When making a call to https://api.music.apple.com/v1/me/library/artists to get a user's library artists, it returns the following (as an example): [ { id: 'r.FCwruQb', type: 'library-artists', href: '/v1/me/library/artists/r.FCwruQb?l=en-US', attributes: { name: 'A Great Big World' } }, { id: 'r.7VSWOgj', type: 'library-artists', href: '/v1/me/library/artists/r.7VSWOgj?l=en-US', attributes: { name: 'Aaliyah' } }, ... ] If I try and use an artist id from that retuned data to look up additional information about the artist by calling https://api.music.apple.com/v1/catalog/us/artists/{id}, it fails. User Library Artists don't seem to equal Catalog Artists. It'd be great if there was a way to use these interchangeably. Am I missing something?

I have a feature requirement: to switch the writer for file writing every 5 minutes, and then quickly merge the last two files. How can I ensure that the merged file is seamlessly combined and that the audio and video information remains synchronized? Currently, the merged video has glitches, and the audio is also out of sync. If there are experts who can provide solutions in this area, I would be extremely grateful.

Hi, I've had a new deck installed in my car for about 1.5 weeks. I'm having compatibility issues with my 15PM. It happens both wired and wirelessly, I get the error "Accessory not supported by this device". It used to happen all the time, now it's 50/50. Sometimes it works. I've removed and added Bluetooth multiple times on phone and deck, I bought a belkin usb-c to usb-a cable today and it seems to fix it but the problem comes back. I've changed the setting "FaceID and passcode-allow access when locked-accessories." The car stereo guy reckons it's definitely an issue with the phone not the deck, I'm inclined to believe him since the error states "by this device". Any advice appreciated.

I'm creating Live Photos programmatically in my app using the Photos and AVFoundation frameworks. While the Live Photos work perfectly in the Photos app (long press shows motion), users cannot set them as motion wallpapers. The system shows "Motion not available" message. Here's my approach for creating Live Photos: // 1. Create video with required metadata let writer = try AVAssetWriter(outputURL: videoURL, fileType: .mov) let contentIdentifier = AVMutableMetadataItem() contentIdentifier.identifier = .quickTimeMetadataContentIdentifier contentIdentifier.value = assetIdentifier as NSString writer.metadata = [contentIdentifier] // Video settings: 882x1920, H.264, 30fps, 2 seconds // Added still-image-time metadata at middle frame // 2. Create HEIC image with asset identifier var makerAppleDict: [String: Any] = [:] makerAppleDict["17"] = assetIdentifier // Required key for Live Photo metadata[kCGImagePropertyMakerAppleDictionary as String] = makerAppleDict // 3. Generate Live Photo PHLivePhoto.request( withResourceFileURLs: [photoURL, videoURL], placeholderImage: nil, targetSize: .zero, contentMode: .aspectFit ) { livePhoto, info in // Success - Live Photo created } // 4. Save to Photos library PHAssetCreationRequest.forAsset().addResource(with: .photo, fileURL: photoURL, options: nil) PHAssetCreationRequest.forAsset().addResource(with: .pairedVideo, fileURL: videoURL, options: nil) What I've Tried Matching exact video specifications from Camera app (882x1920, H.264, 30fps) Adding all documented metadata (content identifier, still-image-time) Testing various video durations (1.5s, 2s, 3s) Different image formats (HEIC, JPEG) Comparing with exiftool against working Live Photos Expected Behavior Live Photos created programmatically should be eligible for motion wallpapers, just like those from the Camera app. Actual Behavior System shows "Motion not available" and only allows setting as static wallpaper. Any insights or workarounds would be greatly appreciated. This is affecting our users who want to use their created content as wallpapers. Questions Are there additional undocumented requirements for Live Photos to be wallpaper-eligible? Is this a deliberate restriction for third-party apps, or a bug? Has anyone successfully created Live Photos that work as motion wallpapers? Environment iOS 17.0 - 18.1 Xcode 16.0 Tested on iPhone 16 Pro

