Hello, My company has an in-store app with FPS SDK 4.x (1024) keys. We've handed those keys over to a trusted third-party and we do not have them. We've been in-store for several years. The person that created the keys in our organization mistakenly stored them encrypted to our third-party's PGP keys, so we cannot decrypt them, and the third party also has no mechanism to provide us with the keys even though it is in their runtime environment. They only have secure mechanisms for us to upload keys onto their servers. We are trying to migrate to a different third-party DRM provider, and would like to obtain new keys. Unfortunately, the developer portal won't let me create new keys, saying that we have exceeded the number of keys allowed, which I assume is one. Additionally, the new DRM provider can only support SDK 4.x keys, and it appears that we can only request SDK 5.x keys on the Apple Developer portal, as the SDK 4.0 option is grayed out. Regardless, it seems that we are not able to request any keys. We've submitted a request to the support e-mail address and received an automated e-mail that the response should take a few days, but may take longer on occasion. It's now been a month. The e-mail says that the reply address is not monitored. Is there any way we can accelerate this? Thank you, Carlos

the problem is when using HLS live stream with AVPlayer on iOS/ tvOS the player chooses first highest bandwidth then slowly steps down to lowest (within 1-3min) and eventually steps up again then repeats to step down. the AVPlayer error log sends events: errorStatusCode: -12888, errorDomain: Optional("CoreMediaErrorDomain"), errorComment: Optional("The operation couldn't be completed. (CoreMediaErrorDomain error -12888 - Playlist File unchanged for longer than 1.5 * target duration we use standard segments in CMAF format, 2sec duration #EXTM3U #EXT-X-VERSION:6 #EXT-X-TARGETDURATION:2 #EXT-X-MEDIA-SEQUENCE:147065903 #EXT-X-MAP:URI="video_1_4660000_init.mp4?device_profile=cmaf_cbcs_verimatrix_cei%26seg_size=2%26cmaf=2" #EXT-X-PROGRAM-DATE-TIME:2025-04-30T12:51:07 #EXTINF:2.000, video_1_4660000_t17460174670001555.mp4?device_profile=cmaf_cbcs_verimatrix_cei%26seg_size=2%26cmaf=2 #EXTINF:2.000, video_1_4660000_t17460174690001555.mp4?device_profile=cmaf_cbcs_verimatrix_cei%26seg_size=2%26cmaf=2 #EXTINF:2.000, video_1_4660000_t17460174710001555.mp4?device_profile=cmaf_cbcs_verimatrix_cei%26seg_size=2%26cmaf=2 when using 6sec segments the player stays stable at highest bandwidth. is there a way to avoid this error? in AVPlayer or HLS configuration?

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.

AVAudioFormat has no Swift concurrency annotations but the documentation states "Instances of this class are immutable." This made me always assume it was safe to pass AVAudioFormat instances around. Is this the case? If so can it be marked as Sendable? Am I missing something?

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!

With older iOS versions, when user taps Mute/Volume button on AVPLayerViewController to unmute, the system restores the sound volume of device to the level when user muted before. On iOS 26, when user taps unmute button on screen, the volume starts from 0 (not restore). (but it still restores if user unmutes by pressing physical volume buttons). As I understand, the Volume bar/button on AVPlayerViewController is MPVolumeView, and I can not control it. So this is a feature of the system. But I got complaints that this is a bug. I did not find documents that describe this change of Mute button behavior. I need some bases to explain this situation. Thank you.

Hi everyone, I’m exploring using the iPhone 17 Pro with the Blackmagic ProDock in a custom capture app. The genlock functionality seems accessible via AVExternalSyncDevice and related APIs, which is great. I’m specifically curious about external timecode coming in from the ProDock: • Is there a public way to access the timecode feed in a custom app via AVFoundation or another Apple API? • If so, what is the recommended approach to read or apply that timecode during capture? • Are there any current limitations or entitlements required to access timecode from ProDock in a third-party app? I’m excited to start integrating synchronized capture in my app, and any guidance or sample patterns would be greatly appreciated. Thanks in advance! — [Artem]

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.

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.

