I am experiencing an issue where my Mac's speakers will crackle and pop when running an app on the Simulator or even when previewing SwiftUI with Live Preview. I am using a 16" MacBook Pro (i9) and I'm running Xcode 12.2 on Big Sur (11.0.1). Killing coreaudiod temporarily fixes the problem however this is not much of a solution. Is anyone else having this problem?

Does anyone have a working example on how to play OGG files with swift? I've been trying for over a year now. I was able to wrap the C Vorbis library in swift. I then used it to parse an OGG file successfully. Then I was required to use Obj-C\++ to fill the PCM because this method seems to only be available in C\++ and that part hangs my app for a good 40 seconds to several minutes depending on the audio file, it then plays for about 2 seconds and then crashes. I can't get the examples on the Vorbis site to work in objective-c and i tried every example on github I could find (most of which are for iOS - I want to play the files on mac) I also tried using Cricket Audio framework below. https://github.com/sjmerel/ck It has a swift example and it can play their proprietary soundbank format but it is also supposed to play OGG and it just doesn't do anything when trying to play OGG as you can see in the posted issue https://github.com/sjmerel/ck/issues/3 Right now I believe every player that can play OGGs on mac is written in Objective-C or C++. Anyway, any help/advice is appreciated. OGG format is very prevalent in the gaming community. I could use unity, which I believe plays oggs through the mono framework but I really really want to stay in swift.

How can I processing on the AudioQueue callback AudioQueueOutputCallback for FFT . Just like this func static void audioQueueOutpuCallBack(void *input, AudioQueueRef inQueue, AudioQueueBufferRef outQueueBuffer) {    SYAudioQueue *aq = (__bridge SYAudioQueue *)input;   dispatch_semaphore_wait(aq-m_mutex, DISPATCH_TIME_FOREVER);   [aq enterQueue:inQueue withBuffer:outQueueBuffer];   dispatch_semaphore_signal(aq-m_mutex); } I know that AVAudioEngine can be processing for FFT under the AVAudioPCMBuffer . or How can I convert AudioQueueBufferRef to AVAudioPCMBuffer

An error is reported when playing h5 audio or video elements in wkwebview: Error acquiring assertion: Error Domain=RBSAssertionErrorDomain Code=3 "Required client entitlement is missing" UserInfo={RBSAssertionAttribute=RBSDomainAttribute| domain:"com.apple.webkit" name:"MediaPlayback" sourceEnvironment:"(null)", NSLocalizedFailureReason=Required client entitlement is missing} Then the performance of the webview will become very poor. There is an audio element and a button button in my HTML file. Click the button to play audio. body button onclick="handleClick()"PLAY/button audio id="audio" src="https://ac-dev.oss-cn-hangzhou.aliyuncs.com/test-2022-music.mp3"/audio script function handleClick() { document.getElementById("audio").play(); } /script /body Create a wkwebview to load the html file in my demo APP. class ViewController: UIViewController , WKUIDelegate{ var webView: WKWebView! override func loadView() { let config = WKWebViewConfiguration() config.preferences.javaScriptEnabled = true config.allowsInlineMediaPlayback = true webView = WKWebView(frame: .zero, configuration: config) //.zero webView.uiDelegate = self view = webView } override func viewDidLoad() { super.viewDidLoad() let myURL = URL(string: "https://ac-dev.oss-cn-hangzhou.aliyuncs.com/test-2022-py.html") let myRequest = URLRequest(url: myURL!) webView.load(myRequest) } } Click the button in the HTML to play the audio, and you can see the error report on the xcode. iPadN[2133:855729] [assertion] Error acquiring assertion: Error Domain=RBSAssertionErrorDomain Code=3 "Required client entitlement is missing" UserInfo={RBSAssertionAttribute=RBSDomainAttribute| domain:"com.apple.webkit" name:"MediaPlayback" sourceEnvironment:"(null)", NSLocalizedFailureReason=Required client entitlement is missing} To sum up, this error will appear when playing audio or video in HTML. Then the app performance will drop a lot, and the interactive response will be very slow.

Where can I find a comprehensive list of all the classes that the built in Sound Classifier model supports?

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.

Hi, I have multiple audio files I want to decide which channel goes to which output. For example, how to route four 2-channel audio files to an 8-channel output. Also If I have an AVAudioPlayerNode playing a 2-channel track through headphones, can I flip the channels on the output for playback, i.e flip left and right? I have read the following thread which seeks to do something similar, but it is from 2012 and I do not quite understand how it would work in modern day. Many thanks, I am a bit stumped.

