Precondition: In system settings, scale the pointer size up to the max. Our SCScreenshotManager code currently works in macOS 15 and earlier to capture the cursor at it's larger size, but broke in one of the minor releases of macOS Tahoe. The error it produces now is "Failed to start stream due to audio/video capture failure". This only seems to happen with the cursor window, not any others. Another way to get the cursor is with https://developer.apple.com/documentation/appkit/nscursor/currentsystem, but that is now deprecated, which makes me think the capture of the cursor is being blocked deliberately. We see this as a critical loss of functionality for our apps, and could use guidance on what to use instead.

I'm developing an audio player app that uses AVAudio​File to read PCM data from various formats. I'm experiencing severe performance issues when seeking in FLAC, while other compressed formats (M4A/AAC) work correctly. I don't intend to use them in my app, but I also tested mp3 files just by curiosity and they also have this issue. Environment: macOS 26 (Tahoe) Xcode 26.3 Apple Silicon (M1) The issue: After setting AVAudio​File​.frame​Position to a position mid-file, the subsequent call to AVAudio​File​.read(into​:frame​Count:) blocks for an unreasonable amount of time for FLAC and MP3 files. The delay scales linearly with the seek target, seeking near the beginning is fast, seeking toward the end is proportionally slower, which suggests the decoder is decoding linearly from the beginning of the file rather than using any seek index. (My app deals with “images” of Audio CDs ripped as a single long audio file.) The issue is particularly severe when reading files from an SMB network share (server on Ethernet, client on Wi-Fi with the access point ~2 meters away in line of sight). Quick Benchmark results: I tested with the same 75-minute audio content (16-bit/44.1 kHz stereo, 200,502,708 frames) encoded in five formats, seeking to the midpoint. Over SMB (Local Network, Server on Ethernet, Client on WiFi): Format | Seek + Read Time ----------|------------------ WAV | 0.007 s AIFF | 0.009 s Apple | 0.015 s Lossless | MP3 | 9.2 s FLAC | 30.2 s Locally (MacBook Air M1 SSD) : Format | Seek + Read Time ----------|------------------ WAV | 0.0005 s AIFF | 0.0004 s Apple | 0.0011 s Lossless | MP3 | 0.1958 s FLAC | 0.7528 s WAV, AIFF, and M4A all seek virtually instantly (< 15 ms). MP3 and FLAC exhibit linear-time behavior, with FLAC being the worst affected. Note that M4A (AAC) is also a compressed format that requires decoding after seeking, yet it completes in 15 ms. This rules out any inherent limitation of compressed formats, the MP4 container's packet index (stts/stco) is clearly being used for fast random access. Both MP3 (Xing/LAME TOC) and FLAC (SEEKTABLE metadata block) have their own seek mechanisms that should provide similar performance. Minimal CLI tool to reproduce: import Foundation guard CommandLine.arguments.count > 1 else { print("Usage: FLACSpeed <audio-file-path>") exit(1) } let path = CommandLine.arguments[1] let fileURL = URL(fileURLWithPath: path) do { let file = try AVAudioFile(forReading: fileURL) let format = file.processingFormat let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: 8192)! let totalFrames = file.length let seekTarget = totalFrames / 2 print("File: \(fileURL.lastPathComponent)") print("Format: \(format)") print("Total frames: \(totalFrames)") print("Seeking to frame: \(seekTarget)") file.framePosition = seekTarget let start = CFAbsoluteTimeGetCurrent() try file.read(into: buffer, frameCount: 8192) let elapsed = CFAbsoluteTimeGetCurrent() - start print("Read after seek took \(elapsed) seconds") } catch { print("Error: \(error.localizedDescription)") exit(1) } Expected behavior: AVAudio​File​.read(into​:frame​Count:) after setting frame​Position should use the available seek mechanisms in FLAC and MP3 files for fast random access, as it already does for M4A (AAC). Even accounting for the fact that seek tables provide approximate (not sample-precise) positioning, the "jump to nearest index point + decode forward" approach should complete in milliseconds, not seconds. Workaround: For FLAC, I've worked around this by using libFLAC directly, which provides instant seeking via FLAC__stream​_decoder​_seek​_absolute(). libFLAC Performance: For comparison, libFLAC's FLAC__stream​_decoder​_seek​_absolute() performs the same seek + read on the same FLAC file in around 0.015, using the FLAC seek table to jump to the nearest preceding seek point, then decoding forward a small number of frames to the exact target sample.

