Hello, I have an iOS camera app that captures exposure brackets and performs custom HDR processing. On iOS 26, I’m observing a visual difference between: a single photo captured at –2 EV, and the –2 EV frame from an exposure bracket (–2 / 0 / +2 EV). On iOS 26: The single –2 EV image looks natural and consistent. The –2 EV image from the bracket appears clamped / distorted, most noticeably in high dynamic range scenes (highlight compression and loss of detail). On iOS 18, both approaches produce visually identical and correct –2 EV images. The issue only appears for bracketed captures on iOS 26. Attachments (examples) iOS 26 Single capture –2 EV (JPEG): /Users/danilobudimir/Downloads/ios26SingleImage/JPEG image-4006-8B77-51-0.jpeg Single capture –2 EV — Capture report (dumped settings): /Users/danilobudimir/Downloads/ios26SingleImage/UnderExposureDebug_CaptureReport_2026-01-09T15-59-20Z.md Bracket capture –2 EV frame (JPEG): /Users/danilobudimir/Downloads/bracket_iOS26/JPEG image-45CE-9793-A5-0.jpeg Bracket capture — Capture report (dumped settings): /Users/danilobudimir/Downloads/bracket_iOS26/UnderExposureDebug_CaptureReport_2026-01-09T15-55-42Z.md iOS 18 Single capture –2 EV (JPEG): /Users/danilobudimir/Downloads/ios18SingleImage/JPEG image-47FD-AF73-28-0.jpeg Single capture –2 EV — Capture report: /Users/danilobudimir/Downloads/ios18SingleImage/UnderExposureDebug_CaptureReport_2026-01-09T16-25-27Z.md Bracket capture — –2 EV frame (JPEG): /Users/danilobudimir/Downloads/bracket_iOS18/JPEG image-4A4C-9E93-46-0.jpeg Bracket capture — Capture report: /Users/danilobudimir/Downloads/bracket_iOS18/UnderExposureDebug_CaptureReport_2026-01-09T16-27-23Z.md Question Is there any new behavior in iOS 26 AVFoundation related to: AVCapturePhotoBracketSettings, tone mapping / HDR preprocessing, or internal image processing applied specifically to bracketed frames? Is there a new flag, format requirement or opt-out mechanism required to preserve linear underexposed frames in exposure brackets?

It's been well over a year since Apple added favoriting of artists back to Apple Music (the little star icon on an artist page), but yet I still haven't seen a way to get this data from an authenticated user from Music API. I was expecting to hear something about this during the WWDC, but there have been no announcements that I've caught. Has anyone else heard anything? People assume when they provide access to their Apple Music account that we can actually get to the data in their Apple Music account, and we end up looking a little dumb not being able to get this core data.

When using AVSampleBufferDisplayLayer to play uncompressed H.264 and H.265 video with B-frames more than 7, frame drops occur. The more B-frames there are, the more noticeable the frame drops become, for example 15 bframes. Use FFmpeg to transcode a video file with visible timestamps and frame numbers (x264 or x265 ): ffmpeg -i test.mp4 -vf "drawtext=fontsize=45:text=%{pts} %{n}:y=400" -c:v libx264 -x264-params "bframes=15:b-adapt=0" -crf 30 -y x264_bf15.mp4 ffmpeg -i test.mp4 -vf "drawtext=fontsize=45:text=%{pts} %{n}:y=400" -c:v libx265 -x265-params "bframes=15:b-adapt=0" -crf 30 -y x265_bf15.mp4 Use the demo player from this repository to reproduce the issue: https://github.com/msfrms/CustomPlayer frame drops can be observed. And following log can be found in devices console. mediaserverd <<<< IQ-CA >>>> piqca_gmstats_dump: FIQCA(0x1266f4000) recent frames: enqueued: 184, displayed: 138, dropped: 42, flushed: 0, evicted: 3, >16ms late: 2 PS. I was using iphone11 iOS14.6, to replay this issue. May I ask why frame drops occur in this case? Is there any configuration or API usage change that could help fix the frame drop issue? Many thanks!

