hi, Is there an Audio Unit logo I can show on my website? I would love to show that my application is able to host Audio Unit plugins. regards, Joël

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?

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

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 wrote a Swift macOS app to control a PCI audio device. The code switches between the default output and input channels. As soon as I launch the Audio-Midi Setup utility, channel switching stops working. The driver properties allow switching, but the system doesn't respond. I have to delete the contents of /Library/Preferences/Audio and reset Core Audio. What am I missing? func setDefaultChannelsOutput() { guard let deviceID = getDeviceIDByName(deviceName: "PCI-424") else { return } let selectedIndex = DefaultChannelsOutput.indexOfSelectedItem if selectedIndex < 0 || selectedIndex >= 24 { return } let channel1 = UInt32(selectedIndex * 2 + 1) let channel2 = UInt32(selectedIndex * 2 + 2) var channels: [UInt32] = [channel1, channel2] var propertyAddress = AudioObjectPropertyAddress( mSelector: kAudioDevicePropertyPreferredChannelsForStereo, mScope: kAudioDevicePropertyScopeOutput, mElement: kAudioObjectPropertyElementWildcard ) let dataSize = UInt32(MemoryLayout<UInt32>.size * channels.count) let status = AudioObjectSetPropertyData(deviceID, &propertyAddress, 0, nil, dataSize, &channels) if status != noErr { print("Error setting default output channels: \(status)") } }

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？

I have used AVQueuePlayer in my music app to play sequence of audios from a remote server, this how I have defined things my player in my ViewModel Variables private var cancellables = Set() private let audioSession = AVAudioSession.sharedInstance() private var avQueuePlayer: AVQueuePlayer? @Published var playbackSpeed: Float = 1.0 before starting playback, I am making sure that audio session is set properly, the code snippet used for that is do { try audioSession.setCategory(.playback, mode: .default, options: []) try audioSession.setActive(true, options: []) } catch { return } and this is the function I am using to update playback speed func updatePlaybackSpeed(_ newSpeed: Float){ if newSpeed > 0.0, newSpeed <= 2.0{ playbackSpeed = newSpeed avQueuePlayer?.rate = newSpeed print("requested speed is (newSpeed) and actual speed is (String(describing: avQueuePlayer?.rate))") } } sometimes whatever speed is set, player seems to play at the same speed as it was set, e.g. Once I got "requested speed is 1.5 and actual speed is 1.5", and player also seemed to play at the speed of 1.5 but another time I got "requested speed is 2.0 and actual speed is 2.0", but player still seemed to play at the speed of 1.0 to observe changes in rate, I used this **private func observeRateChanges() { guard let avQueuePlayer = self.avQueuePlayer else { return } NotificationCenter.default.publisher(for: AVQueuePlayer.rateDidChangeNotification, object: avQueuePlayer) .compactMap { $0.userInfo?[AVPlayer.rateDidChangeReasonKey] as? AVPlayer.RateDidChangeReason } .sink { reason in switch reason { case .appBackgrounded: print("The app transitioned to the background.") case .audioSessionInterrupted: print("The system interrupts the app’s audio session.") case .setRateCalled: print("The app set the player’s rate.") case .setRateFailed: print("An attempt to change the player’s rate failed.") default: break } } .store(in: &cancellables) }** when rate was set properly, I got this "The app set the player’s rate." from the above function, but when it wasn't, I got this "An attempt to change the player’s rate failed.," now I am not able to understand why rate is not being set, and if it gave "requested speed is 2.0 and actual speed is 2.0" from updatePlaybackSpeed function, why does the player seems to play with the speed of 1.0?

Hello, Has anyone else experienced variations in the accuracy of the playbackTime value? After a few seconds of playback, the reported time adjusts by a fraction of a second, making it difficult to calculate the actual playbackTime of the audio. This can be recreated by playing a song in MusicKit, recording the start time of the audio, playing for at least 10-20 seconds, and then comparing the playbackTime value to one calculated using the start time of the audio. In my experience this jump occurs after about 10 seconds of playback. Any help would be appreciated. Thanks!

