Many Apple users own both Bluetooth earphones (AirPods) and traditional wired earphones. While Bluetooth audio provides freedom of movement, some users still prefer wired earphones for comfort, sound profile, or personal preference. However, plugging wired earphones directly into an iPhone can feel restrictive and inconvenient during daily use. This proposal suggests a hybrid audio approach where wired earphones can be connected to a Bluetooth-enabled AirPods charging case (or a similar Apple-designed module), allowing users to enjoy wired earphones without a physical connection to the iPhone. #Problem Statement *Wired earphones offer consistent audio quality and zero latency *Bluetooth earphones provide freedom from cables *Users must currently choose one or the other *Plugging wired earphones into an iPhone limits movement and can feel intrusive in daily scenarios (walking, commuting, working) There is no native Apple solution that allows wired earphones to function wirelessly while maintaining Apple’s audio experience standards. #Proposed Solution Introduce a Wired-to-Wireless Audio Mode through the AirPods charging case or a dedicated Apple Bluetooth audio bridge. How it works: User plugs wired earphones into the AirPods case (or a future AirPods accessory port) The case acts as a Bluetooth audio transmitter Audio is streamed wirelessly from iPhone to the case The case outputs audio to the wired earphones #User experiences: No cable connected to the iPhone Familiar wired earphone sound Freedom of movement similar to Bluetooth earbuds User Experience (UX Flow) Plug wired earphones into the AirPods case iPhone automatically detects: “Wired Earphones via AirPods Case” Seamless pairing using existing AirPods framework Audio controls, volume, and switching handled through iOS No additional apps required #Key Benefits Combines wired sound reliability with wireless convenience Reduces physical cable disturbance during use Extends usefulness of existing wired earphones Minimal learning curve for users Fits naturally into Apple’s ecosystem and design philosophy #Privacy & Performance Considerations On-device audio processing only No cloud involvement Low-latency audio using Apple’s proprietary Bluetooth codecs Power-efficient usage leveraging AirPods case battery #Target Users Users who prefer wired earphones but want wireless freedom Commuters and walkers Developers and professionals who multitask Users sensitive to Bluetooth earbud fit or comfort #Ecosystem Fit Builds on existing AirPods pairing and audio stack Aligns with Apple’s focus on seamless UX Could be implemented via: New AirPods hardware Firmware update + accessory Dedicated Apple audio bridge

Thread 5 Crashed: 0 libobjc.A.dylib 0x19af7b038 objc_msgSend + 56 1 CoreFoundation 0x19dfdb618 cow_cleanup + 135 2 CoreFoundation 0x19dfdb6fc -[__NSDictionaryM dealloc] + 147 3 MediaToolbox 0x1b167636c FigRemotePropertyCacheTeardown + 31 4 MediaToolbox 0x1b1c5b648 remoteXPCAsset_Finalize + 107 5 CoreMedia 0x1b1e9166c FigBaseObjectFinalize + 275 6 CoreFoundation 0x19dfcc5ec _CFRelease + 295 7 AVFCore 0x1b1054d64 -[AVFigAssetTrackInspector dealloc] + 151 8 AVFCore 0x1b0f818d8 -[AVAssetTrack dealloc] + 63 9 CoreFoundation 0x19dfdba28 RELEASE_OBJECTS_IN_THE_ARRAY + 115 10 CoreFoundation 0x19dfdb7e0 -[__NSArrayM dealloc] + 147 11 AVFCore 0x1b0f52e04 -[AVURLAsset dealloc] + 167 12 libobjc.A.dylib 0x19af821f8 object_cxxDestructFromClass(objc_object*, objc_class*) + 115 13 libobjc.A.dylib 0x19af7df20 objc_destructInstance_nonnull_realized(objc_object*) + 75 14 libobjc.A.dylib 0x19af7d4a4 _objc_rootDealloc + 71 15 AVFCore 0x1b0fef988 -[AVAssetReaderOutput dealloc] + 415 16 AVFCore 0x1b0ff11ec -[AVAssetReaderTrackOutput dealloc] + 127 17 CoreFoundation 0x19dfe20a4 -[__NSSingleObjectArrayI dealloc] + 63 18 libobjc.A.dylib 0x19af7d3f8 AutoreleasePoolPage::releaseUntil(objc_object**) + 203

