I’m building a music app using Apple Music streaming via ApplicationMusicPlayer. My goal is to decrease the volume of the current song during the last 10 seconds, and when the next track begins, restore the volume to its normal level. I know that ApplicationMusicPlayer doesn’t expose a volume API, and I want to avoid triggering the system volume HUD. ✅ Using Apple Music streaming (not local files) ❓ Is it possible to implement per-track fade-out/fade-in logic with ApplicationMusicPlayer? Appreciate any clarification or official guidance!

Hi everyone, I’m trying to use AVAssetResourceLoaderDelegate to handle a live radio stream (e.g. Icecast/HTTP stream). My goal is to have access to the last 30 seconds of audio data during playback, so I can analyze it for specific audio patterns in near-real-time. I’ve implemented a custom resource loader that works fine for podcasts and static files, where the file size and content length are known. However, for infinite live streams, my current implementation stops receiving new loading requests after the first one is served. As a result, the playback either stalls or fails to continue. Has anyone successfully used AVAssetResourceLoaderDelegate with a continuous radio stream? Or maybe you can suggest betterapproach for buffering and analyzing live audio? Any tips, examples, or advice would be appreciated. Thanks!

I'm working on a media app that would like to be able to tell if the TV connected to tvOS is running at 59.94hz or 60.00hz, so it can optimize a video stream. It looks like the best I can currently do is to check if the user has Match Content Rate enabled, and based on that, when calling displayManager.preferredDisplayCriteria to change video modes, I could guess which rate their TV might be in. It's not very ideal, because not all TVs support both of these rates, and my request for 59.94 might end up as 60 and vice versa. I dug around and can't find any available method in UIScreen to get this info. The odd thing is, the data is right there in currentMode when I look in the debugger, but it seems to be in a private or undocumented class. Is there any way to get at it?

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

(Note: this is part 3 of a 3 part posting. See Part 1 or Part 2) At WWDC25 we launched a new type of Lab event for the developer community - Group Labs. A Group Lab is a panel Q&A designed for a large audience of developers. Group Labs are a unique opportunity for the community to submit questions directly to a panel of Apple engineers and designers. Here are the highlights from the WWDC25 Group Lab for Camera & Photos. WWDC25 Camera & Photos group lab ran for one hour at 6 PM PST on Tuesday June 10th, 2025 Question 24 What’s the best approach for optimizing barcode scanning using AVFoundation or Vision in low-light or angled scenarios Turn on flash in low-light scenarios Lower framerate to improve exposure and reduce noise Wait until the capture is in focus/notify your user that they need to get closer Question 25 Recent iPhone models introduced macro mode which automatically switch between lenses to take into account of the focal distance difference. Is there official API to implement this, or should I implement them myself using LiDAR values. Using builtInTripleCamera and builtInDualWideCamera will automatically switch to macro when available Question 26 Is there a way to quickly create a thumbnail after the user selects an image with PhotosPicker? File provider API Additional questions from the WWDC25 in-person labs that occurred later in the WWDC week Question 1 When should I build my custom photo picker instead of using the system one? Always start with the system picker -> try embeddable customization APIs -> fallback to custom picker for very special needs Question 2 I'm building a new camera app for pros and I want to give my users the most un-processed image possible, and the most control over the capture as possible. How can I do that with AVCapture? If stills, Brief Bayer RAW capture overview, or Pro RAW if you want Apple's processing and dynamic range If video, talk about prores LOG. Custom exposure settings are available throguh the apis maybe global/local tonemapping discussion?

Hi Team, We are using AVFoundation to read metadata from a stream and have noticed some delay between when the stream provides metadata and when the app receives it. Could someone from the team advise on ways to reduce this? Thanks