I was advised to post here by a Code-Level Support representative. Below will be a copy of my initial issue report, and my minimally reproductive test project can be found at the following GitHub repository URL... https://github.com/PierceLBrooks/vtUudSeiNalCmake DESCRIPTION OF PROBLEM When encoding H264 video codec data using the VTCompressionSession API facilities available through the VideoToolbox framework on MacOS, the resultant bitstream will invariably include Unregistered User Data SEI NAL units that carry the UUID "47564adc-5c4c-433f-94ef-c5113cd143a8". The proprietary decoders we are working with currently struggle with filtering out these NAL units. Can you explain what purpose this serves, what the meaning of the byte-wise unit payloads are, and which configuration settings the VideoToolbox encoder instance specifically depends upon for triggering the insertion of them? STEPS TO REPRODUCE 1. Invoke the instantiation of a new VideoToolbox H264 encoder object by calling VTCompressionSessionCreate with appropriate configuration flags. 2. Push frames through the encoder, receiving their encoded byte buffer counterparts through an asynchronous callback. 3. Write that encoded data to some buffer which will contain the totality of the encoder's output. 4. Inspect the NAL units of the initial portion of this output bitstream buffer. 5. Observe the presence of at least one Unregistered User Data SEI NAL unit carrying the "47564adc-5c4c-433f-94ef-c5113cd143a8" UUID near the beginning of the output segment.

Hello, I have a CarPlay Navigation app and utilize the AVSpeechSynthesizer to speak directions to a user. Everything works great on my CarPlay simulator as well as when plugged into my GMC truck. However, I found out yesterday that one of my users with a Ford truck the audio would cut in an out. After much troubleshooting, I was able to replicate this on my own truck when using Bluetooth to connect to CarPlay. My user was also utilizing Bluetooth. Has anyone else experienced this? Is there a fix to the problem? import SwiftUI import AVFoundation class TextToSpeechService: NSObject, ObservableObject, AVSpeechSynthesizerDelegate { private var speechSynthesizer = AVSpeechSynthesizer() static let shared = TextToSpeechService() override init() { super.init() speechSynthesizer.delegate = self } func configureAudioSession() { speechSynthesizer.delegate = self do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .voicePrompt, options: [.mixWithOthers, .allowBluetooth]) } catch { print("Failed to set audio session category: \(error.localizedDescription)") } } func speak(_ text: String) { Task(priority: .high) { let speechUtterance = AVSpeechUtterance(string: text) speechUtterance.voice = AVSpeechSynthesisVoice(language: AVSpeechSynthesisVoice.currentLanguageCode()) try AVAudioSession.sharedInstance().setActive(true, options: .notifyOthersOnDeactivation) speechSynthesizer.speak(speechUtterance) } } func speechSynthesizer(_ synthesizer: AVSpeechSynthesizer, didFinish utterance: AVSpeechUtterance) { Task { stopSpeech() try AVAudioSession.sharedInstance().setActive(false) } } func stopSpeech() { speechSynthesizer.stopSpeaking(at: .immediate) } }

Hello, As far as I know and in all of my testing there is no way for a user or a developer to change the frame rate of the video output on iPadOS. If you connect an iPad via a USB Hub or a USB to HDMI Adaptor and then connect it to an external monitor it will output at 59.94fps. I have a video app where a user monitors live video at 25fps and 30fps, they often output to an external display and there are times when the external display will stutter due to the mismatch in frame rate, ie. using 25fps and outputting at 59.94fps. I thought it was impossible to change the video output frame rate, then in V3.1 of the Blackmagic Camera App I saw an interesting change in their release notes: ‘Support for HDMI Monitoring at Sensor Rate and Resolution’ This means there is some way to modify it, not sure if this is done via a Private API that Apple has allowed Blackmagic to use. If so, how can we access this or is there a way to enable this that is undocumented? Thanks!

Hello, I am developing a custom player SDK using AVPlayer to support HLS and LL-HLS live streaming. I have some questions about the internal logic of AVPlayer regarding ABR, as this information is not explicitly covered in the documentation. ABR Switching Logic: Does AVPlayer trigger bitrate switching primarily based on stall occurrences (buffer starvation)? I am curious if the switching logic is reactive to stalls or if it proactively switches to prevent them based on throughput estimation. Developer Controls for ABR: To influence or control the ABR selection, are preferredPeakBitRate and preferredForwardBufferDuration the only properties available to developers? Are there any other recommended APIs to assist with ABR decisions? Thank you for your help.

