Hello Apple Developer Community, I am trying to play an HLS stream using the React Native Video player (underneath it's using AvPlayer). I am able to play the stream smoothly, but in some cases the player can not play the stream properly. Behaviour: react-native-video: I am getting the below error. Error details from react-native-video player: Error Code: -12971 Domain: CoreMediaErrorDomain Localised Description: The operation couldn’t be completed. (CoreMediaErrorDomain error -12971.) Target: 2457 The error does not provide a specific failure reason or recovery suggestion, which makes troubleshooting challenging. AvPlayer on native iOS project: Video playback stopped after playing a few seconds. AVPlayer configuration: player.currentItem?.preferredForwardBufferDuration = 1 player.automaticallyWaitsToMinimizeStalling = true N.B.: The same buffer duration is working perfectly for others. Stream properties: video resolution: 1280 x 720 I have attached an overview report generated from MediaStreamValidator. I would appreciate any insights or suggestions on how to address this error. Has anyone in the community experienced a similar issue or have any advice on potential solutions? Thank you for your help!

我在使用 musicKit SDK for Android 1.1.2 时，发现 MediaContainerType 只定义了三种类型： 无 = 0; 专辑 = 1; 播放列表 = 2; 未定义 RADIO_STATION 类型。 但是，com.apple.android.music.playback.model 的文档指出支持 RADIO_STATION 类型。 此问题在我传入 stations ID 后会导致错误： MediaSessionManager com.apple.android.music.sdk.testapp D onPlaybackError（） Quincy java.io.IOException 请问如何解决这个问题？

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 am developing an app that plays HLS audio. When using AVPlayerItem with AVURLAsset, can AVAssetResourceLoaderDelegate correctly handle HLS segments? My goal is to use AVAssetResourceLoaderDelegate to add authentication HTTP headers when accessing HLS .m3u8 and .ts files. I can successfully download the files, but playback fails with errors. Specifically, I am observing the following cases: A. AVAssetResourceLoaderDelegate is canceled, and CoreMediaErrorDomain -12881 occurs In NSURLConnectionDataDelegate’s didReceiveResponse method, set contentInformationRequest In didReceiveData, call dataRequest respondWithData resourceLoader didCancelLoadingRequest is called CoreMediaErrorDomain -12881 occurs B. CoreMediaErrorDomain -12881 occurs In NSURLConnectionDataDelegate’s didReceiveResponse method, set contentInformationRequest In connection didReceiveData, buffer all received data until the end In connectionDidFinishLoading, pass the buffered data to respondWithData Call loadingRequest finishLoading CoreMediaErrorDomain -12881 occurs In both cases, dataRequest.requestsAllDataToEndOfResource is YES. For this use case, I am not using AVURLAssetHTTPHeaderFieldsKey because I need to apply the most up-to-date authentication data at the moment each file is accessed. I would appreciate any advice or suggestions you might have. Thank you in advance!

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, I'm trying to subscribe to AVPlayerItem status updates using Combine and it's bridge to Swift Concurrency – .values. This is my sample code. struct ContentView: View { @State var player: AVPlayer? @State var loaded = false var body: some View { VStack { if let player { Text("loading status: \(loaded)") Spacer() VideoPlayer(player: player) Button("Load") { Task { let item = AVPlayerItem( url: URL(string: "https://sample-videos.com/video321/mp4/360/big_buck_bunny_360p_5mb.mp4")! ) player.replaceCurrentItem(with: item) let publisher = player.publisher(for: \.status) for await status in publisher.values { print(status.rawValue) if status == .readyToPlay { loaded = true break } } print("we are out") } } } else { Text("No video selected") } } .task { player = AVPlayer() } } } After I click on the "load" button it prints out 0 (as the initial status of .unknown) and nothing after – even when the video is fully loaded. At the same time this works as expected (loading status is set to true): struct ContentView: View { @State var player: AVPlayer? @State var loaded = false @State var cancellable: AnyCancellable? var body: some View { VStack { if let player { Text("loading status: \(loaded)") Spacer() VideoPlayer(player: player) Button("Load") { Task { let item = AVPlayerItem( url: URL(string: "https://sample-videos.com/video321/mp4/360/big_buck_bunny_360p_5mb.mp4")! ) player.replaceCurrentItem(with: item) let stream = AsyncStream { continuation in cancellable = item.publisher(for: \.status) .sink { if $0 == .readyToPlay { continuation.yield($0) continuation.finish() } } } for await _ in stream { loaded = true cancellable?.cancel() cancellable = nil break } } } } else { Text("No video selected") } } .task { player = AVPlayer() } } } Is this a bug or something?

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.

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.

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.

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) } }

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.

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.

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!

