When I use musicKit SDK for Android 1.1.2, I found that MediaContainerType only defines three types: NONE = 0; ALBUM = 1; PLAYLIST = 2; The RADIO_STATION type is not defined. However, the documentation of com.apple.android.music.playback.model states that the RADIO_STATION type is supported. This problem causes an error after I pass in the stations ID: MediaSessionManager com.apple.android.music.sdk.testapp D onPlaybackError() Quincy java.io.IOException May I ask how to solve this problem?

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？

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!

Environment→ ・Device: iPad 10th generation ・OS：**iOS18.3.2 I'm using AVAudioSession to record sound in my application. But I recently came to realize that when the app starts a recording session on a tablet, OS automatically sets the tablet volume to 50% and when after recording ends, it doesn't change back to the previous volume level before starting the recording. So I would like to know whether this is an OS default behavior or a bug? If it's a default behavior, I much appreciate if I can get a link to the documentation.

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!

I'm using an AVCaptureSession to send video and audio samples to an AVAssetWriter. When I play back the resultant video, sometimes there is a significant lag between the audio compared with the video, so they're just not in sync. But sometimes they are, with the same code. If I look at the very first presentation time stamps of the buffers being sent to the delegate, via func captureOutput(_: AVCaptureOutput, didOutput sampleBuffer: CMSampleBuffer, from connection: AVCaptureConnection) I see something like this: Adding audio samples for pts time 227711.0855328798, Adding video samples for pts time 227710.778785374 That is, the clock for audio vs video is behind: the first audio sample I receive is at 11.08 something, while the video video sample is earlier in time, at 10.778 something. The times are the presentation time stamps of the buffer, and the outputPresentationTimeStamp is the exact same number. It feels like "video" vs the "audio" clock are just mismatched. This doesn't always happen: sometimes they're synced. Sometimes they're not. Any ideas? The device I'm recording is a webcam, on iPadOS, connected via the usb-c port.

I'm developing a tennis ball tracking feature using Vision Framework in Swift, specifically utilizing VNDetectedObjectObservation and VNTrackObjectRequest. Occasionally (but not always), I receive the following runtime error: Failed to perform SequenceRequest: Error Domain=com.apple.Vision Code=9 "Internal error: unexpected tracked object bounding box size" UserInfo={NSLocalizedDescription=Internal error: unexpected tracked object bounding box size} From my investigation, I suspect the issue arises when the bounding box from the initial observation (VNDetectedObjectObservation) is too small. However, Apple's documentation doesn't clearly define the minimum bounding box size that's considered valid by VNTrackObjectRequest. Could someone clarify: What is the minimum acceptable bounding box width and height (normalized) that Vision Framework's VNTrackObjectRequest expects? Is there any recommended practice or official guidance for bounding box size validation before creating a tracking request? This information would be extremely helpful to reliably avoid this internal error. Thank you!

I've got a problem with my app where I'm testing it on my own phone. I'm using audio kit to generate tones as part of the app. Everything seems to work fine. Sounds start, Stop, etc. They play when the app is closed and when the phone is locked, so background is working. However, I'm seeing an issue where, even when STOP is pressed and the application exited, if I get a notification such as a text message, the base tone for the app starts to play. If I then open the app, check the Start/Stop button - it says start so that. hasnt' been activated. If I click Start, then a 2nd tone starts. This one stops with the Stop button. However the original tone that was set off by an incoming message carries on playing. Until I go to the Open Apps View on the phone and slide the application upwards. For the life of me, I can't figure out whats happening here.