On macOS Sequoia, I'm having the hardest time getting this basic audio output to work correctly. I'm compiling in XCode using C99, and when I run this, I get audio for a split second, and then nothing, indefinitely. Any ideas what could be going wrong? Here's a minimum code example to demonstrate: #include <AudioToolbox/AudioToolbox.h> #include <stdint.h> #define RENDER_BUFFER_COUNT 2 #define RENDER_FRAMES_PER_BUFFER 128 // mono linear PCM audio data at 48kHz #define RENDER_SAMPLE_RATE 48000 #define RENDER_CHANNEL_COUNT 1 #define RENDER_BUFFER_BYTE_COUNT (RENDER_FRAMES_PER_BUFFER * RENDER_CHANNEL_COUNT * sizeof(f32)) void RenderAudioSaw(float* outBuffer, uint32_t frameCount, uint32_t channelCount) { static bool isInverted = false; float scalar = isInverted ? -1.f : 1.f; for (uint32_t frame = 0; frame < frameCount; ++frame) { for (uint32_t channel = 0; channel < channelCount; ++channel) { // series of ramps, alternating up and down. outBuffer[frame * channelCount + channel] = 0.1f * scalar * ((float)frame / frameCount); } } isInverted = !isInverted; } AudioStreamBasicDescription coreAudioDesc = { 0 }; AudioQueueRef coreAudioQueue = NULL; AudioQueueBufferRef coreAudioBuffers[RENDER_BUFFER_COUNT] = { NULL }; void coreAudioCallback(void* unused, AudioQueueRef queue, AudioQueueBufferRef buffer) { // 0's here indicate no fancy packet magic AudioQueueEnqueueBuffer(queue, buffer, 0, 0); } int main(void) { const UInt32 BytesPerSample = sizeof(float); coreAudioDesc.mSampleRate = RENDER_SAMPLE_RATE; coreAudioDesc.mFormatID = kAudioFormatLinearPCM; coreAudioDesc.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked; coreAudioDesc.mBytesPerPacket = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mFramesPerPacket = 1; coreAudioDesc.mBytesPerFrame = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mChannelsPerFrame = RENDER_CHANNEL_COUNT; coreAudioDesc.mBitsPerChannel = BytesPerSample * 8; coreAudioQueue = NULL; OSStatus result; // most of the 0 and NULL params here are for compressed sound formats etc. result = AudioQueueNewOutput(&coreAudioDesc, &coreAudioCallback, NULL, 0, 0, 0, &coreAudioQueue); if (result != noErr) { assert(false == "AudioQueueNewOutput failed!"); abort(); } for (int i = 0; i < RENDER_BUFFER_COUNT; ++i) { uint32_t bufferSize = coreAudioDesc.mBytesPerFrame * RENDER_FRAMES_PER_BUFFER; result = AudioQueueAllocateBuffer(coreAudioQueue, bufferSize, &(coreAudioBuffers[i])); if (result != noErr) { assert(false == "AudioQueueAllocateBuffer failed!"); abort(); } } for (int i = 0; i < RENDER_BUFFER_COUNT; ++i) { RenderAudioSaw(coreAudioBuffers[i]->mAudioData, RENDER_FRAMES_PER_BUFFER, RENDER_CHANNEL_COUNT); coreAudioBuffers[i]->mAudioDataByteSize = coreAudioBuffers[i]->mAudioDataBytesCapacity; AudioQueueEnqueueBuffer(coreAudioQueue, coreAudioBuffers[i], 0, 0); } AudioQueueStart(coreAudioQueue, NULL); sleep(10); // some time to hear the audio AudioQueueStop(coreAudioQueue, true); AudioQueueDispose(coreAudioQueue, true); return 0; }

I tried to modify the AVCam sample code by copying the code here https://developer.apple.com/documentation/avfoundation/adopting-smart-framing-in-your-camera-app#Configure-the-smart-framing-monitor smart framing monitors I can ensure the activeformat supports smart framing, but the supported frames in monitor is always nil. In my another project it has supported value, but the observation has never been triggered, then I tried to keep printing the recommended frame, it's always nil. Could the engineer embed the code into AVCam rather than posting a few code pieces?

Hi, Could anybody share the full list of supported Predicate keys for the PHFetchOptions? I'm aware of the list that is posted in the documentation: https://developer.apple.com/documentation/photos/phfetchoptions However I have reason to believe that this is not an exhaustive list and there also seem to be mistakes in this doc. i.e. isFavorite does not work but favorite does. Through some experimentation I also found that this works: NSPredicate(format: "adjustmentFormatIdentifier == 'com.pixelmatorteam.touch.x.photo.PhotosAdjustmentData.EmbeddedSlimSidecarFileInfo.compressed'") even though adjustmentFormatIdentifier is not listed as a supported key. Are there other secret keys that you are aware of? Specifically I want to filter a fetch result for edited items. Something like this: NSPredicate(format: "hasAdjustments == true") (I tried this, doesn't work) The native Photos app has such a filter which leads me to believe that there probably is a key for this: If one of the Framework Developers reads this: Could you please update the documentation page with this information? Finally if there really aren't any more secret keys, is there a way to achieve this with adjustmentFormatIdentifier? I have tried a bunch of stuff already like adjustmentFormatIdentifier != nil but for some reason that gives me the exact opposite of what I want: all the photos without edits. 🫠 Any tips on the correct syntax here would be much appreciated.

Is there any way for me to use an AutoMix api in my IOS apps, I would play tracks using the Apple Music api and use AutoMix to attempt to merge tracks. Is this feature/api available to developers.

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/

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.

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!