I have an application which records video along with some custom metadata and a chapter track. The resultant video is stored in the Camera Roll. When sharing the video via the Share Sheet or AirDrop, the metadata track is stripped entirely (the chapter markers are preserved) Sharing via AirDrop with the "All Photos Data" option does include the metadata track, as does copying from the device with Image Capture but this is a bad user experience as the user must remember to explicitly select this option, and the filename is lost when sending this way. I have also tried various other approaches (such as encoding my metadata in a subtitle track, which I didn't expect to be stripped as it's an accessibility concern) but it's also removed. Essentially I am looking for a definitive list of things that are not stripped or if there's a way to encode a track in some way to indicate it should be preserved. The metadata is added via AVTimedMetadataGroup containing one AVMutableMetadataItem which has its value as a JSON string. I took a different approach with the Chapter Marker track (mainly because I did it first in a completely different way and didn't rework it when I added the other track). I post-process these after the video is recorded, and add them with addMutableTrack and then addTrackAssociation(to: chapterTrack, type: .chapterList) but I don't think that's the reason the chapter track persists where the custom metadata does not as other tests with video files from other sources containing subtitles etc also had their subtitle data stripped. tl;dr I record videos with metadata that I want to be able to share via Share Sheet and AirDrop, what am I doing wrong?

I’m building a standalone Apple Watch metronome app for running. My goal is for these 3 apps to work at the same time: Runna owns the workout session Spotify plays music my app plays a metronome click in the background So far this is what I've found: Using HKWorkout​Session in my metronome app works well with Spotify, but conflicts with Runna and other workout apps, so I removed that. Using watchOS background audio with longFormAudio allows my app run in the background, and it can coexist with Runna. However, it seems to conflict with Spotify playback, and one app tends to stop the other. Is there any supported watchOS audio/background configuration that allows all 3 at once? More specifically this is what I need: another app owns HKWorkout​Session Spotify keeps playing my app keeps generating metronome clicks in the background Or is this simply not supported by current watchOS session/background rules? My metronome uses AVAudio​Engine / AVAudio​Player​Node with generated click audio. Thank you!

Hi, Our client currently has an FPS deployment certificate generated with SDK version 4 that is still actively used in production. They would like to generate an additional certificate using SDK version 26. Before doing so, they just want to confirm: Will the existing SDK 4 certificate remain unaffected and still visible in the Apple Developer portal? Any considerations they should keep in mind? Thanks!

Hi, Our client currently has an FPS deployment certificate generated with SDK version 4 that is still actively used in production. They would like to generate an additional certificate using SDK version 26. Before doing so, they just want to confirm: Will the existing SDK 4 certificate remain unaffected and still visible in the Apple Developer portal? Any considerations they should keep in mind? Thanks! :)

I'm developing a PHBackgroundResourceUploadExtension and finding it difficult to debug because the system controls when the extension launches. Current experience: The extension starts at unpredictable times (anywhere from 1 minute to several hours after photos are added) By the time I attach the debugger, the upload may have already completed or failed Breakpoints in init() or early lifecycle methods are often missed Questions: Is there a way to force-launch the extension during development (similar to how we can manually trigger Background App Refresh in Xcode)? Are there any launch arguments or environment variables that put the extension in a debug/eager mode? I tried taking photos/videos, but this doesn't trigged app extension in all cases. Any tips for improving the debug cycle would be greatly appreciated. Environment: iOS 26, Xcode 18

I'm implementing PHBackgroundResourceUploadExtension to back up photos and videos to our cloud storage service. During testing, I observed that iCloud-optimized photos (where the full-resolution original is stored in iCloud, not on device) do not upload. The upload job appears to silently skip these assets. Questions: Is this behavior intentional/documented? I couldn't find explicit mention of this limitation. If the device only has the optimized/thumbnail version locally, does the system: - Automatically download the full-resolution asset from iCloud before uploading? - Skip the asset entirely? - Return an error via PHAssetResourceUploadJobChangeRequest? For a complete backup solution, should we: - Pre-fetch full-resolution assets using PHAssetResourceManager.requestData(for:options:) before creating upload jobs? - Use a hybrid approach (this extension for local assets + separate logic for iCloud-only assets)? Environment: iOS 26, Xcode 18

Problem: When the screen is locked, an incoming call does not initiate the launch of the Flutter application required for audio and video communication through Jitsi Meet. In the unlocked state, the application functions correctly. The current implementation does not have a mechanism for activating the Flutter engine when receiving a call via CallKit while the screen is locked. Although CallKit UI displays the call acceptance interface and the audio session is configured, the Flutter application remains in a suspended state, making it impossible to connect to the media server. Audio session activated using didActivateAudioSession method.

