unit and attempted to configure it for recording. Setting Up Callbacks: I set up input and output callbacks to handle the audio processing. Despite my efforts over the last few days, I haven't had much success. The callbacks for the Audio Units don't seem to be invoked correctly, and I'm struggling to achieve simultaneous multi-channel recording. Below is a snippet of my latest attempt: let audioUnitCallback: AURenderCallback = { ( inRefCon: UnsafeMutableRawPointer, ioActionFlags: UnsafeMutablePointer<AudioUnitRenderActionFlags>, inTimeStamp: UnsafePointer<AudioTimeStamp>, inBusNumber: UInt32, inNumberFrames: UInt32, ioData: UnsafeMutablePointer<AudioBufferList>? ) -> OSStatus in guard let ioData = ioData else { return noErr } print("Input callback invoked") let audioUnit = inRefCon.assumingMemoryBound(to: AudioUnit.self).pointee var bufferList = AudioBufferList( mNumberBuffers: 1, mBuffers: AudioBuffer( mNumberChannels: 1, mDataByteSize: 0, mData: nil ) ) let status = AudioUnitRender(audioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList) if status != noErr { print("AudioUnitRender failed: \(status)") return status } // Copy rendered data to output buffer let buffer = UnsafeMutableAudioBufferListPointer(ioData)[0] buffer.mData?.copyMemory(from: bufferList.mBuffers.mData!, byteCount: Int(bufferList.mBuffers.mDataByteSize)) buffer.mDataByteSize = bufferList.mBuffers.mDataByteSize print("Rendered audio data") return noErr } let outputCallback: AURenderCallback = { ( inRefCon: UnsafeMutableRawPointer, ioActionFlags: UnsafeMutablePointer<AudioUnitRenderActionFlags>, inTimeStamp: UnsafePointer<AudioTimeStamp>, inBusNumber: UInt32, inNumberFrames: UInt32, ioData: UnsafeMutablePointer<AudioBufferList>? ) -> OSStatus in guard let ioData = ioData else { return noErr } print("Output callback invoked") // Process the output data if needed return noErr } In essence, I'm stuck and in need of guidance. Has anyone here successfully implemented multi-channel recording on iOS, especially involving both built-in microphones and external audio interfaces? Any shared experiences, insights, or suggestions on how to proceed would be immensely appreciated. Thank you once again for your time and assistance!
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1.2k
Jul ’24
How to show VoIP calls on Apple Watch (WatchOS9) with CallKit?
Now my main app is already invoked voip callkit, I would want to invoke voip on iWatch app, but I have some issue: 1、How to deal audio data and network connect of iWatch voip ? can we depends on iPhone app? 2、How to use voip callkit on iWatch that only via bluetooth connect ? 3、If main app is already support voip callkit, how to support callkit for iwatch? Do we need to repeat and independently implement callkit, network, and audio on iWatch? 4、how to add support dial number on iWatch use by the thirdpart app? the case is the same with on the iPhone use, user can send dial by system call record . Any help is appreciated, thanks in advance.
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Jul ’24
Help Needed with AVAudioSession in Unity for Consistent Sound Output on iOS Devices
Hello, I hope this message finds you well. I am currently working on a Unity-based iOS application that requires continuous microphone input while also producing sound outputs. For this we need to use iOS echo cancellation, so some sounds need to be played via the iOS layer w/ echo cancellation, I am manually setting up the Audio Session after the app starts. Using the .playAndRecord mode of AVAudioSession. However, I am facing an issue where the volume of the sound output is inconsistent across different iOS devices and scenarios. The process is quite simple, for each AudioClip we are about to play via unity, we copy the buffer data to our iOS Swift layer, which then does all the processing then plays the audio via the native layer. Here are the specific issues I am encountering: The volume level for the game sound effects fluctuate between a normal audible volume and a very low volume. The sound output behaves differently depending on whether the app is launched with the device at full volume or on mute, and if the app is put into background and in foreground afterwards. The volume inconsistency affects my game negatively, as it is very hard to hear some audios, regardless of the device or its initial volume state. I have followed the basic setup for AVAudioSession as per the documentation, but the inconsistencies persist. I'm also aware that Unity uses FMOD to set up the audio routing in iOS, we configure our custom routing after that. We tried tweaking the output volume prior to playing an audio so there isn't much discrepancy, this seems to align the output volume, however there is still some places where the volume is super low, I've looked into the waveforms in Unity and they all seem consistent, there is no reason why the volume would take a dip. private var audioPlayer = AVAudioPlayerNode() @objc public func Play() { audioPlayer.volume = AVAudioSession.sharedInstance().outputVolume * 0.25 audioPlayer.play() } We also explored changing the audio session options to see if we had any luck but unfortunately nothing has changed. private func ConfigAudioSession() { let audioSession = AVAudioSession.sharedInstance(); do { try audioSession.setCategory(.playAndRecord, options: [.mixWithOthers, .allowBluetooth, .defaultToSpeaker]); try audioSession.setMode(.spokenAudio) try audioSession.setActive(true); } catch { //Treat error } } Could anyone provide guidance or suggest best practices to ensure a stable and consistent volume output in this scenario? Any advice on this issue would be greatly appreciated. Thank you in advance for your help!
