HI Guys, I'm using Shazamkit in my IOS app and successfully capturing the currently playing track details, when using the devices (iPhone) built-in mic. When I test with AirPods though, my app cannot both send the output to through the AirPods and capture that same output with the AirPods mic, for Shazamkit recognition. I believe this must be possible, because the Shazamkit widget on IOS can do this. Is it restricted in some way for third party apps? If not, I'd appreciate some guidance on how to achieve this in Swift code. Thanks in advance.

I was trying to set custom audio output device for a generated audio on macCatalyst. While using let status = AudioUnitSetProperty(outputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &outputDeviceID, UInt32(MemoryLayout.size)) kAudioOutputUnitProperty_CurrentDevice is invalid, and status = -10879, indicating an error. STEPS TO REPRODUCE Set Run Destination to MacOS and run the program. "AudioUnitSetProperty: 0" should be printed, indicating it works fine. Set Run Destination to Mac Catalyst and run the program. "Error setting output device: -10879" should be printed, indicating an error.

If I call AudioDeviceStart on an AudioDevice in my application then "Hey Siri!" will not wake Siri up. Our users have complained that Siri does not get activated with my application is running. We found that calling AudioDeviceStart is causing the issue. How should we handle this?

Hi all, I have spent a lot of time reading the tech note and watching the WDDC video that introduce the PTTFramework on iOS. I currently have a custom setup where I am using AVAudioEngine to schedule and play buffers that are being streamed through a call. I am looking to use the PTTFramework to allow a user to trigger this push to talk behavior from the lock screen and the various places with the system UI it provides. However I am unsure what the correct behavior is regarding the handling of the audio session. Right now I am using .playback when there is no active voice transmission so that devices such as AirPods can be in AD2P mode where applicable, and then transitioning to .playbackAndRecord category only when the mic input should become active. Following this change in my AVAudioEngine manager I am then manually activating and deactivating the audio session manually when the engine is either playing/recording or idle. In the documentation it states that you should not attempt to activate or deactivate your audio session directly, but allow the framework to handle it. Does that mean that I need to either call the request to transmit delegate function or set an active participant on the channel manager first, and then wait for the didBecomeActive delegate method to trigger before I actually attempt to play or record any audio? (I am using the fullDuplex mode currently.) I noticed that that delegate method will only trigger if the audio session wasn't active before doing one of the above (setting active participant, requesting transmit). Lastly, when using the PTTFramework it also mentions that we get support for PTT devices and I notice on the didBeginTransmittingFrom property we have a handsfreeButton case. Is there any documentation or resources for what is actually supported out of the box for this? I am currently working on handling a lot of the push to talk through bluetooth LE, and wanted to make sure there wasn't overlap with what the system provides. Thank you!

I am unable to access the Int32 error from the errors that CoreAudio throws in Swift type AudioHardwareError. This is critical. There is no way to access the errors or even create an AudioHardwareError to test for errors. do { _ = try AudioHardwareDevice(id: 0).streams // will throw } catch { if let error = error as? AudioHardwareError { // cast to AudioHardwareError print(error) // prints error code but not the errorDescription } } How can get reliably get the error.Int32? Or create a AudioHardwareError with an error constant? There is no way for me to handle these error with code or run tests without knowing what the error is. On top of that, by default the error localizedDescription does not contain the errorDescription unless I extend AudioHardwareError with CustomStringConvertible. extension AudioHardwareError: @retroactive CustomStringConvertible { public var description: String { return self.localizedDescription } }

I've been generating new Audio Unit Extension apps with Xcode 16 (and newer), and although they generally work initially, it is easy (although I'm not sure how to do it reliably) to cause the app to no longer be able to instantiate the audiounit. Generally the call to AVAudioUnit.findComponent fails and SimplePlayEngine hits the fatalError("Failed to find component with type...") In the most recent project, merely adding files to the extension (without making any use of them) caused it to go off the rails. If I "Archive" the app+plugin, there is no audio unit extension in the bundle. If I switch to the audiounit extension and build it it's fine. If I look at the build folder in Library/Developer/Xcode/project_folder the extension_name.appex is there. Any ideas? If I can coax an unmodified audio unit extension project to exhibit this behavior I'll attach it here. Right now what I have has code I don't want to share.

