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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.

hi, i need to read wether the transport is playing or stopped but my current method that works for vst does not work for au. is there a lpx resource available for developers anywhere? if (auto* playHead = processor->getPlayHead()) { juce::AudioPlayHead::CurrentPositionInfo posInfo; if (playHead->getCurrentPosition(posInfo)) { bool isCurrentlyPlaying = posInfo.isPlaying; if (isCurrentlyPlaying != wasTransportPlaying) { if (isCurrentlyPlaying) { wasTransportPlaying = isCurrentlyPlaying; startAllTimers(); } else { wasTransportPlaying = isCurrentlyPlaying; stopAllTimers(); } } } } thanks :)

Hi folks - I'm having trouble finding specific documentation about Audio Unit MIDI plugins - as in MIDI -only. Any suggestions welcome as searches aren't returning much. (too niche? user error?)

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?

Hello there! Is there any list of voices that are always available on iOS/iPadOS devices? It seems that AVSpeechSynthesisVoice(identifier: "com.apple.voice.compact.en-US.Samantha") is always available on all devices. I thought that AVSpeechSynthesisVoice(identifier: "com.apple.ttsbundle.siri_Nicky_en-US_compact") and AVSpeechSynthesisVoice(identifier: "com.apple.ttsbundle.siri_Aaron_en-US_compact") were available by default on certain newer devices. Is this true? I also noticed that on the same iPad where I was using those 2 voices (Nicky and Aaron) - when I updated to the iPadOS 26 beta, those voices were no longer available. Any information you can share about which voices should be reliably available on which devices would be extremely helpful for our development. Thanks so much!

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?

When multiple identical songs are added to a playlist, Playlist.Entry.id uses a suffix-based identifier (e.g. songID_0, songID_1, etc.). Removing one entry causes others to shift, changing their .id values. This leads to diffing errors and collection view crashes in SwiftUI or UIKit when entries are updated. Steps to Reproduce: Add the same song to a playlist multiple times. Observe .id.rawValue of entries (e.g. i.SONGID_0, i.SONGID_1). Remove one entry. Fetch playlist again — note the other IDs have shifted. FB18879062

Since iOS 18, the system setting “Allow Audio Playback” (enabled by default) allows third-party app audio to continue playing while the user is recording video with the Camera app. This has created a problem for the app I’m developing. ➡️ The problem: My app plays continuous audio in both foreground and background states. If the user starts recording video using the iOS Camera app, the app’s audio — still playing in the background — gets captured in the video — obviously an unintended behavior. Yes, the user could stop the app manually before starting the video recording, but that can’t be guaranteed. As a developer, I need a way to stop the app’s audio before the video recording begins. So far, I haven’t found a reliable way to detect when video recording starts if ‘Allow Audio Playback’ is ON. ➡️ What I’ve tried: — AVAudioSession.interruptionNotification → doesn’t fire — devicesChangedEventStream → not triggered I don’t want to request mic permission (app doesn’t use mic). also, disabling the app from playing audio in the background isn’t an option as it is a crucial part of the user experience ➡️ What I need: A reliable, supported way to detect when the Camera app begins video recording, without requiring mic access — so I can stop audio and avoid unintentional overlap with the user’s recordings. Any official guidance, workarounds, or AVFoundation techniques would be greatly appreciated. Thanks.

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 an amateur developer working on a free utility for composers/producers, for which the macOS release needs to create and name RTP-MIDI sessions in Audio MIDI Setup from the command line (so I can ship a small C helper instead of telling users to click through the UI). Here’s what I’ve tried so far, without luck: • Plist hacks: Injecting entries into ~/Library/Audio/MIDI Configurations/*.mcfg works when AMS is closed, but AMS immediately locks and reverts my changes when it’s open. • CoreMIDI C API: I can create virtual ports with MIDISourceCreate, but attempting MIDIObjectGetDataProperty on the apple.midirtp.session plugin always returns err –10836. • Obj-C & Swift: Loading MIDINetworkSession and calling defaultSession, init, setNetworkName: and setting enabled = YES doesn’t produce a new session object in the Network panel. • dlopen/dlsym: I extracted the real CoreMIDI binary out of the dyld shared cache and tried binding _MIDINetworkSessionCreate, _SetName, _SetEnabled, etc., but all the symbols come back null or my tool segfaults. • Plugin registration: I’ve pulled the factory UUID (70C9C5EA-7C65-11D8-B317-000393A34B5A) from /System/Library/Extensions/AppleMIDIRTPDriver.plugin/Contents/Info.plist and called CFPlugInRegisterFactories, but it still never exposes the session-creation calls. At this point I’m convinced I’m either loading the wrong binary or missing one critical step in registering the RTP-MIDI plugin’s private API. Can anyone point me to: The exact path of the dylib or bundle that actually exports the MIDINetworkSessionCreate/MIDINetworkSessionSetName/MIDINetworkSessionSetEnabled symbols? A minimal working snippet (C or Obj-C) that reliably creates and names a Network-MIDI session? Any pointers, sample code, or even ideas about where Apple hides this functionality on macOS 15 would be hugely appreciated. Thanks!