Hi, So I want to use Apple Music API to get the top musics from Apple music. My question would be: Can I place ads between 2 apple music? Use case: User can listen 30 sec previews which I got from the API (link and stuff included of course). Then he/she swipes (like on tiktok) and see a banner or video ad. Then he/she swipe to the next 30 sec preview. (The ad itself is skippable, so it's not necessary to stay just to listen the music) Is this valid, or I shouldn't use ads this way?

I’m implementing a custom playback pipeline using AVSampleBufferAudioRenderer together with AVSampleBufferRenderSynchronizer. hasSufficientMediaDataForReliablePlaybackStart appears to be the intended signal for determining when enough media has been queued to start playback. For local playback, this works well in practice — the property becomes true after a reasonable amount of media is enqueued. However, when the output route is AirPlay, using this property becomes difficult: AirPlay requires significantly more buffered media before the renderer reports sufficient data. The required preroll amount is much larger than for local playback. For short assets, it is possible to enqueue the entire audio track and still never observe hasSufficientMediaDataForReliablePlaybackStart == true. In that situation there is no more media data to enqueue, but the renderer still reports that playback is not ready. Given this behavior, what is the recommended way to determine playback readiness when using AVSampleBufferAudioRenderer with AirPlay?

I am trying to create keys for my personal project with MusicKit and other resources, but MusicKit specifically for now. I want to gather my recent music history and log the time in my system to measure with my other life data to do analysis on. I have created an Identifier with an appropriate Description and Bundle ID and have MusicKit checked in AppServices. I have saved and reset cash and waited all day and the keys have still not update and show "There are no identifiers available that can be associated with the key" in this field. Please help!

Among Japanese end users, audio issues during screen recording—primarily in game applications—have become a topic of discussion. We have confirmed that the trigger for this issue is highly likely to be related to changes to IOBufferDuration. When using setPreferredIOBufferDuration and the IOBufferDuration is set to a value smaller than the default, audio problems occur in the recorded screen capture video. Audio playback is performed using AudioUnit (RemoteIO). https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc This issue was not observed on iOS 18, and it appears to have started occurring after upgrading to iOS 26. We provide an audio middleware solution, and we had incorporated changes to IOBufferDuration into our product to achieve low-latency audio playback. As a result, developers using our product as well as their end users are being affected by this issue. We kindly request that this issue be investigated and addressed in a future update. “This document has been translated by AI. The original text is included below for reference.” 日本のエンドユーザー間で主にゲームアプリケーションにおける画面収録時の音声の問題が話題になっています。 こちらの症状のトリガーが、IOBufferDurationの変更によるものである可能性が高いことを確認しました。 setPreferredIOBufferDurationを使用し、IOBufferDurationがデフォルトより小さい状態の時、画面収録された動画の音声に問題が発生することをしています。 音声の再生にはAudioUnit(RemoteIO)を使用しています。 https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc iOS 18ではこのような問題は確認されておらず、iOS26になってから問題が発生しているようです。 私たちはオーディオミドルウェアを提供しており、低遅延の再生のためにIOBufferDurationの変更を製品に組み込んでいました。 そのため、弊社製品をご利用いただいている開発者およびエンドユーザーの皆様がこの不具合の影響を受けています。 こちらの不具合の調査及び修正対応を検討いただけますでしょうか。

I'm implementing a PHBackgroundResourceUploadExtension to back up photos and videos from the user's library to our cloud storage service. Our existing upload infrastructure uses chunked uploads for large files (splitting videos into smaller byte ranges and uploading each chunk separately). This approach: Allows resumable uploads if interrupted Stays within server-side request size limits Provides granular progress tracking Looking at the PHAssetResourceUploadJobChangeRequest.createJob(destination:resource:) API, I don't see a way to specify byte ranges or create multiple jobs for chunks of the same resource. Questions: Does the system handle large files (1GB+) automatically under the hood, or is there a recommended maximum file size for a single upload job? Is there a supported pattern for chunked/resumable uploads, or should the destination URL endpoint handle the entire file in one request? If our server requires chunked uploads (e.g., BITS protocol with CreateSession → Fragment → CloseSession), is this extension the right mechanism, or should we use a different approach for large videos? Any guidance on best practices for large asset uploads would be greatly appreciated. Environment: iOS 26, Xcode 18

