I'm writing some camera functionality that uses AVCaptureVideoDataOutput. I've set it up so that it calls my AVCaptureVideoDataOutputSampleBufferDelegate on a background thread, by making my own dispatch_queue and configuring the AVCaptureVideoDataOutput. My question is then, if I configure my AVCaptureSession differently, or even stop it altogether, is this guaranteed to flush all pending jobs on my background thread? I have a more practical example below, showing how I am accessing something from the foreground thread from the background thread, but I wonder when/how it's safe to clean up that resource. I have setup similar to the following: // Foreground thread logic dispatch_queue_t queue = dispatch_queue_create("avf_camera_queue", nullptr); AVCaptureSession *captureSession = [[AVCaptureSession alloc] init]; setupInputDevice(captureSession); // Connects the AVCaptureDevice... // Store some arbitrary data to be attached to the frame, stored on the foreground thread FrameMetaData frameMetaData = ...; MySampleBufferDelegate *sampleBufferDelegate = [MySampleBufferDelegate alloc]; // Capture frameMetaData by reference in lambda [sampleBufferDelegate setFrameMetaDataGetter: [&frameMetaData]() { return &frameMetaData; }]; AVCaptureVideoDataOutput *captureVideoDataOutput = [[AVCaptureVideoDataOutput alloc] init]; [captureVideoDataOutput setSampleBufferDelegate:sampleBufferDelegate queue:queue]; [captureSession addOutput:captureVideoDataOutput]; [captureSession startRunning]; [captureSession stopRunning]; // Is it now safe to destroy frameMetaData, or do we need manual barrier? And then in MySampleBufferDelegate: - (void)captureOutput:(AVCaptureOutput *)captureOutput didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection *)connection { // Invokes the callback set above FrameMetaData *frameMetaData = frameMetaDataGetter(); emitSampleBuffer(sampleBuffer, frameMetaData); }

My app is a camera app that supports Picture-in-Picture (PiP) mode. Normally, when the device rotates, I get the device orientation from iOS and use it to rotate the camera feed so that the preview stays correctly aligned. However, when the app enters PiP mode, it is considered to be in the background, and I can no longer receive orientation updates from the system. As a result, I can’t apply rotation corrections to the camera video in PiP mode. Is there any way to retrieve device orientation while the app is in the background (specifically during PiP mode)? Any guidance would be greatly appreciated. Thank you!