I use the Apple Music API to poll my listening history at regular intervals. Every morning between 5:30AM and 7:30AM, I observe a strange pattern in the API responses. During this window, one or more of the regular polling intervals returns a response that differs significantly from the prior history response, even though I had no listening activity at that time. I'm using this endpoint: https://api.music.apple.com/v1/me/recent/played/tracks?types=songs,library-songs&include[library-songs]=catalog&include[songs]=albums,artists Here’s a concrete example from this morning: Time: 5:45AM Fetch 1 Tracks (subset): 1799261990, 1739657416, 1786317143, 1784288789, 1743250261, 1738681804, 1789325498, 1743036755, ... Time: 5:50AM Fetch 2 Tracks (subset): 1799261990, 1739657416, 1786317143, 1623924746, 1635185172, 1574004238, 1198763630, 1621299055, ... Time: 5:55AM Fetch 3 Tracks (subset): 1799261990, 1739657416, 1786317143, 1784288789, 1743250261, 1738681804, 1789325498, 1743036755, ... At 5:50, a materially different history is returned, then it returns back to the prior history at the next poll. I've listened to all of the tracks in each set, but the 5:50 history drops some tracks and returns some from further back in history. I've connected other accounts and the behavior is consistent and repeatable every day across them. It appears the API is temporarily returning a different (possibly outdated or cached?) view of the user's history during that early morning window. Has anyone seen this behavior before? Is this a known issue with the Apple Music API or MusicKit backend? I'd love any insights into what might cause this, or recommendations on how to work around it.

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.

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

Short summary When setting exposureMode to .locked or .custom the brightness of a video stream still changes depending on the composition and contrast of the visible scene. These changes seem to come from contrast enhancements or dynamic range optimizations and totally break any analysis of the image that requires to assess absolute luminance. While exposure lock seems to indeed lock the physical exposure parameters of the camera (shutter speed and ISO), I cannot find any way to control these "soft" modifiers. Details Background I am the developer of the app "phyphox", an educational app that makes the phone's sensors accessible to students as measurement tools in science experiments. Currently I am working on implementing photometric measurements through the camera and one very important aspect of it is luminance measurements. This is particularly relevant since the light sensor of the phone has no publicly accessible API and the camera could to some extend make experiments available to Apple users that are otherwise only possible on Android devices. Implementation The app uses AVFoundation and explicitly picks individual cameras since camera groups do not support custom exposure settings. This means that it handles camera switching during zoom by itself and even implements its own auto exposure routines to optimize for the use in experiments. Therefore it always stays in custom exposure mode. The app uses YUV420 color space and the individual frames are analyzed in Metal using compute shaders. However, the effects discussed here still occur if I remove all code to control the camera and replace it with a simple sequence of setting the exposure mode to custom, setting custom exposure values, setting a fixed white balance and then setting the exposure mode to locked as suggested on stackoverflow. This neither helps on an iPhone 14 Pro nor on an iPhone 8 despite a report on the developer forums that it would resolve the issue for older devices. The app is open source, so the code can be seen in our current development branch (without the changes for the tests here, though) on github. The videos below use the implementation with the suggestion from stackoverflow, but they can be reproduced in the same way with "professional" camera apps that promise manual control over the camera (like the Blackmagic cam to quote a reputable company) as well as the stock camera app after pressing and holding on the preview to enable AE/AF lock. Demonstration These examples were captured on an iPhone 14 Pro. The central part of the image (highlighted by the app using metal shaders after capture) should not change with fixed exposure settings, but significant changes are noticable if there are changes at the edge of the frame when I move a black piece of cardboard in from above: https://share.icloud.com/photos/0b1f_3IB6yAQG-qSH27pm6oDQ The graph above the camera preview is the average luminance (gamma corrected and weighted based on sRGB) across the highlighted central area and as mentioned before it should not change because of something happening at the side of the frame (worst case it should get a bit darker because of the cardboard's shadow). In my opinion, the iPhone changes its mind on the ideal contrast as soon as it has a different exposure histogram because of the dark image part from the cardboard, but that's just me guessing. For completeness here is the same effect in the stock camera app with AE/AF lock enabled: https://share.icloud.com/photos/0cd7QM8ucBZKwPwE9mybnEowg Here you can also see that the iPhone "ramps" the changes. The brightness of the gray area does not change immediately but transitions smoothly, so this is clearly deliberate postprocessing. So... Any suggestion on how to prevent this behavior would be highly appreciated.

I am developing a video streaming app for iPhone. Minimum version is IOS 13. I want to connect an external USB camera to the iPhone app and stream from it. I have looked through a lot of information and have not found how to do this. Is it possible to do this? Is there any documentation on this?