This is the code I use: @MainActor func picker()->UIViewController{ let pickerController = MPMediaPickerController(mediaTypes: .music) print(pickerController) pickerController.showsCloudItems=true pickerController.prompt = NSLocalizedString("Add pieces to queue", comment:""); pickerController.allowsPickingMultipleItems=true; pickerController.delegate=MPMusicPlayerControllerSingleton.sharedController(); return pickerController } MainActor @IBAction func handleBrowserTapped(_ sender: AnyObject){ if let pickerController=contentProvider?.picker(){ self.present(pickerController, animated:true, completion:nil) } } And this his is the crash log: *** Assertion failure in -[MPMediaPickerController_Appex requestRemoteViewController], MPMediaPickerController.m:523 *** Terminating app due to uncaught exception 'NSInternalInconsistencyException', reason: 'remoteViewController cannot be nil -- process will crash inserting in hierarchy. We likely got a nil remoteViewController because Music is crashing.' *** First throw call stack: (0x1869cac70 0x183499224 0x1844f9f50 0x1a6c6a060 0x18d45518c 0x18d4cd410 0x103354544 0x10336dccc 0x10338f748 0x103364204 0x103364144 0x186957a64 0x1868e5288 0x1868e41d0 0x22bde7498 0x18c5a5ca0 0x18c510254 0x18c71ce90 0x103854340 0x1038542b0 0x103854430 0x1834f1c1c) libc++abi: terminating due to uncaught exception of type NSException *** Terminating app due to uncaught exception 'NSInternalInconsistencyException', reason: 'remoteViewController cannot be nil -- process will crash inserting in hierarchy. We likely got a nil remoteViewController because Music is crashing.'

When trying to load an ambisonics file using this project: https://github.com/robertncoomber/NativeiOSAmbisonicPlayback/ I get "Unexpected Ambisonics format". Interestingly, loading a 3rd order ambisonics file works fine: let ambisonicLayoutTag = kAudioChannelLayoutTag_HOA_ACN_SN3D | 16 let AmbisonicLayout = AVAudioChannelLayout(layoutTag: ambisonicLayoutTag) let StereoLayout = AVAudioChannelLayout(layoutTag: kAudioChannelLayoutTag_Stereo) So it's purely related to the kAudioChannelLayoutTag_Ambisonic_B_Format

I am working on offline license support for FairPlay. I am trying to identify offline vs. live streaming requests. How do I do that? Is there any way to identify the request type (offline or streaming) from SPC?