Hello, I'm developing an app that displays a photo library using UICollectionView and PHCachingImageManager. I'd like to achieve a user experience similar to the native iOS Photos app, where low-quality images are shown quickly while scrolling, and higher-quality images are loaded for visible cells once scrolling stops. I'm currently using the following approach: While Scrolling: I'm using the UICollectionViewDataSourcePrefetching protocol. In the prefetchItemsAt method, I call startCachingImages with low-quality options to cache images in advance. After Scrolling Stops: In the scrollViewDidEndDecelerating method, I intend to load high-quality images for the currently visible cells. I have a few questions regarding this approach: What is the best practice for managing both low-quality and high-quality images efficiently with PHCachingImageManager? Is it correct to call startCachingImages with fastFormat options and then call it again with highQualityFormat in scrollViewDidEndDecelerating? How can I minimize the delay when a low-quality image is replaced by a high-quality one? Are there any additional strategies to help pre-load high-quality images more effectively?

I have a question regarding the behavior of AVAudioSession.sharedInstance().outputVolume. Observed behavior: When the app is in the foreground, I read audioSession.outputVolume (for example, 0.1). The app is then moved to the background. While the app is in the background, the user changes the system volume using the hardware buttons (for example, to 0.5). When the app returns to the foreground, audioSession.outputVolume still reports the previous value (0.1). From my testing, outputVolume only seems to update when the system volume is changed while the app is in the foreground. Volume changes made while the app is in the background are not reflected when the app returns to the foreground. Questions: According to Apple’s documentation for AVAudioSession.outputVolume: “The systemwide output volume set by the user.” https://developer.apple.com/documentation/avfaudio/avaudiosession/outputvolume However, based on our testing on iOS 18.6.2 and iOS 18.1, the observed behavior seems to differ from this description. Questions: The documentation states that outputVolume represents the system-wide volume set by the user. In our testing, the value does not reflect volume changes made while the app is in the background and only updates when the app is in the foreground.Is this the expected behavior of AVAudioSession.outputVolume? Is there any other recommended way in Swift to retrieve the current system volume that reflects user changes made both while the app is in the foreground and while it is in the background? Any clarification on the intended behavior or recommended handling would be greatly appreciated.

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?

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!

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.

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

I am developing an iOS application that supports screen mirroring to Google TV (or Chromecast with Google TV). My goal is to mirror the iPhone/iPad screen in real time to a Google TV device. What I Have Tried So Far I have explored multiple approaches but haven't found a direct way to achieve low-latency screen mirroring. Here are some of my findings: Google Cast SDK: Google Cast SDK is primarily designed for casting media (videos, images, audio) rather than real-time mirroring. It supports custom receiver applications, but there are no direct APIs for full screen mirroring. Casting a recorded video is possible, but it introduces latency and is not real-time. ReplayKit for Screen Capture: RPScreenRecorder.shared().startCapture(handler: ...) allows capturing the iPhone screen as a video stream. However, sending this stream to Google TV in real time is a challenge. I could potentially encode the video as HLS and stream it, but the delay is significant. RTSP/UDP Streaming: Some third-party libraries support RTSP/UDP streaming for real-time screen sharing. Google TV does not natively support RTSP, making this approach difficult. My Questions: Is it possible to achieve real-time screen mirroring on Google TV using Google Cast SDK? Does Google TV support WebRTC or any low-latency streaming protocol that can be used from iOS? Are there any alternative approaches to mirror an iOS screen to Google TV with minimal latency? I would appreciate any guidance, code examples, or references to relevant documentation.

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).

Hello, I'm investigating an issue with LL-HLS playback using AVPlayer, specifically during DVR Live seeking (seeking to a past time). I noticed that in certain seeking scenarios, AVPlayer sends a Blocking Playlist Reload request that includes the _HLS_msn parameter but is missing the _HLS_part parameter. While I understand this is compliant with the HLS spec, I would like to know the specific criteria AVPlayer uses to decide when to drop the _HLS_part parameter. Does AVPlayer intentionally omit the part info when it determines that loading a specific partial segment is unnecessary during a seek operation? Clarification on this behavior would help us greatly in debugging our stream delivery. Thanks in advance.

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!