Is anyone developing a way for users to control an iOS or PadOS device playing Apple Music to a DAC via USB to amp from another iOS or PadOS device wirelessly? Specifically, full control. Not Accessibility, not to Apple TV, not HomePods, not firmware downgraded Airport Expresses to a DAC or other hacks mentioned for the past decade this “connect” like feature has been desired by audiophiles seeking exclusive mode on a device with that (iOS/PadOS) but — control it while sitting on a couch or in a wheel chair across the room. Exclusive mode being the key feature iOS and PadOS offer that is desired with full or nearly full Apple Music control.

Hi all, i can successfully match music using shazamkit on Apple using SwiftUI, a simple app that let user to load an audio file and exctracts the relative match, while i am unable to match music using shamzamkit on Android. I am trying to make the same simple app but i cannot match music as i get MATCH_ATTEMPT_FAILED every time i try to. I don't know what i am doing wrong but the shazam part in the kotlin Android code is in this method : suspend fun processAudioFileInBackground( filePath: String, developerTokenProvider: DeveloperTokenProvider ) = withContext(Dispatchers.IO) { val bufferSize = 1024 * 1024 val audioFile = FileInputStream(filePath) val byteBuffer = ByteBuffer.allocate(bufferSize) byteBuffer.order(ByteOrder.LITTLE_ENDIAN) var bytesRead: Int while (audioFile.read(byteBuffer.array()).also { bytesRead = it } != -1) { val signatureGenerator = (ShazamKit.createSignatureGenerator(AudioSampleRateInHz.SAMPLE_RATE_44100) as ShazamKitResult.Success).data signatureGenerator.append(byteBuffer.array(), bytesRead, System.currentTimeMillis()) val signature = signatureGenerator.generateSignature() println("Signature: ${signature.durationInMs}") val catalog = ShazamKit.createShazamCatalog(developerTokenProvider, Locale.ENGLISH) val session = (ShazamKit.createSession(catalog) as ShazamKitResult.Success).data val matchResult = session.match(signature) println("MatchResult : $matchResult") setMatchResult(matchResult) byteBuffer.clear() } audioFile.close() } I noticed that changing Locale in catalog creation results in different result as i get NoMatch without exception. Can you please help me with this?

I noticed the following behavior with CallKit when receiving a VolP push notification: When the app is in the foreground and a CallKit incoming call banner appears, pressing the answer button directly causes the speaker indicator in the CallKit interface to turn on. However, the audio is not actually activated (the iPhone's orange microphone indicator does not light up). In the same foreground scenario, if I expand the CallKit banner before answering the call, the speaker indicator does not turn on, but the orange microphone indicator does light up, and audio works as expected. When the app is in the background or not running, the incoming call banner works as expected: I can answer the call directly without expanding the banner, and the speaker does not turn on automatically. The orange microphone indicator lights up as it should. Why is there a difference in behavior between answering directly from the banner versus expanding it first when the app is in the foreground? Is there a way to ensure consistent audio activation behavior across these scenarios? I tried reconfiguring the audio when answering a call, but an error occurred during setActive, preventing the configuration from succeeding. let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setActive(false) try audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: [.defaultToSpeaker]) try audioSession.setActive(true, options: []) } catch { print("Failed to activate audio session: \(error)") } action.fulfill() } Error Domain=NSOSStatusErrorDomain Code=561017449 "Session activation failed" UserInfo={NSLocalizedDescription=Session activation failed}

Hello, I'm fairly new to AVAudioEngine and I'm trying to connect 2 mono nodes as left/right input to a stereo node. I was successful in splitting the input audio to 2 mono nodes using AVAudioConnectionPoint and channelMap. But I can't figure out how to connect them back to a stereo node. I'll post the code I have so far. The use case for this is that I'm trying to process the left/right channels with separate audio units. Any ideas? let monoFormat = AVAudioFormat(standardFormatWithSampleRate: nativeFormat.sampleRate, channels: 1)! let leftInputMixer = AVAudioMixerNode() let rightInputMixer = AVAudioMixerNode() let leftOutputMixer = AVAudioMixerNode() let rightOutputMixer = AVAudioMixerNode() let channelMixer = AVAudioMixerNode() [leftInputMixer, rightInputMixer, leftOutputMixer, rightOutputMixer, channelMixer].forEach { engine.attach($0) } let leftConnectionR = AVAudioConnectionPoint(node: leftInputMixer, bus: 0) let rightConnectionR = AVAudioConnectionPoint(node: rightInputMixer, bus: 0) plugin.leftInputMixer = leftInputMixer plugin.rightInputMixer = rightInputMixer plugin.leftOutputMixer = leftOutputMixer plugin.rightOutputMixer = rightOutputMixer plugin.channelMixer = channelMixer leftInputMixer.auAudioUnit.channelMap = [0] rightInputMixer.auAudioUnit.channelMap = [1] engine.connect(previousNode, to: [leftConnectionR, rightConnectionR], fromBus: 0, format: monoFormat) // Process right channel, pass through left channel engine.connect(rightInputMixer, to: plugin.audioUnit, format: monoFormat) engine.connect(plugin.audioUnit, to: rightOutputMixer, format: monoFormat) engine.connect(leftInputMixer, to: leftOutputMixer, format: monoFormat) // Mix back to stereo? engine.connect(leftOutputMixer, to: channelMixer, format: stereoFormat) engine.connect(rightOutputMixer, to: channelMixer, format: stereoFormat)

