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’m running HomePod OS 26 on two HomePod minis and OS 18.6 on main HomePod (original) I’ve enabled Crossfade in the Home app. I’m playing Apple Music directly in the HomePod mini. Crossfade just doesn’t work on any HomePod. I can understand it not working on the HomePod - but why isn’t it working on the minis running OS 26? I’ve tried disabling and enabling Crossfade, rebooting HomePods etc but nothing?!

I have an app that displays artwork via MPMediaItem.artwork, requesting an image with a specific size. How do I get a media item's MPMediaItemAnimatedArtwork, and how to get the preview image and video to display to the user?

My app encountered problems when trying to open an x86 audioUnit v2 on a Silicon Mac (although Rosetta is installed). There seems to be a XPC connection issue with the AUHostingService that I don't know how to fix. I observed other host apps opening the same plugins without problem, so there is probably something wrong or incompatible in my codes. I noticed that: The issue occurs whether or not the app is sandboxed. The issue does no longer occur when the app itself runs under Rosetta. There is no error reported by CoreAudio during allocation and initialization of the audio unit. The first notified errors appears when the unit calls AudioUnitRender from the rendering callback. With most x86 plugins, the error is on first call: kAudioUnitErr_RenderTimeout and on any subsequent call: kAudioComponentErr_InstanceInvalidated On the UI side, when the Cocoa View is loaded, it appears shortly, then disappears immediately leaving its superview empty. With another x86 plugin, the Cocoa View is loaded normally, but CoreAudio still emits kAudioUnitErr_NoConnection from AudioUnitRender, whether the view has been loaded or not, and the plugin produces no sound. I also find these messages in the console (printed in that order): CLIENT ERROR: RemoteAUv2ViewController does not override - and thus cannot react to catastrophic errors beyond logging them AUAudioUnit_XPC.mm:641 Crashed AU possible component description: aumu/Helm/Tyte My app uses the AUv2 API and I suspect that working with the AUv3 API would spare me these problems. However, considering how my audio system is built (audio units are wrapped into C++ classes and most connections between units are managed on the fly from the rendering callback), it would be a lot of work to convert, and I’m even not sure that all I do with the AUv2 API would be possible with the AUv3 API. I could possibly find an intermediate solution, but in the immediate future I'm looking for the simplest and fastest possible fix. If I cannot find better, I see two fallback options: In this part of the doc: “Beginning with macOS 11, the system loads audio units into a separate process that depends on the architecture or host preference”, does “host preference” means that it would be possible to disable the “out of process” behavior, for example from the app entitlements or info.plist? Otherwise, as a last resort, I could completely disable the use of x86 audioUnits when my app runs under ARM64, for at least making things cleaner. But the Audio Component API doesn’t give any info about the plugin architecture, how could I found it? Any tip or idea about this issue will be much appreciated. Thanks in advance!

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.

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.

Hi all, I've developed an audio DSP application in C++ using AudioToolbox and CoreAudio on MacOS 14.4.1 with Xcode 15. I use an AudioQueue for input and another for output. This works great. I'm now adding real-time audio analysis eg spectral analysis. I want this to run independently of my audio processing so it can not interfere with audio playback. Taps on AudioQueues seem to be a good way of doing this... Since the analytics won't modify the audio data, I am using a Siphon Tap by setting the AudioQueueProcessingTapFlags to kAudioQueueProcessingTap_PreEffects | kAudioQueueProcessingTap_Siphon; This works fine on my output queue. However, on my input queue the Tap callback is called once and then a EXC_BAD_ACCESS occurs - screen shot below. NB: I believe that a callback should only call AudioQueueProcessingTapGetSourceAudio when not using a Siphon, so I don't call it. Relevant code: AudioQueueProcessingTapCallback tap_callback) { // Makes an audio tap for a queue void * tap_data_ptr = NULL; AudioQueueProcessingTapFlags tap_flags = kAudioQueueProcessingTap_PostEffects | kAudioQueueProcessingTap_Siphon; uint32_t max_frames = 0; AudioStreamBasicDescription asbd; AudioQueueProcessingTapRef tap_ref; OSStatus status = AudioQueueProcessingTapNew(queue_ref, tap_callback, tap_data_ptr, tap_flags, &max_frames, &asbd, &tap_ref); if (status != noErr) printf("Error while making Tap\n"); else printf("Successfully made tap\n"); } void tapper(void * tap_data, AudioQueueProcessingTapRef tap_ref, uint32_t number_of_frames_in, AudioTimeStamp * ts_ptr, AudioQueueProcessingTapFlags * tap_flags_ptr, uint32_t * number_of_frames_out_ptr, AudioBufferList * buf_list) { // Callback function for audio queue tap printf("Tap callback"); }``` Image of exception stack provided by Xcode: ![]("https://developer.apple.com/forums/content/attachment/27479e8d-a118-459b-aa2d-7e30528910e3" "title=Screenshot 2025-06-14 at 1.29.14 PM.png;width=932;height=562") What have I missed? Appreciate any help you learned folks may be able to provide. Best, Geoff.

