AVPictureInPictureControllerContentSource *contentSource = [[AVPictureInPictureControllerContentSource alloc] initWithSampleBufferDisplayLayer:self.renderView.sampleBufferDisplayLayer playbackDelegate:self]; AVPictureInPictureController *pictureInPictureController = [[AVPictureInPictureController alloc] initWithContentSource:contentSource]; pictureInPictureController.delegate = self; (void)pictureInPictureController:(AVPictureInPictureController *)pictureInPictureController failedToStartPictureInPictureWithError:(NSError *)error { //error NSError * domain: @"PGPegasusErrorDomain" - code: -1003 0x00000002819fe3a0 } when first start the PiP play, I got the error "//error NSError * domain: @"PGPegasusErrorDomain" - code: -1003 0x00000002819fe3a0", why? and second start is Ok.

Hello, I'm observing an intermittent memory leak being reported in the iOS Simulator when initializing and starting an AVAudioEngine. Even with minimal setup—just attaching a single AVAudioPlayerNode and connecting it to the mainMixerNode—Xcode's memory diagnostics and Instruments sometimes flag a leak. Here is a simplified version of the code I'm using: // This function is called when the user taps a button in the view controller: #import "ViewController.h" @interface ViewController () @end @implementation ViewController - (void)viewDidLoad { [super viewDidLoad]; } - (IBAction)myButtonAction:(id)sender { NSLog(@"Test"); soundCreate(); } @end // media.m static AVAudioEngine *audioEngine = nil; void soundCreate(void) { if (audioEngine != nil) return; [[AVAudioSession sharedInstance] setCategory:AVAudioSessionCategoryAmbient error:nil]; [[AVAudioSession sharedInstance] setActive:YES error:nil]; audioEngine = [[AVAudioEngine alloc] init]; AVAudioPlayerNode* playerNode = [[AVAudioPlayerNode alloc] init]; [audioEngine attachNode:playerNode]; [audioEngine connect:playerNode to:(AVAudioNode *)[audioEngine mainMixerNode] format:nil]; [audioEngine startAndReturnError:nil]; } In the memory leak report, the following call stack is repeated, seemingly in a loop: ListenerMap::InsertEvent(XAudioUnitEvent const&, ListenerBinding*) AudioToolboxCore ListenerMap::AddParameter(AUListener*, void*, XAudioUnitEvent const&) AudioToolboxCore AUListenerAddParameter AudioToolboxCore addOrRemoveParameterListeners(OpaqueAudioComponentInstance*, AUListenerBase*, AUParameterTree*, bool) AudioToolboxCore 0x180178ddf

In Final Cut Pro, keyframes for transform parameters (such as Position, Scale, and Rotation) are automatically set to “Smooth” interpolation. This often results in undesired easing between keyframes, especially when linear motion is required. Currently, we have to manually adjust each keyframe to "Linear" using the Video Animation Editor, which can be time-consuming when working with many keyframes. Would it be possible to add an option to set the default keyframe interpolation to "Linear"—either globally in Preferences or per parameter in the Inspector? This would greatly streamline the animation workflow for many editors. Thank you for considering this request!

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!

When changing a camera's exposure, AVFoundation provides a callback which offers the timestamp of the first frame captured with the new exposure duration: AVCaptureDevice.setExposureModeCustom(duration:, iso:, completionHandler:). I want to get a similar callback when changing frame duration. After setting AVCaptureDevice.activeVideoMinFrameDuration or AVCaptureDevice.activeVideoMinFrameDuration to a new value, how can I compute the index or the timestamp of the first camera frame which was captured using the newly set frame duration?

How can media resources in my app be recommended to the system media control center, just like TikTok in the picture

Hi, I'm developing a SwiftUI app using RealityKit and ARKit for an AR measuring feature. I’ve noticed that after navigating away from my AR view and performing extensive cleanup (including removing all anchors/entities, pausing the ARSession, and nil-ing out all references), memory usage remains elevated and sometimes grows with repeated AR sessions. Each time I enter and exit the AR view, memory increases The memory does not return to the baseline after cleanup, even though all custom objects are deallocated. Are there best practices beyond what I’ve described to ensure all ARKit/RealityKit resources are released after an AR session?

I’m running HomePod OS 26 on two HomePod minis and OS 18.6 on main HomePod (original) I’ve enabled Crossfade in the Home app. I’m playing Apple Music directly in the HomePod mini. Crossfade just doesn’t work on any HomePod. I can understand it not working on the HomePod - but why isn’t it working on the minis running OS 26? I’ve tried disabling and enabling Crossfade, rebooting HomePods etc but nothing?!