I tried adding watermarks to the recorded video. Appending sample buffers using AVAssetWriterInput's append method fails and when I inspect the AVAssetWriter's error property, I get the following: Error Domain=AVFoundation Error Domain Code=-11800 "This operation cannot be completed" UserInfo={NSLocalizedFailureReason=An unknown error occurred (-12780), NSLocalizedDDescription=This operation cannot be completed, NSUnderlyingError=0x302399a70 {Error Domain=NSOSStatusErrorDomain Code=-12780 "(null)"}} As far as I can tell -11800 indicates an AVErrorUknown, however I have not been able to find information about the -12780 error code, which as far as I can tell is undocumented. Thanks! Here is the code

The problem I have at the moment is that if a phone call comes in during my recording, even if I don't answer, my recording will be interrupted The phenomenon of recording interruption is that the picture is stuck, and the recording can be resumed automatically after the call is over. But it will cause the recorded video sound and painting out of sync Through the AVCaptureSessionWasInterrupted listening, I can get to record the types of alerts and interrupt As far as I can tell, a ringing or vibrating phone can block the audio channel. I found the same scenario in other apps, you can turn off the ring tone or vibration, but I don't know how to do it, I tried a lot of ways, but it doesn't work BlackmagicCam or ProMovie App, when a call comes in during recording, there will only be a notification menu, and there will be no ringtone or vibration, which solves the problem of recording interruption I don't know if this requires some configuration or application, please let me know if it does

Hello, I am trying to follow the getting started guide. I have produced a developer token via the music kit embedding approach and can confirm I'm successfully authorized. When I try to do play music, I'm unable to hear anything. Thought it could be some auto-play problems with the browser, but it doesn't appear to be related, as I can trigger play from a button with no further success. const music = MusicKit.getInstance() try { await music.authorize() // successful const result = await music.api.music(`/v1/catalog/gb/search`, { term: 'Sound Travels', types: 'albums', }) await music.play() } catch (error) { console.error('play error', error) // ! No error triggered } I have searched the forum, have found similar queries but apparently none using V3 of the API. Other potentially helpful information: OS: macos 15.1 (24B83) API version: V3 On localhost Browser: Arc (chromium based), also tried on Safari, The only difference between the two browsers is that safari appears to exit the breakpoint, whereas Arc will continue (without throwing any errors) authorizationStatus: 3 Side note, any reason this is still in beta so many years later?

The following is my playground code. Any of the apple audio units show the plugin view, however anything else (i.e. kontakt, spitfire, etc.) does not. It does not error, just where the visual is expected is blank. import AppKit import PlaygroundSupport import AudioToolbox import AVFoundation import CoreAudioKit let manager = AVAudioUnitComponentManager.shared() let description = AudioComponentDescription(componentType: kAudioUnitType_MusicDevice, componentSubType: 0, componentManufacturer: 0, componentFlags: 0, componentFlagsMask: 0) var deviceComponents = manager.components(matching: description) var names = deviceComponents.map{$0.name} let pluginName: String = "AUSampler" // This works //let pluginName: String = "Kontakt" // This does not var plugin = deviceComponents.filter{$0.name.contains(pluginName)}.first! print("Plugin name: \(plugin.name)") var customViewController:NSViewController? AVAudioUnit.instantiate(with: plugin.audioComponentDescription, options: []){avAudioUnit, error in var ilip = avAudioUnit!.auAudioUnit.isLoadedInProcess print("Loaded in process: \(ilip)") guard error == nil else { print("Error: \(error!.localizedDescription)") return } print("AudioUnit successfully created.") let audioUnit = avAudioUnit!.auAudioUnit audioUnit.requestViewController{ vc in if let viewCtrl = vc { customViewController = vc var b = vc?.view.bounds PlaygroundPage.current.liveView = vc print("Successfully added view controller.") }else{ print("Failed to load controller.") } } }