Hi, I have just implemented an Audio Unit v3 host. AgsAudioUnitPlugin *audio_unit_plugin; AVAudioUnitComponentManager *audio_unit_component_manager; NSArray<AVAudioUnitComponent *> *av_component_arr; AudioComponentDescription description; guint i, i_stop; if(!AGS_AUDIO_UNIT_MANAGER(audio_unit_manager)){ return; } audio_unit_component_manager = [AVAudioUnitComponentManager sharedAudioUnitComponentManager]; /* effects */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_Effect; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } /* instruments */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_MusicDevice; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } But this doesn't show me Audio Unit v2 plugins, why? regards, Joël

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)") } } } } }

Hi, when using ApplicationMusicPlayer from MusicKit my app automatically gets the media controls on the lock screen: Play/ Pause, Skip Buttons, Playback Position etc. I would like to customize these. Tried a bunch of things, e.g. using MPRemoteCommandCenter. So far I haven't had any success. Does anyone know how I can customize the media controls of ApplicationMusicPlayer. Thank you.

In iOS 26 (Developer Beta), the AVCaptureMetadataOutputObjectsDelegate no longer receives callbacks when metadataOutput.metadataObjectTypes = [.face] is set. On earlier iOS versions the issue does not occur. Interestingly, face detection works if I set the sessionPreset to .medium, but not with .high — except on the iPhone 16 Pro Max, where it works regardless.

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?

I am trying to use the new SpeechAnalyzer framework in my Mac app, and am running into an issue for some languages. When I call AssetInstallationRequest.downloadAndInstall() for some languages, it throws an error: Error Domain=SFSpeechErrorDomain Code=1 "transcription.ar asset not found after attempted download." The ".ar" appears to be the language code, which in this case was Arabic. When I call AssetInventory.status(forModules:) before attempting the download, it is giving me a status of "downloading" (perhaps from an earlier attempt?). If this language was completely unsupported, I would expect it to return a status of "unsupported", so I'm not sure what's going on here. For other languages (Polish, for example) SpeechTranscriber.supportedLocale(equivalentTo:) is returning nil, so that seems like a clearly unsupported language. But I can't tell if the languages I'm trying, like Arabic, are supported and something is going wrong, or if this error represents something I can work around. Here's the relevant section of code. The error is thrown from downloadAndInstall(), so I never even get as far as setting up the SpeechAnalyzer itself. private func setUpAnalyzer() async throws { guard let sourceLanguage else { throw Error.languageNotSpecified } guard let locale = await SpeechTranscriber.supportedLocale(equivalentTo: Locale(identifier: sourceLanguage.rawValue)) else { throw Error.unsupportedLanguage } let transcriber = SpeechTranscriber(locale: locale, preset: .progressiveTranscription) self.transcriber = transcriber let reservedLocales = await AssetInventory.reservedLocales if !reservedLocales.contains(locale) && reservedLocales.count == AssetInventory.maximumReservedLocales { if let oldest = reservedLocales.last { await AssetInventory.release(reservedLocale: oldest) } } do { let status = await AssetInventory.status(forModules: [transcriber]) print("status: \(status)") if let installationRequest = try await AssetInventory.assetInstallationRequest(supporting: [transcriber]) { try await installationRequest.downloadAndInstall() } } ...

Any one experience this bug, when playing a video the bluetooth headphones loose audio? my workaround is to select from one of the other audio outputs, and go back again and select the affected headphone.

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

I am trying to use SpeechDetector Module in Speech framework along with SpeechTranscriber. and it is giving me an error Cannot convert value of type 'SpeechDetector' to expected element type 'Array.ArrayLiteralElement' (aka 'any SpeechModule') Below is how I am using it let speechDetector = Speech.SpeechDetector() let transcriber = SpeechTranscriber(locale: Locale.current, transcriptionOptions: [], reportingOptions: [.volatileResults], attributeOptions: [.audioTimeRange]) speechAnalyzer = try SpeechAnalyzer(modules: [transcriber,speechDetector])

When I try to send a DRM-protected video via Airplay to an Apple TV, the license request is made twice instead of once as it normally does on iOS. We only allow one request per session for security reasons, this causes the second request to fail and the video won't play. We've tested DRM-protected videos without token usage limits and it works, but this creates a security hole in our system. Why does it request the license twice in function: func contentKeySession(_ session: AVContentKeySession, didProvide keyRequest: AVContentKeyRequest)? Is there a way to prevent this?