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.") } } }

Hi, I am creating an app that can include videos or images in it's data. While @Attribute(.externalStorage) helps with images, with AVAssets I actually would like access to the URL behind that data. (as it would be stupid to load and then save the data again just to have a URL) One key component is to keep all of this clean enough so that I can use (private) CloudKit syncing with the resulting model. All the best Christoph

Hi, Not sure if this is the right forum to ask this question in, but could you please advise if I can use Apple Digital Masters logo (badge) in my iOS app that is playing music from Apple Music service?

I'm trying to get the item that's assigned to the currentEntry when playing any song which is currently coming up nil when the song is playing. Note currentEntry returns: MusicPlayer.Queue.Entry(id: "evn7UntpH::ncK1NN3HS", title: "SONG TITLE") I'm a bit stumped on the API usage. if the song is playing, how could the queue item be nil? if queueObserver == nil { queueObserver = ApplicationMusicPlayer.shared.queue.objectWillChange .sink { [weak self] in self?.handleNowPlayingChange() } } } private func handleNowPlayingChange() { if let currentItem = ApplicationMusicPlayer.shared.queue.currentEntry { if let song = currentItem.item { self.currentlyPlayingID = song.id self.objectWillChange.send() print("Song ID: \(song.id)") } else { print("NO ITEM: \(currentItem)") } } else { print("No Entries: \(ApplicationMusicPlayer.shared.queue.currentEntry)") } }

I'm trying to setup a listener for kAudioProcessPropertyIsRunningOutput but it's never triggered. I get calls for kAudioProcessPropertyIsRunning and kAudioProcessPropertyDevices but not for kAudioProcessPropertyIsRunningInput or kAudioProcessPropertyIsRunningOutput. class MyDelegate: PropertyListenerDelegate { func propertiesChanged(properties: [AudioObjectPropertyAddress]) { print(properties) } } var myDelegate = MyDelegate() var processes = try AudioHardwareSystem.shared.processes for process in processes { process.delegates += [myDelegate] try process.addListener(forProperties: [AudioObjectPropertyAddress(mSelector: kAudioPropertyWildcardPropertyID, mScope: kAudioObjectPropertyScopeWildcard, mElement: kAudioObjectPropertyElementWildcard)]) } Xcode 16.1 macOS 15.0.1

I’m experiencing an unusual audio issue with AirPods on macOS Sequoia while developing VoIP applications like Zoom and FaceTime. When AirPods are connected, the other party’s voice sometimes sounds unnaturally stretched (approximately twice as long). This problem can be temporarily fixed by switching the sound output settings from AirPods to speakers and then back to AirPods. From our analysis, the issue appears to be related to the sample rate provided by AudioObjectGetPropertyData. Here’s what we’ve observed: When the issue occurs, the AudioStreamBasicDescription.sampleRate for AirPods is reported as 48000. Under normal conditions, it’s reported as 24000. It seems like the system is mistakenly returning a sample rate that doesn’t match the AirPods’ actual settings, perhaps defaulting to a system speaker value. Once the output setting is toggled, the correct sampleRate (24000) is retrieved. This discrepancy causes our application to transmit the audio stream at 48000, leading to the distorted playback. Has anyone encountered a similar issue or knows how to resolve it?