I'm working with modern Core Audio API introduced in macOS Sequoia. I have an AudioHadwareDevice which has several controls of type AudioHardwareControl. I figured out to filter only volume controls I can use classID == kAudioVolumeControlClassID condition. Some devices have volume controls for both input and output. How I can determine the direction of the control? Streams, i.e. AudioHardwareStream object have direction, but I didn't found a way to map controls to streams. There are kAudioObjectPropertyScopeInput and kAudioObjectPropertyScopeOutput property scopes, but no matter what I tried controls always return false to any control.hasProperty(address: whatever). Any other ideas?

Hello! I have a problem with getting album extended info from users library. Note that app authorised to use Apple Music according documentation. I get albums from users library with this code: func getLibraryAlbums() async throws -> MusicItemCollection<Album> { let request = MusicLibraryRequest<Album>() let response = try await request.response() return response.items } This is an example of Albums request respones: { "data" : [ { "meta" : { "musicKit_identifierSet" : { "isLibrary" : true, "id" : "1945382328890400383", "dataSources" : [ "localLibrary", "legacyModel" ], "type" : "Album", "deviceLocalID" : { "databaseID" : "37336CB19CF51727", "value" : "1945382328890400383" }, "catalogID" : { "kind" : "adamID", "value" : "1173535954" } } }, "id" : "1945382328890400383", "type" : "library-albums", "attributes" : { "artwork" : { "url" : "musicKit:\/\/artwork\/transient\/{w}x{h}?id=4A2F444C%2D336D%2D49EA%2D90C8%2D13C547A5B95B", "width" : 0, "height" : 0 }, "genreNames" : [ "Pop" ], "trackCount" : 1, "artistName" : "Сара Окс", "isAppleDigitalMaster" : false, "audioVariants" : [ "lossless" ], "playParams" : { "catalogId" : "1173535954", "id" : "1945382328890400383", "musicKit_persistentID" : "1945382328890400383", "kind" : "album", "musicKit_databaseID" : "37336CB19CF51727", "isLibrary" : true }, "name" : "Нимфомания - Single", "isCompilation" : false } }, { "meta" : { "musicKit_identifierSet" : { "isLibrary" : true, "id" : "-8570883332059662437", "dataSources" : [ "localLibrary", "legacyModel" ], "type" : "Album", "deviceLocalID" : { "value" : "-8570883332059662437", "databaseID" : "37336CB19CF51727" }, "catalogID" : { "kind" : "adamID", "value" : "1618488499" } } }, "id" : "-8570883332059662437", "type" : "library-albums", "attributes" : { "isCompilation" : false, "genreNames" : [ "Pop" ], "trackCount" : 1, "artistName" : "TIMOFEEW & KURYANOVA", "isAppleDigitalMaster" : false, "audioVariants" : [ "lossless" ], "playParams" : { "catalogId" : "1618488499", "musicKit_persistentID" : "-8570883332059662437", "kind" : "album", "id" : "-8570883332059662437", "musicKit_databaseID" : "37336CB19CF51727", "isLibrary" : true }, "artwork" : { "url" : "musicKit:\/\/artwork\/transient\/{w}x{h}?id=BEA6DBD3%2D8E14%2D4A10%2D97BE%2D8908C7C5FC2C", "width" : 0, "height" : 0 }, "name" : "Не звони - Single" } }, ... ] } In AlbumView using task: view modifier I request extended information about the album with this code: func loadExtendedInfo(_ album: Album) async throws -> Album { let response = try await album.with([.tracks, .audioVariants, .recordLabels], preferredSource: .library) return response } but in the response some of the fields are always nil, for example recordLabels, releaseDate, url, editorialNotes, copyright. Please tell me what I'm doing wrong?