I am trying to get MIDI output from the AU Host demo app using the recent MIDI processor example. The processor works correctly in Logic Pro, but I cannot send MIDI from the AUv3 extension in standalone mode using the default host app to another program (e.g., Ableton). The MIDI manager, which is part of the standalone host app, works fine, and I can send MIDI using it directly—Ableton receives it without issues. I have already set the midiOutputNames in the extension, and the midiOutBlock is mapped. However, the MIDI data from the AUv3 extension does not reach Ableton in standalone mode. I suspect the issue is that midiOutBlock might never be called in the plugin, or perhaps an input to the plugin is missing, which prevents it from sending MIDI. I am currently using the default routing. I have modified the MIDI manager such that it works well as described above. Here is a part of my code for SimplePlayEngine.swift and my MIDIManager.swift for reference: @MainActor @Observable public class SimplePlayEngine { private let midiOutBlock: AUMIDIOutputEventBlock = { sampleTime, cable, length, data in return noErr } var scheduleMIDIEventListBlock: AUMIDIEventListBlock? = nil public init() { engine.attach(player) engine.prepare() setupMIDI() } private func setupMIDI() { if !MIDIManager.shared.setupPort(midiProtocol: MIDIProtocolID._2_0, receiveBlock: { [weak self] eventList, _ in if let scheduleMIDIEventListBlock = self?.scheduleMIDIEventListBlock { _ = scheduleMIDIEventListBlock(AUEventSampleTimeImmediate, 0, eventList) } }) { fatalError("Failed to setup Core MIDI") } } func initComponent(type: String, subType: String, manufacturer: String) async -> ViewController? { reset() guard let component = AVAudioUnit.findComponent(type: type, subType: subType, manufacturer: manufacturer) else { fatalError("Failed to find component with type: \(type), subtype: \(subType), manufacturer: \(manufacturer))" ) } do { let audioUnit = try await AVAudioUnit.instantiate( with: component.audioComponentDescription, options: AudioComponentInstantiationOptions.loadOutOfProcess) self.avAudioUnit = audioUnit self.connect(avAudioUnit: audioUnit) return await audioUnit.loadAudioUnitViewController() } catch { return nil } } private func startPlayingInternal() { guard let avAudioUnit = self.avAudioUnit else { return } setSessionActive(true) if avAudioUnit.wantsAudioInput { scheduleEffectLoop() } let hardwareFormat = engine.outputNode.outputFormat(forBus: 0) engine.connect(engine.mainMixerNode, to: engine.outputNode, format: hardwareFormat) do { try engine.start() } catch { isPlaying = false fatalError("Could not start engine. error: \(error).") } if avAudioUnit.wantsAudioInput { player.play() } isPlaying = true } private func resetAudioLoop() { guard let avAudioUnit = self.avAudioUnit else { return } if avAudioUnit.wantsAudioInput { guard let format = file?.processingFormat else { fatalError("No AVAudioFile defined.") } engine.connect(player, to: engine.mainMixerNode, format: format) } } public func connect(avAudioUnit: AVAudioUnit?, completion: @escaping (() -> Void) = {}) { guard let avAudioUnit = self.avAudioUnit else { return } engine.disconnectNodeInput(engine.mainMixerNode) resetAudioLoop() engine.detach(avAudioUnit) func rewiringComplete() { scheduleMIDIEventListBlock = auAudioUnit.scheduleMIDIEventListBlock if isPlaying { player.play() } completion() } let hardwareFormat = engine.outputNode.outputFormat(forBus: 0) engine.connect(engine.mainMixerNode, to: engine.outputNode, format: hardwareFormat) if isPlaying { player.pause() } let auAudioUnit = avAudioUnit.auAudioUnit if !auAudioUnit.midiOutputNames.isEmpty { auAudioUnit.midiOutputEventBlock = midiOutBlock } engine.attach(avAudioUnit) if avAudioUnit.wantsAudioInput { engine.disconnectNodeInput(engine.mainMixerNode) if let format = file?.processingFormat { engine.connect(player, to: avAudioUnit, format: format) engine.connect(avAudioUnit, to: engine.mainMixerNode, format: format) } } else { let stereoFormat = AVAudioFormat(standardFormatWithSampleRate: hardwareFormat.sampleRate, channels: 2) engine.connect(avAudioUnit, to: engine.mainMixerNode, format: stereoFormat) } rewiringComplete() } } and my MIDI Manager @MainActor class MIDIManager: Identifiable, ObservableObject { func setupPort(midiProtocol: MIDIProtocolID, receiveBlock: @escaping @Sendable MIDIReceiveBlock) -> Bool { guard setupClient() else { return false } if MIDIInputPortCreateWithProtocol(client, portName, midiProtocol, &port, receiveBlock) != noErr { return false } for source in self.sources { if MIDIPortConnectSource(port, source, nil) != noErr { print("Failed to connect to source \(source)") return false } } setupVirtualMIDIOutput() return true } private func setupVirtualMIDIOutput() { let virtualStatus = MIDISourceCreate(client, virtualSourceName, &virtualSource) if virtualStatus != noErr { print("❌ Failed to create virtual MIDI source: \(virtualStatus)") } else { print("✅ Created virtual MIDI source: \(virtualSourceName)") } } func sendMIDIData(_ data: [UInt8]) { print("hey") var packetList = MIDIPacketList() withUnsafeMutablePointer(to: &packetList) { ptr in let pkt = MIDIPacketListInit(ptr) _ = MIDIPacketListAdd(ptr, 1024, pkt, 0, data.count, data) if virtualSource != 0 { let status = MIDIReceived(virtualSource, ptr) if status != noErr { print("❌ Failed to send MIDI data: \(status)") } else { print("✅ Sent MIDI data: \(data)") } } } } }

Is there any way we can detect the status of the Show When Muted and Show on Skip Back device settings in code ?