We encounter issue with avplayer in case of EXT-X-DISCONTINUITY misalignment between audio and video produced after insertion of gaps. The initial objective is to introduce an EXT-X-DISCONTINUITY in audio playlist after some missing segments (EXT-X-GAP) which durations are aligned to video segments durations, to handle irregular audio durations. Please find below an example of corresponding video and audio playlists: video: #EXTM3U #EXT-X-VERSION:7 #EXT-X-MEDIA-SEQUENCE:872524632 #EXT-X-INDEPENDENT-SEGMENTS #EXT-X-TARGETDURATION:2 #USP-X-TIMESTAMP-MAP:MPEGTS=7096045027,LOCAL=2025-05-09T12:38:32.369100Z #EXT-X-MAP:URI="hls/StreamingBasic-video=979200.m4s" #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:32.369111Z #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524632.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524633.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524634.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524635.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524636.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524637.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524638.m4s #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:46.383111Z #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524639.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524640.m4s audio: EXTM3U #EXT-X-VERSION:7 #EXT-X-MEDIA-SEQUENCE:872524632 #EXT-X-INDEPENDENT-SEGMENTS #EXT-X-TARGETDURATION:2 #USP-X-TIMESTAMP-MAP:MPEGTS=7096045867,LOCAL=2025-05-09T12:38:32.378400Z #EXT-X-MAP:URI="hls/StreamingBasic-audio_99500_eng=98800.m4s" #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:32.378444Z #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524632.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524633.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524634.m4s #EXTINF:1.984, no desc hls/StreamingBasic-audio_99500_eng=98800-872524635.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524636.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-audio_99500_eng=98800-872524637.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-audio_99500_eng=98800-872524638.m4s #EXT-X-DISCONTINUITY #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:46.778444Z #EXTINF:1.6213, no desc hls/StreamingBasic-audio_99500_eng=98800-872524639.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524640.m4s In this case playback is broken with avplayer. Is it conformed to Http Live Streaming? Is it an avplayer bug? What are the guidelines to handle such gaps?

i have a wallpaper app , i need a photos access when i try to add it in capabilities i don't find it.is there any solution?

I found that the aggregated device correctly obtains input channels in the standard microphone mode. However, in voice isolation mode, it only retrieves channels from the first sub-device in the aggregated device's list. If I want to properly obtain channel information in voice isolation mode, how should I do it?

I have implemented fetching Apple Music preview songs using a Swift framework integrated into a Unity app. My requirement is to fetch full tracks from a user’s Apple Music library and play them inside Unity. To do this, I understand that I need to handle authentication, generate a Developer Token, and then obtain a Music User Token to access the user’s Apple Music content. Currently, I have an Individual Apple Developer account (not Organization). Based on my research, it seems that: With an Individual account, I can implement this functionality and even upload builds to TestFlight for internal testing. However, when releasing the app publicly on the App Store, full-track playback may be restricted for Individual accounts and allowed only for Organization accounts. 👉 Can you confirm if this understanding is correct? 👉 Specifically, is it possible for an Individual account to fetch and play full-length tracks from a subscribed Apple Music user’s library (at least for internal/TestFlight testing)?

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.

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

Hi everyone, I’m testing audio recording on an iPhone 15 Plus using AVFoundation. Here’s a simplified version of my setup: let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatLinearPCM), AVSampleRateKey: 8000, AVNumberOfChannelsKey: 1, AVLinearPCMBitDepthKey: 16, AVLinearPCMIsFloatKey: false ] audioRecorder = try AVAudioRecorder(url: fileURL, settings: settings) audioRecorder?.record() When I check the recorded file’s sample rate, it logs: Actual sample rate: 8000.0 However, when I inspect the hardware sample rate: try session.setCategory(.playAndRecord, mode: .default) try session.setActive(true) print("Hardware sample rate:", session.sampleRate) I consistently get: `Hardware sample rate: 48000.0 My questions are: Is the iPhone mic actually capturing at 8 kHz, or is it recording at 48 kHz and then downsampling to 8 kHz internally? Is there any way to force the hardware to record natively at 8 kHz? If not, what’s the recommended approach for telephony-quality audio (true 8 kHz) on iOS devices? Thanks in advance for your guidance!

Background: For iOS, I have built a custom video record app that records video at the highest possible FPS (supposed to be around 240 FPS but more like ~200 in practice). When I save this video to the user's camera roll, I notice this dynamic playback speed bar. This bar will speed up or slow down the video. My question: How can I save my video such that this playback speed bar is constantly slow or plays at real time speed? For reference I have include the playback speed bar that I am talking about in the screenshot, you can find this in the photo app when you record slow motion video.

I am working to update a live blog in Apple News. As far as I know there is an update endpoint to update a content in Apple News. Is there any feature in Apple News to trigger an event when the original content updated and pull the updated content?

