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.

I've got a web app built with MusicKit that displays a list of songs. I have player controls for play, pause, skip next, skip, previous, toggle shuffle and set repeat mode. All of these work by using music. The play button, when nothing is playing and nothing is in the queue, will enqueue all the tracks and start playing with the below, for example: await music.setQueue({ songs, startPlaying: true }); I've implemented a progress slider based on feedback from the "playbackProgressDidChange" listener. Now, how in the world can I set the volume? This seems like it should be simple, but I am at a complete loss here. The docs say: "The volume of audio playback, which is set directly on the HTMLMediaElement as the HTMLMediaElement.volume property. This value ranges between 0, which would be muting the audio, and 1, which would be the loudest possible." Given that all my controls work off the music instance, I don't understand how I can do that. In this video from WWDC 2022, music web components are touched on briefly. These are also documented very sparsely. The volume docs are here. For the life of me, I can't even get the volume web component to display in the UI. It appears that MusicKit Web is hobbled compared to the native implementation, but surely adjusting volume shouldn't be that hard right? I'd appreciate any insight on how to do this, including how to get web components to work (in a Next JS app). Thanks.

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 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!

I am having issues deploying my iOS app, that uses ShazamKit, to get working on a Mac with Apple silicon. When uploading the archive to App Store Connect I do get ITMS-90863: Macs with Apple silicon support issue - The app links with libraries that aren’t present in macOS: /usr/lib/swift/libswiftShazamKit.dylib Is ShazamKit not supported for iOS apps that can run on Macs with Apple silicon? Or is there something I should fix in my setup / deployment?

I'm encountering errors while using AVAudioEngine with voice processing enabled (setVoiceProcessingEnabled(true)) in scenarios where the input and output audio devices are not the same. This issue arises specifically with mismatched devices, preventing the application from functioning as expected. Works: Paired devices (e.g., MacBook Pro mic → MacBook Pro speakers) Fails: Mismatched devices (e.g., AirPods mic → MacBook Pro speakers) When using paired input and output devices: The setup works as expected. Example: MacBook Pro microphone → MacBook Pro speakers. When using mismatched devices: AVAudioEngine setup fails during aggregate device construction. Example: AirPods microphone → MacBook Pro speakers. Error logs indicate a channel count mismatch. Here are the partial logs. Due to the content limit, I cannot post the entire logs. AUVPAggregate.cpp:1000 client-side input and output formats do not match (err=-10875) AUVPAggregate.cpp:1036 err=-10875 AVAEInternal.h:109 [AVAudioEngineGraph.mm:1344:Initialize: (err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)): error -10875 AggregateDevice.mm:329 Failed expectation of constructed aggregate (312): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (312): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (336): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (336): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AudioHardware-mac-imp.cpp:3484 AudioDeviceSetProperty: no device with given ID AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (348): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (348): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) Is it possible to use voice processing with different input/output devices? If yes, are there any specific configurations required to handle mismatched devices? How can we resolve channel count mismatch errors during aggregate device construction? Are there settings or API adjustments to enforce compatibility between input/output devices? Are there any workarounds or alternative approaches to achieve voice processing functionality with mismatched devices? For instance, can we force an intermediate channel configuration or downmix input/output formats?

I’m currently developing an iOS metronome app using DispatchSourceTimer as the timer. The interval is set very small, around 50 milliseconds, and I’m using CFAbsoluteTimeGetCurrent to calculate the elapsed time to ensure the beat is played within a ±0.003-second margin. The problem is that once the app goes to the background, the timing becomes unstable—it slows down noticeably, then recovers after 1–2 seconds. When coming back to the foreground, it suddenly speeds up, and again, it takes 1–2 seconds to return to normal. It feels like the app is randomly “powering off” and then “overclocking.” It’s super frustrating. I’ve noticed that some metronome apps in the App Store have similar issues, but there’s one called “Professional Metronome” that’s rock solid with no such problems. What kind of magic are they using? Any experts out there who can help? Thanks in advance! P.S. I’ve already enabled background audio permissions. The professional metronome that has no issues: https://link.zhihu.com/?target=https%3A//apps.apple.com/cn/app/pro-metronome-%25E4%25B8%2593%25E4%25B8%259A%25E8%258A%2582%25E6%258B%258D%25E5%2599%25A8/id477960671

I am using an AVAudioPlayer to play a "tick" sound once per second in a SwiftUI app. When running the app on an iPhone 16 (18.2.1) the tick sounds increase in volume after a few seconds. This does not happen in the simulator nor on an iPhone SE 2020 (18.1.1).

Please include the line below in follow-up emails for this request. Case-ID: 11089799 When using AVSpeechUtterance and setting it to play in Mandarin, if Siri is set to Cantonese on iOS 18, it will be played in Cantonese. There is no such issue on iOS 17 and 16. 1.let utterance = AVSpeechUtterance(string: textView.text) let voice = AVSpeechSynthesisVoice(language: "zh-CN") utterance.voice = voice 2.In the phone settings, Siri is set to Cantonese

Please Update Andorid MusicKit，the version 1.1.2 will complied fail。the error msg：•SDKUriHandlerActivity>. Apps targeting Android 12 and higher are required to specify an explicit value for android:exported when the corres

Hi all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!

I recently got some plugins from Universal Audio, and have licensed them properly through both UA and iLok manager. Whenever I try to load up the plugins (specifically from UA) in GarageBand, it first says that "NSCreateObjectFileImageFromMemory-p47UEwps” because the developper can not be verified. After clicking either 'show in finder' or 'okay', it opens the plugin in a form without its GUI and showing that it is not licensed (even though it is). It also displays error code 100001. I have tried only some basic stuff to troubleshoot like restarting the DAW/my computer and reinstalling/relicensing the softwares. I don't know if the macOS version has anything to do with it but for some reason I just can't get it to work.