I have an AUv3 that passes all validation and can be loaded into Logic Pro without issue. The UI for the plug in can be any aspect ratio but Logic insists on presenting it in a view with a fixed aspect ratio. That is when resizing, both the height and width are resized. I have never managed to work out what it is I need to do specify to Logic to allow the user to resize width or height independently of each other. Can anyone tell me what I need to specify in the AU code that will inform Logic that the view can be resized from any side of the window/panel?

Is there a way to install HLS tools on ARM Linux? The only one I can find is for x86 Linux. I'm working on a project on a Raspberry Pi. I'd like to install the tool to generate my hls files, but the alternatives are more complicated to use. Is there a way to run them like Mac OS does with Rosseta?

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.

Hi, Could anybody share the full list of supported Predicate keys for the PHFetchOptions? I'm aware of the list that is posted in the documentation: https://developer.apple.com/documentation/photos/phfetchoptions However I have reason to believe that this is not an exhaustive list and there also seem to be mistakes in this doc. i.e. isFavorite does not work but favorite does. Through some experimentation I also found that this works: NSPredicate(format: "adjustmentFormatIdentifier == 'com.pixelmatorteam.touch.x.photo.PhotosAdjustmentData.EmbeddedSlimSidecarFileInfo.compressed'") even though adjustmentFormatIdentifier is not listed as a supported key. Are there other secret keys that you are aware of? Specifically I want to filter a fetch result for edited items. Something like this: NSPredicate(format: "hasAdjustments == true") (I tried this, doesn't work) The native Photos app has such a filter which leads me to believe that there probably is a key for this: If one of the Framework Developers reads this: Could you please update the documentation page with this information? Finally if there really aren't any more secret keys, is there a way to achieve this with adjustmentFormatIdentifier? I have tried a bunch of stuff already like adjustmentFormatIdentifier != nil but for some reason that gives me the exact opposite of what I want: all the photos without edits. 🫠 Any tips on the correct syntax here would be much appreciated.

Dear Apple Developer Community, I'm encountering a critical issue with the MusicLibrary.shared.createPlaylist() method in MusicKit that's affecting our app's core functionality. Despite implementing all recommended authorization checks, the app consistently freezes for some users when this method is called. What we've already verified before calling createPlaylist(): Network connectivity is properly checked and confirmed Apple Music authorization is explicitly requested via MusicAuthorization.request() User subscription status is verified using MusicSubscription.current.canPlayCatalogContent Despite these precautions, many users report that their app completely freezes when attempting to create a playlist. This is particularly concerning as playlist creation is a core feature of our application. User-reported workarounds (with mixed success): Some users have resolved the issue by restarting their devices or reinstalling our app Others report success after enabling "Sync Library" in Settings → Music Unfortunately, a significant number of users are still experiencing the issue even after trying both solutions above We've reviewed the MusicKit documentation thoroughly and ensured our implementation follows all best practices. Our app correctly handles permissions and uses the async/await pattern as required by the API. Is there a known issue with the createPlaylist() method that might cause it to block indefinitely? Are there additional authorization steps or settings we should be checking before calling this method? Could this be related to how MusicKit communicates with Apple Music servers? Any insights from the developer community or official guidance would be greatly appreciated as this issue is severely impacting our user experience. Thank you for your assistance

Hi everyone! I’ve been working with AVFoundation and trying to use the AVMetricEventStreamPublisher to discover media performance metrics, as described in the Apple documentation. https://developer.apple.com/cn/videos/play/wwdc2024/10113/?time=508 However, when following the example code, I’m not getting the expected results. The performance metrics for both audio and video don’t seem to be captured properly. Has anyone successfully used this example code? If so, could you share your experience or any solutions you’ve found? Any tips or insights would be greatly appreciated. Thanks in advance! Ps. the example code: AVPlayerItem *item = ... AVMetricEventStream *eventStream = [AVMetricEventStream eventStream]; id subscriber = [[MyMetricSubscriber alloc] init]; [eventStream setSubscriber:subscriber queue:mySerialQueue] [eventStream subscribeToMetricEvent:[AVMetricPlayerItemLikelyToKeepUpEvent class]]; [eventStream subscribeToMetricEvent:[AVMetricPlayerItemPlaybackSummaryEvent class]]; [eventStream addPublisher:item];

