I am trying to get MIDI output from the AU Host demo app using the recent MIDI processor example. The processor works correctly in Logic Pro, but I cannot send MIDI from the AUv3 extension in standalone mode using the default host app to another program (e.g., Ableton). The MIDI manager, which is part of the standalone host app, works fine, and I can send MIDI using it directly—Ableton receives it without issues. I have already set the midiOutputNames in the extension, and the midiOutBlock is mapped. However, the MIDI data from the AUv3 extension does not reach Ableton in standalone mode. I suspect the issue is that midiOutBlock might never be called in the plugin, or perhaps an input to the plugin is missing, which prevents it from sending MIDI. I am currently using the default routing. I have modified the MIDI manager such that it works well as described above. Here is a part of my code for SimplePlayEngine.swift and my MIDIManager.swift for reference: @MainActor @Observable public class SimplePlayEngine { private let midiOutBlock: AUMIDIOutputEventBlock = { sampleTime, cable, length, data in return noErr } var scheduleMIDIEventListBlock: AUMIDIEventListBlock? = nil public init() { engine.attach(player) engine.prepare() setupMIDI() } private func setupMIDI() { if !MIDIManager.shared.setupPort(midiProtocol: MIDIProtocolID._2_0, receiveBlock: { [weak self] eventList, _ in if let scheduleMIDIEventListBlock = self?.scheduleMIDIEventListBlock { _ = scheduleMIDIEventListBlock(AUEventSampleTimeImmediate, 0, eventList) } }) { fatalError("Failed to setup Core MIDI") } } func initComponent(type: String, subType: String, manufacturer: String) async -> ViewController? { reset() guard let component = AVAudioUnit.findComponent(type: type, subType: subType, manufacturer: manufacturer) else { fatalError("Failed to find component with type: \(type), subtype: \(subType), manufacturer: \(manufacturer))" ) } do { let audioUnit = try await AVAudioUnit.instantiate( with: component.audioComponentDescription, options: AudioComponentInstantiationOptions.loadOutOfProcess) self.avAudioUnit = audioUnit self.connect(avAudioUnit: audioUnit) return await audioUnit.loadAudioUnitViewController() } catch { return nil } } private func startPlayingInternal() { guard let avAudioUnit = self.avAudioUnit else { return } setSessionActive(true) if avAudioUnit.wantsAudioInput { scheduleEffectLoop() } let hardwareFormat = engine.outputNode.outputFormat(forBus: 0) engine.connect(engine.mainMixerNode, to: engine.outputNode, format: hardwareFormat) do { try engine.start() } catch { isPlaying = false fatalError("Could not start engine. error: \(error).") } if avAudioUnit.wantsAudioInput { player.play() } isPlaying = true } private func resetAudioLoop() { guard let avAudioUnit = self.avAudioUnit else { return } if avAudioUnit.wantsAudioInput { guard let format = file?.processingFormat else { fatalError("No AVAudioFile defined.") } engine.connect(player, to: engine.mainMixerNode, format: format) } } public func connect(avAudioUnit: AVAudioUnit?, completion: @escaping (() -> Void) = {}) { guard let avAudioUnit = self.avAudioUnit else { return } engine.disconnectNodeInput(engine.mainMixerNode) resetAudioLoop() engine.detach(avAudioUnit) func rewiringComplete() { scheduleMIDIEventListBlock = auAudioUnit.scheduleMIDIEventListBlock if isPlaying { player.play() } completion() } let hardwareFormat = engine.outputNode.outputFormat(forBus: 0) engine.connect(engine.mainMixerNode, to: engine.outputNode, format: hardwareFormat) if isPlaying { player.pause() } let auAudioUnit = avAudioUnit.auAudioUnit if !auAudioUnit.midiOutputNames.isEmpty { auAudioUnit.midiOutputEventBlock = midiOutBlock } engine.attach(avAudioUnit) if avAudioUnit.wantsAudioInput { engine.disconnectNodeInput(engine.mainMixerNode) if let format = file?.processingFormat { engine.connect(player, to: avAudioUnit, format: format) engine.connect(avAudioUnit, to: engine.mainMixerNode, format: format) } } else { let stereoFormat = AVAudioFormat(standardFormatWithSampleRate: hardwareFormat.sampleRate, channels: 2) engine.connect(avAudioUnit, to: engine.mainMixerNode, format: stereoFormat) } rewiringComplete() } } and my MIDI Manager @MainActor class MIDIManager: Identifiable, ObservableObject { func setupPort(midiProtocol: MIDIProtocolID, receiveBlock: @escaping @Sendable MIDIReceiveBlock) -> Bool { guard setupClient() else { return false } if MIDIInputPortCreateWithProtocol(client, portName, midiProtocol, &port, receiveBlock) != noErr { return false } for source in self.sources { if MIDIPortConnectSource(port, source, nil) != noErr { print("Failed to connect to source \(source)") return false } } setupVirtualMIDIOutput() return true } private func setupVirtualMIDIOutput() { let virtualStatus = MIDISourceCreate(client, virtualSourceName, &virtualSource) if virtualStatus != noErr { print("❌ Failed to create virtual MIDI source: \(virtualStatus)") } else { print("✅ Created virtual MIDI source: \(virtualSourceName)") } } func sendMIDIData(_ data: [UInt8]) { print("hey") var packetList = MIDIPacketList() withUnsafeMutablePointer(to: &packetList) { ptr in let pkt = MIDIPacketListInit(ptr) _ = MIDIPacketListAdd(ptr, 1024, pkt, 0, data.count, data) if virtualSource != 0 { let status = MIDIReceived(virtualSource, ptr) if status != noErr { print("❌ Failed to send MIDI data: \(status)") } else { print("✅ Sent MIDI data: \(data)") } } } } }

