Hello,We have been working on FPS for a while and managed to play an encrypted asset with it, using version 2 of FairPlay Streaming Server SDK 2.03.Now we are working on "Download and Play Offline" functionality. For this I have downloaded the FairPlay Streaming Server SDK 3.0 and in SDK Folder there is "OfflineHLSGuide_withFPS.pdf" document beside "FairPlayStreaming_PG.pdf" where they tell about how it works and what need to be done.In that document ("OfflineHLSGuide_withFPS.pdf") there is new version of Content Key Duration TLLV which is slightly different than the structure shown in "FairPlayStreaming_PG.pdf"For Content Key Duration TLLV (0x47acf6a418cd091a) in "FairPlayStreaming_PG" there is "Lease Duration" between bytes 16-19.But in the document about OfflineHLSGuide, it says "Reserved" for bytes between 16-19.Where has LeaseDuration gone for offline? ORShould I use one ContentKeyDuration TLLV for non-persist and other version of ContentKeyDuration TLLV for (persist)offline?Thanks

The same H265 encrypted Fairplay content can be played in all Apple devices except A1625. The clear H265 content is played in A1625. The question is: will this model (A1625) support H265 Fairplay encrypted content? A ticket was created here: https://discussions.apple.com/thread/255658006?sortBy=best

I am playing FairPlay + Multi-Key content (fMP4) in Safari browser. I want to implement the implementation to distinguish between SD and HD video quality, and play it in HD if HDCP is supported, and in SD if HDCP is not supported. I have already confirmed that HDCP support is the default, and that a black screen is output in non-HDCP environments. What I want is to improve the user experience by appropriately switching to SD/HD depending on HDCP support when playing DRM content. Question: Is there an API or function that can detect HDCP support in Safari through JavaScript or other methods? Or is there a way to indirectly guess it?

I’m building a professional camera app where users can customize the video recording format and color grading. In the func captureOutput(_ output: AVCaptureOutput, didOutput sampleBuffer: CMSampleBuffer, from connection: AVCaptureConnection) method, I handle video frames and use Metal for real-time color grading. This works well when device.activeColorSpace is sRGB or P3, and the results are great. However, when the color space is HLG_BT2020 or appleLog, the MTKTextureLoader.newTexture(cgImage: cgImage, options: options) method throws an error. After researching, I found that the video frame in these color spaces has a bit-per-channel (bpc) greater than 8 after being converted to CGImage, causing the texture creation to fail. I tried converting the CGImage to a lower bpc to successfully create the texture, but the final output image is garbled and not as expected. Is there a solution to this issue?

We are encountering an issue where AVPlayer throws the error: Error Domain=AVFoundationErrorDomain Code=-11819 "Cannot Complete Action" > Underlying Error Domain[(null)]:Code[0]:Desc[(null)] This error seems to occur intermittently during video playback, especially after extended usage or when switching between different streams. We observe Error 11819 (AVFoundationErrorDomain) in Conviva platform that some of our users experience it but we couldn't reproduce it so far and we’re need support to determine the root cause and/or best practices to prevent it. Some questions we have: What typically triggers this error? Could it be related to memory/resource constraints, network instability, or backgrounding? Are there any recommended ways to handle or recover from this error gracefully? Any insights or guidance would be greatly appreciated. Thanks!

Hello! I have been following the UsingAVFoundationToPlayAndPersistHTTPLiveStreams sample code in order to test persisting streams to disk. In addition to support for m3u8, I have noticed in testing that this also seems to work for MP3 Audio, simply by changing the plist entries to point to remote URLs with audio/mpeg content. Is this expected, or are there caveats that I should be aware of? Thanks you!

The app registers a periodic time observer to the AVPlayer when the playback starts and it works fine. When switching to AirPlay during playback, the periodic time observation continues working as expected. However, when switching back to local playback, the periodic time observer does not fire anymore until a seek is performed. The app removes the periodic time observer only when the playback stops. I can see that when switching back to local playback, the timeControlStatus successively changes to .waitingToPlayAtSpecifiedRate (reason: .evaluatingBufferingRate) then to .waitingToPlayAtSpecifiedRate (reason: .toMinimizeStalls) and finally to .playing But the time observation does not work anymore. Also, the issue is systematic with Live and VOD streams providing a program date (with HLS property #EXT-X-PROGRAM-DATE-TIME), with or without any DRM, and is never reproduced with other VOD streams.

