Hi, I'm trying to setup a AVAudioEngine for USB Audio recording and monitoring playthrough. As soon as I try to setup playthough I get an error in the console: AVAEInternal.h:83 required condition is false: [AVAudioEngineGraph.mm:1361:Initialize: (IsFormatSampleRateAndChannelCountValid(outputHWFormat))] Any ideas how to fix it? // Input-Device setzen try? setupInputDevice(deviceID: inputDevice) let input = audioEngine.inputNode // Stereo-Format erzwingen let inputHWFormat = input.inputFormat(forBus: 0) let stereoFormat = AVAudioFormat(commonFormat: inputHWFormat.commonFormat, sampleRate: inputHWFormat.sampleRate, channels: 2, interleaved: inputHWFormat.isInterleaved) guard let format = stereoFormat else { throw AudioError.deviceSetupFailed(-1) } print("Input format: \(inputHWFormat)") print("Forced stereo format: \(format)") audioEngine.attach(monitorMixer) audioEngine.connect(input, to: monitorMixer, format: format) // MonitorMixer -> MainMixer (Output) // Problem here, format: format also breaks. audioEngine.connect(monitorMixer, to: audioEngine.mainMixerNode, format: nil)

Hello, I'm working on a Flutter app targeting both Android and iOS, where I implemented ShazamKit. In order to achieve that, I first tried with the flutter_shazam_kit package, but since it's not maintained anymore, I forked it here, and tried to update it to meet the Google Play Store requirements, as you can see here: https://github.com/mregnauld/flutter_shazam_kit/tree/fix-16k Unfortunately, after trying everything, my app still doesn't meet the (not so) new 16 KB native library alignment. Also, I'm 100% sure it comes from that because the error message disappears if I remove that package from my app. So after investigating, it seems that the problem comes from the ShazamKit for Android (that you can find here: https://developer.apple.com/download/all/?q=Android%20ShazamKit), and especially the .so files in the .aar file. Is there anything I can do to fix that, or should I wait before the ShazamKit team fix that? I'm totally stuck with that so any help is highly appreciated. Thanks.

Hey there, I just upgraded to Mac OS Tahoe ,son an apple MacBook Pro 2019 16inch. am using IntellijIDEA and Flutter to develop a mobile app which I test on the simulator app running iOS 18.4 . the issue: when I start the simulator app. ( while in the loading phase and in the operation phase as well ), the audio from an already open YouTube tab on safari (this happens on chrome browser as well). the sound glitches and becomes Noise. a fix I found online is to kill the audio deamon on Mac OS, This works using the command: "sudo killall coreaudiod" this kills the audio process, (while the emulator is operational), then the macOS restarts the audio deamon then the audio works fine alongside with the simulator being open. I just want to ask is there a permanent fix for this? is Apple working on a fix for this in the upcoming update?

I am trying to use SpeechTranscriber from Speech framework. Is it possible to use it on Simulator of iOS 26 (Mac OS Tahoe)? Function "supportedLocales" returns an empty array.

I am using https://developer.apple.com/documentation/applemusicapi/add-tracks-to-a-library-playlist to add tracks to playlists. This endpoint works fine for all playlists except for collaborative playlists. For collaborative playlist I get the following 500 error as a response: "errors": [ { "id": "<some id>", "title": "Upstream Service Error", "detail": "Unable to update tracks", "status": "500", "code": "50001" } ] } Steps to reproduce: Create a playlist in your library. Use the api to add a song. Confirm that it works. Make that same playlist collaborative. Update the playlist ID in your api request (as making a playlist collaborative changes its id) Confirm that you get the 500 error.

Just downloaded iOS 26.1 and my phone keeps ringing after the call has been answered. Any fixes for this?

We build mobile apps for creators to edit their videos. Post editing the video, the creator has to export the video so that it can be uploaded to Youtube. The export is a time consuming and GPU intensive process. The creator can exit the app due to various reasons like receiving the call, putting the app in background etc. This causes the export to fail :( Keeping this limitation in mind there was an announcement from Apple that with the IOS 26 launch would start to support background GPU access. Here is the official documentation: https://developer.apple.com/documentation/BundleResources/Entitlements/com.apple.developer.background-tasks.continued-processing.gpu When we tried using this feature, we were not able to get it to work on IOS 26. We stumbled upon this ticket(https://developer.apple.com/forums/thread/797538?answerId=854825022#854825022) in the Apple Developer forum, in which possibly an Apple engineer claims it is supported ONLY for iPadOS 26. This is a very big bummer for us. 96% of the users are on iPhone(compared to iPad), and if we refer to the official documentation above, it claims that this feature should work on IOS 26. This feature is extremely important for having the best user experience and reducing user frustration and will be useful for other video editing apps. Looking forward to a resolution.

Hi all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!

