I am analysing sounds by tapping the mic on the Mac. All is working well, but it disrupts other (what I assume) are low priority sounds e.g. dragging an item off the dock, sending a message is messages, speaking something in Shortcuts or Terminal. Other sounds like music.app playing, Siri speaking are not disrupted. The disruption sounds like the last part of the sound being repeated two extra times, very noticeable. This is the code: import Cocoa import AVFAudio class AudioHelper: NSObject { let audioEngine = AVAudioEngine() func start() async throws { audioEngine.inputNode.installTap(onBus: 0, bufferSize: 8192, format: nil) { buffer, time in } try audioEngine.start() } } I have tried increasing the buffer, changing the qos to utility (in the hope the sound analysis would become less important than the disrupted sounds),running on a non-main thread, but no luck. MacOS 13.4.1 Any assistance would be appreciated.

I am using AVSpeechSynthesizer to get audio buffer and play, I am using AVAudioEngine and AVAudioPlayerNode to play the buffer. But I am getting error. [avae] AVAEInternal.h:76 required condition is false: [AVAudioPlayerNode.mm:734:ScheduleBuffer: (_outputFormat.channelCount == buffer.format.channelCount)] 2023-05-02 03:14:35.709020-0700 AudioPlayer[12525:308940] *** Terminating app due to uncaught exception 'com.apple.coreaudio.avfaudio', reason: 'required condition is false: _outputFormat.channelCount == buffer.format.channelCount' Can anyone please help me to play the AVAudioBuffer from AVSpeechSynthesizer write method?

I work on a video conferencing application, which makes use of AVAudioEngine and the videoChat AVAudioSession.Mode This past Friday, an internal user reported an "audio cutting in and out" issue with their new iPhone 14 Pro, and I was able to reproduce the issue later that day on my iPhone 14 Pro Max. No other iOS devices running iOS 16 are exhibiting this issue. I have narrowed down the root cause to the videoChat AVAudioSession.Mode after changing line 53 of the ViewController.swift file in Apple's "Using Voice Processing" sample project (https://developer.apple.com/documentation/avfaudio/audio_engine/audio_units/using_voice_processing) from: try session.setCategory(.playAndRecord, options: .defaultToSpeaker) to try session.setCategory(.playAndRecord, mode: .videoChat, options: .defaultToSpeaker) This only causes issues on my iPhone 14 Pro Max device, not on my iPhone 13 Pro Max, so it seems specific to the new iPhones only. I am also seeing the following logged to the console using either device, which appears to be specific to iOS 16, but am not sure if it is related to the videoChat issue or not: 2022-09-19 08:23:20.087578-0700 AVEchoTouch[2388:1474002] [as] ATAudioSessionPropertyManager.mm:71  Invalid input size for property 1684431725 2022-09-19 08:23:20.087605-0700 AVEchoTouch[2388:1474002] [as] ATAudioSessionPropertyManager.mm:225  Invalid input size for property 1684431725 I am assuming 1684431725 is 'dfcm' but I am not sure what Audio Session Property that might be.

I've noticed that enabling voice processing on AVAudioInputNode change the node's format - most noticeably channel count. let inputNode = avEngine.inputNode print("Format #1: \(inputNode.outputFormat(forBus: 0))") // Format #1: <AVAudioFormat 0x600002bb4be0:  1 ch,  44100 Hz, Float32> try! inputNode.setVoiceProcessingEnabled(true) print("Format #2: \(inputNode.outputFormat(forBus: 0))") // Format #2: <AVAudioFormat 0x600002b18f50:  3 ch,  44100 Hz, Float32, deinterleaved> Is this expected? How can I interpret these channels? My input device is an aggregate device where each channel comes from a different microphone. I then record each channels to separate files. But when voice processing messes up with the channels layout, I cannot rely on this anymore.

I’m developing a voice communication app for the iPad with both playback and record and using AudioUnit of type kAudioUnitSubType_VoiceProcessingIO to have echo cancellation. When playing the audio before initializing the recording audio unit, volume is high. But if I'm playing the audio after initializing the audio unit or when switching to remoteio and then back to vpio the playback volume is low. It seems like a bug in iOS, any solution or workaround for this? Searching the net I only found this post without any solution: https://developer.apple.com/forums/thread/671836

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?