Hi everyone, We’re currently developing a music-based app using MusicKit, and we recently noticed that iOS 26 beta introduces a new “Automix” feature in the Apple Music app. This enables seamless DJ-style transitions between songs—beyond the standard crossfade functionality. We’re trying to understand: Will this Automix feature be accessible to third-party apps that use MusicKit? If not available in the initial iOS 26 release, is there a plan to expose it through public APIs in a future update? Is there any technical documentation, WWDC session, or roadmap info regarding Automix support via MusicKit? This functionality would be a significant enhancement for our app, especially for intelligent audio transitions and curated playlists. Thanks.

AVCaptureVideoDataOutput.preparesCellularRadioForNetworkConnection requires com.apple.developer.avfoundation.video-data-output-prepares-cellular-radio-for-machine-readable-code-scanning. But I cannot acquire its entitlement. I can't find its entitlement on 'Certificates, Identifiers & Profiles'. Any solutions? Provisioning profile "iOS Team Provisioning Profile: ......" doesn't include the com.apple.developer.avfoundation.video-data-output-prepares-cellular-radio-for-machine-readable-code-scanning entitlement.

it will use about 300MB memory, it cause a memory peak

Issue: After iOS 18.5 release, our app is experiencing a significant increase in AVCaptureSessionInterruptionReason.videoDeviceNotAvailableWithMultipleForegroundApps errors. Details: Our camera-related code has not been updated recently.However, we've observed that the error rate has significantly increased starting from May 2025. The error rate has risen from approximately 0.02% (2 in 10,000 users) to 0.1% (1 in 1,000 users). This represents a 5x increase in error occurrence. The frequency has increased noticeably since iOS 18.5 This is affecting our app's camera functionality and user experience Questions: Are there any known changes in iOS 18.5 regarding camera access management? What are the recommended best practices to handle this interruption reason? Are there any API changes we should be aware of? Best, Shay

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?

We have a React website build to scan qr codes. The website is properly working for Android devices but for Iphone we see a camera glitch causing delay in scan which is unexpected. Website URL : https://react-qr-code-scanner-app.vercel.app/

I have sent in a feedback report (FB18222398) but I have no idea if anyone has looked at it. I know from past experiences that Apple devs do look at these forums. This applies to each of the betas, 1, 2 and 3. I have created a new Personal Voice with each beta. I create a personal voice in English. When it's done processing, I tap Preview and it says in English what is expected. But after some time, an hour or a day, the language of the voice file changes languages and no longer works properly. If I press Preview it is no longer intelligible. I have a text to speech app and initially the created voice works but then when the language of the file changes, it no longer works. I have run an app on my iphone through Xcode that prints to the console the voices installed on the device with the language. Currently this is the voice file: Voice Identifier: com.apple.speech.personalvoice.AAA9C6F2-9125-475F-BA2F-22C63274991D Language: es-MX and on a second device the same personal voice is in a different language: Voice Identifier: com.apple.speech.personalvoice.AAA9C6F2-9125-475F-BA2F-22C63274991D Language: zh-CN Although, a previous personal voice file that listed as Spanish-Mexican played in English with a Spanish accent or when playing Spanish text, it sounded almost perfect. This current personal voice doesn't do that, and is unintelligible. Previous attempts have converted to Chinese. I hope someone can look into this.

Our Final Cut Pro workflow extension built with ProExtensionHost framework uses an advanced NSPasteboardItemDataProvider system with multi-version FCPXML support (1.9, 1.10, 1.13) and proper relative path UIDs for Motion templates. We've implemented clip wrapper approach with placeholder assets and elements containing effects to enable direct timeline drag functionality. However, drag and drop from our Final Cut Pro workflow extension directly to timeline is still not working despite proper element structure in our FCPXML. Our implementation creates valid clip elements with effects applied, but Final Cut Pro timeline doesn't accept them during drag operations from our ProExtensionHost-based workflow extension. Steps to Reproduce: Create Final Cut Pro workflow extension using ProExtensionHost framework with NSPasteboardItemDataProvider implementation Generate FCPXML with proper element structure: Expected Result: Clip should be accepted by timeline and effect applied from workflow extension Actual Result: Timeline rejects drag operation from ProExtensionHost-based workflow extension Question: Are there additional requirements or ProExtensionHost API calls needed beyond standard NSPasteboardItemDataProvider for Final Cut Pro workflow extension timeline drag functionality?

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.