Hello everyone, I'm working on implementing a screen sharing feature using RPSystemBroadcastPickerView and a Broadcast Upload Extension to share the entire app screen in an iOS application. The Broadcast Upload Extension is set up following Apple's ReplayKit guidelines. However, I’m encountering an issue during the broadcast startup sequence: ❗ Problem Description The Screen Broadcast UI appears as expected I tap “Start Broadcast” The countdown (3 → 2 → 1) completes Then it immediately reverts to the "Start Broadcast" screen, and screen sharing does not begin No error messages are displayed None of the extension lifecycle methods (broadcastStarted(withSetupInfo:), processSampleBuffer, etc.) are called There are no logs or crash reports, neither in the main app nor in the extension ✅ What Has Been Verified Info.plist of the Broadcast Upload Extension includes: NSExtensionPointIdentifier = com.apple.broadcast-services-upload NSExtensionPrincipalClass set correctly RPBroadcastProcessMode = RPBroadcastProcessModeSampleBuffer preferredExtension is set properly to the extension’s bundle identifier Extension is listed in the main app's build settings under "Frameworks, Libraries, and Embedded Content" ⚠️ Additional Concern We noticed that in Xcode (latest version), the Broadcast Upload Extension is listed under "Embedded Frameworks" with the setting "Embed Without Signing", and there is no option to change it to "Embed & Sign". We're wondering if this could be the reason the extension fails to launch correctly at runtime, despite being detected by the broadcast picker. ❓ Questions Has anyone faced similar issues where the broadcast never starts despite correct setup? Could the "Embed Without Signing" be causing the system to silently cancel or ignore the extension at runtime? Are there any provisioning profile or entitlement requirements specific to Broadcast Upload Extensions that might trigger this behavior silently? Any insights, suggestions, or workarounds would be greatly appreciated. Thank you in advance!

Hello, Starting in iOS 17, our application started having some issue publishing to our video session. More specifically the video capture seems to be broken in some, but not all sessions. What's troubling is that we're seeing that it fails consistently every 4 sessions. It also fails silently, without reporting any problems to the app. We only notice that there are no frames being rendered or sent to the remote devices. Here's what shows-up in the console: <<<< FigCaptureSourceRemote >>>> Fig assert: "! storage->connectionDied" at bail (FigCaptureSourceRemote.m:235) - (err=0) <<<< FigCaptureSourceRemote >>>> Fig assert: "err == 0 " at bail (FigCaptureSourceRemote.m:253) - (err=-16453) Anyone seeing this? Any idea what could be the cause? Our sessions work perfectly on iOS16 and below. Thanks

using iOS 26.2; Airpods 4 Long press stem to launch Siri Speak "Record Voice Memo" -> Recording starts Recording in progress... Long press stem to launch Siri -> Nothing happens. To stop recording need use phone. is this intended behaviour? i would like to be able to stop recording with Siri I am able to launch Siri from phone while recording, but point is to keep phone in pocket and start/stop recordings only via Airpods.

Is it possible to stream video from a UVC (USB Video Class) camera on an iPhone 15? If so, are there any specific hardware or software requirements to enable this functionality?

I'm working on adding CarPlay support to an audio app and am running into an issue. Occasionally, when a user opens the app from CarPlay while the main app scene is either not connected or is currently in the background, I will receive an error when attempting to activate the audio session. The code below mimics my setup: do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio) try AVAudioSession.sharedInstance().setActive(true) } catch { print(error) // NSOSStatusErrorDomain - 560557684: Session activation failed } That error code maps to AVAudioSession.ErrorCode.cannotInterruptOthers. Once in this state, all subsequent attempts to play different pieces of content will fail. However, things will start working normally if the user opens the app on their phone and tries again from CarPlay (while the app is in the foreground on their phone). I'm not sure why it would behave this way and want to note that I do have the audio background mode capability enabled. Has anyone else encountered this? Are there any workarounds or changes I could make to prevent this from happening?

I'm receiving output from avcapturesession and capturing an image using Vision, but the image is output in landscape orientation instead of portrait. Even when I set the orientation to up in ciimage, cgimage, and uiimage, the image is still output in landscape orientation. On iPhones 16 and below, the image is output in portrait orientation. But on iPhones 17 and above, the image is output in landscape orientation. Please help.

The two ScreenCaptureKit WWDC22 sessions show how to capture with the new framework but the retina factor is hardcoded to 2 in SCStreamConfiguration. When using on a non-retina display, the screencapture is floating on the upper-left corner of the image buffer. There does not seem to be a simple way to retrieve the retina factor from the SCShareableContent data (when configuring the capture). When processing the streaming output, the SCStreamFrameInfo attachment is supposed to have a scaleFactor property but .scaleFactor does not return a value. I have found out that the attachement dictionary contains SCStreamUpdateFrameDisplayResolution. This entry gives me the retina factor but it is not an official SCStreamFrameInfo key. I list the keys to access it. What is the proper way with ScreenCapture to handle the retina factors ?