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1.2k
Jul ’24
iOS18 web audio lock foucs and not release
在我们App中,打开一个H5页面,使用webplayer播放H5中的视频。 然后再去播放App的播放器,播放视频、或音频文件, 都存在抢不到音频焦点问题,声音响一下就停了,播放器还在运行。 尝试在每次App播放都先调用setCategory、setActive也不生效。 这个问题,在beta1~beta3都存在。 请问,webkit的 player做了什么处理,会一直锁定着音频焦点,App要怎么处理才能把焦点拿过来? In our App, open an H5 page and use webplayer to play the video in H5. Then go to the PlayApp player to play the video or audio file. There is a problem of not being able to grab the audio focus. The sound stops as soon as it sounds, but the player is still running. Trying to call setCategory and setActive every time in AppPlay does not work either. This problem exists in beta1~beta3. I would like to ask, what processing has been done by the webkit player to keep the audio focus locked? How can the app handle it so that it can take the focus?
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Jul ’24
Device Volume Changes After Setting AVAudioSession Category
Hi there, I am encountering an issue in my project which utilizes a speech recognizer and occasionally plays audio files. The problem arises when I configure the AVAudioSession and enable voice processing. The system volume changes unexpectedly and becomes uncontrollable. Specifically, the volume is excessively loud on iPhone but quite low on iPad let audioSession = AVAudioSession.sharedInstance() try audioSession.setCategory(.playAndRecord, mode: .default, options: [.defaultToSpeaker, .allowBluetooth, .interruptSpokenAudioAndMixWithOthers]) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) try audioEngine.inputNode.setVoiceProcessingEnabled(true) try audioEngine.outputNode.setVoiceProcessingEnabled(true) I have provided a sample project here: Sample Project. To reproduce the issue, please follow these steps on a real device: Click on "Play recording" to hear the sound at normal volume. Click on "Start recording" to set up the category and speech recognizer. Click on "Stop recording" to stop the recording. Click on "Play recording" again and observe that the sound volume has changed. Thank you for your assistance.
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1.2k
Jun ’24
USB microphone with high samplerate and AVAudioEngine
Hello, I can't get my head wrapped around the following problem: I have an external USB microphone capable of samplerates of up to 500 kHz. I want to capture the samples and do analysis and display - no playback required. I can not find a way to run the microphone with its maximum samplerate, I always get 48 kHz. I would like to stick to AVAudioEngine if possible. Any pointer welcome. thx! volker
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1.2k
May ’24
Voice Processing in multiple apps simultaneously
Hi everyone! We are wondering whether it's possible to have two macOS apps use the Voice Processing from Audio Engine at the same time, since we have had issues trying to do so. Specifically, our app seems to cut off the input stream from the other, only if it has Voice Processing enabled. We are developing a macOS app that records microphone input simultaneously with videoconference apps like Zoom. We are utilizing the Voice Processing from Audio Engine like in this sample: https://developer.apple.com/documentation/avfaudio/audio_engine/audio_units/using_voice_processing We have also noticed this behaviour in Safari recording audios with the Javascript Web Audio API, which also seems to use Voice Processing under the hood due to the Echo Cancellation. Any leads on this would be greatly appreciated! Thanks
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781
May ’24
watchOS: Resume recording from AudioInterruption in background mode
Hi, I have a watchOS app that records audio for an extended period of time and because the mic is active, continues to record in background mode when the watch face is off. However, when a call comes in or Siri is activated, recording stops because of an audio interruption. Here is my code for setting up the session: private func setupAudioSession() { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .default, options: [.overrideMutedMicrophoneInterruption]) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) } catch { print("Audio Session error: \(error)") } } Before this I register an interruption handler that holds a reference to my AudioEngine (which I start and stop each time recording is activated by the user): _audioInterruptionHandler = AudioInterruptionHandler(audioEngine: _audioEngine) And here is how this class implements recovery: fileprivate class AudioInterruptionHandler { private let _audioEngine: AVAudioEngine public init(audioEngine: AVAudioEngine) { _audioEngine = audioEngine // Listen to interrupt notifications NotificationCenter.default.addObserver(self, selector: #selector(handleAudioInterruption(notification:)), name: AVAudioSession.interruptionNotification, object: nil) } @objc private func handleAudioInterruption(notification: Notification) { guard let userInfo = notification.userInfo, let interruptionTypeRawValue = userInfo[AVAudioSessionInterruptionTypeKey] as? UInt, let interruptionType = AVAudioSession.InterruptionType(rawValue: interruptionTypeRawValue) else { return } switch interruptionType { case .began: print("[AudioInterruptionHandler] Interruption began") case .ended: print("[AudioInterruptionHandler] Interruption ended") print("Interruption ended") do { try AVAudioSession.sharedInstance().setActive(true) } catch { print("[AudioInterruptionHandler] Error resuming audio session: \(error.localizedDescription)") } default: print("[AudioInterruptionHandler] Unknown interruption: \(interruptionType.rawValue)") } } } Unfortunately, it fails with: Error resuming audio session: Session activation failed Is this even possible to do on watchOS? This code worked for me on iOS. Thank you, -- B.