I developed an educational app that implements audio-video communication through RTC, while using WebView to display course materials during classes. However, some users are experiencing an issue where the audio playback from WebView is very quiet. I've checked that the AVAudioSessionCategory is set by RTC to AVAudioSessionCategoryPlayAndRecord, and the AVAudioSessionCategoryOption also includes AVAudioSessionCategoryOptionMixWithOthers. What could be causing the WebView audio to be suppressed, and how can this be resolved?

Capturing more than one display is no longer working with macOS Sequoia. We have a product that allows users to capture up to 2 displays/screens. Our application is using gstreamer which in turn is based on AVFoundation. I found a quick way to replicate the issue by just running 2 captures from separate terminals. Assuming display 1 has device index 0, and display 2 has device index 1, here are the steps: install gstreamer with brew install gstreamer Then open 2 terminal windows and launch the following processes: terminal 1 (device-index:0): gst-launch-1.0 avfvideosrc -e device-index=0 capture-screen=true ! queue ! videoscale ! video/x-raw,width=640,height=360 ! videoconvert ! osxvideosink terminal 2 (device-index:1): gst-launch-1.0 avfvideosrc -e device-index=1 capture-screen=true ! queue ! videoscale ! video/x-raw,width=640,height=360 ! videoconvert ! osxvideosink The first process that is launched will show the screen, the second process launched will not. Testing this on macOS Ventura and Sonoma works as expected, showing both screens. I submitted the same issue on Feedback Assistant: FB15900976

I’m building a SwiftUI app whose primary job is to play audio. I manage all of the Now-Playing metadata and Command center manually via the available shared instances: MPRemoteCommandCenter.shared() MPNowPlayingInfoCenter.default().nowPlayingInfo In certain parts of the app I also need to display videos, but as soon as I attach another AVPlayer, it automatically pushes its own metadata into the Control Center and overwrites my audio info. What I need: a way to show video inline without ever having that video player update the system’s Now-Playing info (or Control Center). In my app, I start by configuring the shared audio session do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .default, options: [ .allowAirPlay, .allowBluetoothA2DP ]) try AVAudioSession.sharedInstance().setActive(true) } catch { NSLog("%@", "**** Failed to set up AVAudioSession \(error.localizedDescription)") } and then set the MPRemoteCommandCenter commands and MPNowPlayingInfoCenter nowPlayingInfo like mentioned above. All this works without any issues as long as I only have one AVPlayer in my app. But when I add other AVPlayers to display some videos (and keep the main AVPlayer for the sound) they push undesired updates to MPNowPlayingInfoCenter: struct VideoCardView: View { @State private var player: AVPlayer let videoName: String init(player: AVPlayer = AVPlayer(), videoName: String) { self.player = player self.videoName = videoName guard let path = Bundle.main.path(forResource: videoName, ofType: nil) else { return } let url = URL(fileURLWithPath: path) let item = AVPlayerItem(url: url) self.player.replaceCurrentItem(with: item) } var body: some View { VideoPlayer(player: player) .aspectRatio(contentMode: .fill) .onAppear { player.isMuted = true player.allowsExternalPlayback = false player.actionAtItemEnd = .none player.play() MPNowPlayingInfoCenter.default().nowPlayingInfo = nil MPNowPlayingInfoCenter.default().playbackState = .stopped NotificationCenter.default.addObserver(forName: .AVPlayerItemDidPlayToEndTime, object: player.currentItem, queue: .main) { notification in guard let finishedItem = notification.object as? AVPlayerItem, finishedItem === player.currentItem else { return } player.seek(to: .zero) player.play() } } .onDisappear { player.pause() } } } Which is why I tried adding: MPNowPlayingInfoCenter.default().nowPlayingInfo = nil MPNowPlayingInfoCenter.default().playbackState = .stopped // or .interrupted, .unknown But that didn't work. I also tried making a wrapper around the AVPlayerViewController in order to set updatesNowPlayingInfoCenter to false, but that didn’t work either: struct CustomAVPlayerView: UIViewControllerRepresentable { let player: AVPlayer func makeUIViewController(context: Context) -> AVPlayerViewController { let vc = AVPlayerViewController() vc.player = player vc.updatesNowPlayingInfoCenter = false vc.showsPlaybackControls = false return vc } func updateUIViewController(_ controller: AVPlayerViewController, context: Context) { controller.player = player } } Hence any help on how to embed video in SwiftUI without its AVPlayer touching MPNowPlayingInfoCenter would be greatly appreciated. All this was tested on an actual device with iOS 18.4.1, and built with Xcode 16.2 on macOS 15.5