I am getting an error in iOS 16. This error doesn't appear in previous iOS versions. I am using RemoteIO to playback live audio at 4000 hz. The error is the following: Input data proc returned inconsistent 2 packets for 186 bytes; at 2 bytes per packet, that is actually 93 packets This is how the audio format and the callback is set: // Set the Audio format AudioStreamBasicDescription audioFormat; audioFormat.mSampleRate = 4000; audioFormat.mFormatID = kAudioFormatLinearPCM; audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; audioFormat.mFramesPerPacket = 1; audioFormat.mChannelsPerFrame = 1; audioFormat.mBitsPerChannel = 16; audioFormat.mBytesPerPacket = 2; audioFormat.mBytesPerFrame = 2; AURenderCallbackStruct callbackStruct; // Set output callback callbackStruct.inputProc = playbackCallback; callbackStruct.inputProcRefCon = (__bridge void * _Nullable)(self); status = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, kOutputBus, &callbackStruct, sizeof(callbackStruct)); Note that the mSampleRate I set is 4000 Hz. In iOS 15 I get 0.02322 seconds of buffer duration (IOBufferDuration) and 93 frames in each callback. This is expected, because: number of frames * buffer duration = sampling rate 93 * 0.02322 = 4000 Hz However, in iOS 16 I am getting the aforementioned error in the callback. Input data proc returned inconsistent 2 packets for 186 bytes; at 2 bytes per packet, that is actually 93 packets Since the number of frames is equal to the number of packets, I am getting 1 or 2 frames in the callback and the buffer duration is of 0.02322 seconds. This didn't affect the playback of the "raw" signal, but it did affect the playback of the "processed" signal. number of frames * buffer duration = sampling rate 2 * 0.02322 = 0.046 Hz That doesn't make any sense. This error appears for different sampling rates (8000, 16000, 32000), but not for 44100. However I would like to keep 4000 as my sampling rate. I have also tried to set the sampling rate by using the setPreferredSampleRate(_:) function of AVAudioSession, but the attempt didn't succeed. The sampling rate was still 44100 after calling that function. Any help on this issue would be appreciated.

Is there a way to implement controls for background audio using ActivityKit like in the Apple Music application? I didn't found anything in your documentation about handling actions like this except deep links, but they're not suitable for this use case

We noticed iOS 16 doesn't seem to support these commands anymore: MPRemoteCommandCenter.shared().likeCommand MPRemoteCommandCenter.shared().dislikeCommand MPRemoteCommandCenter.shared().bookmarkCommand Or is there another way to show a menu in lieu of the previous button on the lock screen?

Hello, Here is an issue I encountered recently. Does anybody have feedback on this? Issue encountered AVAudioFile throws when opening WAV files and MPEG-DASH files with .mp3 extension, works fine with many other tested combinations of formats and extension (for example, an AIFF file with .mp3 extension is read by AVAudioFile without error). The Music app, AVAudioFile and ExtAudioFile all fail on the same files. However, previewing an audio file in Finder (select the file and hit the space bar) works regardless of the file extension. Why do I consider this an issue? AVAudioFile seems to rely on extension sometimes but not always to guess the audio format of the file, which leads to unexpected errors. I would expect AVAudioFile to deal properly with wrong extensions for all supported audio formats. ⚠️ This behaviour can cause real trouble in iOS and macOS applications using audio files coming from the user, which often have unreliable extensions. I published some code to easily reproduce the issue: https://github.com/ThomasHezard/AVAudioFileFormatIssue Thank you everybody, have a great day 😎

Thank you for this new API. Today, when using AUVoiceIO, voice gets processed ahead of rendering to ensure the echo canceller is capable of filtering it out from the input. Will other audio be processed in the same way? For example, rendered as mono in a 16kHz sampling rate? I'm asking because I'm wondering if this API will unlock the ability to use wide-band, stereo, high quality other audio (for example game audio) simultaneously while using voice. Thanks!

New on iOS17 we can control the amount of 'other audio' ducking through the AVAudioEngine. Is this also possible on AVAudioSession? In my app I don't voice input, but I do play voice audio while music from other apps plays in the background. Often the music either drowns to voice, if I use the .mixWithOthers category, or it's not loud enough if I use .duckOthers. It would be awesome to have the level of control that AVAudioEngine has.

A: iPhone SE 2nd (iOS 16.5) Used bluetooth model: Shokz OpenRun S803 B: Any mobile device A uses bluetooth microphone/speaker, and make a call to B using iPhone app. Mute the A's headphone. (The bluetooth device support mute by hardware). While A mutes, B speaks. Unmute A's headphone. Every time B speaks, B can hear the echo. Since there is no audio data during the hardware muted, VPIO don't recognize audio reference data to remove echo signal. Is there any alternative to resolve this echo in VoIP software using VPIO?

Building an audio streamer macOS app to record system audio and other inputs/outputs: seeking guidance on frameworks and APIs to use. Need suggestions for capturing and saving audio data efficiently. Any recommendations or insights?

I use openAl to play music. It works on iOS15. x phones, but there is no sound and no error reported on iOS16. x. Can someone help me

With numerous audio channel available from Dolby Atmos encoded audio files. Is it possible to address, and increase the volume of a preset channel when engaging in a specific head movement? For example, toggle on/off right surround bass channel when sweeping your nose quickly from Centerpoint to right shoulder.

I try to match a microphone audio with a custom catalog which I created via ShazamKit. What is the code for extracting and displaying "matchedMediaItem.artist" information on my iPhone screen after finding a song match with an entry of my custom-built catalog? I am a beginner.

Does anyone know if there will be a problem with publishing an app that contains music under a CC (royalty-free) license? Will adding the credentials of the author be sufficient for the review team?