I'm streaming mp3 audio data using URLSession/AudioFileStream/AVAudioConverter and getting occasional silent buffers and glitches (little bleeps and whoops as opposed to clicks). The issues are present in an offline test, so this isn't an issue of underruns. Doing some buffering on the input coming from the URLSession (URLSessionDataTask) reduces the glitches/silent buffers to rather infrequent, but they do still happen occasionally. var bufferedData = Data() func parseBytes(data: Data) { bufferedData.append(data) // XXX: this buffering reduces glitching // to rather infrequent. But why? if bufferedData.count > 32768 { bufferedData.withUnsafeBytes { (bytes: UnsafeRawBufferPointer) in guard let baseAddress = bytes.baseAddress else { return } let result = AudioFileStreamParseBytes(audioStream!, UInt32(bufferedData.count), baseAddress, []) if result != noErr { print("❌ error parsing stream: \(result)") } } bufferedData = Data() } } No errors are returned by AudioFileStream or AVAudioConverter. func handlePackets(data: Data, packetDescriptions: [AudioStreamPacketDescription]) { guard let audioConverter else { return } var maxPacketSize: UInt32 = 0 for packetDescription in packetDescriptions { maxPacketSize = max(maxPacketSize, packetDescription.mDataByteSize) if packetDescription.mDataByteSize == 0 { print("EMPTY PACKET") } if Int(packetDescription.mStartOffset) + Int(packetDescription.mDataByteSize) > data.count { print("❌ Invalid packet: offset \(packetDescription.mStartOffset) + size \(packetDescription.mDataByteSize) > data.count \(data.count)") } } let bufferIn = AVAudioCompressedBuffer(format: inFormat!, packetCapacity: AVAudioPacketCount(packetDescriptions.count), maximumPacketSize: Int(maxPacketSize)) bufferIn.byteLength = UInt32(data.count) for i in 0 ..< Int(packetDescriptions.count) { bufferIn.packetDescriptions![i] = packetDescriptions[i] } bufferIn.packetCount = AVAudioPacketCount(packetDescriptions.count) _ = data.withUnsafeBytes { ptr in memcpy(bufferIn.data, ptr.baseAddress, data.count) } if verbose { print("handlePackets: \(data.count) bytes") } // Setup input provider closure var inputProvided = false let inputBlock: AVAudioConverterInputBlock = { packetCount, statusPtr in if !inputProvided { inputProvided = true statusPtr.pointee = .haveData return bufferIn } else { statusPtr.pointee = .noDataNow return nil } } // Loop until converter runs dry or is done while true { let bufferOut = AVAudioPCMBuffer(pcmFormat: outFormat, frameCapacity: 4096)! bufferOut.frameLength = 0 var error: NSError? let status = audioConverter.convert(to: bufferOut, error: &error, withInputFrom: inputBlock) switch status { case .haveData: if verbose { print("✅ convert returned haveData: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(haveData) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } case .inputRanDry: if verbose { print("🔁 convert returned inputRanDry: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(inputRanDry) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } return // wait for next handlePackets case .endOfStream: if verbose { print("✅ convert returned endOfStream") } return case .error: if verbose { print("❌ convert returned error") } if let error = error { print("error converting: \(error.localizedDescription)") } return @unknown default: fatalError() } } }

Hi, I downloaded and ran https://developer.apple.com/documentation/realitykit/rendering-stereoscopic-video-with-realitykit and noticed that memory usage grows linearly. I replaced the sample video with a different 8k side by side video, and the app crashed almost immediately due to memory leak. it looks like the culprit is from makeMutablePixelBuffer() function and the allocated pixelBuffers are not recycled after being used. screenshot is from a physical device.

I'm working in Swift/SwiftUI, running XCode 16.3 on macOS 15.4 and I've seen this when running in the iOS simulator and in a macOS app run from XCode. I've also seen this behaviour with 3 different audio files. Nothing in the documentation says that the speechRecognitionMetadata property on an SFSpeechRecognitionResult will be nil until isFinal, but that's the behaviour I'm seeing. I've stripped my class down to the following: private var isAuthed = false // I call this in a .task {} in my SwiftUI View public func requestSpeechRecognizerPermission() { SFSpeechRecognizer.requestAuthorization { authStatus in Task { self.isAuthed = authStatus == .authorized } } } public func transcribe(from url: URL) { guard isAuthed else { return } let locale = Locale(identifier: "en-US") let recognizer = SFSpeechRecognizer(locale: locale) let recognitionRequest = SFSpeechURLRecognitionRequest(url: url) // the behaviour occurs whether I set this to true or not, I recently set // it to true to see if it made a difference recognizer?.supportsOnDeviceRecognition = true recognitionRequest.shouldReportPartialResults = true recognitionRequest.addsPunctuation = true recognizer?.recognitionTask(with: recognitionRequest) { (result, error) in guard result != nil else { return } if result!.isFinal { //speechRecognitionMetadata is not nil } else { //speechRecognitionMetadata is nil } } } } Further, and this isn't documented either, the SFTranscriptionSegment values don't have correct timestamp and duration values until isFinal. The values aren't all zero, but they don't align with the timing in the audio and they change to accurate values when isFinal is true. The transcription otherwise "works", in that I get transcription text before isFinal and if I wait for isFinal the segments are correct and speechRecognitionMetadata is filled with values. The context here is I'm trying to generate a transcription that I can then highlight the spoken sections of as audio plays and I'm thinking I must be just trying to use the Speech framework in a way it does not work. I got my concept working if I pre-process the audio (i.e. run it through until isFinal and save the results I need to json), but being able to do even a rougher version of it 'on the fly' - which requires segments to have the right timestamp/duration before isFinal - is perhaps impossible?