Hi all, I've developed an audio DSP application in C++ using AudioToolbox and CoreAudio on MacOS 14.4.1 with Xcode 15. I use an AudioQueue for input and another for output. This works great. I'm now adding real-time audio analysis eg spectral analysis. I want this to run independently of my audio processing so it can not interfere with audio playback. Taps on AudioQueues seem to be a good way of doing this... Since the analytics won't modify the audio data, I am using a Siphon Tap by setting the AudioQueueProcessingTapFlags to kAudioQueueProcessingTap_PreEffects | kAudioQueueProcessingTap_Siphon; This works fine on my output queue. However, on my input queue the Tap callback is called once and then a EXC_BAD_ACCESS occurs - screen shot below. NB: I believe that a callback should only call AudioQueueProcessingTapGetSourceAudio when not using a Siphon, so I don't call it. Relevant code: AudioQueueProcessingTapCallback tap_callback) { // Makes an audio tap for a queue void * tap_data_ptr = NULL; AudioQueueProcessingTapFlags tap_flags = kAudioQueueProcessingTap_PostEffects | kAudioQueueProcessingTap_Siphon; uint32_t max_frames = 0; AudioStreamBasicDescription asbd; AudioQueueProcessingTapRef tap_ref; OSStatus status = AudioQueueProcessingTapNew(queue_ref, tap_callback, tap_data_ptr, tap_flags, &max_frames, &asbd, &tap_ref); if (status != noErr) printf("Error while making Tap\n"); else printf("Successfully made tap\n"); } void tapper(void * tap_data, AudioQueueProcessingTapRef tap_ref, uint32_t number_of_frames_in, AudioTimeStamp * ts_ptr, AudioQueueProcessingTapFlags * tap_flags_ptr, uint32_t * number_of_frames_out_ptr, AudioBufferList * buf_list) { // Callback function for audio queue tap printf("Tap callback"); }``` Image of exception stack provided by Xcode: ![]("https://developer.apple.com/forums/content/attachment/27479e8d-a118-459b-aa2d-7e30528910e3" "title=Screenshot 2025-06-14 at 1.29.14 PM.png;width=932;height=562") What have I missed? Appreciate any help you learned folks may be able to provide. Best, Geoff.

Hi everyone, I’m testing audio recording on an iPhone 15 Plus using AVFoundation. Here’s a simplified version of my setup: let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatLinearPCM), AVSampleRateKey: 8000, AVNumberOfChannelsKey: 1, AVLinearPCMBitDepthKey: 16, AVLinearPCMIsFloatKey: false ] audioRecorder = try AVAudioRecorder(url: fileURL, settings: settings) audioRecorder?.record() When I check the recorded file’s sample rate, it logs: Actual sample rate: 8000.0 However, when I inspect the hardware sample rate: try session.setCategory(.playAndRecord, mode: .default) try session.setActive(true) print("Hardware sample rate:", session.sampleRate) I consistently get: `Hardware sample rate: 48000.0 My questions are: Is the iPhone mic actually capturing at 8 kHz, or is it recording at 48 kHz and then downsampling to 8 kHz internally? Is there any way to force the hardware to record natively at 8 kHz? If not, what’s the recommended approach for telephony-quality audio (true 8 kHz) on iOS devices? Thanks in advance for your guidance!

I have integrated the ShazamKit SDK into my iOS app and would like to implement the same functionality in my Android app. My question is: Can I use the Android version of the ShazamKit SDK for commercial purposes? After extensive research, I could not find any official information regarding the license of the Android version of the ShazamKit SDK. Could you please provide a formal license statement?

3 I am working on an application to get when input audio device is being used. Basically I want to know the application using the microphone (built-in or external) This app runs on macOS. For Mac versions starting from Sonoma I can use this code: int getAudioProcessPID(AudioObjectID process) { pid_t pid; if (@available(macOS 14.0, *)) { constexpr AudioObjectPropertyAddress prop { kAudioProcessPropertyPID, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMain }; UInt32 dataSize = sizeof(pid); OSStatus error = AudioObjectGetPropertyData(process, &prop, 0, nullptr, &dataSize, &pid); if (error != noErr) { return -1; } } else { // Pre sonoma code goes here } return pid; } which works. However, kAudioProcessPropertyPID was added in macOS SDK 14.0. Does anyone know how to achieve the same functionality on previous versions?

