After investing more than a week into getting a bunch of audio unit projects converted into app + appex + framework, they all are now correctly loaded in-process in the demo host app that is part of Xcode's template. However, Logic Pro adamantly refuses to load them in-process. Does Logic Pro simply not do that ever, or is there some hint or configuration my plugins need to provide to enable that? If it is unsupported, will it be supported in some future version of Logic? The entire point of investing that week was performance, which is moot if it is impossible to test the impact of loading in-process in a real-world usage scenario.

Hi community, I'm trying to setup an AVAudioFormat with AVAudioPCMFormatInt16. But, i've an error : AVAEInternal.h:125 [AUInterface.mm:539:SetFormat: ([[busArray objectAtIndexedSubscript:(NSUInteger)element] setFormat:format error:&nsErr])] returned false, error Error Domain=NSOSStatusErrorDomain Code=-10868 "(null)" If i understand the error code 10868, the format is not correct. But, how i can use PCM Int16 format ? Here is my method : - (void)setupAudioDecoder:(double)sampleRate audioChannels:(double)audioChannels { if (self.isRunning) { return; } self.audioEngine = [[AVAudioEngine alloc] init]; self.audioPlayerNode = [[AVAudioPlayerNode alloc] init]; [self.audioEngine attachNode:self.audioPlayerNode]; AVAudioChannelCount channelCount = (AVAudioChannelCount)audioChannels; self.audioFormat = [[AVAudioFormat alloc] initWithCommonFormat:AVAudioPCMFormatInt16 sampleRate:sampleRate channels:channelCount interleaved:YES]; NSLog(@"Audio Format: %@", self.audioFormat); NSLog(@"Audio Player Node: %@", self.audioPlayerNode); NSLog(@"Audio Engine: %@", self.audioEngine); // Error on this line [self.audioEngine connect:self.audioPlayerNode to:self.audioEngine.mainMixerNode format:self.audioFormat]; /**NSError *error = nil; if (![self.audioEngine startAndReturnError:&error]) { NSLog(@"Erreur lors de l'initialisation du moteur audio: %@", error); return; } [self.audioPlayerNode play]; self.isRunning = YES;*/ } Also, i see the audioEngine seem not running ? Audio Engine: ________ GraphDescription ________ AVAudioEngineGraph 0x600003d55fe0: initialized = 0, running = 0, number of nodes = 1 Anyone have already use this format with AVAudioFormat ? Thank you !

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

At Apple Developer documentation, https://developer.apple.com/documentation/avfoundation/avcapturedevice/discoverysession you can find the sentence You can also key-value observe this property to monitor changes to the list of available devices. But how to use it? I tried it with the code above and tested on my MacBook with EarPods. When I disconnect the EarPods, nothing was happened. MacBook Air M2 macOS Sequoia 15.0.1 Xcode 16.0 import Foundation import AVFoundation let discovery_session = AVCaptureDevice.DiscoverySession.init(deviceTypes: [.microphone], mediaType: .audio, position: .unspecified) let devices = discovery_session.devices for device in devices { print(device.localizedName) } let device = devices[0] let observer = Observer() discovery_session.addObserver(observer, forKeyPath: "devices", options: [.new, .old], context: nil) let input = try! AVCaptureDeviceInput(device: device) let queue = DispatchQueue(label: "queue") var output = AVCaptureAudioDataOutput() let delegate = OutputDelegate() output.setSampleBufferDelegate(delegate, queue: queue) var session = AVCaptureSession() session.beginConfiguration() session.addInput(input) session.addOutput(output) session.commitConfiguration() session.startRunning() let group = DispatchGroup() let q = DispatchQueue(label: "", attributes: .concurrent) q.async(group: group, execute: DispatchWorkItem() { sleep(10) session.stopRunning() }) _ = group.wait(timeout: .distantFuture) class Observer: NSObject { override func observeValue(forKeyPath keyPath: String?, of object: Any?, change: [NSKeyValueChangeKey : Any]?, context: UnsafeMutableRawPointer?) { print("Change") if keyPath == "devices" { if let newDevices = change?[.newKey] as? [AVCaptureDevice] { print("New devices: \(newDevices.map { $0.localizedName })") } if let oldDevices = change?[.oldKey] as? [AVCaptureDevice] { print("Old devices: \(oldDevices.map { $0.localizedName })") } } } } class OutputDelegate : NSObject, AVCaptureAudioDataOutputSampleBufferDelegate { func captureOutput(_ output: AVCaptureOutput, didOutput sampleBuffer: CMSampleBuffer, from connection: AVCaptureConnection) { print("Output") } }