I have a PCM audio buffer (AVAudioPCMFormatInt16). When I try to play it using AVPlayerNode / AVAudioEngine an exception is thrown: "[[busArray objectAtIndexedSubscript:(NSUInteger)element] setFormat:format error:&nsErr]: returned false, error Error Domain=NSOSStatusErrorDomain Code=-10868 (related thread https://forums.developer.apple.com/forums/thread/700497?answerId=780530022#780530022) If I convert the buffer to AVAudioPCMFormatFloat32 playback works. My questions are: Does AVAudioEngine / AVPlayerNode require AVAudioPCMBuffer to be in the Float32 format? Is there a way I can configure it to accept another format instead for my application? If 1 is YES is this documented anywhere? If 1 is YES is this required format subject to change at any point? Thanks! I was looking to watch the "AVAudioEngine in Practice" session video from WWDC 2014 but I can't find it anywhere (https://forums.developer.apple.com/forums/thread/747008).

I want the audio session to always use the built-in microphone. However, when using the setPreferredInput() method like in this example private func enableBuiltInMic() { // Get the shared audio session. let session = AVAudioSession.sharedInstance() // Find the built-in microphone input. guard let availableInputs = session.availableInputs, let builtInMicInput = availableInputs.first(where: { $0.portType == .builtInMic }) else { print("The device must have a built-in microphone.") return } // Make the built-in microphone input the preferred input. do { try session.setPreferredInput(builtInMicInput) } catch { print("Unable to set the built-in mic as the preferred input.") } } and calling that function once in the initializer, the audio session still switches to the external microphone once one is plugged in. The session's preferredInput is nil again at that point, even if the built-in microphone is still listed in availableInputs. So, why is the preferredInput suddenly reset? when would be the appropriate time to set the preferredInput again? Observing the session’s availableInputs did not work and setting the preferredInput again in the routeChangeNotification handler seems a bad choice as it’s already a bit too late then.

I have an iPadOS M-processor application with two different running configurations. In config1, the shared AVAudioSession is configured for .videoChat mode using the built-in microphone. The input/output nodes of the AVAudioEngine are configured with voice processing enabled. The built-in mic is formatted for 1 channel at 48KHz. In config2, the shared AVAudioSession is configured for .measurement mode using an external USB microphone. The input/output nodes of the AVAudioEngine are configured with voice processing disabled. The external mic is formatted for 2 channels at 44.1KHz I've written a configuration manager designed to safely switch between these two configurations. It works by stopping AVAudioEngine and detaching all but the input and output nodes, updating the shared audio session for the desired mic and sample-rates, and setting the appropriate state for voice processing to either true or false as required by the configuration. Finally the new audio graph is constructed by attaching appropriate nodes, connecting them, and re-starting AVAudioEngine I'm experiencing what I believe is a race-condition between switching voice processing on or off and then trying to re-build and start the new audio graph. Even though notifications, which are dumped to the console indicate that my requested input and sample-rate settings are in place, I crash when trying to start the audio engine because the sample-rate is wrong. Investigating further it looks like the switch from remote I/O to voice-processing I/O or vice-versa has not yet actually completed. I introduced a 100ms second delay and that seems to help but is obviously not a reliable way to build software that must work consistently. How can I make sure that what are apparently asynchronous configuration changes to the shared audio session and the input/output nodes have completed before I go on? I tried using route change notifications from the shared AVAudioSession but these lie. They say my preferred mic input and sample-rate setting is in place but when I dump the AVAudioEngine graph to the debugger console, I still see the wrong sample rate assigned to the input/output nodes. Also these are the wrong AU nodes. That is, VPIO is still in place when RIO should be, or vice-versa. How can I make the switch reliable without arbitrary time delays? Is my configuration manager approach appropriate (question for Apple engineers)?

My app has been using the iTunes Search API (itunes.apple.com/search) for a few years now, but at some point over the last week or so (late Sept. 2025) it is no longer returning track results with explicit content, regardless of whether I provide "explicit=Yes" (which is the default anyway, according to the API documentation - https://performance-partners.apple.com/search-api). Has anyone else experienced this with this API and have you figured out a workaround? FYI, I do also use the more robust Apple Music API in another part of my app, which isn't going through this issue, so I know it's technically an alternative. I just need to stick with iTunes Search API in this particular case. Thanks.

I've been using CGWindowListCreateImage which automatically creates an image with the size of the captured window. But SCScreenshotManager.captureImage(contentFilter:configuration:) always creates images with the width and height specified in the provided SCStreamConfiguration. I could be setting the size explicitly by reading SCWindow.frame or SCContentFilter.contentRect and multiplying the width and height by SCContentFilter.pointPixelScale , but it won't work if I want to keep the window shadow with SCStreamConfiguration.ignoreShadowsSingleWindow = false. Is there a way and what's the best way to take full-resolution screenshots of the correct size? import Cocoa import ScreenCaptureKit class ViewController: NSViewController { @IBOutlet weak var imageView: NSImageView! override func viewDidAppear() { imageView.imageScaling = .scaleProportionallyUpOrDown view.wantsLayer = true view.layer!.backgroundColor = .init(red: 1, green: 0, blue: 0, alpha: 1) Task { let windows = try await SCShareableContent.excludingDesktopWindows(false, onScreenWindowsOnly: true).windows let window = windows[0] let filter = SCContentFilter(desktopIndependentWindow: window) let configuration = SCStreamConfiguration() configuration.ignoreShadowsSingleWindow = false configuration.showsCursor = false configuration.width = Int(Float(filter.contentRect.width) * filter.pointPixelScale) configuration.height = Int(Float(filter.contentRect.height) * filter.pointPixelScale) print(filter.contentRect) let windowImage = try await SCScreenshotManager.captureImage(contentFilter: filter, configuration: configuration) imageView.image = NSImage(cgImage: windowImage, size: CGSize(width: windowImage.width, height: windowImage.height)) } } }