Hello, I’m new here. I'm developing an iOS app and I’d like to know whether it is possible to detect if a phone call is being recorded by another app running in the background. I’ve already reviewed the documentation for CallKit and AVAudioSession, but I couldn’t find anything related. My expectation was that iOS might provide some callback or API to indicate if a call is being recorded (third-party apps), but so far I haven’t found a way. My questions are: Does iOS expose any API to detect if a call is being recorded? If not, is there any indirect, Apple's policy compliant method (e.g., microphone usage events) that can be relied upon? Or is this something that iOS explicitly prevents for privacyreasons? Expecting solutions that align with Apple’s policies and would be accepted under the App Store Review Guidelines. Thanks in advance for any guidance.

AVAudioSessionCategoryOptionAllowBluetooth is marked as deprecated in iOS 8 in iOS 26 beta 5 when this option was not deprecated in iOS 18.6. I think this is a mistake and the deprecation is in iOS 26. Am I right? It seems that the substitute for this option is "AVAudioSessionCategoryOptionAllowBluetoothHFP". The documentation does not make clear if the behaviour is exactly the same or if any difference should be expected... Has anyone used this option in iOS 26? Should I expect any difference with the current behaviour of "AVAudioSessionCategoryOptionAllowBluetooth"? Thank you.

I am working to update a live blog in Apple News. As far as I know there is an update endpoint to update a content in Apple News. Is there any feature in Apple News to trigger an event when the original content updated and pull the updated content?

I'm experiencing a significant limitation with MusicKit's Dolby Atmos implementation on macOS and would appreciate clarification on whether this is intended behavior or if there are solutions available. When streaming Dolby Atmos content through MusicKit's ApplicationMusicPlayer, the output is limited to 2-channel stereo, even when: Audio MIDI Setup is configured for 7.1.4 (12-channel) output The same tracks play in full multichannel through the native Apple Music app Dolby Atmos is set to "Automatic" in Apple Music preferences Please let me know if there is anyway to enable this. If not, is this documented anywhere? Thanks!