Hi there, I want to set the iphone camera to "S mode, or Shutter Priority" in camera terminology. Which is a semi-auto exposure model with shutter speed fixed, or set manually. However, when setting the shutter speed manually, it disables the auto exposure. So is there a way to keep the auto exposure on while restrict the shutter speed? Also, I would like to keep a low frame rate, e.g. 30 fps. Would I be able to set shutter speed independent of frame rate? Here's the code for setting up the camera Best,

Hi, On macOS I used to open MP3 and MP4 files with ExtAudioFile. For a few years it doesn't work anymore. So I decided to try different macOS API using the AudioFileID of AudioToolbox framework. I decided to write a test: https://gist.github.com/joelkraehemann/7f5b241b52ca38c3a765c138fb647588 It fails right here: AudioFileOpenWithCallbacks() By telling OSStatus error 1954115647, which means kAudioFileUnsupportedFileTypeError. The filename was set to an MP4 file: ~/Music/test.mp4 Howto fix this? regards, Joël

I'm experiencing an issue with my app when saving images to the camera roll. This is intermittent, but it happens several times a day. The error I receive is the following: Connection to assetsd was interrupted - assetsd exited, died, or closed the photo library Error getting remote object proxy for -[PLNonBindingAssetsdPhotoKitClient sendChangesRequest:reply:]_block_invoke: Error Domain=NSCocoaErrorDomain Code=4097 "connection to service named com.apple.photos.service" UserInfo={NSDebugDescription=connection to service named com.apple.photos.service} PhotoKit XPC proxy is invalid. Dropping request on the floor and returning an error: Error Domain=PHPhotosErrorDomain Code=3301 "(null)" (underlying error Error Domain=NSCocoaErrorDomain Code=4097 "connection to service named com.apple.photos.service" UserInfo={NSDebugDescription=connection to service named com.apple.photos.service}) CoreData: error: XPC: synchronousRemoteObjectProxyWithErrorHandler: store 'file:///var/mobile/Media/PhotoData/Photos.sqlite' encountered error: Error Domain=NSCocoaErrorDomain Code=4099 "The connection to service created from an endpoint was invalidated: failed to check-in, peer may have been unloaded: mach_error=10000003." UserInfo={NSDebugDescription=The connection to service created from an endpoint was invalidated: failed to check-in, peer may have been unloaded: mach_error=10000003.} CoreData: error: XPC: synchronousRemoteObjectProxyWithErrorHandler: store 'file:///var/mobile/Media/PhotoData/Photos.sqlite' encountered error: Error Domain=NSCocoaErrorDomain Code=4099 "The connection to service created from an endpoint was invalidated: failed to check-in, peer may have been unloaded: mach_error=10000003." UserInfo={NSDebugDescription=The connection to service created from an endpoint was invalidated: failed to check-in, peer may have been unloaded: mach_error=10000003.} My code is unchanged from using my app daily on an iPhone 16 Pro with iOS 26. I never saw the issue on this device. Here is an excerpt from my code for saving the image: var localIdentifier = String() PHPhotoLibrary.shared().performChanges({ let albumChangeRequest = PHAssetCollectionChangeRequest(for: album) let assetCreationRequest = PHAssetCreationRequest.forAsset() let options = PHAssetResourceCreationOptions() assetCreationRequest.addResource(with: .photo, data: imageData, options: options) assetCreationRequest.creationDate = Date.now let placeHolder = assetCreationRequest.placeholderForCreatedAsset albumChangeRequest?.addAssets([placeHolder!] as NSArray) if placeHolder != nil { localIdentifier = (placeHolder?.localIdentifier)! } }) { (didSucceed, error) in OperationQueue.main.addOperation({ didSucceed ? success(localIdentifier) : failure(error) }) } I'm not sure why this would be device specific but I have had users with iPhone 17 Pro and iPhone Air reporting the issue. Alex

Since the last update to IOS 26.0 (23A5276f) the AirPods connect to my IPhone and the Audio is still running through the phone. They are shown in the Bluetooth Icon that they’re paired.