Hi all, I’ve implemented the new Core Audio Tap API (AudioHardwareCreateProcessTap with CATapDescription) and I’m seeing consistent level attenuation that scales with the number of stereo output pairs exposed by the target device. What I observe Device with 4 stereo pairs (8 outs) → tap shows −12.04 dB relative to source. True 2-ch devices (built-in speakers, AirPods) → ~0 dB attenuation. The attenuation appears regardless of whether I: Create a global (default-output) tap via initStereoGlobalTapButExcludeProcesses: Or create a per-process/per-device tap via initWithProcesses:andDeviceUID:withStream: Additionally, the routing choice inside the sending app matters: App output to “System/Default Output” → I often see no attenuation. App output directly to a multi-out interface (e.g., RME Fireface) → I see the pair-count-scaled attenuation. I can query Core Audio for the number of output channels/pairs and gain-compensate (+20·log10(N_pairs) dB) and that matches my measurements for many cases. However, this compensation is not universally correct because it seems to depend on where each process routes its audio (Default Output vs. direct device), even when those processes are included in the same tap aggregate. Question Is there a supported way to obtain the raw, unattenuated streams for all processes through the Tap API—i.e., to bypass this automatic headroom/attenuation behavior entirely? If this attenuation is expected by design: Is there a documented rule for when it applies (global vs. device taps, per-process taps, stream selection, etc.)? Is there a property/flag to disable it, or a reliable, official method to compute the exact compensation (beyond counting stereo pairs)? Any guidance on ensuring consistent levels when multiple processes route differently (Default Output vs. direct device) but are captured by the same tap? Environment API: AudioHardwareCreateProcessTap + CATapDescription Devices: built-in output (2-ch), RME Fireface (8+ outs / 4+ stereo pairs) Behavior reproducible with both global and per-process/per-device tap descriptions. Attenuation example: 4 stereo pairs → −12.04 dB observed. Happy to provide a minimal sample, measurements, and device logs. Thanks! — David