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1.2k
Apr ’24
How to use Swift and AVFoundation to stream/record USB microphone input?
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?
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2.3k
Apr ’24
Other Audio Ducking in AVAudio session
https://developer.apple.com/videos/play/wwdc2023/10235/ - In this WWDC session, at 3:19 - Apple has introduced **Other audio ducking ** feature In iOS17, we can control the amount of 'other audio' ducking through the AVAudioEngine. Is this also possible on AVAudioSession ? We are using an AVAudioSession for a VOIP call while concurrently attempting to play a video through an AVPlayer. However, the volume of the AVPlayer is considerably low. Does anyone have any ideas on how to achieve the level of control that AVAudioEngine offers?
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Apr ’24
AVAudioEngine: Is there a way to play audio at full volume while having an active input tap?
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
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1.3k
Mar ’24
How to Add .sf2 Instruments to Multiple Channels Using AVFAudio?
Hello everyone, I'm relatively new to iOS development, and I'm currently working on a Flutter plugin package. I want to use the AVFAudio package to load instrument sounds from an SF2 file into different channels. Specifically, I'd like to load individual instruments from the SF2 file onto separate channels. However, I've been struggling to find a way to achieve this. Could someone guide me on how to load SF2 instrument sounds into different channels using AVFAudio? I've tried various combinations of parameters (program number, soundbank MSB, and soundbank LSB), but none seem to work. If anyone has experience with AVFAudio and SF2 files, I'd greatly appreciate your help. Perhaps there's a proven approach or a way to determine the correct values for these parameters? Should I use a soundfont editor to inspect specific values within the SF2 file? Thank you in advance for any assistance! Best regards, Melih
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1.4k
Mar ’24
USB audio multi route input - AVAudioEngine inputNode
Hi everybody, I'm trying to use the multi input of an usb device using the AVAudioEngine. My aim is to connect different inputNode channels to 2 or more different audionode (f.e. mixer). I'm able to get a spefic input channel from the engine inputNode with OSStatus err = AudioUnitSetProperty(avEngine.inputNode.audioUnit, kAudioOutputUnitProperty_ChannelMap, kAudioUnitScope_Output, 1, outputChannelMap, propSize); but this will change the routing to all the input node and to all the destination mixer nodes. How to send channel 1 of inputNode to a mixerNode1 and channel 2 to another mixerNode2?
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969
Feb ’24
How to obtain an AVAudioFormat for a canonical format?
I receive a buffer from[AVSpeechSynthesizer convertToBuffer:fromBuffer:] and want to schedule it on an AVPlayerNode. The player node's output format need to be something that the next node could handle and as far as I understand most nodes can handle a canonical format. The format provided by AVSpeechSynthesizer is not something thatAVAudioMixerNode supports. So the following:   AVAudioEngine *engine = [[AVAudioEngine alloc] init];   playerNode = [[AVAudioPlayerNode alloc] init];   AVAudioFormat *format = [[AVAudioFormat alloc] initWithSettings:utterance.voice.audioFileSettings];   [engine attachNode:self.playerNode];   [engine connect:self.playerNode to:engine.mainMixerNode format:format]; Throws an exception: Thread 1: "[[busArray objectAtIndexedSubscript:(NSUInteger)element] setFormat:format error:&nsErr]: returned false, error Error Domain=NSOSStatusErrorDomain Code=-10868 \"(null)\"" I am looking for a way to obtain the canonical format for the platform so that I can use AVAudioConverter to convert the buffer. Since different platforms have different canonical formats, I imagine there should be some library way of doing this. Otherwise each developer will have to redefine it for each platform the code will run on (OSX, iOS etc) and keep it updated when it changes. I could not find any constant or function which can make such format, ASDB or settings. The smartest way I could think of, which does not work:   AudioStreamBasicDescription toDesc;   FillOutASBDForLPCM(toDesc, [AVAudioSession sharedInstance].sampleRate,                      2, 16, 16, kAudioFormatFlagIsFloat, kAudioFormatFlagsNativeEndian);   AVAudioFormat *toFormat = [[AVAudioFormat alloc] initWithStreamDescription:&toDesc]; Even the provided example for iPhone, in the documentation linked above, uses kAudioFormatFlagsAudioUnitCanonical and AudioUnitSampleType which are deprecated. So what is the correct way to do this?