I've been wondering if there is a way to modify or even disable tones for indicating channel states. The behaviour regarding tones seems like a black box with little documentation. During migration to Apple's PT Framework we've noticed that there are few scenarios where a tone is played which doesn't match certain certifications. For example; moving from a channel to another produces a tone which would fail a test case. I understand the reasoning fully, as it marks that the channel is ready to transmit or receive, but this doesn't mirror the behaviour of TETRA which would be wanted in this case. I'm also wondering if there would be any way to directly communicate feedback regarding PT Framework?

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'm writing a simple app for iOS and I'd like to be able to do some text to speech in it. I have a basic audio manager class with a "speak" function: import Foundation import AVFoundation class AudioManager { static let shared = AudioManager() var audioPlayer: AVAudioPlayer? var isPlaying: Bool { return audioPlayer?.isPlaying ?? false } var playbackPosition: TimeInterval = 0 func playSound(named name: String) { guard let url = Bundle.main.url(forResource: name, withExtension: "mp3") else { print("Sound file not found") return } do { if audioPlayer == nil || !isPlaying { audioPlayer = try AVAudioPlayer(contentsOf: url) audioPlayer?.currentTime = playbackPosition audioPlayer?.prepareToPlay() audioPlayer?.play() } else { print("Sound is already playing") } } catch { print("Error playing sound: \(error.localizedDescription)") } } func stopSound() { if let player = audioPlayer { playbackPosition = player.currentTime player.stop() } } func speak(text: String) { let synthesizer = AVSpeechSynthesizer() let utterance = AVSpeechUtterance(string: text) utterance.voice = AVSpeechSynthesisVoice(language: "en-GB") synthesizer.speak(utterance) } } And my app shows text in a ScrollView: ScrollView { Text(self.description) .padding() .foregroundColor(.black) .font(.headline) .background(Color.gray.opacity(0)) }.onAppear { AudioManager.shared.speak(text: self.description) } However, the text doesn't get read out (in the simulator). I see some output in the console: Error fetching voices: Swift.DecodingError.dataCorrupted(Swift.DecodingError.Context(codingPath: [], debugDescription: "Invalid container metadata for _UnkeyedDecodingContainer, found keyedGraphEncodingNodeID", underlyingError: nil)). Using fallback voices. I'm probably doing something wrong here, but not sure what.

Hi everyone, I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms. Problem: When the app is recording audio and an interruption occurs: I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began). On .ended, I check for .shouldResume and call audioRecorder?.record() again. The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder. Repro: Start a recording with AVAudioRecorder Simulate a system interruption (e.g., incoming call) Resume recording after the interruption Stop and inspect the output audio file Expected: Full audio (before and after interruption) should be saved. Actual: Only the audio after interruption is saved; the earlier part is missing Notes: According to the documentation, calling .record() after .pause() should resume recording into the same file. I confirmed that the file URL does not change, and I do not recreate the recorder instance. No error is thrown by the system during this process. This behavior happens consistently when the app is interrupted and resumed. Question: Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen? Thanks in advance!

Hi all, I'm using Apple Sample Code below to create application using dockkit. "Controlling a DockKit accessory using your camera app" https://developer.apple.com/documentation/dockkit/controlling-a-dockkit-accessory-using-your-camera-app?changes=_8 I used vision hand recognition and put the observation data to dockAccessory.track, but Belkin or Insta360 devices never move on iPhone 16 Pro Max with iOS 18.3. If I use other functions like face search (system tracking) in the app, those work ok. I used Belkin and Insta360 Flow 2 Pro to reproduce the problem. My friend is also saying that the custom tracking feature was working fine on the OS 18 beta, but on recent iOS 18.3 that feature does not work. If I can get the iOS 18.0 beta then we can test that feature. But I cannot revert my iOS from 18.3 to the iOS 18.0 Beta. Regards, TO

For iOS17 we've had no problem playing Apple Fairplay encrypted content with keys delivered from our key server running on FairPlay Streaming Server SDK 5.1 and subsequently FairPlay Streaming Server SDK 26. It's built and deployed using Xcode Version 26.1.1 (17B100) with no changes to the code and - as expected - the content continued to be successfully decrypted and played (so far so good). However, as soon as a device was updated to iOS26, that device would no longer play the encrypted content. Devices remaining on iOS17 continue to work normally and the debugging logs are a sanity-check that proves that. Is anyone else experiencing this issue? Here's the code (you should be able to drop it into a fresh iOS Xcode project and provide a server url, content url and certificate).