I'm using AVAudioEngine to play AVAudioPCMBuffers. I'd like to synchronize some events with the playback. For example if the audio's frame position is >= some point && less than some point trigger some code. So I'm looking at - (void)installTapOnBus:(AVAudioNodeBus)bus bufferSize:(AVAudioFrameCount)bufferSize format:(AVAudioFormat * __nullable)format block:(AVAudioNodeTapBlock)tapBlock; Now I have frame positions calculated (predetermined before audio is scheduled I already made all necessary computations) . So I just need to fire code at certain points during playback: [playerNode installTapOnBus:bus bufferSize:bufferSize format:format block:^(AVAudioPCMBuffer * _Nonnull buffer, AVAudioTime * _Nonnull when) { //Inspect current audio here and fire... }]; [playerNode scheduleBuffer:fullbuffer atTime:startTime options:0 completionCallbackType:AVAudioPlayerNodeCompletionDataPlayedBack completionHandler:^(AVAudioPlayerNodeCompletionCallbackType callbackType) { // some code is here, not important to this question. }]; The problem I'm having is figuring out at what point in full buffer I'm at within the tap block. The tap block passes chunks (not the full audio buffer). I tried using the when parameter of the block to calculate the frame position relative to the entire audio but have be unsuccessful so far. I'm assuming the when parameter is relative to the buffer passed in the tap block (not my entire audio buffer I scheduled). Not installing a tap and just using a timer before scheduling my fullBuffer has given me good results but I'd rather avoid using a timer if possible and use sample time.

I'd like to find out: Can backgrounded apps record audio? In the past as I recall, I found that backgrounded apps were pretty restricted and couldn't do much of anything. However I'm not familiar with the current state of affairs. With iOS 15.8 and above, can backgrounded apps record audio if they've been given permission by the user to access the microphone? Thanks.

Description As of iOS 18, AVAudioSession.setPreferredIOBufferDuration ignores the requested buffer size when Sound Recognition or Vocal Shortcuts is enabled. This results in 1) much larger buffer sizes and 2) mismatched buffer sizes between input and output buffers, which causes ‘glitchy’ audio and increased latency. Additionally, when this issue occurs AVAudioSession.setPreferredIOBufferDuration continues to return ‘true’ and no error is produced. Steps to Reproduce: Enable Vocal Shortcuts on a device running iOS 18. Enable at least one shortcut (e.g. Control Center). Open or clone the example project (https://github.com/cwalo/SoundRecognitionBug) Build and install the example project Attach a headset and launch the application Observe console logs showing a requested buffer size of 0.005805 (256 samples @ 48k) an actual buffer size of 0.023220 (1104 samples @48k - this is regularly the resulting buffer size in all of our tests) Quit the app and detach the headset. Enable mutesOutput in AudioSystem.mm (to avoid feedback) Launch the application Observe Same result from step 4 Mismatched hardware buffer size of 1104 and recorded frame count of 1024 Mismatched playbackCount and recordCount Quit the app and disable vocal shortcuts Launch the app Observe IOBufferDuration matching the requested duration and matched buffer sizes (expected behavior) Expected results: Requested IOBufferDuration is respected or AVAudioSession returns false or error is produced Input and output buffer sizes match Device(s): iPhone 11 Pro, iPad Pro OS: iOS 18.0.1 Environment: Xcode 16.1 FB: FB15715421 Related to: https://forums.developer.apple.com/forums/thread/765477

In our app we have implemented a AVAssetResourceLoaderDelegate to handle encrypted downloaded files. We have it working on all iOS versions but we are seeing issues on iOS 15 (15.8.3) with large files (> 1 Gb). We have so far seen two cases where either the load method on the AVURLAsset fails early and throws an unknown error code or starts requesting more data than the device has available RAM. The CPU usage is almost always over 100%, even after pausing playback. The memory issue can happen even though the player has successfully started playback. When running this on devices running iOS 16 and above we set the isEntireLengthAvailableOnDemand to true on the AVAssetResourceLoadingContentInformationRequest. This seems to be key to solving the issue those devices that support it. If we set the property to false we see the same memory issue as on iOS 15. So we have a solution for iOS 16 and upwards but are at a loss for how to handle iOS 15. Is there something we have overlooked or is it in fact an issue with that iOS version?

Hi! I am creating a aumi AUv3 extension and I am trying to achieve simultaneous connections to multiple other avaudionodes. I would like to know it is possible to route the midi to different outputs inside the render process in the AUv3. I am using connectMIDI(_:to:format:eventListBlock:) to connect the output of the AUv3 to multiple AvAudioNodes. However, when I send midi out of the AUv3, it gets sent to all the AudioNodes connected to it. I can't seem to find any documentation on how to route the midi only to one of the connected nodes. Is this possible?