With iOS 18.1 having call recording out of the box, is it now possible to build apps that can record calls? I could not find anything in the swift ios docs yet.

I'm seeking information about the original file schema for an m4a file recorded directly on an iPhone (iPhone 5 running iOS 9.2.0). I currently have two files from which I extracted metadata using ExifTool. The first file was provided to me by someone who claims it was recorded on an iPhone 5 with iOS 9.2.0. I would like to verify whether this file has been edited. File Permissions: -rwx------ Content Create Date: 2016:03:01 14:21:08+07:00 The second file was recorded by me on the same device model and iOS version. File Permissions: -rw-r--r-- Date/Time Original: 2024:10:03 11:44:16+07:00 As you can see, the file permissions differ, and the key for the recording date also differs: one uses "Content Create Date" while the other uses "Date/Time Original." I would like to determine if the first file was edited, but I haven't been able to find any official documentation on the m4a schema or metadata structure from audio recorder apps. I reached out to support, and they directed me to this forum. Any insights or help would be appreciated.

Currently we tested iOS AAC LC encoder using AudioToolbox framework, no matter we set mManufacturer to kAppleHardwareAudioCodecManufacturer or kAppleSoftwareAudioCodecManufacturer, it always run on CPU.

I’m working on an iOS app for a client, and I have a question regarding a specific feature we're looking to implement. We want the app to respond to a user pressing the volume button three times while the app is in the background. The goal is to allow users to discreetly trigger a safety feature without drawing attention, particularly in situations where they may be in danger or at risk. This feature is critical for the app and would be a valuable addition, as it could potentially help protect users in emergency situations. However, I haven’t found much information on whether iOS allows background listening for volume button presses. Therefore, I would greatly appreciate your insights on the following: Is it possible to listen for volume button presses when the app is in the background, or are there system-level restrictions that prevent this? If it's not directly possible, are there any special provisions, APIs, or entitlements that can be requested from Apple to enable this functionality? In case this feature is not supported, are there alternative approaches to achieve a similar discreet activation mechanism? If this is something that requires special permission or a process, could you please guide me on how to proceed? I understand that maintaining user privacy and security is a priority for iOS, and I want to ensure that any implementation fully complies with Apple's guidelines. Thanks in advance for your help!

When we tested the audio quality of our VoIP App, we found that when the iOS18.0 device was played with AirPods Pro 2, we could hear noises similar to peak clipping and distortion, especially when the sound source played was loud and high-pitched. Here is the device information we tested: Model: iPhone 16 Pro Max, iPhone 15 Pro System version: iOS 18.0 (22A3354) Bluetooth headset model: AirPods Pro 2 Bluetooth firmware version: 6F8 We tested multiple apps (including phone calls, FaceTime, Zoom, WeChat, Tencent Meeting), and they all had the above noise problem. We also found two phenomena: If we use the same iOS 18 device to connect HUAWEI FreeBuds Pro or FreeBuds 2, there is no such noise problem; If we use an iOS 17 device to connect to the same AirPods Pro 2 for testing, there is no such noise problem; Therefore, we suspect that it is caused by the compatibility problem between iOS 18.0 and AirPods firmware 6F8. The firmware version of our AirPods Pro 2 is 6F8, which was released on June 26, and iOS 18.0 was released on September 16. Maybe they are not very compatible. I hope that subsequent firmware updates can fix this problem.