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

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've been trying to use AVMIDIControlChangeEvent with a bankSelect message type to change the instrument the sequencer uses on a AVMusicTrack with no luck. I started with the Apple AVAEMixerSample, converting the initial setup/loading and portions dealing with the sequencer to Swift. I got that working and playing the "bluesyRiff" and then modified it to play individual notes. So my createAndSetupSequencer looked like func createAndSetupSequencer() { sequencer = AVAudioSequencer(audioEngine: engine) // guard let midiFileURL = Bundle.main.url(forResource: "bluesyRiff", withExtension: "mid") else { // print (" failed guard trying to get URL for bluesyRiff") // return // } let track = sequencer.createAndAppendTrack() var currTime = 1.0 for i: UInt32 in 0...8 { let newNoteEvent = AVMIDINoteEvent(channel: 0, key: 60+i, velocity: 64, duration: 2.0) track.addEvent(newNoteEvent, at: AVMusicTimeStamp(currTime)) currTime += 2.0 } The notes played, so then I also replaced the gs_instruments sound bank with GeneralUser GS MuseScore v1.442 first by trying guard let soundBankURL = Bundle.main.url(forResource: "GeneralUser GS MuseScore v1.442", withExtension: "sf2") else { return} do { try sampler.loadSoundBankInstrument(at: soundBankURL, program: 0x001C, bankMSB: 0x79, bankLSB: 0x08) } catch{.... } This appears to work, the instrument (8 which is "Funk Guitar") plays. If I change to bankLSB: 0x00 I get the "Palm Muted guitar". So I know that the soundfont has these instruments Stuff goes off the rails when I try to change the instruments in createAndSetupSequencer. Putting let programChange = AVMIDIProgramChangeEvent(channel: 0, programNumber: 0x001C) let bankChange = AVMIDIControlChangeEvent(channel: 0, messageType: AVMIDIControlChangeEvent.MessageType.bankSelect, value: 0x00) track.addEvent(programChange, at: AVMusicTimeStamp(1.0)) track.addEvent(bankChange, at: AVMusicTimeStamp(1.0)) just before my add note loop doesn't produce any change. Loading bankLSB 8 (Funk) in sampler.loadSoundBankInstrument and trying to change with bankSelect 0 (Palm muted) in createAndSetupSequencer results in instrument 8 (Funk) playing not Palm Muted. Loading bankLSB 0 (Palm muted) and trying to change with bankSelect 8 (Funk) doesn't work, 0 (Palm muted) plays I also tried sampler.loadInstrument(at: soundBankURL) and then I always get the first instrument in the sound font file (piano)no matter what values I put in my programChange/bankChange I've also changed the time in the track.addEvent to be 0, 1.0, 3.0 etc to no success The sampler.loadSoundBankInstrument specifies two UInt8 parameters, bankMSB and BankLSB while the AVMIDIControlChangeEvent bankSelect value is UInt32 suggesting it might be some combination of bankMSB and BankLSB. But the documentation makes no mention of what this should look like. I tried various combinations of 0x7908, 0X0879 etc to no avail I will also point out that I am able to successfully execute other control change events For example adding if i == 1 { let portamentoOnEvent = AVMIDIControlChangeEvent(channel: 0, messageType: AVMIDIControlChangeEvent.MessageType.portamento, value: 0xFF) track.addEvent(portamentoOnEvent, at: AVMusicTimeStamp(currTime)) let portamentoRateEvent = AVMIDIControlChangeEvent(channel: 0, messageType: AVMIDIControlChangeEvent.MessageType.portamentoTime, value: 64) track.addEvent(portamentoRateEvent, at: AVMusicTimeStamp(currTime)) } does produce a change in the sound. (As an aside, a definition of what portamento time is, other than "the rate of portamento" would be welcome. is it notes/seconds? freq/minute? beats/hour?) I was able to get the instrument to change in a different program using MusicPlayer and a series of MusicTrackNewMIDIChannelEvent on a track but these operate on a MusicTrack not the AVMusicTrack which the sequencer uses. Has anyone been successful in switching instruments through an AVMIDIControlChangeEvent or have any feedback on how to do this?

Hi, I'm working on a project that uses the AVSpeechSynthesizer and AVSpeechUtterance. I discovered by chance that the AVSpeechSynthesizer automatically completes some words instead of just outputting what it's supposed to. These are abbreviations for days of the week or months, but not all of them. I don't want either of them automatically completed, but only the specified text. The completion transcends languages. I have written a short example program for demonstration purposes. import SwiftUI import AVFoundation import Foundation let synthesizer: AVSpeechSynthesizer = AVSpeechSynthesizer() struct ContentView: View { var body: some View { VStack { Button { utter("mon") } label: { Text("mon") } .buttonStyle(.borderedProminent) Button { utter("tue") } label: { Text("tue") } .buttonStyle(.borderedProminent) Button { utter("thu") } label: { Text("thu") } .buttonStyle(.borderedProminent) Button { utter("feb") } label: { Text("feb") } .buttonStyle(.borderedProminent) Button { utter("feb", lang: "de-DE") } label: { Text("feb DE") } .buttonStyle(.borderedProminent) Button { utter("wed") } label: { Text("wed") } .buttonStyle(.borderedProminent) } .padding() } private func utter(_ text: String, lang: String = "en-US") { let utterance = AVSpeechUtterance(string: text) let voice = AVSpeechSynthesisVoice(language: lang) utterance.voice = voice synthesizer.speak(utterance) } } #Preview { ContentView() } Thank you Christian

Is it possible to play WebM audio on iOS? Either with AVPlayer, AVAudioEngine, or some other API? Safari has supported this for a few releases now, and I'm wondering if I missed something about how to do this. By default these APIs don't seem to work (nor does ExtAudioFileOpen). Our usecase is making it possible for iOS users to play back audio recorded in our webapp (desktop versions of Chrome & Firefox only support webm as a destination format for MediaRecorder)