Since many users like me use Apple Music on Android, the app is almost as feature-rich as iOS. It would be fantastic if the developers could add the new iOS 26 features to the Android app, along with a minor UI change. I know it’s challenging to implement liquid glass on Android hardware or design, but features like auto-mix, pronunciation, and translation could be added. kindly consider this request !!!!

It's been an ask for a few years and I'm wondering if there are any plans, or whether the '26 SDKs/Tools allow Apple Music to work in the simulator? I develop for the Vision Pro so the usual 'fix' of running on the device is a bit of a hard ask. At the very least a small sample library that works in the simulator would be welcome (similar to how photos works) Cheers

I'm developing a Final Cut Pro X workflow extension that transcribes audio and creates a text output. I need to allow users to drag this text directly from my extension into FCPX's timeline as titles. Current Implementation: Using NSFilePromiseProvider as per Apple's guidelines for drag and drop Generating valid FCPXML (v1.10) with proper structure: Complete resources section with format and asset references Event and project hierarchy Asset clip with connected title elements Proper timing and duration calculations Supporting multiple pasteboard types: com.apple.finalcutpro.xml.v1-10 com.apple.finalcutpro.xml.v1-9 com.apple.finalcutpro.xml What's Working: Drag operation initiates correctly File promise provider is set up properly FCPXML generation is successful (verified content) All required pasteboard types are registered Proper logging confirms data is being requested and provided Current Pasteboard Types Offered: com.apple.NSFilePromiseItemMetaData com.apple.pasteboard.promised-file-name com.apple.pasteboard.promised-suggested-file-name com.apple.pasteboard.promised-file-content-type Apple files promise pasteboard type com.apple.pasteboard.NSFilePromiseID com.apple.pasteboard.promised-file-url com.apple.finalcutpro.xml.v1-10 com.apple.finalcutpro.xml.v1-9 com.apple.finalcutpro.xml What additional requirements or considerations are needed to make FCPX accept the dragged FCPXML content? Are there specific requirements for workflow extensions regarding drag and drop operations with titles that aren't documented? Any insights, especially from those who have implemented similar functionality in FCPX workflow extensions, would be greatly appreciated. Technical Details: macOS Version: 15.5 (24F74) FCPX Version: 11.1.1 Extension built with SwiftUI and AppKit integration Using NSFilePromiseProvider and NSPasteboardItemDataProvider Full pasteboard type support for FCPXML versions

Since MacOS 26 Apple Music has inconsitent drops to the Quality of some Tracks indiscrimantly. I don't know if others Expereinced it. It doesn't happen on the Speakers or connected via Bluetooth, but the AUX I/O has it quite often. It is more noticable on Headphones with 48kHz and higher Frequency Bandwidth. Here is the FB18062589

AVCaptureSession's startRunning method is thread blocking and seems to be slow. What is this method doing behind the scenes? For context: I'm working on Simulator Camera support and I have a 'fake' AVCaptureDevice that might be causing this. My hypothesis is that AVCaptureSession tries to connect to the device and waits for a notification to be posted back. I'd love to find a way to let my fake device message AVCaptureSession that it's connected.

I'm using AVFoundation to make a multi-track editor app, which can insert multiple track and clip, including scale some clip to change the speed of the clip, (also I'm not sure whether AVFoundation the best choice for me) but after making the scale with scaleTimeRange API, there is some short noise sound in play back. Also, sometimes it's fine when play AVMutableCompostion using AVPlayer with AVPlayerItem, but after exporting with AVAssetReader, will catch some short noise sounds in result file.... Not sure why. Here is the example project, which can build and run directly. https://github.com/luckysmg/daily_images/raw/refs/heads/main/TestDemo.zip

Before you post —Camera doesn't work on the Simulator— that's no longer true. I've made a solution that makes the Simulator believe there's an actual hardware device connected, allowing users to stream the macOS camera to the iOS Simulator (see for more info RocketSim's documentation: https://docs.rocketsim.app/features/hzQMSrSga7BGWvxdNVdwYs/simulator-camera-support/58tQ5jvevLNSnyUEA7VgAv) Now, it works for VNDocumentCameraViewController, but when I try opening DataScannerViewController, I directly run into: Failed to start scanning: The operation couldn’t be completed. (VisionKit.DataScannerViewController.ScanningUnavailable error 0.) My question: How does this view controller determine whether scanning is available? Is there a certain capability the available AVCaptureDevice's need to support maybe? Any direction would be helpful for me to make this work for developers, making them build apps faster!