Here is the demo from Apple's site This issues is specific to iOS 18. When running this demo, we are getting new text when we have a gap in speaking, the recognitionTask(with:resultHandler:) provides new text which is only spoken after the gap and not the concatenation of old text and the new spoken text.

Hi! I have an AVAudioSequencer with some AVMusicTracks that are filled with AVParameterEvents. If I toggle the isMuted property of a track, it will instantly mute when changed to true. However, after turning the muting to false, the events will only triggers on the next round of a loop and not instantly. Is this intended behaviour, and is there some way to get the events to trigger immediately after toggling the isMuted to be false?

Hi everyone, I’m experiencing an issue where audio interruptions (e.g., phone calls) are not being intercepted while running sound classification in an app that uses the AVAudioSession. Classification works fine, but interruptions aren’t handled, even though I’ve followed Apple’s guidelines on handling audio interruptions [1_Document]. The classification was initially based on [2_Classifer], where it worked perfectly. However, when I adopted classification in a more camera-focused app using [3_Cam], the interruption behavior stopped working. The classification setup is functioning with [3_Cam], but audio interruptions are not triggered. The listener is invoked before starting sound analysis as suggested in [2_Classifier]. startListeningForAudioSessionInterruptions() try startAnalyzing([(request, observer)]) FYI, one change I have made for classifications is following. This works fine in all cases. // try audioSession.setCategory(.record, mode: .default) try audioSession.setCategory(.playAndRecord, mode: .default, options: [.defaultToSpeaker, .allowBluetooth]) I suspect the issue might be related to the AVAudioSession configuration or how the app handles recording and playback together. Is there anything else I should check related to AVAudioSession? Are there additional APIs I could use to pre-check or better handle audio interruptions? Any suggestions or guidance would be greatly appreciated! Platform: Swift 5, Xcode 16, iOS 18. References: Document Classifier Cam Best Regards

After updating iOS18.1 Beta Version, I have a lot of issues with my Apple Car Play as per following. Audio Quality Really Bad (it’s not playing with media instead voice channel. Sometime, it’s playing with the phone speakers even though I have connected the car play via cable. Its not operating well with car steering control such as Volume up & down, skip button. After receiving the phone call, it’s go back to the original audio quality but when my phone screen locked, its go bad again. I expect to fix these problems asap as I love to play music when I drive around.

Hello, We have noticed an issue with SoundRecognition that causes glitching with our AudioUnit setup in Smule. Input and output frame sizes are inconsistent. Input frame size does not match [AVAudioSession sharedInstance].IOBufferDuration My best guess is that SoundRecognition influences the input frame size and not the output frame size. To reproduce use the example app here: https://github.com/MarkoGill/SoundRecognitionBug Hardware/OS iPhone 14 Pro on iOS 18 -> Experiences the problem iPhone 11 on iOS 18 -> Experiences the problem iPhone 15 on iOS 18 -> Not experiencing the problem Reproduction Steps Enable Sound Recognition (Settings > Accessibility > Sound Recognition > On) Enable a Sound for detection (Sounds > Dog > On) Open the example app with headset (it routes input to output) Notice glitching occurs Check the logs. Record and Playback buffer sizes vary Example Log: AU input sample rate: 48000.000000 AU output sample rate: 48000.000000 hardware sample rate: 48000.000000 hardware buffer size: 1104.000000 updated record frame counts: 1024 updated playback frame counts: 1104 Notes: You can disable Sound Recognition, restart the app, and playback behaves correctly.