Hi, I’m an iOS developer building an app with an use case that needs advanced playback on Apple Music subscription streams, specifically: • Real-time tempo change (BPM) during playback — i.e., time-stretch with key-lock, not just crossfade. • Beat-matched transitions between tracks. From what I can tell, this capability seems to exist only for approved partners and isn’t available through public MusicKit. Question: What’s the official request path to be evaluated for that restricted partner entitlement (application form, questionnaire, NDA, or internal team/BD contact)? If the entitlement identifier is internal, how can I get my account routed to the right Apple Music team? For reference, publicly announced partners include Algoriddim djay, Serato DJ Pro, rekordbox (AlphaTheta), and Engine DJ—all of which appear to implement mixing features that imply advanced playback (tempo/beat-matching) on Apple Music content. I’d prefer not to share product details publicly for the moment and can provide specifics privately if needed. Thanks in advance!

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'm trying to write 16-bit interleaved 2-channel data captured from a LiveSwitch audio source to a AVAudioFile. The buffer and file formats match but I get a bad parameter error from the API. Does this API not support the specified format or is there some other issue? Here is the debugger output. (lldb) po audioFile.url ▿ file:///private/var/mobile/Containers/Data/Application/1EB14379-0CF2-41B6-B742-4C9A80728DB3/tmp/Heart%20Sounds%201 - _url : file:///private/var/mobile/Containers/Data/Application/1EB14379-0CF2-41B6-B742-4C9A80728DB3/tmp/Heart%20Sounds%201 - _parseInfo : nil - _baseParseInfo : nil (lldb) po error Error Domain=com.apple.coreaudio.avfaudio Code=-50 "(null)" UserInfo={failed call=ExtAudioFileWrite(_impl->_extAudioFile, buffer.frameLength, buffer.audioBufferList)} (lldb) po buffer.format <AVAudioFormat 0x302a12b20: 2 ch, 44100 Hz, Int16, interleaved> (lldb) po audioFile.fileFormat <AVAudioFormat 0x302a515e0: 2 ch, 44100 Hz, Int16, interleaved> (lldb) po buffer.frameLength 882 (lldb) po buffer.audioBufferList ▿ 0x0000000300941e60 - pointerValue : 12894608992 This code handles the details of converting the Live Switch frame into an AVAudioPCMBuffer. extension FMLiveSwitchAudioFrame { func convertedToPCMBuffer() -> AVAudioPCMBuffer { Self.convertToAVAudioPCMBuffer(from: self)! } static func convertToAVAudioPCMBuffer(from frame: FMLiveSwitchAudioFrame) -> AVAudioPCMBuffer? { // Retrieve the audio buffer and format details from the FMLiveSwitchAudioFrame guard let buffer = frame.buffer(), let format = buffer.format() as? FMLiveSwitchAudioFormat else { return nil } // Extract PCM format details from FMLiveSwitchAudioFormat let sampleRate = Double(format.clockRate()) let channelCount = AVAudioChannelCount(format.channelCount()) // Determine bytes per sample based on bit depth let bitsPerSample = 16 let bytesPerSample = bitsPerSample / 8 let bytesPerFrame = bytesPerSample * Int(channelCount) let frameLength = AVAudioFrameCount(Int(buffer.dataBuffer().length()) / bytesPerFrame) // Create an AVAudioFormat from the FMLiveSwitchAudioFormat guard let avAudioFormat = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: sampleRate, channels: channelCount, interleaved: true) else { return nil } // Create an AudioBufferList to wrap the existing buffer let audioBufferList = UnsafeMutablePointer<AudioBufferList>.allocate(capacity: 1) audioBufferList.pointee.mNumberBuffers = 1 audioBufferList.pointee.mBuffers.mNumberChannels = channelCount audioBufferList.pointee.mBuffers.mDataByteSize = UInt32(buffer.dataBuffer().length()) audioBufferList.pointee.mBuffers.mData = buffer.dataBuffer().data().mutableBytes // Directly use LiveSwitch buffer // Transfer ownership of the buffer to AVAudioPCMBuffer let pcmBuffer = AVAudioPCMBuffer(pcmFormat: avAudioFormat, bufferListNoCopy: audioBufferList) /* { buffer in // Ensure the buffer is freed when AVAudioPCMBuffer is deallocated buffer.deallocate() // Only call this if LiveSwitch allows manual deallocation } */ pcmBuffer?.frameLength = frameLength return pcmBuffer } } This is the handler that is invoked with every frame in order to convert it for use with AVAudioFile and optionally update a scrolling signal display on the screen. private func onRaisedFrame(obj: Any!) -> Void { // Bail out early if no one is interested in the data. guard isMonitoring else { return } // Convert LS frame to AVAudioPCMBuffer (no-copy) let frame = obj as! FMLiveSwitchAudioFrame let buffer = frame.convertedToPCMBuffer() // Hand subscribers a reference to the buffer for rendering to display. bufferPublisher?.send(buffer) // If we have and output file, store the data there, as well. guard let audioFile = self.audioFile else { return } do { try audioFile.write(from: buffer) // FIXME: This call is throwing error -50 } catch { FMLiveSwitchLog.error(withMessage: "Failed to write buffer to audio file at \(audioFile.url): \(error)") self.audioFile = nil } } This is how the audio file is being setup. static var recordingFormat: AVAudioFormat = { AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44_100, channels: 2, interleaved: true)! }() let audioFile = try AVAudioFile(forWriting: outputURL, settings: Self.recordingFormat.settings)