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2.8k
Feb ’24
Using AVAudioEngine to monitor audio in VisionOS
I am writing code to monitor the incoming audio levels in VisionOS. It works properly in the simulator, but gets an error on the device. Curious if anyone has any tips. I took out some of the code so it's a bit shorter, as it fails in setupAudioEngine when I try to start the engine with this error: Error starting audio engine: The operation couldn’t be completed. (com.apple.coreaudio.avfaudio error 561145187.) Thanks in advance! Here is my code: class AudioInputMonitor: ObservableObject { private var audioEngine: AVAudioEngine? @Published var inputLevel: Float = 0 init() { requestMicrophonePermission() } private func requestMicrophonePermission() { AVAudioApplication.requestRecordPermission { granted in DispatchQueue.main.async { if granted { self.setupAudioSessionAndEngine() } else { print("Microphone permission not granted") // Handle the case where permission is not granted } } } } private func setupAudioSessionAndEngine() { do { let audioSession = AVAudioSession.sharedInstance() try audioSession.setCategory(.playAndRecord, mode: .measurement, options: []) try audioSession.setActive(true) self.setupAudioEngine() } catch { print("Failed to set up the audio session: \(error)") } } private func setupAudioEngine() { audioEngine = AVAudioEngine() guard let inputNode = audioEngine?.inputNode else { print("Failed to get the audio input node") return } let recordingFormat = inputNode.outputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { [weak self] (buffer, _) in self?.analyzeAudio(buffer: buffer) } do { try audioEngine?.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } private func analyzeAudio(buffer: AVAudioPCMBuffer) { // removed to be brief } func stopMonitoring() { // removed to be brief } }
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1.7k
Feb ’24
Configuring Apple Vision Pro's microphones to effectively pick up other speaker's voice
I am developing a visionOS app that captions speech in real environments. Currently, I am using Apple's built-in speech recognizer. However, when I was testing the app with a Vision Pro, the device seemed to only pick up the user's voice (in other words, the voices of the wearer of the Vision Pro device). For example, when the speech recognition task is running, and another person in front of me is talking, the system does not pick up the speech well. I tried to set the AVAudioSession to be equally sensitive to all directions: private func configureAudioSession() { do { try audioSession.setCategory(.record, mode: .measurement) try audioSession.setActive(true) if #available(visionOS 1.0, *) { let availableDataSources = audioSession.availableInputs?.first?.dataSources if let omniDirectionalSource = availableDataSources?.first(where: {$0.preferredPolarPattern == .omnidirectional}) { try audioSession.setInputDataSource(omniDirectionalSource) } } } catch { print("Failed to set up audio session: \(error)") } } And here is how I set up the speech recognition and configure the microphone inputs: private func startSpeechRecognition(completion: @escaping (String) -> Void) { do { // Cancel the previous task if it's running. if let recognitionTask = recognitionTask { recognitionTask.cancel() self.recognitionTask = nil } // The AudioSession is already active, creating input node. let inputNode = audioEngine.inputNode try inputNode.setVoiceProcessingEnabled(false) // Create and configure the speech recognition request recognitionRequest = SFSpeechAudioBufferRecognitionRequest() guard let recognitionRequest = recognitionRequest else { fatalError("Unable to create a recognition request") } recognitionRequest.shouldReportPartialResults = true // Keep speech recognition data on device if #available(iOS 13, *) { recognitionRequest.requiresOnDeviceRecognition = true } // Create a recognition task for speech recognition session. // Keep a reference to the task so that it can be canceled. recognitionTask = speechRecognizer?.recognitionTask(with: recognitionRequest) { result, error in // var isFinal = false if let result = result { // Update the recognizedText completion(result.bestTranscription.formattedString) } else if let error = error { completion("Recognition error: \(error.localizedDescription)") } if error != nil || result?.isFinal == true { // Stop recognizing speech if there is a problem self.audioEngine.stop() inputNode.removeTap(onBus: 0) self.recognitionRequest = nil self.recognitionTask = nil } } // Configure the microphone input let recordingFormat = inputNode.outputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { (buffer, when) in self.recognitionRequest?.append(buffer) } audioEngine.prepare() try audioEngine.start() } catch { completion("Audio engine could not start: \(error.localizedDescription)") } }
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1.3k
Activity
Jul ’24
Setting Audio Input node for AVAudioEngine causes outside audio to stop
I'm building an app that will allow users to record voice notes. The functionality of all that is working great; I'm trying to now implement changes to the AudioSession to manage possible audio streams from other apps. I want it so that if there is audio playing from a different app, and the user opens my app; the audio keep playing. When we start recording, any third party app audio should stop, and can then can resume again when we stop recording. This is my main audio setup code: private var audioEngine: AVAudioEngine! private var inputNode: AVAudioInputNode! func setupAudioEngine() { audioEngine = AVAudioEngine() inputNode = audioEngine.inputNode audioPlayerNode = AVAudioPlayerNode() audioEngine.attach(audioPlayerNode) let format = AVAudioFormat(standardFormatWithSampleRate: AUDIO_SESSION_SAMPLE_RATE, channels: 1) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: format) } private func setupAudioSession() { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .default, options: [.defaultToSpeaker, .allowBluetooth]) try audioSession.setPreferredSampleRate(AUDIO_SESSION_SAMPLE_RATE) try audioSession.setPreferredIOBufferDuration(0.005) // 5ms buffer for lower latency try audioSession.setActive(true) // Add observers setupInterruptionObserver() } catch { audioErrorMessage = "Failed to set up audio session: \(error)" } } This is all called upon app startup so we're ready to record whenever the user presses the record button. However, currently when this happens, any outside audio stops playing. I isolated the issue to this line: inputNode = audioEngine.inputNode When that's commented out, the audio will play -- but I obviously need this for recording functionality. Is this a bug? Expected behavior?
