Why voice processing enabled on AVAudioInputNode makes output audio noticable lower than without it and how to overcome it using voice processing enabled

What's the best way of dealing with presentation mode (when the phone screen is captured by Google Meet or any other presenter software) when AudioSession is activated with .playAndRecord mode? The goal is to share the audio output between the phone and Google Meet so participants can hear it as well, while also enabling a microphone for future interactions within the app, for, let's say, voice chat.

What's the best way of dealing with AVAudioInputNode from resources perspective? Ideally, to install it as soon as it is necessary for microphone access and use it as long as it is necessary (for example, when you have a long voice chat where users exchange voice messages intermittently AVAudioInputNode should be installed at the beginning and kept until the chat ends) Or is it better to setup it inputNode.installTap( onBus: bus, bufferSize: bufferSize, format: format, block: { buffer, _ in } ) everytime user accessing microphone with push to talk The question is what's better from a performance perspective

What's the best way of detecting if player node / audio engine is in broken state (for example as a result of AVAudioSession.mediaServicesWereResetNotification). Sometimes it requires being reset, but from time to time it is not enought to just reset it for example playerNode.reset() engine.reset() But also required to reinitialize it playerNode.reset() playerNode = .init() engine.reset() engine = .init() Can we subscribe on the broken states and reinitialize them proactively?

Looks like dispatch has been broken