A bit of a novice to app development here but I have a paid developer account, I have registered the identifier for MusicKit on the developer website (using the bundle identifier I've selected in Xcode) but the option to add MusicKit as a capability is not available in Xcode? I've manually updated the certificates, closed the app and reopened it, started a new project and tried with a different demo project? Apologies if I am missing something obvious but could someone help me get this capability added?

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.

Howdy. I'm trying to access media from a users song library and receive: <ICUserIdentityStoreACAccountBackend: 0x148f8af30> Failed to initialize active account, error=Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store} I'm told I need to add a Media Library Access Capability. Nothing like this shows up in Xcode under Signing & Capabilities > +Capabilities. Also I can't find anything like this in my account in dev.apple.com. How do I enable myself and a test user using another iPhone device to access my music and their music respectively? Thanks!

In our logging tools (Firebase) I see a lot of errors reported when users are playing content and the app transitions to the background. A AVPlayerItemFailedToPlayToEndTime notification is fired with an error containing error codes like -1102 and 1852797029 which seem to correspond to NSURLErrorNoPermissionsToReadFile and kCMIOHardwareIllegalOperationError respectively. To me, it looks like these might have something to do with caching logic. The items being played are HLS streams and we make use of AVAssetDownloadTask to make any streamed content offline available. Our setup is similar to the sample provided here: https://developer.apple.com/documentation/avfoundation/using-avfoundation-to-play-and-persist-http-live-streams. Whenever an item is selected for playback the app will check if a cached version is available and if so gets the url to the stored file like the "localAssetForStream()" method in the example, or get the asset from a currently running AVAssetDownloadTask for the item, or else, starts a new AVAssetDownloadTask and returns a AVAsset from that task to play. This seems to work fine, and I can't reproduce the issues our users and our logging tools are reporting. Is there some case I am missing where AVAssetDownloadTask and associated AVAssets might become unreadable when the app transitions to the background? Or do these errors indicate a different problem entirely?

In iOS 26 When we download any DRM content first time it is downloading again when we edit audios and Video Quality and start downloading it is freezing complete app. Neither it is crashing not giving any error.

In iOS 18, CarPlay shows an error: “There was a problem loading this content” after playback starts. Audio works fine, but the Now Playing screen doesn’t load. I’m using MPPlayableContentManager. This worked fine in iOS 17. Anyone else seeing this error in iOS 18?

Since iOS 18, the system setting “Allow Audio Playback” (enabled by default) allows third-party app audio to continue playing while the user is recording video with the Camera app. This has created a problem for the app I’m developing. ➡️ The problem: My app plays continuous audio in both foreground and background states. If the user starts recording video using the iOS Camera app, the app’s audio — still playing in the background — gets captured in the video — obviously an unintended behavior. Yes, the user could stop the app manually before starting the video recording, but that can’t be guaranteed. As a developer, I need a way to stop the app’s audio before the video recording begins. So far, I haven’t found a reliable way to detect when video recording starts if ‘Allow Audio Playback’ is ON. ➡️ What I’ve tried: — AVAudioSession.interruptionNotification → doesn’t fire — devicesChangedEventStream → not triggered I don’t want to request mic permission (app doesn’t use mic). also, disabling the app from playing audio in the background isn’t an option as it is a crucial part of the user experience ➡️ What I need: A reliable, supported way to detect when the Camera app begins video recording, without requiring mic access — so I can stop audio and avoid unintentional overlap with the user’s recordings. Any official guidance, workarounds, or AVFoundation techniques would be greatly appreciated. Thanks.