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819
Activity
Jul ’24
Implementing Multi-Channel Audio Recording on iOS with Built-In and External Mics
Hi there community, First and foremost, a big thank you to everyone who takes the time to read this. TL;DR: How, if even possible, can I record multiple audio streams simultaneously on an iOS application (iPad/iPhone)? I'm working on a recorder for the iPad to gather data for a machine learning project focused on speech recognition. Our goal is to capture extensive speech data, which requires recording from multiple microphones. Specifically, I need to record from all mics connected to our Scarlett 4i4 audio interface and, most importantly, also record from the built-in mic on the iPad or iPhone at the same time. As a newcomer to Swift development, I initially explored AVAudioRecorder. However, I quickly realized that it only supports one active audio node at a time, making multi-channel recording impossible. (perhaps you can proof me wrong, would make my day) Next, I transitioned to using AVAudioEngine, but encountered the same limitation: I couldn't manage to get input nodes for both the built-in mic and the Scarlett interface channels simultaneously. The application started behaving oddly, often resulting in identical audio data being recorded across all files. Determined to find a solution, I delved deeper into the Core Audio framework, specifically using Audio Toolbox. My approach involved creating and configuring multiple Audio Units, each corresponding to a different audio input device. Here's a brief overview of my current implementation: Listing Available Input Devices: I used AVAudioSession to enumerate all available input devices. Creating Audio Units: For each device, I created an Audio Unit and attempted to configure it for recording. Setting Up Callbacks: I set up input and output callbacks to handle the audio processing. Despite my efforts over the last few days, I haven't had much success. The callbacks for the Audio Units don't seem to be invoked correctly, and I'm struggling to achieve simultaneous multi-channel recording. Below is a snippet of my latest attempt: let audioUnitCallback: AURenderCallback = { ( inRefCon: UnsafeMutableRawPointer, ioActionFlags: UnsafeMutablePointer<AudioUnitRenderActionFlags>, inTimeStamp: UnsafePointer<AudioTimeStamp>, inBusNumber: UInt32, inNumberFrames: UInt32, ioData: UnsafeMutablePointer<AudioBufferList>? ) -> OSStatus in guard let ioData = ioData else { return noErr } print("Input callback invoked") let audioUnit = inRefCon.assumingMemoryBound(to: AudioUnit.self).pointee var bufferList = AudioBufferList( mNumberBuffers: 1, mBuffers: AudioBuffer( mNumberChannels: 1, mDataByteSize: 0, mData: nil ) ) let status = AudioUnitRender(audioUnit, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList) if status != noErr { print("AudioUnitRender failed: \(status)") return status } // Copy rendered data to output buffer let buffer = UnsafeMutableAudioBufferListPointer(ioData)[0] buffer.mData?.copyMemory(from: bufferList.mBuffers.mData!, byteCount: Int(bufferList.mBuffers.mDataByteSize)) buffer.mDataByteSize = bufferList.mBuffers.mDataByteSize print("Rendered audio data") return noErr } let outputCallback: AURenderCallback = { ( inRefCon: UnsafeMutableRawPointer, ioActionFlags: UnsafeMutablePointer<AudioUnitRenderActionFlags>, inTimeStamp: UnsafePointer<AudioTimeStamp>, inBusNumber: UInt32, inNumberFrames: UInt32, ioData: UnsafeMutablePointer<AudioBufferList>? ) -> OSStatus in guard let ioData = ioData else { return noErr } print("Output callback invoked") // Process the output data if needed return noErr } In essence, I'm stuck and in need of guidance. Has anyone here successfully implemented multi-channel recording on iOS, especially involving both built-in microphones and external audio interfaces? Any shared experiences, insights, or suggestions on how to proceed would be immensely appreciated. Thank you once again for your time and assistance!