I’m facing a problem while trying to achieve spatial audio effects in my iOS 18 app. I have tried several approaches to get good 3D audio, but the effect never felt good enough or it didn’t work at all. Also what mostly troubles me is I noticed that AirPods I have doesn’t recognize my app as one having spatial audio (in audio settings it shows "Spatial Audio Not Playing"). So i guess my app doesn't use spatial audio potential. First approach uses AVAudioEnviromentNode with AVAudioEngine. Chaining position of player as well as changing listener’s doesn’t seem to change anything in how audio plays. Here's simple how i initialize AVAudioEngine import Foundation import AVFoundation class AudioManager: ObservableObject { // important class variables var audioEngine: AVAudioEngine! var environmentNode: AVAudioEnvironmentNode! var playerNode: AVAudioPlayerNode! var audioFile: AVAudioFile? ... //Sound set up func setupAudio() { do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } audioEngine = AVAudioEngine() environmentNode = AVAudioEnvironmentNode() playerNode = AVAudioPlayerNode() audioEngine.attach(environmentNode) audioEngine.attach(playerNode) audioEngine.connect(playerNode, to: environmentNode, format: nil) audioEngine.connect(environmentNode, to: audioEngine.mainMixerNode, format: nil) environmentNode.listenerPosition = AVAudio3DPoint(x: 0, y: 0, z: 0) environmentNode.listenerAngularOrientation = AVAudio3DAngularOrientation(yaw: 0, pitch: 0, roll: 0) environmentNode.distanceAttenuationParameters.referenceDistance = 1.0 environmentNode.distanceAttenuationParameters.maximumDistance = 100.0 environmentNode.distanceAttenuationParameters.rolloffFactor = 2.0 // example.mp3 is mono sound guard let audioURL = Bundle.main.url(forResource: "example", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: audioURL) } catch { print("Failed to load audio file: \(error)") } } ... //Playing sound func playSpatialAudio(pan: Float ) { guard let audioFile = audioFile else { return } // left side playerNode.position = AVAudio3DPoint(x: pan, y: 0, z: 0) playerNode.scheduleFile(audioFile, at: nil, completionHandler: nil) do { try audioEngine.start() playerNode.play() } catch { print("Failed to start audio engine: \(error)") } ... } Second more complex approach using PHASE did better. I’ve made an exemplary app that allows players to move audio player in 3D space. I have added reverb, and sliders changing audio position up to 10 meters each direction from listener but audio seems to only really change left to right (x axis) - again I think it might be trouble with the app not being recognized as spatial. //Crucial class Variables: class PHASEAudioController: ObservableObject{ private var soundSourcePosition: simd_float4x4 = matrix_identity_float4x4 private var audioAsset: PHASESoundAsset! private let phaseEngine: PHASEEngine private let params = PHASEMixerParameters() private var soundSource: PHASESource private var phaseListener: PHASEListener! private var soundEventAsset: PHASESoundEventNodeAsset? // Initialization of PHASE init{ do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } // Init PHASE Engine phaseEngine = PHASEEngine(updateMode: .automatic) phaseEngine.defaultReverbPreset = .mediumHall phaseEngine.outputSpatializationMode = .automatic //nothing helps // Set listener position to (0,0,0) in World space let origin: simd_float4x4 = matrix_identity_float4x4 phaseListener = PHASEListener(engine: phaseEngine) phaseListener.transform = origin phaseListener.automaticHeadTrackingFlags = .orientation try! self.phaseEngine.rootObject.addChild(self.phaseListener) do{ try self.phaseEngine.start(); } catch { print("Could not start PHASE engine") } audioAsset = loadAudioAsset() // Create sound Source // Sphere soundSourcePosition.translate(z:3.0) let sphere = MDLMesh.newEllipsoid(withRadii: vector_float3(0.1,0.1,0.1), radialSegments: 14, verticalSegments: 14, geometryType: MDLGeometryType.triangles, inwardNormals: false, hemisphere: false, allocator: nil) let shape = PHASEShape(engine: phaseEngine, mesh: sphere) soundSource = PHASESource(engine: phaseEngine, shapes: [shape]) soundSource.transform = soundSourcePosition print(soundSourcePosition) do { try phaseEngine.rootObject.addChild(soundSource) } catch { print ("Failed to add a child object to the scene.") } let simpleModel = PHASEGeometricSpreadingDistanceModelParameters() simpleModel.rolloffFactor = rolloffFactor soundPipeline.distanceModelParameters = simpleModel let samplerNode = PHASESamplerNodeDefinition( soundAssetIdentifier: audioAsset.identifier, mixerDefinition: soundPipeline, identifier: audioAsset.identifier + "_SamplerNode") samplerNode.playbackMode = .looping do {soundEventAsset = try phaseEngine.assetRegistry.registerSoundEventAsset( rootNode: samplerNode, identifier: audioAsset.identifier + "_SoundEventAsset") } catch { print("Failed to register a sound event asset.") soundEventAsset = nil } } //Playing sound func playSound(){ // Fire new sound event with currently set properties guard let soundEventAsset else { return } params.addSpatialMixerParameters( identifier: soundPipeline.identifier, source: soundSource, listener: phaseListener) let soundEvent = try! PHASESoundEvent(engine: phaseEngine, assetIdentifier: soundEventAsset.identifier, mixerParameters: params) soundEvent.start(completion: nil) } ... } Also worth mentioning might be that I only own personal team account

Hello, Using ShazamKit, based on a shazam catalog result, would it be possible to detect the audio-recorded FPS (speed)? I'm thinking that the shazam catalog which was created from an audio file can be used to compare the speed of a live recorded audio. Thank you!