I’m working on a macOS app, written in Swift. My goal is to record audio from an external microphone, e.g., one connected via USB. For this, I’m using an AVCaptureSession and recording its output with an AVAssetWriter. This works perfectly in principle (and reliably with internal microphones, for example). The problem occurs after my app has successfully completed the first recording and I then want to make additional recordings (which makes me think it might be process-dependent, because it works again after restarting the app). The problem: Noisy or distorted-sounding audio files. In addition, the following error message appears in the Console from CoreAudio / its AudioConverter: Input data proc returned inconsistent 512 packets for 2048 bytes; at 3 bytes per packet, that is actually 682 packets It is easy to reproduce. This problem is reproducible even if I don’t configure the AVAssetWriter manually and instead let it receive its audioSettings using a preset from an AVOutputSettingsAssistant. I’m running on macOS 15.0 (24A335). I’ve filed a feedback including a demo project → FB15333298 🎟️ I would greatly appreciate any help! Have a great day, Martin

Hi, I'm trying to insert CMSampleBuffers into an AVAssetWriterInput that has been configured with expectsMediaDataInRealTime = false with pauses. That is, I insert fixed-length audio at specific (increasing and non-overlapping) time points with large gaps in between. E.g., 5 seconds of audio at t=3.0, 5 seconds of audio at t=12.0, etc. The first audio sample plays at t=3 in the final output video as expected. But then all the other samples are bunched up immediately after it instead of being scheduled at the correct time. Below is my code. I'm just loading the asset and then readjusting its timestamps to be correct in the absolute timeline. Why do they not get scheduled correctly when the timestamps and durations are definitely correct and non-overlapping? func addFrame(_ pixelBuffer: CVPixelBuffer) { guard CGSize(width: pixelBuffer.width, height: pixelBuffer.height) == outputSize else { return } let frameTime = CMTimeMake(value: frameCount, timescale: frameRate) if videoInput?.isReadyForMoreMediaData == true { pixelBufferAdaptor?.append(pixelBuffer, withPresentationTime: frameTime) frameCount += 1 currentTime = frameTime } } func addMP3AudioClip(_ audioData: Data) async throws { let tempURL = FileManager.default.temporaryDirectory.appendingPathComponent(UUID().uuidString + ".mp3") defer { try? FileManager.default.removeItem(at: tempURL) } try audioData.write(to: tempURL) let asset = AVAsset(url: tempURL) let duration = try await asset.load(.duration) let audioTrack = try await asset.loadTracks(withMediaType: .audio).first! let audioReader = try AVAssetReader(asset: asset) let outputSettings: [String: Any] = [ AVFormatIDKey: kAudioFormatLinearPCM, AVSampleRateKey: 44100, AVNumberOfChannelsKey: 2, AVLinearPCMBitDepthKey: 16, AVLinearPCMIsFloatKey: false, AVLinearPCMIsBigEndianKey: false, AVLinearPCMIsNonInterleaved: false ] let audioReaderOutput = AVAssetReaderTrackOutput(track: audioTrack, outputSettings: outputSettings) audioReader.add(audioReaderOutput) guard audioReader.startReading() else { throw NSError(domain: "AudioReaderError", code: 0, userInfo: [NSLocalizedDescriptionKey: "Failed to start reading audio"]) } let baseInsertionTime = currentTime.convertScale(duration.timescale, method: .default) // Capture the current video time when the method is called print("Adding audio clip at \(baseInsertionTime.seconds) seconds, duration: \(duration.seconds) seconds") var audioTime = CMTime.zero var totalDuration: Double = 0 while let sampleBuffer = audioReaderOutput.copyNextSampleBuffer() { let bufferDuration = CMSampleBufferGetDuration(sampleBuffer) let adjustedBuffer = adjustTimeStamp(of: sampleBuffer, by: baseInsertionTime) while !audioInput!.isReadyForMoreMediaData { try await Task.sleep(nanoseconds: 100_000_000) // 0.1 second } audioInput!.append(adjustedBuffer) print(" Adjusted time: \(adjustedBuffer.presentationTimeStamp.seconds)") audioTime = CMTimeAdd(audioTime, bufferDuration) totalDuration += bufferDuration.seconds } print("Finished adding audio clip. Last sample at: \(CMTimeAdd(baseInsertionTime, audioTime).seconds) seconds") print(" totalDuration=\(totalDuration)") } private func adjustTimeStamp(of sampleBuffer: CMSampleBuffer, by timeOffset: CMTime) -> CMSampleBuffer { var count: CMItemCount = 0 CMSampleBufferGetSampleTimingInfoArray(sampleBuffer, entryCount: 0, arrayToFill: nil, entriesNeededOut: &count) var timingInfo = [CMSampleTimingInfo](repeating: CMSampleTimingInfo(), count: count) CMSampleBufferGetSampleTimingInfoArray(sampleBuffer, entryCount: count, arrayToFill: &timingInfo, entriesNeededOut: nil) for i in 0..<count { timingInfo[i].presentationTimeStamp = CMTimeAdd(timingInfo[i].presentationTimeStamp, timeOffset) if timingInfo[i].decodeTimeStamp != .invalid { timingInfo[i].decodeTimeStamp = CMTimeAdd(timingInfo[i].decodeTimeStamp, timeOffset) } else { timingInfo[i].decodeTimeStamp = timingInfo[i].presentationTimeStamp } } var adjustedBuffer: CMSampleBuffer? CMSampleBufferCreateCopyWithNewTiming(allocator: nil, sampleBuffer: sampleBuffer, sampleTimingEntryCount: count, sampleTimingArray: &timingInfo, sampleBufferOut: &adjustedBuffer) return adjustedBuffer! }