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Jul ’24
How to show VoIP calls on Apple Watch (WatchOS9) with CallKit?
Now my main app is already invoked voip callkit, I would want to invoke voip on iWatch app, but I have some issue: 1、How to deal audio data and network connect of iWatch voip ? can we depends on iPhone app? 2、How to use voip callkit on iWatch that only via bluetooth connect ? 3、If main app is already support voip callkit, how to support callkit for iwatch? Do we need to repeat and independently implement callkit, network, and audio on iWatch? 4、how to add support dial number on iWatch use by the thirdpart app? the case is the same with on the iPhone use, user can send dial by system call record . Any help is appreciated, thanks in advance.
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Jul ’24
Help Needed with AVAudioSession in Unity for Consistent Sound Output on iOS Devices
Hello, I hope this message finds you well. I am currently working on a Unity-based iOS application that requires continuous microphone input while also producing sound outputs. For this we need to use iOS echo cancellation, so some sounds need to be played via the iOS layer w/ echo cancellation, I am manually setting up the Audio Session after the app starts. Using the .playAndRecord mode of AVAudioSession. However, I am facing an issue where the volume of the sound output is inconsistent across different iOS devices and scenarios. The process is quite simple, for each AudioClip we are about to play via unity, we copy the buffer data to our iOS Swift layer, which then does all the processing then plays the audio via the native layer. Here are the specific issues I am encountering: The volume level for the game sound effects fluctuate between a normal audible volume and a very low volume. The sound output behaves differently depending on whether the app is launched with the device at full volume or on mute, and if the app is put into background and in foreground afterwards. The volume inconsistency affects my game negatively, as it is very hard to hear some audios, regardless of the device or its initial volume state. I have followed the basic setup for AVAudioSession as per the documentation, but the inconsistencies persist. I'm also aware that Unity uses FMOD to set up the audio routing in iOS, we configure our custom routing after that. We tried tweaking the output volume prior to playing an audio so there isn't much discrepancy, this seems to align the output volume, however there is still some places where the volume is super low, I've looked into the waveforms in Unity and they all seem consistent, there is no reason why the volume would take a dip. private var audioPlayer = AVAudioPlayerNode() @objc public func Play() { audioPlayer.volume = AVAudioSession.sharedInstance().outputVolume * 0.25 audioPlayer.play() } We also explored changing the audio session options to see if we had any luck but unfortunately nothing has changed. private func ConfigAudioSession() { let audioSession = AVAudioSession.sharedInstance(); do { try audioSession.setCategory(.playAndRecord, options: [.mixWithOthers, .allowBluetooth, .defaultToSpeaker]); try audioSession.setMode(.spokenAudio) try audioSession.setActive(true); } catch { //Treat error } } Could anyone provide guidance or suggest best practices to ensure a stable and consistent volume output in this scenario? Any advice on this issue would be greatly appreciated. Thank you in advance for your help!
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Jul ’24
iOS18 web audio lock foucs and not release
在我们App中,打开一个H5页面,使用webplayer播放H5中的视频。 然后再去播放App的播放器,播放视频、或音频文件, 都存在抢不到音频焦点问题,声音响一下就停了,播放器还在运行。 尝试在每次App播放都先调用setCategory、setActive也不生效。 这个问题,在beta1~beta3都存在。 请问,webkit的 player做了什么处理,会一直锁定着音频焦点,App要怎么处理才能把焦点拿过来? In our App, open an H5 page and use webplayer to play the video in H5. Then go to the PlayApp player to play the video or audio file. There is a problem of not being able to grab the audio focus. The sound stops as soon as it sounds, but the player is still running. Trying to call setCategory and setActive every time in AppPlay does not work either. This problem exists in beta1~beta3. I would like to ask, what processing has been done by the webkit player to keep the audio focus locked? How can the app handle it so that it can take the focus?
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Jul ’24
Device Volume Changes After Setting AVAudioSession Category
Hi there, I am encountering an issue in my project which utilizes a speech recognizer and occasionally plays audio files. The problem arises when I configure the AVAudioSession and enable voice processing. The system volume changes unexpectedly and becomes uncontrollable. Specifically, the volume is excessively loud on iPhone but quite low on iPad let audioSession = AVAudioSession.sharedInstance() try audioSession.setCategory(.playAndRecord, mode: .default, options: [.defaultToSpeaker, .allowBluetooth, .interruptSpokenAudioAndMixWithOthers]) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) try audioEngine.inputNode.setVoiceProcessingEnabled(true) try audioEngine.outputNode.setVoiceProcessingEnabled(true) I have provided a sample project here: Sample Project. To reproduce the issue, please follow these steps on a real device: Click on "Play recording" to hear the sound at normal volume. Click on "Start recording" to set up the category and speech recognizer. Click on "Stop recording" to stop the recording. Click on "Play recording" again and observe that the sound volume has changed. Thank you for your assistance.