Good day, ladies and gents. I have an application that reads audio from the microphone. I'd like it to also be able to read from the Mac's audio output stream. (A bonus would be if it could detect when the Mac is playing music.) I'd eventually be able to figure it out reading docs, but if someone can give a hint, I'd be very grateful, and would owe you the libation of your choice. Here's the code used to set up the AudioUnit: -(NSString*) configureAU { AudioComponent component = NULL; AudioComponentDescription description; OSStatus err = noErr; UInt32 param; AURenderCallbackStruct callback; if( audioUnit ) { AudioComponentInstanceDispose( audioUnit ); audioUnit = NULL; } // was CloseComponent // Open the AudioOutputUnit description.componentType = kAudioUnitType_Output; description.componentSubType = kAudioUnitSubType_HALOutput; description.componentManufacturer = kAudioUnitManufacturer_Apple; description.componentFlags = 0; description.componentFlagsMask = 0; if( component = AudioComponentFindNext( NULL, &description ) ) { err = AudioComponentInstanceNew( component, &audioUnit ); if( err != noErr ) { audioUnit = NULL; return [ NSString stringWithFormat: @"Couldn't open AudioUnit component (ID=%d)", err] ; } } // Configure the AudioOutputUnit: // You must enable the Audio Unit (AUHAL) for input and output for the same device. // When using AudioUnitSetProperty the 4th parameter in the method refers to an AudioUnitElement. // When using an AudioOutputUnit for input the element will be '1' and the output element will be '0'. param = 1; // Enable input on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &param, sizeof(UInt32) ); chkerr("Couldn't set first EnableIO prop (enable inpjt) (ID=%d)"); param = 0; // Disable output on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(UInt32) ); chkerr("Couldn't set second EnableIO property on the audio unit (disable ootpjt) (ID=%d)"); param = sizeof(AudioDeviceID); // Select the default input device AudioObjectPropertyAddress OutputAddr = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; err = AudioObjectGetPropertyData( kAudioObjectSystemObject, &OutputAddr, 0, NULL, &param, &inputDeviceID ); chkerr("Couldn't get default input device (ID=%d)"); // Set the current device to the default input unit err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDeviceID, sizeof(AudioDeviceID) ); chkerr("Failed to hook up input device to our AudioUnit (ID=%d)"); callback.inputProc = AudioInputProc; // Setup render callback, to be called when the AUHAL has input data callback.inputProcRefCon = self; err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(AURenderCallbackStruct) ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); param = sizeof(AudioStreamBasicDescription); // get hardware device format err = AudioUnitGetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &deviceFormat, &param ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); audioChannels = MAX( deviceFormat.mChannelsPerFrame, 2 ); // Twiddle the format to our liking actualOutputFormat.mChannelsPerFrame = audioChannels; actualOutputFormat.mSampleRate = deviceFormat.mSampleRate; actualOutputFormat.mFormatID = kAudioFormatLinearPCM; actualOutputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved; if( actualOutputFormat.mFormatID == kAudioFormatLinearPCM && audioChannels == 1 ) actualOutputFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved; #if __BIG_ENDIAN__ actualOutputFormat.mFormatFlags |= kAudioFormatFlagIsBigEndian; #endif actualOutputFormat.mBitsPerChannel = sizeof(Float32) * 8; actualOutputFormat.mBytesPerFrame = actualOutputFormat.mBitsPerChannel / 8; actualOutputFormat.mFramesPerPacket = 1; actualOutputFormat.mBytesPerPacket = actualOutputFormat.mBytesPerFrame; // Set the AudioOutputUnit output data format err = AudioUnitSetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &actualOutputFormat, sizeof(AudioStreamBasicDescription)); chkerr("Could not change the stream format of the output device (ID=%d)"); param = sizeof(UInt32); // Get the number of frames in the IO buffer(s) err = AudioUnitGetProperty( audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &audioSamples, &param ); chkerr("Could not determine audio sample size (ID=%d)"); err = AudioUnitInitialize( audioUnit ); // Initialize the AU chkerr("Could not initialize the AudioUnit (ID=%d)"); // Allocate our audio buffers audioBuffer = [self allocateAudioBufferListWithNumChannels: actualOutputFormat.mChannelsPerFrame size: audioSamples * actualOutputFormat.mBytesPerFrame]; if( audioBuffer == NULL ) { [ self cleanUp ]; return [NSString stringWithFormat: @"Could not allocate buffers for recording (ID=%d)", err]; } return nil; } (...again, it would be nice to know if audio output is active and thereby choose the clean output stream over the noisy mic, but that would be a different chunk of code, and my main question may just be a quick edit to this chunk.) Thanks for your attention! ==Dave [p.s. if i get more than one useful answer, can i "Accept" more than one, to spread the credit around?] {pps: of course, the code lines up prettier in a monospaced font!}

Problem: When the screen is locked, an incoming call does not initiate the launch of the Flutter application required for audio and video communication through Jitsi Meet. In the unlocked state, the application functions correctly. The current implementation does not have a mechanism for activating the Flutter engine when receiving a call via CallKit while the screen is locked. Although CallKit UI displays the call acceptance interface and the audio session is configured, the Flutter application remains in a suspended state, making it impossible to connect to the media server. Audio session activated using didActivateAudioSession method.