Our app, Universalis (Apple ID 284942719) plays audio successfully on all versions of iOS up to and including iOS 17. It uses the old MediaPlayer interface because it is targeted at versions all the way down to iOS 12. On iOS 18, it plays audio but CarPlay fails to show the Now Playing screen. Instead, a message box pops up in CarPlay saying "There was a problem loading this content", with an OK button. Nevertheless the audio plays correctly. This has been reported in the wild by a user of iOS 18 with CarPlay. I am also able to reproduce it locally, running the app in Xcode with the CarPlay Simulator, with an iPhone using iOS 18.0 or iOS 18.1. Earlier versions work correctly. Looking at the console log in CarPlay, the following error message appears about 10 seconds before the error message pops up: MSVEntitlementUtilities - Process Universalis PID[1173] - Group: (null) - Entitlement: com.apple.mediaremote.external-artwork-validation - Entitled: NO - Error: (null) The message has an orange background which appears to mean that it does not come directly from NSLog in the app. The message appears immediately after the request handler of MPMediaItemArtwork has been called requesting a 128 x 128 image and has successfully returned a 128x128 UIImage object. This has been reported through Feedback Assistant: Bug report ID is FB15343941 How can we work round this error?

Hi, I've recently been working with the Apple Music API and have had success in loading all the playlists on my account, loading songs from each playlist, and adding songs to the ApplicationMusicPlayer.share.queue. The problem I'm running into is that not all songs from the playlist are being added to the queue, despite confirming all the songs are being based on the PlaybackView.swift I'm about to share with you. I would also like to answer other underlying questions if possible. I am also open to any other suggestions. In this scenario were also assuming isShuffled is true every time. How can I determine when a song has ended? How can I get the album title information? How can I get the current song title, album information, and artist information without pressing play? I can only seem to update the screen when I select my play meaning ApplicationMusicPlayer.shared.play() is being called. How do I get the endTime of the song? I believe it should be ApplicationMusicPlayer.shared.queue.currentEntry.endTime but this doesn't seem to work. // // PlayBackView.swift // // Created by Justin on 8/16/24. // import SwiftUI import MusicKit import Foundation enum PlayState { case play case pause } struct PlayBackView: View { @State var song: Track @Binding var songs: [Track]? @State var isShuffled = false @State private var playState: PlayState = .pause @State private var isFirstPlay = true private let player = ApplicationMusicPlayer.shared private var isPlaying: Bool { return (player.state.playbackStatus == .playing) } var body: some View { VStack { // Album Cover HStack(spacing: 20) { if let artwork = player.queue.currentEntry?.artwork { ArtworkImage(artwork, height: 100) } else { Image(systemName: "music.note") .resizable() .frame(width: 100, height: 100) } VStack(alignment: .leading) { // Song Title Text(player.queue.currentEntry?.title ?? "Song Title Not Found") .font(.title) .fixedSize(horizontal: false, vertical: true) // How do I get AlbumTitle from here?? // Artist Name Text(player.queue.currentEntry?.subtitle ?? "Artist Name Not Found") .font(.caption) } } .padding() Spacer() // Progress View // endTime doesn't work and not sure why. ProgressView(value: player.playbackTime, total: player.queue.currentEntry?.endTime ?? 