I have a simple AVAudioEngine graph as follows: AVAudioPlayerNode -> AVAudioUnitEQ -> AVAudioUnitTimePitch -> AVAudioUnitReverb -> Main mixer node of AVAudioEngine. I noticed that whenever I have AVAudioUnitTimePitch or AVAudioUnitVarispeed in the graph, I noticed a very distinct crackling/popping sound in my Airpods Pro 2 when starting up the engine and playing the AVAudioPlayerNode and unable to find the reason why this is happening. When I remove the node, the crackling completely goes away. How do I fix this problem since i need the user to be able to control the pitch and rate of the audio during playback. import AVKit @Observable @MainActor class AudioEngineManager { nonisolated private let engine = AVAudioEngine() private let playerNode = AVAudioPlayerNode() private let reverb = AVAudioUnitReverb() private let pitch = AVAudioUnitTimePitch() private let eq = AVAudioUnitEQ(numberOfBands: 10) private var audioFile: AVAudioFile? private var fadePlayPauseTask: Task<Void, Error>? private var playPauseCurrentFadeTime: Double = 0 init() { setupAudioEngine() } private func setupAudioEngine() { guard let url = Bundle.main.url(forResource: "Song name goes here", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: url) } catch { print("Failed to load audio file: \(error)") return } reverb.loadFactoryPreset(.mediumHall) reverb.wetDryMix = 50 pitch.pitch = 0 // Increase pitch by 500 cents (5 semitones) engine.attach(playerNode) engine.attach(pitch) engine.attach(reverb) engine.attach(eq) // Connect: player -> pitch -> reverb -> output engine.connect(playerNode, to: eq, format: audioFile?.processingFormat) engine.connect(eq, to: pitch, format: audioFile?.processingFormat) engine.connect(pitch, to: reverb, format: audioFile?.processingFormat) engine.connect(reverb, to: engine.mainMixerNode, format: audioFile?.processingFormat) } func prepare() { guard let audioFile else { return } playerNode.scheduleFile(audioFile, at: nil) } func play() { DispatchQueue.global().async { [weak self] in guard let self else { return } engine.prepare() try? engine.start() DispatchQueue.main.async { [weak self] in guard let self else { return } playerNode.play() fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: true) // Ramp up volume until 1 is reached if volume >= 1 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 1 } } } } func pause() { fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: false) // Ramp down volume until 0 is reached if volume <= 0 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 0 playerNode.pause() // Shut down engine once ramp down completes DispatchQueue.global().async { [weak self] in guard let self else { return } engine.pause() } } } private func updateVolume(for x: Double, rising: Bool) -> Float { if rising { // Fade in return Float(pow(x, 2) * (3.0 - 2.0 * (x))) } else { // Fade out return Float(1 - (pow(x, 2) * (3.0 - 2.0 * (x)))) } } func setPitch(_ value: Float) { pitch.pitch = value } func setReverbMix(_ value: Float) { reverb.wetDryMix = value } } struct ContentView: View { @State private var audioManager = AudioEngineManager() @State private var pitch: Float = 0 @State private var reverb: Float = 0 var body: some View { VStack(spacing: 20) { Text("🎵 Audio Player with Reverb & Pitch") .font(.title2) HStack { Button("Prepare") { audioManager.prepare() } Button("Play") { audioManager.play() } .padding() .background(Color.green) .foregroundColor(.white) .cornerRadius(10) Button("Pause") { audioManager.pause() } .padding() .background(Color.red) .foregroundColor(.white) .cornerRadius(10) } VStack { Text("Pitch: \(Int(pitch)) cents") Slider(value: $pitch, in: -2400...2400, step: 100) { _ in audioManager.setPitch(pitch) } } VStack { Text("Reverb Mix: \(Int(reverb))%") Slider(value: $reverb, in: 0...100, step: 1) { _ in audioManager.setReverbMix(reverb) } } } .padding() } }