I'm reaching out regarding a recurring issue I'm experiencing with MusicKit developer tokens. I'm using a valid .p8 private key to sign JWTs for Apple MusicKit integration. Each token I generate includes the appropriate claims (iss, iat, exp) and is signed with the ES256 algorithm, with an expiration date set approximately 6 months ahead. Everything works as expected immediately after generating the token. However, after a few days, the same JWT (still well within its expiration period) suddenly begins returning invalid/unauthorized responses when used in Postman and other API clients. Importantly: I did not delete or revoke the .p8 key during this time. I verified the JWT contains valid claims and a proper structure. The issue consistently resolves only when I create a new .p8 file and regenerate a fresh JWT with it—after which the cycle repeats. This issue occurs even when the environment and app identifiers remain unchanged. I would greatly appreciate it if you could help me understand: Why these tokens become invalid after a few days, despite having a long exp value and an unchanged key. Whether there's any automatic revocation or timeout policy on .p8 keys that could explain this behavior. If there's a better way to maintain long-lived developer tokens without requiring new .p8 key generation every few days. Thank you for your help and clarification on this issue. Best regards, Liad Altif

1, Devices Apple TV HD tvOS 18.5 (22L572) Mode(A1625) HomePod Gen1 OS18.5 (22L572) 2, Test cases: 2.1 UIViewController + AVPlayerLayer + AVPlayer Result: Can not AirPlay to HomePod Sample code for UIViewController + AVPlayer 2.1 AVPlayerViewController + AVPlayer Result: Can AirPlay to HomePod Sample code for AVPlayerViewController + AVPlayer

I've got a setup using AVAudioEngine with several tone generator nodes, each with a chain of processing nodes, the chains then mixed into the main output. Generator ➡️ Effect ➡️... ➡️ .mainMixerNode ➡️ .outputNode). Generator ➡️ Effect ➡️... ⤴️ ... Generator ➡️ Effect ➡️... ⤴️ The user should be able to mute any chain individually. I've found several potential approaches to muting, but not terribly happy with any of them. Adjust the amplitudes directly in my tone generators. Issue: Consumes CPU even when completely muted. 4 generators adds ~15% cpu, even when all chains are muted. Detach/attach chains that are muted/unmuted. Issue: Causes loud clicking/popping sounds whenever muted/unmuted. Fade mixer output volume while detaching/attaching a chain (just cutting the volume immediately to 0 doesn't get rid of the clicking/popping). Issue: Causes all channels to fade during the transition, so not ideal. The rest of these ideas are variations on making volume control+detatch/attach work for individual chains, since approach #3 worked well. Add an AVAudioMixer to the end of each chain (just for volume control). Issue: Only the mixer on the final chain functions -- the others block all output. Not sure what's going on there. Use matrix mixer (for multi-input volume control). Plus detach/attach to reduce CPU if necessary. Not yet attempted, due to perceived complexity and reports of fragility in order of wiring in. A bunch of effort before I even know if it's going to work. Develop my own fader node to put on the end of each channel. Unlike the tone generator (simple AVSourceNode), developing an effect node seems complex and time consuming. Might not even fix CPU use. I'm not completely averse to the learning curve of either 5 or 6, but would rather get some guidance on best approach before diving in. They both seem likely to take more effort than I'd like for the simple behavior I'm trying to achieve.

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.

I need to implement a solution through an API or custom driver to completely block out the built-in speakers and microphone of Mac, because I need other apps to use specified external devices as audio input and output. Is there a way to achieve this requirement? What I mean is that even in system preferences, it should not be possible to choose the built-in microphone and speakers; only my external device can be used.

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!

Hey everyone, I'm encountering an issue with audio sample rate conversion that I'm hoping someone can help with. Here's the breakdown: Issue Description: I've installed a tap on an input device to convert audio to an optimal sample rate. There's a converter node added on top of this setup. The problem arises when joining Zoom or FaceTime calls—the converter gets deallocated from memory, causing the program to crash. Symptoms: The converter node is being deallocated during video calls. The program crashes entirely when this happens. Traditional methods of monitoring sample rate changes (tracking nominal or actual sample rates) aren't working as expected. The Big Challenge: I can't figure out how to properly monitor sample rate changes. Listeners set up to track these changes don't trigger when the device joins a Zoom or FaceTime call. Please, if anyone has experience with this or knows a solution, I'd really appreciate your help. Thanks in advance! ⁠

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

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!