Hi everyone, I’m trying to use AVAssetResourceLoaderDelegate to handle a live radio stream (e.g. Icecast/HTTP stream). My goal is to have access to the last 30 seconds of audio data during playback, so I can analyze it for specific audio patterns in near-real-time. I’ve implemented a custom resource loader that works fine for podcasts and static files, where the file size and content length are known. However, for infinite live streams, my current implementation stops receiving new loading requests after the first one is served. As a result, the playback either stalls or fails to continue. Has anyone successfully used AVAssetResourceLoaderDelegate with a continuous radio stream? Or maybe you can suggest betterapproach for buffering and analyzing live audio? Any tips, examples, or advice would be appreciated. Thanks!

I’m developing a macOS audio monitoring app using AVAudioEngine, and I’ve run into a critical issue on macOS 26 beta where AVFoundation fails to detect any input devices, and AVAudioEngine.start() throws the familiar error 10877. FB#: FB19024508 Strange Behavior: AVAudioEngine.inputNode shows no channels or input format on bus 0. AVAudioEngine.start() fails with -10877 (AudioUnit connection error). AVCaptureDevice.DiscoverySession returns zero audio devices. Microphone permission is granted (authorized), and the app is properly signed and sandboxed with com.apple.security.device.audio-input. However, CoreAudio HAL does detect all input/output devices: Using AudioObjectGetPropertyDataSize and AudioObjectGetPropertyData with kAudioHardwarePropertyDevices, I can enumerate 14+ devices, including AirPods, USB DACs, and BlackHole. This suggests the lower-level audio stack is functional. I have tried: Resetting CoreAudio with sudo killall coreaudiod Rebuilding and re-signing the app Clearing TCC with tccutil reset Microphone Running on Apple Silicon and testing Rosetta/native detection via sysctl.proc_translated Using a fallback mechanism that logs device info from HAL and rotates logs for submission via Feedback Assistant I have submitted logs and a reproducible test case via Feedback Assitant : FB#: FB19024508]