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Jun ’24
USB microphone with high samplerate and AVAudioEngine
Hello, I can't get my head wrapped around the following problem: I have an external USB microphone capable of samplerates of up to 500 kHz. I want to capture the samples and do analysis and display - no playback required. I can not find a way to run the microphone with its maximum samplerate, I always get 48 kHz. I would like to stick to AVAudioEngine if possible. Any pointer welcome. thx! volker
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May ’24
Voice Processing in multiple apps simultaneously
Hi everyone! We are wondering whether it's possible to have two macOS apps use the Voice Processing from Audio Engine at the same time, since we have had issues trying to do so. Specifically, our app seems to cut off the input stream from the other, only if it has Voice Processing enabled. We are developing a macOS app that records microphone input simultaneously with videoconference apps like Zoom. We are utilizing the Voice Processing from Audio Engine like in this sample: https://developer.apple.com/documentation/avfaudio/audio_engine/audio_units/using_voice_processing We have also noticed this behaviour in Safari recording audios with the Javascript Web Audio API, which also seems to use Voice Processing under the hood due to the Echo Cancellation. Any leads on this would be greatly appreciated! Thanks
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781
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May ’24
watchOS: Resume recording from AudioInterruption in background mode
Hi, I have a watchOS app that records audio for an extended period of time and because the mic is active, continues to record in background mode when the watch face is off. However, when a call comes in or Siri is activated, recording stops because of an audio interruption. Here is my code for setting up the session: private func setupAudioSession() { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .default, options: [.overrideMutedMicrophoneInterruption]) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) } catch { print("Audio Session error: \(error)") } } Before this I register an interruption handler that holds a reference to my AudioEngine (which I start and stop each time recording is activated by the user): _audioInterruptionHandler = AudioInterruptionHandler(audioEngine: _audioEngine) And here is how this class implements recovery: fileprivate class AudioInterruptionHandler { private let _audioEngine: AVAudioEngine public init(audioEngine: AVAudioEngine) { _audioEngine = audioEngine // Listen to interrupt notifications NotificationCenter.default.addObserver(self, selector: #selector(handleAudioInterruption(notification:)), name: AVAudioSession.interruptionNotification, object: nil) } @objc private func handleAudioInterruption(notification: Notification) { guard let userInfo = notification.userInfo, let interruptionTypeRawValue = userInfo[AVAudioSessionInterruptionTypeKey] as? UInt, let interruptionType = AVAudioSession.InterruptionType(rawValue: interruptionTypeRawValue) else { return } switch interruptionType { case .began: print("[AudioInterruptionHandler] Interruption began") case .ended: print("[AudioInterruptionHandler] Interruption ended") print("Interruption ended") do { try AVAudioSession.sharedInstance().setActive(true) } catch { print("[AudioInterruptionHandler] Error resuming audio session: \(error.localizedDescription)") } default: print("[AudioInterruptionHandler] Unknown interruption: \(interruptionType.rawValue)") } } } Unfortunately, it fails with: Error resuming audio session: Session activation failed Is this even possible to do on watchOS? This code worked for me on iOS. Thank you, -- B.
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Apr ’24
How to use Swift and AVFoundation to stream/record USB microphone input?
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?
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Apr ’24
Other Audio Ducking in AVAudio session
https://developer.apple.com/videos/play/wwdc2023/10235/ - In this WWDC session, at 3:19 - Apple has introduced **Other audio ducking ** feature In iOS17, we can control the amount of 'other audio' ducking through the AVAudioEngine. Is this also possible on AVAudioSession ? We are using an AVAudioSession for a VOIP call while concurrently attempting to play a video through an AVPlayer. However, the volume of the AVPlayer is considerably low. Does anyone have any ideas on how to achieve the level of control that AVAudioEngine offers?
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Apr ’24
How to get realtime headphone levels in decibel unit
Hello, I am currently developing an application for audiogram testing. What methods can I use to obtain the dB values of headphone levels in real-time?
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Apr ’24
AVAudioEngine: Is there a way to play audio at full volume while having an active input tap?
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
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Mar ’24
How to Add .sf2 Instruments to Multiple Channels Using AVFAudio?