Summary On iOS 26, HLS alternate audio track selection behaves inconsistently on both VOD and live streams: the French track falls back to the DEFAULT=YES (English) track after manual selection, and in some cases switching to a non-default track appears to work but it is then impossible to switch back to English. Environment iOS version: 26 Players affected: native Safari on iOS 26 and THEOplayer (issue also reproducible on THEOplayer's own demo page) Stream type: HLS/CMAF with demuxed alternate audio renditions (CMFC container) Affected stream types: both VOD and live streaming Issue NOT present on iOS 17/18 Manifest #EXTM3U #EXT-X-VERSION:4 #EXT-X-INDEPENDENT-SEGMENTS #EXT-X-STREAM-INF:BANDWIDTH=8987973,AVERAGE-BANDWIDTH=8987973,VIDEO-RANGE=SDR,CODECS="avc1.640028",RESOLUTION=1920x1080,FRAME-RATE=29.970,AUDIO="program_audio" video_1080p.m3u8 #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="program_audio",LANGUAGE="en",ASSOC-LANGUAGE="en",NAME="English",AUTOSELECT=YES,DEFAULT=YES,CHANNELS="2",URI="audio_ENG.m3u8" #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="program_audio",LANGUAGE="de",ASSOC-LANGUAGE="de",NAME="Deutsch",AUTOSELECT=YES,DEFAULT=NO,CHANNELS="2",URI="audio_DEU.m3u8" #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="program_audio",LANGUAGE="fr",ASSOC-LANGUAGE="fr",NAME="Francais",AUTOSELECT=YES,DEFAULT=NO,CHANNELS="2",URI="audio_FRA.m3u8" Steps to Reproduce Load the HLS manifest (VOD or live) in Safari on iOS 26, or in any AVFoundation-backed player Start playback — English plays correctly as DEFAULT Manually select "Francais" from the audio track selector Observe that English audio continues playing (French does not play) In a separate scenario: manually select "Deutsch" — German plays correctly Attempt to switch back to English — English does not resume; audio remains on the previously selected track Expected behavior Selecting any track should immediately switch to that language Switching back to English (DEFAULT=YES) should work at any time Behavior should be consistent across VOD and live streams Actual behavior Two distinct anomalies observed, reproducible on both VOD and live streams, in both native Safari and THEOplayer: French-specific fallback: selecting the French track causes playback to fall back to English. This does not happen with German. Cannot return to English: in cases where a non-default track plays correctly, attempting to switch back to the DEFAULT=YES track (English) fails — the previous non-default track continues playing. The fact that the issue reproduces in native Safari confirms this is an AVFoundation/WebKit-level regression, not a third-party player bug. What we have already verified and ruled out LANGUAGE codes are BCP-47 compliant (en, fr, de) ✓ EXT-X-VERSION:4 is present ✓ Audio codec removed from STREAM-INF CODECS (video-only) ✓ ASSOC-LANGUAGE attribute added matching LANGUAGE value ✓ Container metadata verified via ffprobe: mdhd box correctly contains language tags (e.g. "fra") ✓ Audio segment content verified via ffplay: correct audio in each language file ✓ French audio source file contains correct French audio content ✓ Issue reproduces in native Safari on iOS 26, confirming it is not a THEOplayer-specific bug Issue does NOT reproduce on iOS 17/18 with the same manifest and segments Additional notes The VOD stream is packaged with AWS MediaConvert, CMAF output group, SEGMENTED_FILES, AAC-LC codec (mp4a.40.2), 128kbps, 48kHz stereo. English uses AudioTrackType ALTERNATE_AUDIO_AUTO_SELECT_DEFAULT; French and German use ALTERNATE_AUDIO_AUTO_SELECT. The live stream uses AWS MediaPackage with a similar CMAF/HLS output configuration. Please advise whether this is a known regression in AVFoundation on iOS 26 and whether a fix is planned.

Hello, I am having an issue with a very small percentage of our users not being able to view the UI of our MIDI Plugin Chord Prism. I have looked this up and seen to where it has been resolved within Logic for AU Instrument and Effect plugins by switching out of "Controls" view, but my situation is different and there is no option on what is displayed to switch out of "Controls" view. Is this something that can be fixed by adjusting settings within Logic?

I have a bunch of Audio Unit v3 plugins that are approaching release, and I was considering using subscription-model pricing, as I have done in a soon to be released iOS app. However, whether this is possible or not is not at all obvious. Specifically: The plugin can, depending on the host app, be loaded in-process or out-of-process - yes, I know, Logic Pro and Garage Band will not load a plug-in in-process anymore, but I am not going to rule that out for other audio apps and force on them the overhead of IPC (I spent two solid weeks deciphering the process to actually make it possible for an AUv3 to run in-process - see this - https://github.com/timboudreau/audio_unit_rust_demo - example with notes) Depending on how it is loaded, the value of Bundle.main.bundleIdentifier will vary. If I use the StoreKit API, will that return product results for my bundle identifier when being called as a library from a foreign application? I would expect it would be a major security hole if random apps could query about purchases of other random apps, so I assume not. Even if I restricted the plugins to running out-of-process, I have to set up the in-app purchases on the app store for the App container's ID, not the extension's ID, and the extension is what run - the outer app that is what you purchase is just a toy demo that exists solely to register the audio unit. I have similar questions with regard to MetricKit, which I would similarly like to use, but which may be running inside some random app. If there were some sort of signed token, or similar mechanism, that could be bundled or acquired by the running plugin extension that could be used to ensure both StoreKit and MetricKit operate under the assumption that purchases and metrics should be accessed as if called from the container app, that would be very helpful. This is the difference between having a one-and-done sales model and something that provides ongoing revenue to maintain these products - I am a one-person shop - if I price these products where they would need to be to pay the bills assuming a single sale per customer ever, the price will be too high for anyone to want to try products from a small vendor they've never heard of. So, being able to do a free trial period and then subscription is the difference between this being a viable business or not.