My app - natively iOS but built with the "Designed for iPad" option to run on Mac - does not recognise an attached USB microphone when running on a Mac. This line int32_t items = (int32_t) [[[AVAudioSession sharedInstance] availableInputs] count ]; returns 1, which is the Mac internal mic. On iPad and iPhone it sees both the internal mic and the USB mic. Is this an inherent "Designed for iPad" restriction, and is there some trick I can pull to get the USB microphone to be recognised by the system?

I’m facing a strange audio routing issue that seems specific to iPhone 14 Pro / Pro Max. I’m using LiveKit (WebRTC) in a React Native app, which uses AVAudioSession internally for audio capture (VoIP / call-style usage). 🔍 What’s happening: I’m using an external USB microphone. On these devices: iPhone 11 → ✅ USB mic works iPhone 13 → ✅ USB mic works iPhone 17 Pro → ✅ USB mic works iPhone 14 Pro Max → ❌ USB mic does NOT work On iPhone 14 Pro Max: The same USB mic: ✅ Works in Voice Memos ✅ Works in Instagram Live ❌ Does NOT appear as an input option in my app ❌ Does NOT work in WhatsApp / Instagram calls Also: In my app on iPhone 14 Pro Max, iOS does not show the audio input selector UI On iPhone 17 Pro, the same app and same build does show the selector and the USB mic works ⚙️ My audio session config ( LiveKit ): await AudioSession.setAppleAudioConfiguration({ audioCategory: 'playAndRecord', audioMode: 'default', audioCategoryOptions: ['allowBluetooth', 'defaultToSpeaker'], }); await AudioSession.startAudioSession(); ❓ My questions: Is this a known limitation or behavior specific to iPhone 14 Pro / Pro Max? Does iPhone 14 Pro have different audio routing rules for call / VoIP mode compared to other devices? Why does the same USB mic work in recording apps (Voice Memos, Instagram Live) but not in call-style apps (LiveKit, WhatsApp, Instagram call)? Is there any documented difference in AVAudioSession behavior on iPhone 14 Pro regarding external USB audio inputs?

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() } }

I might have misunderstood the docs, but is Call Translation going to be available for VOIP applications? Eg in an already connected VOIP call, would it be possible for Call Translations to be enabled on an iOS 26 and Apple Intelligence supported device? I have personally tried it and it doesn’t look like it supported VOIP but would love to confirm this. reference: https://developer.apple.com/documentation/callkit/cxsettranslatingcallaction/

I am developing an app with transcription and I am exploring ways to improve the transcription from the SpeechAnalyzer/Transcriber for technical terms. SFSpeech... recognition had the capability of being augmented by contextualStrings. Does something similar exist for SpeechAnalyzer/Transcriber? If so please point me towards the documentation and any sample code that may exist for this. If there are other options, please let me know.

Is there a way to destroy MIDIUMPMutableEndpoint again? In my app, the user has a setting to enable and disable MIDI 2.0. If MIDI 2.0 should not be supported (or if iOS version < 18), it creates a virtual destination and a virtual source. And if MIDI 2.0 should be enabled, it instead creates a MIDIUMPMutableEndpoint, which itself creates the virtual destination and source automatically. So here is my problem: I didn't find any way to destroy the MIDIUMPMutableEndpoint again. There is a method to disable it (setEnabled:NO), but that doesn't destroy or hide the virtual destination and source. So when the user turns MIDI 2.0 support off, I will have two virtual destinations and sources, and cannot get rid of the 2.0 ones. What is the correct way to get rid of the MIDIUMPMutableEndpoint once it is created?