Hi everyone, I’m running into an issue with PHPickerFilter when using PHPickerViewController. When I configure the picker with a .videos and .livePhotos filter, it seems to work correctly in the Photos tab. However, when I switch to the Collections tab, the filter doesn’t always apply — users can still see and select static image assets in certain collections (e.g. from one of the People & Pets sections). Here’s a simplified snippet of my setup: var configuration = PHPickerConfiguration(photoLibrary: .shared()) configuration.selectionLimit = 1 var filters = [PHPickerFilter]() filters.append(.videos) filters.append(.livePhotos) configuration.filter = PHPickerFilter.any(of: filters) configuration.preferredAssetRepresentationMode = .current let picker = PHPickerViewController(configuration: configuration) picker.delegate = self present(picker, animated: true) Expected behavior: The picker should consistently respect the filter across both Photos and Collections tabs, only showing assets that match the filter. Actual behavior: The filter seems to apply correctly in the Photos tab, but in the Collections tab, other asset types are still visible/selectable. Has anyone else encountered this behavior? Is this expected or a known issue, or am I missing something in the configuration? Thanks in advance!

My audio app shows a control bar at the bottom of the window. The controls show nicely, but there is a black "slab" appearing behind the inline controls, the same size as the playerView. Setting the player view background color does nothing: playerView.wantsLayer = true playerView.layer?.backgroundColor = NSColor.clear.cgColor How can I clear the background? If I use .floating controlsStyle, I don't get the background "slab".

I discovered when editing photos with the PhotoKit API, PHContentEditingOutput's renderedContentURL is a file in the app container's tmp directory with a filename that seems to follow the format render.<uuid>.JPG, and that file does not get deleted if the edit does not complete successfully (the user cancels the edit request, an error occurs, the app crashes, etc). I understand the system is supposed to automatically delete tmp files every once in a while, but some users are noticing my app's Documents & Data inflates, so I'm considering deleting these render files each time the app is launched. But I don't want to delete everything in the tmp directory as there could possibly be other data in there. What's the best way to remove those temporary files? Does the filename always start with render. no matter the device language? I thought I'd delete files in NSTemporaryDirectory() with that prefix but then I discovered in Mac Catalyst the location is not the tmp directory directly, they're in tmp/TemporaryItems/<bundleid>. Thanks!

Using an iPhone Pro 12 running iOS 26.0.1, with AirPods Pro 3. Camera app does capture video with what seems to be "Studio Quality Recording". Am trying to replicate that SQR with my own Camera like app, and while I can pull audio in from the APP3 mic, and my video capture app is recording a 48,000Hz high-bitrate video, the audio still sounds non-SQR. I'm seeing bluetoothA2DP , bluetoothLE , bluetoothHFP as portType, and not sure if SQR depends on one of those? Is there sample code demonstrating a SQR capture? Nevermind video and camera, just audio even? Also, I don't understand what SQR is doing between the APP3 and the iPhone. What codec is that? What bitrate is that? If I capture video using Capture and inspect the audio stream I see mono 74.14 kbit/s MPEG-4 AAC, 48000 Hz. But I assume that's been recompressed and not really giving me any insight into the APP3 H2 transmission?

We have a Low-Latency HLS stream, and on iOS 26, even though the bandwidth is sufficient, it still selects a low-bandwidth resolution (e.g., RESOLUTION=640x360) for playback instead of using a higher-bandwidth resolution (e.g., RESOLUTION=1920x1080) when using AVPlayerViewController with AVPlayer. This works fine on iOS version 18 and previous versions. What could be the solution to this issue?

Hi, In the iOS13 and macOS Catalina release notes it says: Metal CIKernel instances now support arguments with arbitrarily structured data. I've been trying to use this functionality in a CIKernel with mixed results. I'm particularly interested in passing data in the form of a dynamically sized array. It seems to work up to a certain size. Beyond the threshold excessive data is discarded and the kernel becomes unstable. I assume there is some kind of memory alignment issue going on, but I've tried various types in my array and always get a similar result. I have not found any documentation or sample code regarding this. It would be great to know how this is intended to work and what the limitations are. In the forums there are two similar unanswered questions about data arguments, so I'm sure there are a few out there with similar issues. Thanks! Michael

Currently, I am using the Broadcast UploadExtension function to obtain samplebuffer data through APP Group and IPC (based on the local Unix Domain Socket) The screen recording data transmission method of the domain socket is transmitted to the APP. However, when the APP goes back to the background to view videos in the album or other audio and video, the data transmission stops and the APP cannot obtain the screen recording data. I would like to ask how to solve this problem. I suspect that the system has suspended the extended screen recording