Hello everyone, I'm relatively new to iOS development, and I'm currently working on a Flutter plugin package. I want to use the AVFAudio package to load instrument sounds from an SF2 file into different channels. Specifically, I'd like to load individual instruments from the SF2 file onto separate channels. However, I've been struggling to find a way to achieve this. Could someone guide me on how to load SF2 instrument sounds into different channels using AVFAudio? I've tried various combinations of parameters (program number, soundbank MSB, and soundbank LSB), but none seem to work. If anyone has experience with AVFAudio and SF2 files, I'd greatly appreciate your help. Perhaps there's a proven approach or a way to determine the correct values for these parameters? Should I use a soundfont editor to inspect specific values within the SF2 file? Thank you in advance for any assistance! Best regards, Melih
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Mar ’24
USB audio multi route input - AVAudioEngine inputNode
Hi everybody, I'm trying to use the multi input of an usb device using the AVAudioEngine. My aim is to connect different inputNode channels to 2 or more different audionode (f.e. mixer). I'm able to get a spefic input channel from the engine inputNode with OSStatus err = AudioUnitSetProperty(avEngine.inputNode.audioUnit, kAudioOutputUnitProperty_ChannelMap, kAudioUnitScope_Output, 1, outputChannelMap, propSize); but this will change the routing to all the input node and to all the destination mixer nodes. How to send channel 1 of inputNode to a mixerNode1 and channel 2 to another mixerNode2?
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Feb ’24
AVAudioEngine in Practice WWDC 2014 Session Video ?
I'm looking for the AVAudioEngine in Practice video from WWDC 2014 (session 502) but can't seem to find it anywhere. Does anyone have a link to this session video? I can only find the slides. Thanks.
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Feb ’24
How to obtain an AVAudioFormat for a canonical format?
I receive a buffer from[AVSpeechSynthesizer convertToBuffer:fromBuffer:] and want to schedule it on an AVPlayerNode. The player node's output format need to be something that the next node could handle and as far as I understand most nodes can handle a canonical format. The format provided by AVSpeechSynthesizer is not something thatAVAudioMixerNode supports. So the following:   AVAudioEngine *engine = [[AVAudioEngine alloc] init];   playerNode = [[AVAudioPlayerNode alloc] init];   AVAudioFormat *format = [[AVAudioFormat alloc] initWithSettings:utterance.voice.audioFileSettings];   [engine attachNode:self.playerNode];   [engine connect:self.playerNode to:engine.mainMixerNode format:format]; Throws an exception: Thread 1: "[[busArray objectAtIndexedSubscript:(NSUInteger)element] setFormat:format error:&nsErr]: returned false, error Error Domain=NSOSStatusErrorDomain Code=-10868 \"(null)\"" I am looking for a way to obtain the canonical format for the platform so that I can use AVAudioConverter to convert the buffer. Since different platforms have different canonical formats, I imagine there should be some library way of doing this. Otherwise each developer will have to redefine it for each platform the code will run on (OSX, iOS etc) and keep it updated when it changes. I could not find any constant or function which can make such format, ASDB or settings. The smartest way I could think of, which does not work:   AudioStreamBasicDescription toDesc;   FillOutASBDForLPCM(toDesc, [AVAudioSession sharedInstance].sampleRate,                      2, 16, 16, kAudioFormatFlagIsFloat, kAudioFormatFlagsNativeEndian);   AVAudioFormat *toFormat = [[AVAudioFormat alloc] initWithStreamDescription:&toDesc]; Even the provided example for iPhone, in the documentation linked above, uses kAudioFormatFlagsAudioUnitCanonical and AudioUnitSampleType which are deprecated. So what is the correct way to do this?
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Feb ’24
Using AVAudioEngine to monitor audio in VisionOS
I am writing code to monitor the incoming audio levels in VisionOS. It works properly in the simulator, but gets an error on the device. Curious if anyone has any tips. I took out some of the code so it's a bit shorter, as it fails in setupAudioEngine when I try to start the engine with this error: Error starting audio engine: The operation couldn’t be completed. (com.apple.coreaudio.avfaudio error 561145187.) Thanks in advance! Here is my code: class AudioInputMonitor: ObservableObject { private var audioEngine: AVAudioEngine? @Published var inputLevel: Float = 0 init() { requestMicrophonePermission() } private func requestMicrophonePermission() { AVAudioApplication.requestRecordPermission { granted in DispatchQueue.main.async { if granted { self.setupAudioSessionAndEngine() } else { print("Microphone permission not granted") // Handle the case where permission is not granted } } } } private func setupAudioSessionAndEngine() { do { let audioSession = AVAudioSession.sharedInstance() try audioSession.setCategory(.playAndRecord, mode: .measurement, options: []) try audioSession.setActive(true) self.setupAudioEngine() } catch { print("Failed to set up the audio session: \(error)") } } private func setupAudioEngine() { audioEngine = AVAudioEngine() guard let inputNode = audioEngine?.inputNode else { print("Failed to get the audio input node") return } let recordingFormat = inputNode.outputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { [weak self] (buffer, _) in self?.analyzeAudio(buffer: buffer) } do { try audioEngine?.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } private func analyzeAudio(buffer: AVAudioPCMBuffer) { // removed to be brief } func stopMonitoring() { // removed to be brief } }
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Feb ’24
Is there an available API that can be used to get a sense of voice recognition?
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)
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Jan ’24