My app is properly configured with MusicKit. I've generated a JWT using my valid credentials (Team ID, Key ID, private key), and I’ve ensured the time settings are correct via NTP. When I call: https://api.music.apple.com/v1/catalog/jp/search?term=ado&types=songs I consistently receive a 500 Internal Server Error. The JWT is generated using ES256 with valid iat and exp values. I’ve confirmed the token decodes properly using jwt.io, and it's passed via the Authorization: Bearer header. Things I’ve confirmed: Key ID, Team ID, private key are correct App ID is configured with MusicKit capability JWT is generated and signed correctly macOS time is synced via NTP Used both curl and Python to test — same result Is there anything else I should check on the Apple Developer Console (like App ID, Certificates, or provisioning profile)? Or could this be a backend issue on Apple’s side? Any guidance would be appreciated.

Hi everyone, I'm developing a visionOS app for Apple Vision Pro, and I've encountered an issue related to window resizing at runtime when using AVPlayer to play a live HLS stream. ✅ What I'm Trying to Do Play a live HLS stream (from Wowza) inside my app using AVPlayer. Support resizing the immersive window using Vision Pro’s built-in runtime scaling gesture. Stream works fine at default window size when the app launches. ❌ Problem If I resize the app’s window at runtime (using the Vision Pro pinch-drag gesture), then try to start the stream, it does not play. Instead, it just shows the "Loading live stream..." state and never proceeds to playback. This issue only occurs after resizing the window — if I don’t resize, the stream works perfectly every time. 🧪 What I’ve Tried Verified the HLS URL — it’s working and plays fine in Safari and in the app before resizing. Set .automaticallyWaitsToMinimizeStalling = false on AVPlayer. Observed that .status on AVPlayerItem never reaches .readyToPlay after resizing. Tried to force window size back using UIWindowScene.requestGeometryUpdate(...), but behavior persists.