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.") } } }

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 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 } }

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 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?

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.

We are using a VoiceProcessingIO audio unit in our VoIP application on Mac. In certain scenarios, the AudioComponentInstanceNew call blocks for up to five seconds (at least two). We are using the following code to initialize the audio unit: OSStatus status; AudioComponentDescription desc; AudioComponent inputComponent; desc.componentType = kAudioUnitType_Output; desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO; desc.componentFlags = 0; desc.componentFlagsMask = 0; desc.componentManufacturer = kAudioUnitManufacturer_Apple; inputComponent = AudioComponentFindNext(NULL, &desc); status = AudioComponentInstanceNew(inputComponent, &unit); We are having the issue with current MacOS versions on a host of different Macs (x86 and x64 alike). It takes two to three seconds until AudioComponentInstanceNew returns. We also see the following errors in the log multiple times: AUVPAggregate.cpp:2560 AggInpStreamsChanged wait failed and those right after (which I don't know if they matter to this issue): KeystrokeSuppressorCore.cpp:44 ERROR: KeystrokeSuppressor initialization was unsuccessful. Invalid or no plist was provided. AU will be bypassed. vpStrategyManager.mm:486 Error code 2003332927 reported at GetPropertyInfo

After updating to 18.3 my iPad 7th generation has no sound and will not play videos

I have tried everything. The songs load unto the playlists and on searches, but when prompted to play, they just won't play. I have a wrapper since my main player (which carries the buttons for play/rewind/forward/etc.), is in Objc. // // ApplePlayerWrapper.swift // UniversallyMac // // Created by Dorian Mattar on 11/10/24. // import Foundation import MusicKit import MediaPlayer @objc public class MusicKitWrapper: NSObject { @objc public static let shared = MusicKitWrapper() private let player = ApplicationMusicPlayer.shared // Play the current track @objc public func play() { guard !player.queue.entries.isEmpty else { print("Queue is empty. Cannot start playback.") return } logPlayerState(message: "Before play") Task { do { try await player.prepareToPlay() try await player.play() print("Playback started successfully.") } catch { if let nsError = error as NSError? { print("NSError Code: \(nsError.code), Domain: \(nsError.domain)") } } logPlayerState(message: "After play") } } // Log the current player state @objc public func logPlayerState(message: String = "") { print("Player State - \(message):") print("Playback Status: \(player.state.playbackStatus)") print("Queue Count: \(player.queue.entries.count)") // Only log current track details if the player is playing if player.state.playbackStatus == .playing { if let currentEntry = player.queue.currentEntry { print("Current Track: \(currentEntry.title)") print("Current Position: \(player.playbackTime) seconds") print("Track Length: \(currentEntry.endTime ?? 0.0) seconds") } else { print("No current track.") } } else { print("No track is playing.") } print("----------") } // Debug the queue @objc public func debugQueue() { print("Debugging Queue:") for (index, entry) in player.queue.entries.enumerated() { print("\(index): \(entry.title)") } } // Ensure track availability in the queue public func queueTracks(_ tracks: [Track]) { Task { do { for track in tracks { // Validate Play Parameters guard let playParameters = track.playParameters else { print("Track \(track.title) has no Play Parameters.") continue } // Log the Play Parameters print("Track Title: \(track.title)") print("Play Parameters: \(playParameters)") print("Raw Values: \(track.id.rawValue)") // Ensure the ID is valid if track.id.rawValue.isEmpty { print("Track \(track.title) has an invalid or empty ID in Play Parameters.") continue } // Queue the track try await player.queue.insert(track, position: .afterCurrentEntry) print("Queued track: \(track.title)") } print("Tracks successfully added to the queue.") } catch { print("Error queuing tracks: \(error)") } debugQueue() } } // Clear the current queue @objc public func resetMusicPlayer() { Task { player.stop() player.queue.entries.removeAll() print("Queue cleared.") print("Apple Music player reset successfully.") } } } I opened an Apple Dev. ticket, but I'm trying here as well. Thanks!

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) } } }

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 recently installed a rear-view camera in my car, and ever since, I've been experiencing a frustrating issue with my CarPlay. After about 15 seconds of playing audio via Bluetooth, the sound stops coming out of the speakers, even though the song continues to run in the background. For context, my stereo system is an aftermarket unit that I installed to enable CarPlay functionality. Everything worked perfectly before adding the rear-view camera. Unfortunately, my unit does not have a port for a wired connection, so I can't test the audio using a cable. Has anyone experienced a similar issue? Could the camera installation be interfering with the Bluetooth or audio system somehow? Any advice or troubleshooting tips would be greatly appreciated!