Dear Sirs, I'd like to add an icon to my audio driver based on AudioDriverKit. This icon should show up left of my audio device in the audio devices dialog. For an Audio Server Plugin I managed to do this using the property kAudioDevicePropertyIcon and CFBundleCopyResourceURL(...) but how would you do this with AudioDriverKit? Should I use IOUserAudioCustomProperty or IOUserAudioControl and how would I refer to the Bundle? Is there an example available somewhere? Thanks and best regards, Johannes

Hi all! I have been experiencing some issues when using the AVAudioEngine to play audio and record input while doing a voice chat (through the PTT Interface). I noticed if I connect any players to the AudioGraph OR call start that the audio session becomes active (this is on iOS). I don't see anything in the docs or the header files in the AVFoundation, but is it possible that calling the stop method on an engine deactivates the audio session too? In a normal app this behavior seems logical, but when using PTT all activation and deactivation of the audio session must go through the framework and its delegate methods. The issue I am debugging is that when the engine with the input node tapped gets stopped, and there is a gap between the input and when the server replies with inbound audio to be played and something seems to be getting the hardware/audio session into a jammed state. Thanks for any feedback and/or confirmation on this behavior!

Issue Description I'm implementing a system audio capture feature using AudioHardwareCreateProcessTap and AudioHardwareCreateAggregateDevice. The app successfully creates the tap and aggregate device, but when starting the IO procedure with AudioDeviceStart, it sometimes fails with OSStatus error 1852797029. (The operation couldn’t be completed. (OSStatus error 1852797029.)) The error occurs inconsistently, which makes it particularly difficult to debug and reproduce. Questions Has anyone encountered this intermittent "nope" error code (0x6e6f7065) when working with system audio capture? Are there specific conditions or system states that might trigger this error sporadically? Are there any known workarounds for handling this intermittent failure case? Any insights or guidance would be greatly appreciated. I'm wondering if anyone else has encountered this specific "nope" error code (0x6e6f7065) when working with system audio capture.

Hello, We are developing a real-time speech recognition application and are utilizing AVAudioEngine with voice processing enabled on the input node. However, we have observed that enabling this mode interferes with the built-in iOS screen recording feature - specifically, the recorded video does not capture any audio when this mode is active. Since we want users to be able to record their experience within our app, this issue significantly impacts our functionality. Is there a known workaround or recommended approach to ensure that both voice processing and screen recording can function simultaneously? Any guidance would be greatly appreciated. Thank you!

Hi everyone, I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms. Problem: When the app is recording audio and an interruption occurs: I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began). On .ended, I check for .shouldResume and call audioRecorder?.record() again. The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder. Repro: Start a recording with AVAudioRecorder Simulate a system interruption (e.g., incoming call) Resume recording after the interruption Stop and inspect the output audio file Expected: Full audio (before and after interruption) should be saved. Actual: Only the audio after interruption is saved; the earlier part is missing Notes: According to the documentation, calling .record() after .pause() should resume recording into the same file. I confirmed that the file URL does not change, and I do not recreate the recorder instance. No error is thrown by the system during this process. This behavior happens consistently when the app is interrupted and resumed. Question: Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen? Thanks in advance!

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?

Hi. I want to read ADPCM encoded audio data, coming from an external device to my Mac via serial port (/dev/cu.usbserial-0001) as 256 byte chunks, and feed it into an audio player. So far I am using Swift and SwiftSerial (GitHub - mredig/SwiftSerial: A Swift Linux and Mac library for reading and writing to serial ports. 3) to get the data via serialPort.asyncBytes() into a AsyncStream but I am struggling to understand how to feed the stream to a AVAudioPlayer or similar. I am new to Swift and macOS audio development so any help to get me on the right track is greatly appreciated. Thx

I'm capturing audio from other applications on macOS to mix them with other sources in a real time streaming application. I noticed that audio data captured via the new tapping mechanism introduced in macOS 14.2 arrives delayed in my app, when the macOS system device is a Bluetooth headphone, e.g. Apple AirPods. Sometimes this delay is about 300-400 milliseconds, which makes it unusable for live streaming, because the audio is out of sync with the video and also audio captured from other devices. What is confusing to me, is that this also happens when my app does not even use that output device. Is this a known issue? Is there a way around this?