I found that the aggregated device correctly obtains input channels in the standard microphone mode. However, in voice isolation mode, it only retrieves channels from the first sub-device in the aggregated device's list. If I want to properly obtain channel information in voice isolation mode, how should I do it?

Hi all, with my app ScreenFloat, you can record your screen, along with system- and microphone audio. Those two audio feeds are recorded into separate audio tracks in order to individually remove or edit them later on. Now, these recordings you create with ScreenFloat can be drag-and-dropped to other apps instantly. So far, so good, but some apps, like Slack, or VLC, or even websites like YouTube, do not play back multiple audio tracks, just one. So what I'm trying to do is, on dragging the video recording file out of ScreenFloat, instantly baking together the two individual audio tracks into one, and offering that new file as the drag and drop file, so that all audio is played in the target app. But it's slow. I mean, it's actually quite fast, but for drag and drop, it's slow. My approach is this: "Bake together" the two audio tracks into a one-track m4a audio file using AVMutableAudioMix and AVAssetExportSession Take the video track, add the new audio file as an audio track to it, and render that out using AVAssetExportSession For a quick benchmark, a 3'40'' movie, step 1 takes ~1.7 seconds, and step two adds another ~1.5 seconds, so we're at ~3.2 seconds. That's an eternity for a drag and drop, where the user might cancel if there's no immediate feedback. I could also do it in one step, but then I couldn't use the AV*Passthrough preset, and that makes it take around 32 seconds then, because I assume it touches the video data (which is unnecessary in this case, so I think the two-step approach here is the fastest). So, my question is, is there a faster way? The best idea I can come up with right now is, when initially recording the screen with system- and microphone audio as separate tracks, to also record both of them into a third, muted, "hidden" track I could use later on, basically eliminating the need for step one and just ripping the two single audio tracks out of the movie and only have the video and the "hidden" track (then unmuted), but I'd still have a ~1.5 second delay there. Also, there's the processing and data overhead (basically doubling the movie's audio data). All this would be great for an export operation (where one expects it to take a little time), but for a drag-and-drop operation, it's not ideal. I've discarded the idea of doing a promise file drag, because many apps do not accept those, and I want to keep wide compatibility with all sorts of apps. I'd appreciate any ideas or pointers. Thank you kindly, Matthias

I ran 5.1 audio tests in both YouTube and Apple Music, and I noticed that when sound is supposed to play from the rear or front surround speakers, it’s also duplicated in the front left and right channels. I’m absolutely sure the issue is with the Apple TV, because I played the same video directly through my TV’s native system, and the channel separation was correct. Everything used to work perfectly before, so this must be a software issue. I’m currently on tvOS 26 Developer Beta 5, but I’m certain the problem also existed on the stable tvOS 18.5. I’ve already reset and updated my Apple TV, and I also tried switching the audio format to forced Dolby Atmos 5.1. On the forums, I mostly see complaints about Dolby Atmos not working at all — in my case, everything technically works, but not the way it’s supposed to.