I did watch WWDC 2019 Session 716 and understand that an active audio session is key to unlocking low‑level networking on watchOS. I’m configuring my audio session and engine as follows: private func configureAudioSession(completion: @escaping (Bool) -> Void) { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: []) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) // Retrieve sample rate and configure the audio format. let sampleRate = audioSession.sampleRate print("Active hardware sample rate: \(sampleRate)") audioFormat = AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1) // Configure the audio engine. audioInputNode = audioEngine.inputNode audioEngine.attach(audioPlayerNode) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: audioFormat) try audioEngine.start() completion(true) } catch { print("Error configuring audio session: \(error.localizedDescription)") completion(false) } } private func setupUDPConnection() { let parameters = NWParameters.udp parameters.includePeerToPeer = true connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupTCPConnection() { let parameters = NWParameters.tcp connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupWebSocketConnection() { guard let url = URL(string: "ws://***.***.xxxxx.***:0000") else { print("Invalid WebSocket URL") return } let session = URLSession(configuration: .default) webSocketTask = session.webSocketTask(with: url) webSocketTask?.resume() print("WebSocket connection initiated") sendAudioToServer() receiveDataFromServer() sendWebSocketPing(after: 0.6) } private func setupNWConnectionHandlers() { connection?.stateUpdateHandler = { [weak self] state in DispatchQueue.main.async { switch state { case .ready: print("Connected (NWConnection)") self?.isConnected = true self?.failToConnect = false self?.receiveDataFromServer() self?.sendAudioToServer() case .waiting(let error), .failed(let error): print("Connection error: \(error.localizedDescription)") DispatchQueue.main.asyncAfter(deadline: .now() + 2) { self?.setupNetwork() } case .cancelled: print("NWConnection cancelled") self?.isConnected = false default: break } } } connection?.start(queue: .main) } Duplex in this context refers to two-way audio transmission simultaneously recording and sending audio while also receiving and playing back incoming audio, similar to a VoIP/SIP call. The setup works fine on the simulator, which suggests that the core logic is correct. However, since the simulator doesn’t fully replicate WatchOS hardware behavior especially for audio sessions and networking issues might arise when running on a real device. The problem likely lies in either the Watch’s actual hardware limitations, permission constraints, or specific audio session configurations. I am reaching out to seek further assistance regarding the challenges I've been experiencing with establishing a UDP, TCP & web socket connection on watchOS using NWConnection for duplex audio streaming. Despite implementing the recommendations provided earlier, I am still encountering difficulties From what I can see, your implementation is focused on streaming audio playback with the server. In my case, I'm looking for a slightly different approach: I want to capture audio and send buffers of a specific size to the server while playing audio simultaneously, essentially achieving full duplex streaming similar to a VOIP call. Additionally, I’d like to ensure that if no external audio route is connected, the Apple Watch speaker is used by default. Any thoughts or insights on adapting this setup for those requirements would be very welcome.

i need to make app for iOS to get code for XCode

who can help me to make app for iOS with XCode

Hello, I'm trying to create a webbrowser but currently when signed into apple music webplayer I get the following message when I attempt to play on any versions of my webbrowser: Not available on the web You can listen to this in the Apple Music app. Is there a way to setup DRM (assuming this is the issue) with apple to allow my webbrowser to play this content? I believe Apple TV is also affected. Thank you ahead of time.

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

Hello, We're seeing an intermittent issue when playing back FairPlay-protected HLS downloads while the device is offline. Assets are downloaded using AVAggregateAssetDownloadTask with FairPlay protection. After download, asset.assetCache.isPlayableOffline == true. On first playback attempt (offline), ~8% of downloads fail. Retrying playback always works. We recreate the asset and player on each attempt. During the playback setup, we try to load variants via: try await asset.load(.variants) This call sometimes fails with: Error Domain=NSURLErrorDomain Code=-1009 “The Internet connection appears to be offline.” UserInfo={NSUnderlyingError=0x105654a00 {Error Domain=NSURLErrorDomain Code=-1009 “The Internet connection appears to be offline.” UserInfo={NSDescription=The Internet connection appears to be offline.}}, NSErrorFailingURLStringKey=file:///private/var/mobile/Containers/Data/Application/2DDF9D7C-9197-46BE-8690-C23EE75C9E90/Library/com.apple.UserManagedAssets.XVvqfh/Baggage_9DD4E2D3F9C0E68F.movpkg/, NSErrorFailingURLKey=file:///private/var/mobile/Containers/Data/Application/2DDF9D7C-9197-46BE-8690-C23EE75C9E90/Library/com.apple.UserManagedAssets.XVvqfh/Baggage_9DD4E2D3F9C0E68F.movpkg/, NSURL=file:///private/var/mobile/Containers/Data/Application/2DDF9D7C-9197-46BE-8690-C23EE75C9E90/Library/com.apple.UserManagedAssets.XVvqfh/Baggage_9DD4E2D3F9C0E68F.movpkg/, AVErrorFailedDependenciesKey=( “assetProperty_HLSAlternates” ), NSLocalizedDescription=The Internet connection appears to be offline.} This variant load is used to determine available audio tracks, check for Dolby support, and apply user language preferences. After this step, the AVPlayerItem also fails via Combine’s publisher for .status. However, retrying the entire process immediately after (same offline conditions, same asset path, new AVURLAsset) results in successful playback. Assets are represented using the following class: public class DownloadedAsset: AVURLAsset { public let id: String public let localFileUrl: URL public let fairplayLicenseUrlString: String? public let drmToken: String? var isProtected: Bool { return fairplayLicenseUrlString != nil } public init(id: String, localFileUrl: URL, fairplayLicenseUrlString: String?, drmToken: String?) { self.id = id self.localFileUrl = localFileUrl self.fairplayLicenseUrlString = fairplayLicenseUrlString self.drmToken = drmToken super.init(url: localFileUrl, options: nil) } } We use user-selected quality levels to control bitrate and multichannel (e.g. Dolby 5.1) downloads: let downloadQuality = UserDefaults.standard.downloadVideoQuality let bitrate: Int let shouldDownloadMultichannelTracks: Bool switch downloadQuality { case .dataSaver: shouldDownloadMultichannelTracks = false bitrate = 596564 case .standard: shouldDownloadMultichannelTracks = false bitrate = 1503844 case .best: shouldDownloadMultichannelTracks = true bitrate = 7038970 } var selections = multichannelIdentifiedMediaSelections if !shouldDownloadMultichannelTracks { selections = selections.filter { !$0.isMultichannel } } let task = session.aggregateAssetDownloadTask( with: asset, mediaSelections: selections.map { $0.mediaSelection }, assetTitle: title, assetArtworkData: nil, options: [AVAssetDownloadTaskMinimumRequiredMediaBitrateKey: bitrate] ) Seen on devices running iOS 16, iOS 17, and iOS 18. What could cause the initial failure of an otherwise valid, offline-ready FairPlay HLS asset? Could .load(.variants) internally trigger a failed network resolution, even when offline? Is there an internal caching or initialization behavior in AVFoundation that might explain why the second attempt works? Any guidance would be appreciated.