Hi everyone, Our team is encountering a reproducible crash when using VTLowLatencyFrameInterpolation on iOS 26.3 while processing a live LL-HLS input stream. 🤖 Environment Device: iPhone 16 OS: iOS 26.3 Xcode: Xcode 26.3 Framework: VideoToolbox 💥 Crash Details The application crashes with the following fatal error: Fatal error: Swift/ContiguousArrayBuffer.swift:184: Array index out of range The stack trace highlights the following: VTLowLatencyFrameInterpolationImplementation processWithParameters:frameOutputHandler: Called from VTFrameProcessor.process(parameters:) Here is the simplified implementation block where the crash occurs. (Note: PrismSampleBuffer and PrismLLFIError are our internal custom wrapper types). // Create `VTFrameProcessorFrame` for the source (previous) frame. let sourcePTS = sourceSampleBuffer.presentationTimeStamp var sourceFrame: VTFrameProcessorFrame? if let pixelBuffer = sourceSampleBuffer.imageBuffer { sourceFrame = VTFrameProcessorFrame(buffer: pixelBuffer, presentationTimeStamp: sourcePTS) } // Validate the source VTFrameProcessorFrame. guard let sourceFrame else { throw PrismLLFIError.missingImageBuffer } // Create `VTFrameProcessorFrame` for the next frame. let nextPTS = nextSampleBuffer.presentationTimeStamp var nextFrame: VTFrameProcessorFrame? if let pixelBuffer = nextSampleBuffer.imageBuffer { nextFrame = VTFrameProcessorFrame(buffer: pixelBuffer, presentationTimeStamp: nextPTS) } // Validate the next VTFrameProcessorFrame. guard let nextFrame else { throw PrismLLFIError.missingImageBuffer } // Calculate interpolation intervals and allocate destination frame buffers. let intervals = interpolationIntervals() let destinationFrames = try framesBetween(firstPTS: sourcePTS, lastPTS: nextPTS, interpolationIntervals: intervals) let interpolationPhase: [Float] = intervals.map { Float($0) } // Create VTLowLatencyFrameInterpolationParameters. // This sets up the configuration required for temporal frame interpolation between the previous and current source frames. guard let parameters = VTLowLatencyFrameInterpolationParameters( sourceFrame: nextFrame, previousFrame: sourceFrame, interpolationPhase: interpolationPhase, destinationFrames: destinationFrames ) else { throw PrismLLFIError.failedToCreateParameters } try await send(sourceSampleBuffer) // Process the frames. // Using progressive callback here to get the next processed frame as soon as it's ready, // preventing the system from waiting for the entire batch to finish. for try await readOnlyFrame in self.frameProcessor.process(parameters: parameters) { // Create an interpolated sample buffer based on the output frame. let newSampleBuffer: PrismSampleBuffer = try readOnlyFrame.frame.withUnsafeBuffer { pixelBuffer in try PrismLowLatencyFrameInterpolation.createSampleBuffer(from: pixelBuffer, readOnlyFrame.timeStamp) } // Pass the newly generated frame to the output stream. try await send(newSampleBuffer) } 🙋 Questions Are there any known limitations or bugs regarding VTLowLatencyFrameInterpolation when handling live 60fps streams? Are there any undocumented constraints we should be aware of regarding source/previous frame timing, pixel buffer attributes, or how destinationFrames and interpolationPhase arrays must be allocated? Is a "warm-up" sequence recommended after startSession() before making the first process(parameters:) call?

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

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.

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'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.

I'm using an AVAudioConverter object to decode an OPUS stream for VoIP. The decoding itself works well, however, whenever the stream stalls (no more audio packet is available to decode because of network instability) this can be heard in crackling / abrupt stop in decoded audio. OPUS can mitigate this by indicating packet loss by passing a null pointer in the C-library to int opus_decode_float (OpusDecoder * st, const unsigned char * data, opus_int32 len, float * pcm, int frame_size, int decode_fec), see https://opus-codec.org/docs/opus_api-1.2/group__opus__decoder.html#ga9c554b8c0214e24733a299fe53bb3bd2. However, with AVAudioConverter using Swift I'm constructing an AVAudioCompressedBuffer like so:         let compressedBuffer = AVAudioCompressedBuffer(             format: VoiceEncoder.Constants.networkFormat,             packetCapacity: 1,             maximumPacketSize: data.count         )         compressedBuffer.byteLength = UInt32(data.count)         compressedBuffer.packetCount = 1   compressedBuffer.packetDescriptions! .pointee.mDataByteSize = UInt32(data.count)         data.copyBytes(             to: compressedBuffer.data .assumingMemoryBound(to: UInt8.self),             count: data.count         ) where data: Data contains the raw OPUS frame to be decoded. How can I specify data loss in this context and cause the AVAudioConverter to output PCM data whenever no more input data is available? More context: I'm specifying the audio format like this:         static let frameSize: UInt32 = 960         static let sampleRate: Float64 = 48000.0         static var networkFormatStreamDescription = AudioStreamBasicDescription(             mSampleRate: sampleRate,             mFormatID: kAudioFormatOpus,             mFormatFlags: 0,             mBytesPerPacket: 0,             mFramesPerPacket: frameSize,             mBytesPerFrame: 0,             mChannelsPerFrame: 1,             mBitsPerChannel: 0,             mReserved: 0         )         static let networkFormat = AVAudioFormat( streamDescription: &networkFormatStreamDescription )! I've tried 1) setting byteLength and packetCount to zero and 2) returning nil but setting .haveData in the AVAudioConverterInputBlock I'm using with no success.