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

I'm able to get text to speech to audio file using the following code for iOS 12 iPhone 8 to create a car file: audioFile = try AVAudioFile( forWriting: saveToURL, settings: pcmBuffer.format.settings, commonFormat: .pcmFormatInt16, interleaved: false) where pcmBuffer.format.settings is: [AVAudioFileTypeKey: kAudioFileMP3Type, AVSampleRateKey: 48000, AVEncoderBitRateKey: 128000, AVNumberOfChannelsKey: 2, AVFormatIDKey: kAudioFormatLinearPCM] However, this code does not work when I run the app in iOS 18 on iPhone 13 Pro Max. The audio file is created, but it doesn't sound right. It has a lot of static and it seems the speech is very low pitch. Can anyone give me a hint or an answer?

I have some tried-and-tested code that records and plays back audio via AUHAL which breaks on Tahoe on Intel. The same code works fine on Sequioa and also works on Tahoe on Apple Silicon. To start with something simple, the following code to request access to the Microphone doesn't work as it should: bool RequestMicrophoneAccess () { __block AVAuthorizationStatus status = [AVCaptureDevice authorizationStatusForMediaType: AVMediaTypeAudio]; if (status == AVAuthorizationStatusAuthorized) return true; __block bool done = false; [AVCaptureDevice requestAccessForMediaType: AVMediaTypeAudio completionHandler: ^ (BOOL granted) { status = (granted) ? AVAuthorizationStatusAuthorized : AVAuthorizationStatusDenied; done = true; }]; while (!done) CFRunLoopRunInMode (kCFRunLoopDefaultMode, 2.0, true); return status == AVAuthorizationStatusAuthorized; } On Tahoe on Intel, the code runs to completion but granted is always returned as NO. Tellingly, the popup to ask the user to grant microphone access is never displayed, even though the app is not present in the Privacy pane and never appears there. On Apple Silicon, everything works fine. There are some other problems, but I'm hoping they have a common underlying cause and that the Apple guys can figure out what's wrong from the information in this post. I'd be happy to test any potential fix. Thanks.

Not able to record audio in AAC format with 96 kHz sample rate using AVAudioRecorder or Extended Audio File services with 96 kHz input audio from input device. The audio recording settings used are let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatMPEG4AAC), AVSampleRateKey: sampleRate AVNumberOfChannelsKey: 1 AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue ] When tried using AVAudioEngine using AVAudioFile, AVAudioFile(forWriting: fileURL, // file extension .m4a settings: fileSettings, commonFormat: AVAudioCommonFormat.pcmFormatFloat32, interleaved: interleaved) else { return } got error CodecConverterFactory.cpp:977 unable to select compatible encoder sample rate AudioConverter.cpp:1017 Failed to create a new in process converter -> from 1 ch, 96000 Hz, Float32 to 1 ch, 96000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame, with status 1718449215

Hi, when a CoreMIDI driver controls physical HW it is probably quite commune to have to control the amount of MIDI data received from the system. What comes to mind is to just delay returning control of the MIDIDriverInterface::Send() callback to the calling process. While the application trying to send MIDI really stalls until the callback returns it seems only to be a side effect of a generally stalled CoreMIDI server. Between the callbacks the application can send as much MIDI data as it wants to CoreMIDI, it's buffering seems to be endless... However the HW might not be able to play out all the data. It seems there is no way to indicate an overflow/full buffer situation back the application/CoreMIDI. How is this supposed to work? Thanks, any hints or pointers are highly appreciated! Hagen.

I develop a application with an uvc camera, this camera is a webcam, I use the AVFoundation library ,but when I run the code "[self.mCaptureSession startRunning]" ,I can not get the buffer, I already set the delegate, any answer will help.