TL;DR How to solve possible racing issue of EXT-X-SESSION-KEY request and encrypted media segment request? I'm having trouble using custom AVAssetResourceLoaderDelegate with video manifest containing VideoProtectionKey(VPK). My master manifest contains rendition manifest url and VPK url. When not using custom resource delegate, everything works fine. My custom resource delegate is implemented in way where it first append prefix to scheme of the master manifest url before creating the asset. And during handling master manifest, it puts back original scheme, make the request, modify the scheme for rendition manifest url in the response content by appending the same prefix again, so that rendition manifest request also goes into custom resource loader delegate. Same goes for VPK request. The AES-128 key is stored in memory within custom resource loader delegate object. So far so good. The VPK is requested before segment request. But the problem comes where the media segment requests happen. The media segment request url from rendition manifest goes into custom resource loader as well and those are encrypted. I can see segment request finish first then the related VPK requests kick in after a few seconds. The previous VPK value is cached in memory so it is not network causing the delay but some mechanism that I'm not aware of causing this. So could anyone tell me what would be the proper way of handling this situation? The native library is handling it well so I just want to know how. Thanks in advance!

We are seeing logs were on iOS devices we see some keyframes request. but on safari browser don’t see any request like this. could you please explain what is it. /d8ceb9244ff889b42b82eb807327531-c27dbcb10e0bbf3cde6c-1/d8ceb9244ff88e9b42b82eb807327531-c27dbcb10e0bbf3cde6c-1/keyframes/hls/.

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!

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!

While validating a Dolby Vision Profile 5 playlist in CMAF format (with segments in MP4), the Media Stream Validator reported the following error in the MUXT-FIX-ISSUES list: However, the playlist correctly specifies Dolby Vision Profile 5 in both the EXT-X-STREAM-INF and EXT-X-I-STREAM-INF tags. Playlist: #EXTM3U #EXT-X-VERSION:8 #EXT-X-INDEPENDENT-SEGMENTS #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio-ec3",LANGUAGE="und",NAME="Undetermined",AUTOSELECT=YES,CHANNELS="6",URI="var14711339/aud1257/playlist.m3u8?device_profile=hls&seg_size=6&cmaf=1" #EXT-X-STREAM-INF:BANDWIDTH=14680000,AVERAGE-BANDWIDTH=14676380,VIDEO-RANGE=PQ,CODECS="ec-3,dvh1.05.06",RESOLUTION=3840x2160,AUDIO="audio-ec3" var14711339/vid/playlist.m3u8?device_profile=hls&seg_size=6&cmaf=1 #EXT-X-I-FRAME-STREAM-INF:BANDWIDTH=1419881,URI="trk14711339/playlist.m3u8?device_profile=hls&cmaf=1",VIDEO-RANGE=PQ,CODECS="dvh1.05.06",RESOLUTION=3840x2160 Could you please review this and clarify: Why is the Media Stream Validator reporting this error, even though the playlist correctly includes Dolby Vision Profile 5 parameters in CMAF format? Why is this error not reported when using a playlist with TS segments instead of CMAF (MP4)?

I'm capturing video stream from GoPro camera (via UDP MPEG-TS packets) but unable to play in iOS app. can someone provide a source code to decode MPEG-TS data to CMSampleBuffer.

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.