Hey folks, I'm running into an odd issue suddenly with an app that had a working MusicKit integration before. I'm using ApplicationMusicPlayer to play Apple Music albums and songs. I'm testing on a physical device, signed in to Apple ID, and with a valid subscription. Apple Music via the first-party app works entirely fine on this device. Attempting to play back any content at all gives the log: <ICUserIdentityStoreACAccountBackend: 0x1070bf3e0> Failed to initialize primary apple account, error=Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store} [ICUserIdentityStore] - initializing account histories with activeAccountDSID = nil, activeLockerAccountDSID = nil, timestamp = 14605951908 [ICUserIdentityStore] Failed to fetch local store account with error: Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store}. The album artwork, track names, etc, all appear in the control center playback controls, but the music doesn't play. Trying to trigger playback with control center just results in it skipping to the next track, which doesn't play either. This exact code used to work. I have the MusicKit service selected in Apple Connect. Since this isn't entitlement-based, I'm not sure how else to check that I'm set up correctly. I've tried deleting/reinstalling the app, restarting the device, cleaning/rebuilding, and deleting DerivedData, to no avail. Any help? Running Xcode 16.4 (16F6), testing on iOS 18.5 (22F76)

According to the documentation (https://developer.apple.com/documentation/avfoundation/avplayeritem/externalmetadata), AVPlayerItem should have an externalMetadata property. However it does not appear to be visible to my app. When I try, I get: Value of type 'AVPlayerItem' has no member 'externalMetadata' Documentation states iOS 12.2+; I am building with a minimum deployment target of iOS 18. Code snippet: import Foundation import AVFoundation /// ... in function ... // create metadata as described in https://developer.apple.com/videos/play/wwdc2022/110338 var title = AVMutableMetadataItem() title.identifier = .commonIdentifierAlbumName title.value = "My Title" as NSString? title.extendedLanguageTag = "und" var playerItem = await AVPlayerItem(asset: composition) playerItem.externalMetadata = [ title ]

Hello everyone, I’m new to Swift development and have been working on an audio module that plays a specific sound at regular intervals - similar to a workout timer that signals switching exercises every few minutes. Following AVFoundation documentation, I’m configuring my audio session like this: let session = AVAudioSession.sharedInstance() try session.setCategory( .playback, mode: .default, options: [.interruptSpokenAudioAndMixWithOthers, .duckOthers] ) self.engine.attach(self.player) self.engine.connect(self.player, to: self.engine.outputNode, format: self.audioFormat) try? session.setActive(true) When it’s time to play cues, I schedule playback on a DispatchQueue: // scheduleAudio uses DispatchQueue self.scheduleAudio(at: interval.start) { do { try audio.engine.start() audio.node.play() for sample in interval.samples { audio.node.scheduleBuffer(sample.buffer, at: AVAudioTime(hostTime: sample.hostTime)) } } catch { print("Audio activation failed: \(error)") } } This works perfectly in the foreground. But once the app goes into the background, the scheduled callback runs, yet the audio engine fails to start, resulting in an error with code 561015905. Interestingly, if the app is already playing audio before going to the background, the scheduled sounds continue to play as expected. I have added the required background audio mode to my Info plist file by including the key UIBackgroundModes with the value audio. Is there anything else I should configure? What is the best practice to play periodic audio when the app runs in the background? How do apps like turn-by-turn navigation handle continuous audio playback in the background? Any advice or pointers would be greatly appreciated!

Hi everyone! I’ve developed a location-based Audio AR app in Unity with FMOD &amp; Resonance Audio and AirPods Pro Head-Tracking to create a ubiquitous augmented soundscape experience. Think of it as an audio version of Pokémon Go, but with a more precise location requirement to ensure spatial audio is placed correctly. I want this experience to run in the background on iOS, but from what I’ve gathered, it seems Unity doesn’t support this well. So, I’m considering developing a Swift version instead. Since this is primarily for research purposes, privacy concerns are not a major issue in my case. However, I’ve come across some potential challenges: Real-time precise location updates – Can iOS provide fully instantaneous, high-accuracy location updates in the background? Continuous real-time data processing – Can an app continuously process spatial audio, head-tracking, and location data while running in the background? I’m not sure if newer iOS versions have improved in these areas or if there are workarounds to achieve this. Would this kind of experience be feasible to run in the background on iOS? Any insights or pointers would be greatly appreciated! I’m very new to iOS development, so apologies if this is a basic question. Thanks in advance!