Using an iPhone Pro 12 running iOS 26.0.1, with AirPods Pro 3. Camera app does capture video with what seems to be "Studio Quality Recording". Am trying to replicate that SQR with my own Camera like app, and while I can pull audio in from the APP3 mic, and my video capture app is recording a 48,000Hz high-bitrate video, the audio still sounds non-SQR. I'm seeing bluetoothA2DP , bluetoothLE , bluetoothHFP as portType, and not sure if SQR depends on one of those? Is there sample code demonstrating a SQR capture? Nevermind video and camera, just audio even? Also, I don't understand what SQR is doing between the APP3 and the iPhone. What codec is that? What bitrate is that? If I capture video using Capture and inspect the audio stream I see mono 74.14 kbit/s MPEG-4 AAC, 48000 Hz. But I assume that's been recompressed and not really giving me any insight into the APP3 H2 transmission?

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.

I am trying to use AVAudioEngine for recording and playing for a voice chat kind of app, but when the speaker plays any audio while recording, the recording take the speaker audio as input. I want to filter that out. Are there any suggestions for the swift code

I have an app that records a health provider’s conversation with a patient. I am using Audio Queue Services for this. If a phone call comes in while recording, the doctor wants to be able to ignore the call and continue the conversation without touching the phone. If the doctor answers the call, that’s fine – I will stop the recording. I can detect when the call comes in and ends using CXCallObserver and AVAudioSession.interruptionNotification. Unfortunately, when a call comes in and before it is answered or dismissed, the audio is suppressed. After the call is dismissed, the audio continues to be suppressed. How can I continue to get audio from the mic as long as the user does not answer the phone call?

Hi, everyone, I downloaded the source code EditingSpatialAudioWithAnAudioMix.zip from https://developer.apple.com/documentation/Cinematic/editing-spatial-audio-with-an-audio-mix, when I carried out one of the actions named "process" in command line the program crashed!! Form the source code, I found that the value of componentType is set to kAudioUnitType_FormatConverter: // The actual `AudioUnit`. public var auAudioMix = AVAudioUnitEffect() init() { // Generate a component description for the audio unit. let componentDescription = AudioComponentDescription( componentType: kAudioUnitType_FormatConverter, componentSubType: kAudioUnitSubType_AUAudioMix, componentManufacturer: kAudioUnitManufacturer_Apple, componentFlags: 0, componentFlagsMask: 0) auAudioMix=AVAudioUnitEffect(audioComponentDescription: componentDescription) } But in the document from https://developer.apple.com/documentation/avfaudio/avaudiouniteffect/init(audiocomponentdescription:), it seems that componentType can not be set to kAudioUnitType_FormatConverter and : Has everyone encountered this problem?

I am developing a VOD playback app, but when I stream video to an external monitor connected via HDMI with Lightning on iOS 18 or later, the screen goes dark and I cannot confirm playback. The app I am developing does not detect the HDMI and display the Player separately, but simply mirrors the video. We have confirmed that the same phenomenon occurs with other services, but we were able to confirm playback with some services such as Apple TV. Please let us know if there are any other necessary settings such as video certificates required for video playback. We would also like to know if the problem occurs with iOS 18 or later.