I’m an amateur developer working on a free utility for composers/producers, for which the macOS release needs to create and name RTP-MIDI sessions in Audio MIDI Setup from the command line (so I can ship a small C helper instead of telling users to click through the UI). Here’s what I’ve tried so far, without luck: • Plist hacks: Injecting entries into ~/Library/Audio/MIDI Configurations/*.mcfg works when AMS is closed, but AMS immediately locks and reverts my changes when it’s open. • CoreMIDI C API: I can create virtual ports with MIDISourceCreate, but attempting MIDIObjectGetDataProperty on the apple.midirtp.session plugin always returns err –10836. • Obj-C & Swift: Loading MIDINetworkSession and calling defaultSession, init, setNetworkName: and setting enabled = YES doesn’t produce a new session object in the Network panel. • dlopen/dlsym: I extracted the real CoreMIDI binary out of the dyld shared cache and tried binding _MIDINetworkSessionCreate, _SetName, _SetEnabled, etc., but all the symbols come back null or my tool segfaults. • Plugin registration: I’ve pulled the factory UUID (70C9C5EA-7C65-11D8-B317-000393A34B5A) from /System/Library/Extensions/AppleMIDIRTPDriver.plugin/Contents/Info.plist and called CFPlugInRegisterFactories, but it still never exposes the session-creation calls. At this point I’m convinced I’m either loading the wrong binary or missing one critical step in registering the RTP-MIDI plugin’s private API. Can anyone point me to: The exact path of the dylib or bundle that actually exports the MIDINetworkSessionCreate/MIDINetworkSessionSetName/MIDINetworkSessionSetEnabled symbols? A minimal working snippet (C or Obj-C) that reliably creates and names a Network-MIDI session? Any pointers, sample code, or even ideas about where Apple hides this functionality on macOS 15 would be hugely appreciated. Thanks!

Environment Windows 11 [edition/build]: [e.g., 23H2, 22631.x] Apple Music for Windows version: [e.g., 1.x.x from Microsoft Store] Library folder: C:\Users<user>\Music\Apple Music\Apple Music Library.musiclibrary Summary I need a supported way to programmatically enumerate the local Apple Music library on Windows (track file paths, playlists, etc.) for reconciliation with the on-disk Media folder. On macOS this used to be straightforward via scripting/export; on Windows I can’t find an equivalent. What I’m seeing in the library bundle Library.musicdb → not SQLite. First 4 bytes: 68 66 6D 61 ("hfma"). Library Preferences.musicdb → also starts with "hfma". artwork.sqlite → SQLite but appears to be artwork cache only (no track file paths). Extras.itdb → has SQLite format 3 header but (from a quick scan) not seeing track locations. Genius.itdb → not a SQLite database on this machine. What I’ve tried Attempted to open Library.musicdb with SQLite providers → error: “file is not a database.” Binary/string scans (ASCII, UTF-16LE/BE, null-stripped) of Library.musicdb → did not reveal file paths or obvious plist/XML/JSON blobs. The Windows Apple Music UI doesn’t appear to expose “Export Library / Export Playlist” like legacy iTunes did, and I can’t find a public API for local library enumeration on Windows. What I’m trying to accomplish Read local track entries (absolute or relative paths), detect broken links, and reconcile against the Media folder. A read-only solution is fine; I do not need to modify the library. Questions for Apple Is the Library.musicdb file format documented anywhere, or is there a supported SDK/API to enumerate the local library on Windows? Is there a supported export mechanism (CLI, UI, or API) on Windows Apple Music to dump the local library and/or playlists (XML/CSV/JSON)? Is there a Windows-specific equivalent to the old iTunes COM automation or any MusicKit surface that can return local library items (not streaming catalog) and their file locations? If none of the above exist today, is there a recommended workaround from Apple for library reconciliation on Windows (e.g., documented support for importing M3U/M3U8 to rebuild the local library from disk)? Are there any plans/timeline for adding Windows feature parity with iTunes/Music on macOS for exporting or scripting the local library? Why this matters For large personal libraries, users occasionally end up with orphaned files on disk or broken links in the app. Without an export or API, it’s difficult to audit and fix at scale on Windows. Reference details (in case it helps triage) Library.musicdb header bytes: 68-66-6D-61-A0-00-00-00-10-26-34-00-15-00-01-00 (ASCII shows hfma…). artwork.sqlite is readable but doesn’t contain track file paths (appears limited to artwork). I can supply a minimal repro tool and logs if that’s helpful. Feature request (if no current API) Add an official Export Library/Playlists action on Windows Apple Music, or Provide a read-only Windows API (or schema doc) that surfaces track file locations and playlist membership from the local library. Thanks in advance for any guidance or pointers to docs I might have missed.