Getting MatchError "MATCH_ATTEMPT_FAILED" everytime when matchstream on Android Studio Java+Kotlin project. My project reads the samples from the mic input using audioRecord class and sents them to the Shazamkit to matchstream. I created a kotlin class to handle to Shazamkit. The audioRecord is build to be mono and 16 bit. My Kotlin Class class ShazamKitHelper { val shazamScope = CoroutineScope(Dispatchers.IO + SupervisorJob()) lateinit var streaming_session: StreamingSession lateinit var signature: Signature lateinit var catalog: ShazamCatalog fun createStreamingSessionAsync(developerTokenProvider: DeveloperTokenProvider, readBufferSize: Int, sampleRate: AudioSampleRateInHz ): CompletableFuture<Unit>{ return CompletableFuture.supplyAsync { runBlocking { runCatching { shazamScope.launch { createStreamingSession(developerTokenProvider,readBufferSize,sampleRate) }.join() }.onFailure { throwable -> }.getOrThrow() } } } private suspend fun createStreamingSession(developerTokenProvider:DeveloperTokenProvider,readBufferSize: Int,sampleRateInHz: AudioSampleRateInHz) { catalog = ShazamKit.createShazamCatalog(developerTokenProvider) streaming_session = (ShazamKit.createStreamingSession( catalog, sampleRateInHz, readBufferSize ) as ShazamKitResult.Success).data } fun startMatching() { val audioData = sharedAudioData ?: return // Return if sharedAudioData is null CoroutineScope(Dispatchers.IO).launch { runCatching { streaming_session.matchStream(audioData.data, audioData.meaningfulLengthInBytes, audioData.timestampInMs) }.onFailure { throwable -> Log.e("ShazamKitHelper", "Error during matchStream", throwable) } } } @JvmField var sharedAudioData: AudioData? = null; data class AudioData(val data: ByteArray, val meaningfulLengthInBytes: Int, val timestampInMs: Long) fun startListeningForMatches() { CoroutineScope(Dispatchers.IO).launch { streaming_session.recognitionResults().collect { matchResult -> when (matchResult) { is MatchResult.Match -> { val match = matchResult.matchedMediaItems println("Match found: ${match.get(0).title} by ${match.get(0).artist}") } is MatchResult.NoMatch -> { println("No match found") } is MatchResult.Error -> { val error = matchResult.exception println("Match error: ${error.message}") } } } } } } My code in java reads the samples from a thread: shazam_create_session(); while (audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING){ if (shazam_session_created){ byte[] buffer = new byte[288000];//max_shazam_seconds * sampleRate * 2]; audioRecord.read(buffer,0,buffer.length,AudioRecord.READ_BLOCKING); helper.sharedAudioData = new ShazamKitHelper.AudioData(buffer,buffer.length,System.currentTimeMillis()); helper.startMatching(); if (!listener_called){ listener_called = true; helper.startListeningForMatches(); } } else{ SystemClock.sleep(100); } } private void shazam_create_session() { MyDeveloperTokenProvider provider = new MyDeveloperTokenProvider(); AudioSampleRateInHz sample_rate = AudioSampleRateInHz.SAMPLE_RATE_48000; if (sampleRate == 44100) sample_rate = AudioSampleRateInHz.SAMPLE_RATE_44100; CompletableFuture<Unit> future = helper.createStreamingSessionAsync(provider, 288000, sample_rate); future.thenAccept(result -> { shazam_session_created = true; }); future.exceptionally(throwable -> { Toast.makeText(mine, "Failure", Toast.LENGTH_SHORT).show(); return null; }); } I Implemented the developer token in java as follows public static class MyDeveloperTokenProvider implements DeveloperTokenProvider { DeveloperToken the_token = null; @NonNull @Override public DeveloperToken provideDeveloperToken() { if (the_token == null){ try { the_token = generateDeveloperToken(); return the_token; } catch (NoSuchAlgorithmException | InvalidKeySpecException e) { throw new RuntimeException(e); } } else{ return the_token; } } @NonNull private DeveloperToken generateDeveloperToken() throws NoSuchAlgorithmException, InvalidKeySpecException { PKCS8EncodedKeySpec priPKCS8 = new PKCS8EncodedKeySpec(Decoders.BASE64.decode(p8)); PrivateKey appleKey = KeyFactory.getInstance("EC").generatePrivate(priPKCS8); Instant now = Instant.now(); Instant expiration = now.plus(Duration.ofDays(90)); String jwt = Jwts.builder() .header().add("alg", "ES256").add("kid", keyId).and() .issuer(teamId) .issuedAt(Date.from(now)) .expiration(Date.from(expiration)) .signWith(appleKey) // Specify algorithm explicitly .compact(); return new DeveloperToken(jwt); } }