After updating to iOS 18.5, we’ve observed that outgoing audio from our app intermittently stops being transmitted during VoIP calls using AVAudioSession configured with .playAndRecord and .voiceChat. The session is set active without errors, and interruptions are handled correctly, yet audio capture suddenly ceases mid-call. This was not observed in earlier iOS versions (≤ 18.4). We’d like to confirm if there have been any recent changes in AVAudioSession, CallKit, or related media handling that could affect audio input behavior during long-running calls. func configureForVoIPCall() throws { try setCategory( .playAndRecord, mode: .voiceChat, options: [.allowBluetooth, .allowBluetoothA2DP, .defaultToSpeaker]) try setActive(true) }

The device is connected to Bluetooth A and Bluetooth B, currently the audio is played through Bluetooth A, click the interface button, how to realize the code to switch to Bluetooth B？

Environment Device: iPhone 16e iOS Version: 18.4.1 - 18.7.1 Framework: AVFoundation (AVAudioEngine) Problem Summary On iPhone 16e (iOS 18.4.1-18.7.1), the installTap callback stops being invoked after resuming from a phone call interruption. This issue is specific to phone call interruptions and does not occur on iPhone 14, iPhone SE 3, or earlier devices. Expected Behavior After a phone call interruption ends and audioEngine.start() is called, the previously installed tap should continue receiving audio buffers. Actual Behavior After resuming from phone call interruption: Tap callback is no longer invoked No audio data is captured No errors are thrown Engine appears to be running normally Note: Normal pause/resume (without phone call interruption) works correctly. Steps to Reproduce Start audio recording on iPhone 16e Receive or make a phone call (triggers AVAudioSession interruption) End the phone call Resume recording with audioEngine.start() Result: Tap callback is not invoked Tested devices: iPhone 16e (iOS 18.4.1-18.7.1): Issue reproduces ✗ iPhone 14 (iOS 18.x): Works correctly ✓ iPhone SE 3 (iOS 18.x): Works correctly ✓ Code Initial Setup (Works) let inputNode = audioEngine.inputNode inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in self.processAudioBuffer(buffer, at: time) } audioEngine.prepare() try audioEngine.start() Interruption Handling NotificationCenter.default.addObserver( forName: AVAudioSession.interruptionNotification, object: AVAudioSession.sharedInstance(), queue: nil ) { notification in guard let userInfo = notification.userInfo, let typeValue = userInfo[AVAudioSessionInterruptionTypeKey] as? UInt, let type = AVAudioSession.InterruptionType(rawValue: typeValue) else { return } if type == .began { self.audioEngine.pause() } else if type == .ended { try? self.audioSession.setActive(true) try? self.audioEngine.start() // Tap callback doesn't work after this on iPhone 16e } } Workaround Full engine restart is required on iPhone 16e: func resumeAfterInterruption() { audioEngine.stop() inputNode.removeTap(onBus: 0) inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in self.processAudioBuffer(buffer, at: time) } audioEngine.prepare() try audioSession.setActive(true) try audioEngine.start() } This works but adds latency and complexity compared to simple resume. Questions Is this expected behavior on iPhone 16e? What is the recommended way to handle phone call interruptions? Why does this only affect iPhone 16e and not iPhone 14 or SE 3? Any guidance would be appreciated!

Since the last update to IOS 26.0 (23A5276f) the AirPods connect to my IPhone and the Audio is still running through the phone. They are shown in the Bluetooth Icon that they’re paired.

Hi guys, I am having issue in live-streaming audio from Bluetooth headset and playing it live on the iPhone speaker. I am able to redirect audio back to the headset but this is not what I want. The issue happens when I am trying to override output - the iPhone switches to speaker but also switches a microphone. This is example of the code: import AVFoundation class AudioRecorder { let player: AVAudioPlayerNode let engine:AVAudioEngine let audioSession:AVAudioSession let audioSessionOutput:AVAudioSession init() { self.player = AVAudioPlayerNode() self.engine = AVAudioEngine() self.audioSession = AVAudioSession.sharedInstance() self.audioSessionOutput = AVAudioSession() do { try self.audioSession.setCategory(AVAudioSession.Category.playAndRecord, options: [.defaultToSpeaker]) try self.audioSessionOutput.setCategory(AVAudioSession.Category.playAndRecord, options: [.allowBluetooth]) // enables Bluetooth HFP profile try self.audioSession.setMode(AVAudioSession.Mode.default) try self.audioSession.setActive(true) // try self.audioSession.overrideOutputAudioPort(.speaker) // doens't work } catch { print(error) } let input = self.engine.inputNode self.engine.attach(self.player) let bus = 0 let inputFormat = input.inputFormat(forBus: bus) self.engine.connect(self.player, to: engine.mainMixerNode, format: inputFormat) input.installTap(onBus: bus, bufferSize: 512, format: inputFormat) { (buffer, time) -> Void in self.player.scheduleBuffer(buffer) print(buffer) } } public func start() { try! self.engine.start() self.player.play() } public func stop() { self.player.stop() self.engine.stop() } } I am not sure if this is a bug or not. Can somebody point me into the right direction? I there a way to design a custom audio routing? I would also appreciate some good documentation besides AVFoundation docs.