Hi, I just started to develop audio unit hosting support in my application. Offline rendering seems to work except that I hear no output, but why? I suspect with the player goes something wrong. I connect to CoreAudio in a different location in the code. Here are some error messages I faced so far: 2025-08-14 19:42:04.132930+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:42:04.151171+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:43:08.344530+0200 com.gsequencer.GSequencer[34358:18614927] AUAudioUnit.mm:1417 Cannot set maximumFramesToRender while render resources allocated. 2025-08-14 19:43:08.346583+0200 com.gsequencer.GSequencer[34358:18614927] [avae] AVAEInternal.h:104 [AVAudioSequencer.mm:121:-[AVAudioSequencer(AVAudioSequencer_Player) startAndReturnError:]: (impl->Start()): error -10852 ** (<unknown>:34358): WARNING **: 19:43:08.346: error during audio sequencer start - -10852 I have implemented an AVAudioEngine based AudioUnit host. Here I instantiate player and effect: /* audio engine */ audio_engine = [[AVAudioEngine alloc] init]; fx_audio_unit_audio->audio_engine = (gpointer) audio_engine; av_format = (AVAudioFormat *) fx_audio_unit_audio->av_format; /* av audio player node */ av_audio_player_node = [[AVAudioPlayerNode alloc] init]; /* av audio unit */ av_audio_unit_effect = [[AVAudioUnitEffect alloc] initWithAudioComponentDescription:[((AVAudioUnitComponent *) AGS_AUDIO_UNIT_PLUGIN(base_plugin)->component) audioComponentDescription]]; av_audio_unit = (AVAudioUnit *) av_audio_unit_effect; fx_audio_unit_audio->av_audio_unit = av_audio_unit; /* audio sequencer */ av_audio_sequencer = [[AVAudioSequencer alloc] initWithAudioEngine:audio_engine]; fx_audio_unit_audio->av_audio_sequencer = (gpointer) av_audio_sequencer; /* output node */ [[AVAudioOutputNode alloc] init]; /* audio player and audio unit */ [audio_engine attachNode:av_audio_player_node]; [audio_engine attachNode:av_audio_unit]; [audio_engine connect:av_audio_player_node to:av_audio_unit format:av_format]; [audio_engine connect:av_audio_unit to:[audio_engine outputNode] format:av_format]; ns_error = NULL; [audio_engine enableManualRenderingMode:AVAudioEngineManualRenderingModeOffline format:av_format maximumFrameCount:buffer_size error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("enable manual rendering mode error - %d", [ns_error code]); } ns_error = NULL; [[av_audio_unit AUAudioUnit] allocateRenderResourcesAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("Audio Unit allocate render resources returned error - ErrorCode %d", [ns_error code]); } Then I render in a dedicated thread. ns_error = NULL; [audio_engine startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio engine start - %d", [ns_error code]); } [av_audio_sequencer prepareToPlay]; ns_error = NULL; [av_audio_sequencer startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio sequencer start - %d", [ns_error code]); } [av_audio_player_node play]; while(is_running){ /* pre sync */ /* IO buffers */ av_output_buffer = (AVAudioPCMBuffer *) scope_data->av_output_buffer; av_input_buffer = (AVAudioPCMBuffer *) scope_data->av_input_buffer; /* fill input buffer */ /* schedule av input buffer */ frame_position = 0; // (gint64) ((note_offset * absolute_delay) + delay_counter) * buffer_size; av_audio_player_node = (AVAudioPlayerNode *) fx_audio_unit_audio->av_audio_player_node; AVAudioTime *av_audio_time = [[AVAudioTime alloc] initWithHostTime:frame_position sampleTime:frame_position atRate:((double) samplerate)]; [av_audio_player_node scheduleBuffer:av_input_buffer atTime:av_audio_time options:0 completionHandler:nil]; /* render */ ns_error = NULL; status = [audio_engine renderOffline:AGS_FX_AUDIO_UNIT_AUDIO_FIXED_BUFFER_SIZE toBuffer:av_output_buffer error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("render offline error - %d", [ns_error code]); } } regards, Joël

Two issues: No matter what I set in try audioSession.setPreferredSampleRate(x) the sample rate on both iOS and macOS is always 48000 when the output goes through the speaker, and 24000 when my Airpods connect to an iPhone/iPad. Now, I'm checking the current output loudness to animate a 3D character, using mixerNode.installTap(onBus: 0, bufferSize: y, format: nil) { [weak self] buffer, time in Task { @MainActor in // calculate rms and animate character accordingly but any buffer size under 4800 is just ignored and the buffers I get are 4800 sized. This is ok, when the sampleRate is currently 48000, as 10 samples per second lead to decent visual results. But when AirPods connect, the samplerate is 24000, which means only 5 samples per second, so the character animation looks lame. My AVAudioEngine setup is the following: audioEngine.connect(playerNode, to: pitchShiftEffect, format: format) audioEngine.connect(pitchShiftEffect, to: mixerNode, format: format) audioEngine.connect(mixerNode, to: audioEngine.outputNode, format: nil) Now, I'd be fine if the outputNode runs at whatever if it needs, as long as my tap would get at least 10 samples per second. PS: Specifying my favorite format in the let format = AVAudioFormat(standardFormatWithSampleRate: 48_000, channels: 2)! mixerNode.installTap(onBus: 0, bufferSize: y, format: format) doesn't change anything either