Does Phase support creating new sound events at runtime? Is that implemented in the plugin for Unity as well? Does Phase support Unity's addressable system, are they compatible?

I am trying to get access to raw audio samples from mic. I've written a simple example application that writes the values to a text file. Below is my sample application. All the input samples from the buffers connected to the input tap is zero. What am I doing wrong? I did add the Privacy - Microphone Usage Description key to my application target properties and I am allowing microphone access when the application launches. I do find it strange that I have to provide permission every time even though in Settings > Privacy, my application is listed as one of the applications allowed to access the microphone. class AudioRecorder { private let audioEngine = AVAudioEngine() private var fileHandle: FileHandle? func startRecording() { let inputNode = audioEngine.inputNode let audioFormat: AVAudioFormat #if os(iOS) let hardwareSampleRate = AVAudioSession.sharedInstance().sampleRate audioFormat = AVAudioFormat(standardFormatWithSampleRate: hardwareSampleRate, channels: 1)! #elseif os(macOS) audioFormat = inputNode.inputFormat(forBus: 0) // Use input node's current format #endif setupTextFile() inputNode.installTap(onBus: 0, bufferSize: 1024, format: audioFormat) { [weak self] buffer, _ in self!.processAudioBuffer(buffer: buffer) } do { try audioEngine.start() print("Recording started with format: \(audioFormat)") } catch { print("Failed to start audio engine: \(error.localizedDescription)") } } func stopRecording() { audioEngine.stop() audioEngine.inputNode.removeTap(onBus: 0) print("Recording stopped.") } private func setupTextFile() { let tempDir = FileManager.default.temporaryDirectory let textFileURL = tempDir.appendingPathComponent("audioData.txt") FileManager.default.createFile(atPath: textFileURL.path, contents: nil, attributes: nil) fileHandle = try? FileHandle(forWritingTo: textFileURL) } private func processAudioBuffer(buffer: AVAudioPCMBuffer) { guard let channelData = buffer.floatChannelData else { return } let channelSamples = channelData[0] let frameLength = Int(buffer.frameLength) var textData = "" var allZero = true for i in 0..<frameLength { let sample = channelSamples[i] if sample != 0 { allZero = false } textData += "\(sample)\n" } if allZero { print("Got \(frameLength) worth of audio data on \(buffer.stride) channels. All data is zero.") } else { print("Got \(frameLength) worth of audio data on \(buffer.stride) channels.") } // Write to file if let data = textData.data(using: .utf8) { fileHandle!.write(data) } } }

Hello all! I've been having this issue for a while, on my iPhone 12 Pro. The volume when listening to music, watching YouTube, TikTok, etc. It will randomly lower, but the actual audio slider won't it will still be at max volume but get very quiet. I've followed other instructions such as turn off audio awareness, and other settings but nothing seems to be working. And phone calls too Has anyone else had this issue and managed to fix it?