Hello. I am attempting to display the music inside of my app in Now Playing. I've tried a few different methods and keep running into unknown issues. I'm new to Objective-C and Apple development so I'm at a loss of how to continue. Currently, I have an external call to viewDidLoad upon initialization. Then, when I'm ready to play the music, I call playMusic. I have it hardcoded to play an mp3 called "1". I believe I have all the signing set up as the music plays after I exit the app. However, there is nothing in Now Playing. There are no errors or issues that I can see while the app is running. This is the only file I have in Xcode relating to this feature. Please let me know where I'm going wrong or if there is another object I need to use! #import <Foundation/Foundation.h> #import <UIKit/UIKit.h> #import <MediaPlayer/MediaPlayer.h> #import <AVFoundation/AVFoundation.h> @interface ViewController : UIViewController <AVAudioPlayerDelegate> @property (nonatomic, strong) AVPlayer *player; @property (nonatomic, strong) MPRemoteCommandCenter *commandCenter; @property (nonatomic, strong) MPMusicPlayerController *controller; @property (nonatomic, strong) MPNowPlayingSession *nowPlayingSession; @end @implementation ViewController - (void)viewDidLoad { [super viewDidLoad]; NSLog(@"viewDidLoad started."); [self setupAudioSession]; [self initializePlayer]; [self createNowPlayingSession]; [self configureNowPlayingInfo]; NSLog(@"viewDidLoad completed."); } - (void)setupAudioSession { AVAudioSession *audioSession = [AVAudioSession sharedInstance]; NSError *setCategoryError = nil; if (![audioSession setCategory:AVAudioSessionCategoryPlayback error:&setCategoryError]) { NSLog(@"Error setting category: %@", [setCategoryError localizedDescription]); } else { NSLog(@"Audio session category set."); } NSError *activationError = nil; if (![audioSession setActive:YES error:&activationError]) { NSLog(@"Error activating audio session: %@", [activationError localizedDescription]); } else { NSLog(@"Audio session activated."); } } - (void)initializePlayer { NSString *soundFilePath = [NSString stringWithFormat:@"%@/base/game/%@",[[NSBundle mainBundle] resourcePath], @"bgm/1.mp3"]; if (!soundFilePath) { NSLog(@"Audio file not found."); return; } NSURL *soundFileURL = [NSURL fileURLWithPath:soundFilePath]; self.player = [AVPlayer playerWithURL:soundFileURL]; NSLog(@"Player initialized with URL: %@", soundFileURL); } - (void)createNowPlayingSession { self.nowPlayingSession = [[MPNowPlayingSession alloc] initWithPlayers:@[self.player]]; NSLog(@"Now Playing Session created with players: %@", self.nowPlayingSession.players); } - (void)configureNowPlayingInfo { MPNowPlayingInfoCenter *infoCenter = [MPNowPlayingInfoCenter defaultCenter]; CMTime duration = self.player.currentItem.duration; Float64 durationSeconds = CMTimeGetSeconds(duration); CMTime currentTime = self.player.currentTime; Float64 currentTimeSeconds = CMTimeGetSeconds(currentTime); NSDictionary *nowPlayingInfo = @{ MPMediaItemPropertyTitle: @"Example Title", MPMediaItemPropertyArtist: @"Example Artist", MPMediaItemPropertyPlaybackDuration: @(durationSeconds), MPNowPlayingInfoPropertyElapsedPlaybackTime: @(currentTimeSeconds), MPNowPlayingInfoPropertyPlaybackRate: @(self.player.rate) }; infoCenter.nowPlayingInfo = nowPlayingInfo; NSLog(@"Now Playing info configured: %@", nowPlayingInfo); } - (void)playMusic { [self.player play]; [self createNowPlayingSession]; [self configureNowPlayingInfo]; } - (void)pauseMusic { [self.player pause]; [self configureNowPlayingInfo]; } @end

ApplicationMusicPlayer is not available on watchOS but all other platforms. Is there a technical reason for that like battery life? Same goes for SystemMusicPlayer and MPMusicPlayerController. I already filed feedbacks for that.

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

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)

I’ve been researching how to achieve a recording playback effect in iOS similar to the hands-free calling effect in the system’s phone app. How can this be implemented? I tried using the voice chat recording method, but found that the volume of the speaker output is too low. How should this issue be addressed? I couldn’t find a suitable API. Could you provide me with some documentation or sample code? Thank you.

Your draft looks great! Here's a refined version with the iOS 17 comparison emphasized and slightly better flow: Hi Apple Engineers and fellow developers, I'm experiencing a critical regression with ShazamKit's background operation on iOS 18. ShazamKit's SHManagedSession stops identifying songs in the background after approximately 20 seconds on iOS 18, while the exact same code works perfectly on iOS 17. The behavior is consistent: the app works perfectly in the foreground, but when backgrounded or device is locked, it initially works for about 20 seconds then stops identifying new songs. The microphone indicator remains active suggesting audio access is maintained, but ShazamKit doesn't send identified songs in the background until you open the app again. Detection immediately resumes when bringing the app to foreground. My technical setup uses SHManagedSession for continuous matching with background modes properly configured in Info.plist including audio mode, and Background App Refresh enabled. I've tested this on physical devices running iOS 18.0 through 18.5 with the same results across all versions. The exact same code running on iOS 17 devices works flawlessly in the background. To reproduce: initialize SHManagedSession and start matching, begin song identification in foreground, background the app or lock device, play different songs which are initially detected for about 20 seconds, then after the timeout period new songs are no longer identified until you bring the app to foreground. This regression has impacted my production app as users who rely on continuous background music identification are experiencing a broken feature. I submitted this as Feedback ID FB15255903 last September with no solution so far. I've created a minimal demo project that reproduces this issue: https://github.com/tfmart/ShazamKitBackground Has anyone else experienced this ShazamKit background regression on iOS 18? Are there any known workarounds or alternative approaches? Given the time this issue has persisted, could we please get acknowledgment of this regression, expected timeline for a fix, or any recommended workarounds? Testing environment is Xcode 16.0+ on iOS 18.0-18.5 across multiple physical device models. Any guidance would be greatly appreciated.