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

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

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.

Hi! I have a music app using AVAudioEngine. Right now, I have set it up to play multi channel tracks and show "Multichannel" in the volume controls. However, I am unable to figure out how to get it to use Dolby Atmos. Is there something that needs to be enabled? Is it even possible for AVAudioEngine? I saw some apps that are able of playing with Dolby Atmos, but they do not have EQ feature, so I'm guessing that they are not using AVAudioEngine.

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?

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?

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, I'm relatively new to iOS development and kindly ask for some feedback on a strategy to achieve this desired behavior in my app. My Question: What would be the best strategy for sound effect playback when an app is in the background with precise timing? Is this even possible? Context: I created a basic countdown timer app (targeting iOS 17 with Swift/SwiftUI.). Countdown sessions can last up to 30-60 mins. When the timer is started it progresses through a series of sub-intervals and plays a short sound for each one. I used AVAudioPlayer and everything works fine when the app is in the foreground. I'm considering switching to AVAudioEngine b/c precise timing is very important and the AIs tell me this would have better precision. I'm already setting "App plays audio or streams audio/video using AirPlay" in my Plist, and have configured: AVAudioSession.sharedInstance().setCategory(.playback, mode: .default, options: .mixWithOthers) Curiously, when testing on my iPhone 13 mini, sounds sometimes still play when the app is in the background, but not always. What I've considered: Background Tasks: Would they make any sense for this use-case? Seems like not if the allowed time is short & limited by the system. Pre-scheduling all Sounds: Not sure this would even work and seems like a lot of memory would be needed (could be hundreds of intervals). ActivityKit Alerts: works but with a ~50ms delay which is too long for my purposes. Pre-Render all SFX to 1 large audio file: Seems like a lot of work and processing time and probably not worth it. I hope there's a better solution. I'd really appreciate any feedback.

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?

I’m using AVAudioEngine to get a stream of AVAudioPCMBuffers from the device’s microphone using the usual installTap(onBus:) setup. To distribute the audio stream to other parts of the program, I’m sending the buffers to a Combine publisher similar to the following: private let publisher = PassthroughSubject<AVAudioPCMBuffer, Never>() I’m starting to suspect I have some kind of concurrency or memory management issue with the buffers, because when consuming the buffers elsewhere I’m getting a range of crashes that suggest some internal pointer in a buffer is NULL (specifically, I’m seeing crashes in vDSP.convertElements(of:to:) when I try to read samples from the buffer). These crashes are in production and fairly rare — I can’t reproduce them locally. I never modify the audio buffers, only read them for analysis. My question is: should it be possible to put AVAudioPCMBuffers into a Combine pipeline? Does the AVAudioPCMBuffer class not retain/release the underlying AudioBufferList’s memory the way I’m assuming? Is this a fundamentally flawed approach?