Hi everyone, I wanted to bring up a question about Core Audio and its potential for future updates or improvements, specifically regarding latency optimization. As someone who relies on Core Audio for real-time audio processing, any enhancements in this area would be incredibly beneficial for professionals in the industry. Does anyone know if Apple has shared any plans or updates regarding Core Audio’s performance, particularly for low-latency applications? I’d appreciate any insights or advice from the community! Thanks so much! Best, Michael

I'm experiencing audio issues while developing for visionOS when playing PCM data through AVAudioPlayerNode. Issue Description: Occasionally, the speaker produces loud popping sounds or distorted noise This occurs during PCM audio playback using AVAudioPlayerNode The issue is intermittent and doesn't happen every time Technical Details: Platform: visionOS Device: vision pro / simulator Audio Framework: AVFoundation Audio Node: AVAudioPlayerNode Audio Format: PCM I would appreciate any insights on: Common causes of audio distortion with AVAudioPlayerNode Recommended best practices for handling PCM playback in visionOS Potential configuration issues that might cause this behavior Has anyone encountered similar issues or found solutions? Any guidance would be greatly helpful. Thank you in advance!

private var audioEngine = AVAudioEngine() private var inputNode: AVAudioInputNode! func startAnalyzing() { inputNode = audioEngine.inputNode let recordingFormat = inputNode.outputFormat(forBus: 0) let hardwareSampleRate = recordingSession.sampleRate inputNode.removeTap(onBus: 0) if recordingFormat.sampleRate != hardwareSampleRate { print("。") let newFormat = AVAudioFormat(commonFormat: recordingFormat.commonFormat, sampleRate: hardwareSampleRate, channels: recordingFormat.channelCount, interleaved: recordingFormat.isInterleaved) inputNode.installTap(onBus: 0, bufferSize: 1024, format: newFormat) { buffer, time in self.processAudioBuffer(buffer, time: time) } } else { inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { buffer, time in self.processAudioBuffer(buffer, time: time) } } do { audioEngine.prepare() try audioEngine.start() } catch { print(": \(error)") } } I back the app to the background and then call startAnalyzing(), which reports an error and the background recording permissions are configured。 error： [10429:570139] [aurioc] AURemoteIO.cpp:1668 AUIOClient_StartIO failed (561145187) [10429:570139] [avae] AVAEInternal.h:109 [AVAudioEngineGraph.mm:1545:Start: (err = PerformCommand(*ioNode, kAUStartIO, NULL, 0)): error 561145187 Audio engine couldn't start. Is background boot not allowed？

I'm trying to load Music Kit on the server with solid js. I can confirm that my implementation has been sufficient to return authentication tokens and for MusicKit.isAuthorized to return true. My issue is that if I reload the page, it only succeeds intermittently (perhaps 25% of the time?). My question is - what is wrong with my implementation? Removing the async keyword ensures it loads every time but playing and queuing music no longer works. I'm currently assuming this is an SSR issue but the docs haven't explicitly specified this isn't possible. I have the following boilerplate: export default createHandler( () => ( <StartServer document={({ assets, children, scripts }) => { return ( <html lang="en"> <head> <meta name="apple-music-developer-token" content={authResult.token} /> <meta name="apple-music-app-name" content="app name" /> <meta name="apple-music-app-build" content="1978.4.1" /> {assets} <script src="https://js-cdn.music.apple.com/musickit/v3/musickit.js" async /> </head> <body> <div id="app">{children}</div> {scripts} </body> </html> ) }} /> )) When I first load my app, I'll encounter: musickit.js:13 Uncaught TypeError: Cannot read properties of undefined (reading 'node') at musickit.js:13:10194 at musickit.js:13:140 at musickit.js:13:209 The intermittence signals an issue relating to the async keyword. An expansion on this issue can be found here.