1.00) .progressViewStyle(.linear) .tint(.red.opacity(0.5)) // Duration View HStack { Text(durationStr(from: player.playbackTime)) .font(.caption) Spacer() if let duration = player.queue.currentEntry?.endTime { Text(durationStr(from: duration)) .font(.caption) } } Spacer() Button { Task { do { try await player.skipToNextEntry() } catch { print(error.localizedDescription) } } } label: { Label("", systemImage: "forward.fill") .tint(.white) } Spacer() // Play/Pause Button Button(action: { handlePlayButton() isFirstPlay = false }, label: { Text(playState == .play ? "Pause" : isFirstPlay ? "Play" : "Resume") .frame(maxWidth: .infinity) }) .buttonStyle(.borderedProminent) .padding() .font(.largeTitle) .tint(.red) } .padding() .onAppear { if isShuffled { songs = songs?.shuffled() if let songs, let firstSong = songs.first { player.queue = .init(for: songs, startingAt: firstSong) player.state.shuffleMode = .songs } } } .onDisappear { player.stop() player.queue = [] player.playbackTime = .zero } } private func handlePlayButton() { Task { if isPlaying { player.pause() playState = .pause } else { playState = .play await playTrack() } } } @MainActor private func playTrack() async { do { try await player.play() } catch { print(error.localizedDescription) } } private func durationStr(from duration: TimeInterval) -> String { let seconds = Int(duration) let minutes = seconds / 60 let remainder = seconds % 60 // Format the string to ensure two digits for the remainder (seconds) return String(format: "%d:%02d", minutes, remainder) } } //#Preview { // PlayBackView() //}

Hello, As explained in this link, the AVAssetReaderTrackOutput.copyNextSampleBuffer() returns a CMSampleBuffer in linear PCM audio format. I want to place this audio buffer into an AVAssetWriterInput of type kAudioFormatMPEG4AAC, but I can't manage the conversion. Could you help me by providing an extension that returns a CMSampleBuffer converted from linear PCM audio format to kAudioFormatMPEG4AAC? Example: extension CMSampleBuffer { func fromPCMToAAC() -> CMSampleBuffer? { // Here, get a new AudioStreamBasicDescription, create a CMSampleBuffer and a CMBlockBuffer } } I've tried multiple times but without success. Software: iOS 18.1 XCode: 16.0 Thank you!

I am using the AVAudioEngine to play back samples in an iOS game. I would like to change the play back rate of a sample in real time. When using AVAudioUnitVarispeed for chaging the play back rate it creates stutters in the game as it isn't processed in real time (as stated here:AVAudioUnitTimeEffect) The other option I found to change the rate is by using an AVAudioEnvironmentNode and change the rate of the AVAudioPlayerNode. That works without creating stutters but limits the valid values for the rate from 0.5 to 2.0 (I need higher rates then 2.0). See here: AVAudio3DMixing. Are there any other ways to play back a sample with a rate control in real time?