I'm encountering numerous crashes involving the com.apple.coreaudio.AQClient thread on our application. The crash details are as follows: #10 com.apple.coreaudio.AQClient SIGSEGV SEGV_ACCERR 0 libobjc.A.dylib _objc_msgSend + 44 1 AudioToolbox ClientMessageHandler::PropertyChanged(unsigned int) + 872 2 AudioToolbox ClientAudioQueue::FetchAndDeliverPendingCallbacks(unsigned int) + 924 3 AudioToolbox __XCallbackNotificationsAvailable + 212 4 libAudioToolboxUtility.dylib _mshMIGPerform + 260 5 CoreFoundation ___CFRUNLOOP_IS_CALLING_OUT_TO_A_SOURCE1_PERFORM_FUNCTION__ + 56 6 CoreFoundation ___CFRunLoopDoSource1 + 596 7 CoreFoundation ___CFRunLoopRun + 2392 8 CoreFoundation _CFRunLoopRunSpecific + 572 9 AudioToolbox CADeprecated::GenericRunLoopThread::Entry(void*) + 156 10 libAudioToolboxUtility.dylib CADeprecated::CAPThread::Entry(CADeprecated::CAPThread*) + 88 11 libsystem_pthread.dylib __pthread_start + 116 All these crashes occur on system versions below iOS/iPadOS 17, primarily when the device's available RAM is low. What steps can I take to resolve this issue? Any insights would be greatly appreciated!

On Apple TV 4K 3rd generation, with tvOS 26 beta 2, when two HomePod 2 are paired to the device, music and movie sources with Dolby Atmos can only be listened to in stereo. dolby atmos not supported

Hi there! We have a suite of AudioUnit v2 plugins that have been shipped for some time as aufx plugins, and we are looking into MIDI-related platform upgrades, so we need a way to update these plugins to request MIDI from Logic (and other AU hosts) but avoid changing our AU type and subtype so we don't break existing sessions. Any ideas on how we can do this?

Hi, macOS (latest macOS, latest HW, but doesn't matter) seems to prevent CoreMIDI driver logging with standard logging procedures (syslog, unified logging). The only chance to log something is writing to a file at one of the rare write-accessible locations for CoreMIDI. How is this supposed to work? Any hint is highly appreciated. Thanks!

I’m running the iOS 26.2 Public Beta update and my album artwork is missing from the music app (I’m not using Apple Music). I use google to get my album artwork. Do I need to wait for a new update?

Hi, I believe I've found a potential error in the sample code on the documentation page for creating and using a process tap with an aggregate device. The issue is in the section explaining how to add a tap to the aggregate device. I have already filed a Feedback Assistant ticket on this (ID: FB17411663) but haven't heard back for months. Capturing system audio with Core Audio taps The sample code for modifying the kAudioAggregateDevicePropertyTapList incorrectly uses the tapID as the target AudioObjectID when calling AudioObjectSetPropertyData. // (Code to get the list and potentially modify listAsArray) if var listAsArray = list as? [CFString] { // ... (modification logic) ... // Set the list back on the aggregate device. <--- The comment is correct list = listAsArray as CFArray _ = withUnsafeMutablePointer(to: &list) { list in // INCORRECT: This call uses tapID as the target object. AudioObjectSetPropertyData(tapID, &propertyAddress, 0, nil, propertySize, list) } } The kAudioAggregateDevicePropertyTapList is a property that belongs to the aggregate device, not the individual tap. Therefore, to set this property, the AudioObjectSetPropertyData function must target the AudioObjectID of the aggregate device itself. Using tapID as the first argument is logically incorrect for this operation and will not update the aggregate device as intended. Furthermore, the preceding AudioObjectGetPropertyData call to fetch the list also appears to use the incorrect tapID as its target in the sample. The AudioObjectID for both getting and setting this property should be the ID of the aggregate device. _ = AudioObjectGetPropertyData(aggregateDeviceID, &propertyAddress, 0, nil, &propertySize, &list) _ = AudioObjectSetPropertyData(aggregateDeviceID, &propertyAddress, 0, nil, propertySize, newList) Thank you!

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!

My audio app shows a control bar at the bottom of the window. The controls show nicely, but there is a black "slab" appearing behind the inline controls, the same size as the playerView. Setting the player view background color does nothing: playerView.wantsLayer = true playerView.layer?.backgroundColor = NSColor.clear.cgColor How can I clear the background? If I use .floating controlsStyle, I don't get the background "slab".

Hello everyone, I've written an audio unit plugin that needs to be aware of any upstream latency caused by heavy plugins before it on the channel. Is there any way to query this? I know that Logic applies PDC at the channel's output (summing point), but I need to know what the accumulated latency is at the point the audio enters my plugin. Thanks!

After upgrading to watchOS 26, users report that when playing music on Apple Watch, if a fitness reminder is received, the music automatically pauses and users need to manually tap the play button to resume music playback. This phenomenon occurs with multiple music and podcast apps. This issue did not exist before the upgrade. We would like to know if this is an Apple bug or if there are any special development configurations needed?"