This is my native module code implementation I'm getting base64 encoded string from server and passing this to my native module of pcm player to play audio App.tsx PcmPlayer.writeChunk(e.data); PcmPlayer.swift import AVFoundation @objc(PcmPlayer) class PcmPlayer: RCTEventEmitter { private var engine: AVAudioEngine? private var playerNode: AVAudioPlayerNode? private var format: AVAudioFormat? private var bufferQueue = [Data]() private var isPlaying = false private var hasEnded = false private var scheduledBufferCount = 0 private let minBufferBytes = 50000 private let pcmQueue = DispatchQueue(label: "pcm.queue") override init() { super.init() } override func supportedEvents() -> [String]! { return ["onStatus", "onMessage"] } @objc(initPlayer:channels:bitsPerSample:) func initPlayer(_ sampleRate: NSNumber, channels: NSNumber, bitsPerSample: NSNumber) { pcmQueue.async { self.stopInternal() let session = AVAudioSession.sharedInstance() do { try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true, options: .notifyOthersOnDeactivation) try session.setMode(.default) print("🔈 Audio session active. Output route:", session.currentRoute.outputs) } catch { print("❌ Audio session setup failed:", error) return } self.engine = AVAudioEngine() self.playerNode = AVAudioPlayerNode() guard let engine = self.engine, let playerNode = self.playerNode else { print("❌ Engine or playerNode is nil") return } engine.attach(playerNode) self.format = AVAudioFormat(commonFormat: .pcmFormatFloat32, sampleRate: sampleRate.doubleValue, channels: AVAudioChannelCount(channels.uintValue), interleaved: false) guard let format = self.format else { print("❌ Failed to create AVAudioFormat") return } engine.connect(playerNode, to: engine.mainMixerNode, format: format) do { try engine.start() playerNode.play() engine.mainMixerNode.outputVolume = 1.0 print("✅ AVAudioEngine started with format:", format) } catch { print("❌ Engine start failed:", error) } self.hasEnded = false } } @objc(writeChunk:) func writeChunk(_ base64Pcm: String) { pcmQueue.async { guard base64Pcm.count >= 10 else { print("⚠️ Skipping short base64 string") return } var padded = base64Pcm let mod4 = base64Pcm.count % 4 if mod4 > 0 { padded += String(repeating: "=", count: 4 - mod4) } guard let data = Data(base64Encoded: padded, options: .ignoreUnknownCharacters) else { print("❌ Failed to decode base64") return } self.bufferQueue.append(data) print("📥 Received PCM chunk (\(data.count) bytes)") print("📥 writeChunk called. isPlaying=\(self.isPlaying), bufferQueue.count=\(self.bufferQueue.count)") if !self.isPlaying { self.isPlaying = true self.waitForBufferAndStartPlayback() } else if self.scheduledBufferCount == 0 { self.isPlaying = true self.waitForBufferAndStartPlayback() } } } private func waitForBufferAndStartPlayback() { DispatchQueue.global().async { while self.queueSize() < self.minBufferBytes && !self.hasEnded { Thread.sleep(forTimeInterval: 0.01) } self.writeLoop() } } private func writeLoop() { DispatchQueue.global().async { writeLoop: while self.isPlaying { if self.bufferQueue.isEmpty { for _ in 0..<100 { Thread.sleep(forTimeInterval: 0.01) if !self.bufferQueue.isEmpty { break } } if self.bufferQueue.isEmpty { print("🔇 No more data to play after waiting") self.isPlaying = false break writeLoop } } var data: Data? self.pcmQueue.sync { if !self.bufferQueue.isEmpty { data = self.bufferQueue.removeFirst() } } guard let chunk = data else { print("⚠️ No data to process") continue } if let buffer = self.pcmBufferFromData(chunk) { self.scheduledBufferCount += 1 self.playerNode?.scheduleBuffer(buffer, completionHandler: { self.pcmQueue.async { self.scheduledBufferCount -= 1 if self.bufferQueue.isEmpty && self.scheduledBufferCount == 0 { print("ℹ️ Playback idle - waiting for more data") self.isPlaying = false } } }) } } } } private func pcmBufferFromData(_ data: Data) -> AVAudioPCMBuffer? { guard let format = self.format else { return nil } let frameCount = UInt32(data.count / 2) guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: frameCount) else { print("❌ Failed to create AVAudioPCMBuffer") return nil } buffer.frameLength = frameCount guard let floatChannelData = buffer.floatChannelData?[0] else { print("❌ floatChannelData is nil") return nil } data.withUnsafeBytes { (rawBuffer: UnsafeRawBufferPointer) in let int16Buffer = rawBuffer.bindMemory(to: Int16.self) let count = min(int16Buffer.count, Int(frameCount)) for i in 0..<count { floatChannelData[i] = Float32(int16Buffer[i]) / Float32(Int16.max) } } return buffer } @objc(stopPlayer) func stopPlayer() { pcmQueue.async { self.stopInternal() } } private func stopInternal() { print("🛑 stopInternal called") self.playerNode?.stop() self.engine?.stop() self.engine?.reset() self.playerNode = nil self.engine = nil self.format = nil self.bufferQueue.removeAll() self.isPlaying = false self.hasEnded = true self.scheduledBufferCount = 0 } @objc(canWrite:rejecter:) func canWrite(_ resolve: @escaping RCTPromiseResolveBlock, rejecter reject: RCTPromiseRejectBlock) { pcmQueue.async { resolve(self.bufferQueue.count < 20) } } @objc(flushPlayer:rejecter:) func flushPlayer(_ resolve: @escaping RCTPromiseResolveBlock, rejecter reject: RCTPromiseRejectBlock) { pcmQueue.async { self.bufferQueue.removeAll() resolve(nil) } } @objc static override func requiresMainQueueSetup() -> Bool { return false } private func queueSize() -> Int { return pcmQueue.sync { return self.bufferQueue.reduce(0) { $0 + $1.count } } } } I couldn't able to hear any audio via my real iOS device also it is working fine on emulator.

Hello everyone, I’m new to Swift development and have been working on an audio module that plays a specific sound at regular intervals - similar to a workout timer that signals switching exercises every few minutes. Following AVFoundation documentation, I’m configuring my audio session like this: let session = AVAudioSession.sharedInstance() try session.setCategory( .playback, mode: .default, options: [.interruptSpokenAudioAndMixWithOthers, .duckOthers] ) self.engine.attach(self.player) self.engine.connect(self.player, to: self.engine.outputNode, format: self.audioFormat) try? session.setActive(true) When it’s time to play cues, I schedule playback on a DispatchQueue: // scheduleAudio uses DispatchQueue self.scheduleAudio(at: interval.start) { do { try audio.engine.start() audio.node.play() for sample in interval.samples { audio.node.scheduleBuffer(sample.buffer, at: AVAudioTime(hostTime: sample.hostTime)) } } catch { print("Audio activation failed: \(error)") } } This works perfectly in the foreground. But once the app goes into the background, the scheduled callback runs, yet the audio engine fails to start, resulting in an error with code 561015905. Interestingly, if the app is already playing audio before going to the background, the scheduled sounds continue to play as expected. I have added the required background audio mode to my Info plist file by including the key UIBackgroundModes with the value audio. Is there anything else I should configure? What is the best practice to play periodic audio when the app runs in the background? How do apps like turn-by-turn navigation handle continuous audio playback in the background? Any advice or pointers would be greatly appreciated!