Hi, I'm trying to plan out development of an app and am wondering if it is possible to have user generated content automatically populate into a custom shazamkit catalogue and be able to query this catalogue non-locally? Storing all the submissions locally would obviously not scale.

I'm working on a project to support spatial audio editing, using this sample project as a reference: https://developer.apple.com/documentation/Cinematic/editing-spatial-audio-with-an-audio-mix This sample works well on an unedited capture, but does not work for a capture that has already been edited. The failure is occurring at "let audioInfo = try await CNAssetSpatialAudioInfo(asset: myAsset)", which is throwing "no eligible audio tracks in asset". I also find that for already edited captures, if i use CNAssetSpatialAudioInfo.assetContainsSpatialAudio, it returns false. What i mean by "already edited" is that if I take a spatial capture with my iPhone 16, and then edit that capture in the Photos app using the Cinematic effect, and then save the edited output (e.g. edited_capture.mov), I can't import that edited_capture.mov into my project as a spatial audio asset. Is this intentional behavior or a bug? If it's intentional, can you describe why?

I'm developing a TTS Audio Unit Extension that needs to write trace/log files to a shared App Group container. While the main app can successfully create and write files to the container, the extension gets sandbox denied errors despite having proper App Group entitlements configured. Setup: Main App (Flutter) and TTS Audio Unit Extension share the same App Group App Group is properly configured in developer portal and entitlements Main app successfully creates and uses files in the container Container structure shows existing directories (config/, dictionary/) with populated files Both targets have App Group capability enabled and entitlements set Current behavior: Extension can access/read the App Group container Extension can see existing directories and files All write attempts are blocked with "sandbox deny(1) file-write-create" errors Code example: const char* createSharedGroupPathWithComponent(const char* groupId, const char* component) { NSString* groupIdStr = [NSString stringWithUTF8String:groupId]; NSString* componentStr = [NSString stringWithUTF8String:component]; NSURL* url = [[NSFileManager defaultManager] containerURLForSecurityApplicationGroupIdentifier:groupIdStr]; NSURL* fullPath = [url URLByAppendingPathComponent:componentStr]; NSError *error = nil; if (![[NSFileManager defaultManager] createDirectoryAtPath:fullPath.path withIntermediateDirectories:YES attributes:nil error:&amp;error]) { NSLog(@"Unable to create directory %@", error.localizedDescription); } return [[fullPath path] UTF8String]; } Error output: Sandbox: simaromur-extension(996) deny(1) file-write-create /private/var/mobile/Containers/Shared/AppGroup/36CAFE9C-BD82-43DD-A962-2B4424E60043/trace Key questions: Are there additional entitlements required for TTS Audio Unit Extensions to write to App Group containers? Is this a known limitation of TTS Audio Unit Extensions? What is the recommended way to handle logging/tracing in TTS Audio Unit Extensions? If writing to App Group containers is not supported, what alternatives are available? Current entitlements: &lt;dict&gt; &lt;key&gt;com.apple.security.application-groups&lt;/key&gt; &lt;array&gt; &lt;string&gt;group.com.&lt;company&gt;.&lt;appname&gt;&lt;/string&gt; &lt;/array&gt; &lt;/dict&gt;

Hi 👋! We have a SpriteKit-based app where we play AVAudio sounds in three different ways: Effects (incl. UI sounds) with AVAudioPlayer. Long looping tracks with AVAudioPlayer. Short animation effects on the timeline of SpriteKit's SKScene files (effectively SKAudioNode nodes). We've found that when you exit the app or otherwise interrupt audio plays, future audio plays often fail. For example, there's a WebKit-based video trailer inside the app, and if you play it, our looping background music track (2.) will stop playing, and won't resume as you close the trailer (return from WebKit). This is probably due to us not manually restarting the track (so may well be easily fixed). Periodically played AVAudioPlayer audio (1.) are not affected. However, the more concerning thing is that the audio tracks on SKScene file timelines (3.) will no longer play. My hypothesis is that AVAudioEngine gets interrupted, and needs to be restarted for those AVAudioNode elements to regain functionality. Thing is, we don't deal with AVAudioEngine at all currently in the app, meaning it is never initiated to begin with. Obviously things return to normal when you remove the app from short-term memory and restart it. However, it seems many of our users aren't doing this, and often report audio failing presumably due to some interruption in the past without the app ever being cleared from memory. Any idea why timeline-run SKAudioNodes would fail like this? Should the app react to app backgrounding/foregrounding regarding audio? Any help would be very much appreciated ✌️!