Is there a way to permanently disable PHASE SDK logging? It seems to be a lot chattier than Apple's other SDKs. While developing a RealityKit app that uses AudioPlaybackController, I must manually hide the PHASE SDK log output several times each day so I can see my app's log messages. Thank you.

I'm streaming mp3 audio data using URLSession/AudioFileStream/AVAudioConverter and getting occasional silent buffers and glitches (little bleeps and whoops as opposed to clicks). The issues are present in an offline test, so this isn't an issue of underruns. Doing some buffering on the input coming from the URLSession (URLSessionDataTask) reduces the glitches/silent buffers to rather infrequent, but they do still happen occasionally. var bufferedData = Data() func parseBytes(data: Data) { bufferedData.append(data) // XXX: this buffering reduces glitching // to rather infrequent. But why? if bufferedData.count > 32768 { bufferedData.withUnsafeBytes { (bytes: UnsafeRawBufferPointer) in guard let baseAddress = bytes.baseAddress else { return } let result = AudioFileStreamParseBytes(audioStream!, UInt32(bufferedData.count), baseAddress, []) if result != noErr { print("❌ error parsing stream: \(result)") } } bufferedData = Data() } } No errors are returned by AudioFileStream or AVAudioConverter. func handlePackets(data: Data, packetDescriptions: [AudioStreamPacketDescription]) { guard let audioConverter else { return } var maxPacketSize: UInt32 = 0 for packetDescription in packetDescriptions { maxPacketSize = max(maxPacketSize, packetDescription.mDataByteSize) if packetDescription.mDataByteSize == 0 { print("EMPTY PACKET") } if Int(packetDescription.mStartOffset) + Int(packetDescription.mDataByteSize) > data.count { print("❌ Invalid packet: offset \(packetDescription.mStartOffset) + size \(packetDescription.mDataByteSize) > data.count \(data.count)") } } let bufferIn = AVAudioCompressedBuffer(format: inFormat!, packetCapacity: AVAudioPacketCount(packetDescriptions.count), maximumPacketSize: Int(maxPacketSize)) bufferIn.byteLength = UInt32(data.count) for i in 0 ..< Int(packetDescriptions.count) { bufferIn.packetDescriptions![i] = packetDescriptions[i] } bufferIn.packetCount = AVAudioPacketCount(packetDescriptions.count) _ = data.withUnsafeBytes { ptr in memcpy(bufferIn.data, ptr.baseAddress, data.count) } if verbose { print("handlePackets: \(data.count) bytes") } // Setup input provider closure var inputProvided = false let inputBlock: AVAudioConverterInputBlock = { packetCount, statusPtr in if !inputProvided { inputProvided = true statusPtr.pointee = .haveData return bufferIn } else { statusPtr.pointee = .noDataNow return nil } } // Loop until converter runs dry or is done while true { let bufferOut = AVAudioPCMBuffer(pcmFormat: outFormat, frameCapacity: 4096)! bufferOut.frameLength = 0 var error: NSError? let status = audioConverter.convert(to: bufferOut, error: &error, withInputFrom: inputBlock) switch status { case .haveData: if verbose { print("✅ convert returned haveData: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(haveData) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } case .inputRanDry: if verbose { print("🔁 convert returned inputRanDry: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(inputRanDry) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } return // wait for next handlePackets case .endOfStream: if verbose { print("✅ convert returned endOfStream") } return case .error: if verbose { print("❌ convert returned error") } if let error = error { print("error converting: \(error.localizedDescription)") } return @unknown default: fatalError() } } }