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.

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.

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 !

As a straightforward example, I've taken Apple's MV-HEVC sample project and added two lines. First, after the AVAssetWriterInput is created: frameInput.performsMultiPassEncodingIfSupported = true Second, after the call to multiviewWriter.startWriting(): print("canPerformMultiplePasses: \(frameInput.canPerformMultiplePasses)") Which prints true. This leads me to believe that the first encoding pass should proceed as-normal (even though I haven't handled the logic for the completion of the first pass, etc.). However, I receive this error when the code attempts to appendTaggedBuffers to the AVAssetWriterInputTaggedPixelBufferGroupAdaptor: Fatal error: Failed to append tagged buffers to multiview output Am I missing a step? Or is the multi-pass encoding only supported for standard sample/pixel buffers (and not tagged buffers)?

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.

We are working on an app for the vision pro which has high polygons count and lots of high resolution textures. Everything looks smooth and in general very well, The issue is the moment we turn on voice control even if it is not being used the visuals at the center start to stutter left to right. Has anyone seen this ?, it must be a bug, any workaround ? Thanks, Guillermo

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.

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!

Hi, for the implementation of an audio player with signed URL's, I need to be able to set an authorization header to the request for an AVURLAsset. This works but not on Airplay when trying to stream multiple songs in a queue. For each item I do: let headerFields: [String: String] = ["Authorization": getIdToken()!] super.init(url: url, options: ["AVURLAssetHTTPHeaderFieldsKey": headerFields]) But only the first 2 songs in the queue actually get this authorization header sent along, somehow it is removed for subsequent songs. Any ideas on how I can fix this? thanks, Thomas

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() //}

Title says it all.

Hello, I am a deaf-blind wheelchair user, and I program in Swift using a braille display. I’m reaching out for your help on an issue I’ve been struggling to solve. Basically, when I extract a CMSampleBuffer from an AVAsset of a video, it comes with the Audio Format ID as Linear PCM. However, when I try to pass this CMSampleBuffer to write another video using AVAssetWriter, the video ends up muted. The audio settings of the output video are configured to MPEG-4 AAC, but the input CMSampleBuffer has the Audio Format ID as Linear PCM. I would like to request an extension for CMSampleBuffer that converts Linear PCM audio to MPEG-4 AAC. I’ve searched extensively and couldn’t find anything. Looking forward to your help. Thank you.

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

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.

I have a USB audio interface that is causing kernel traps and the audio output to "skip" or dropout every few seconds. This behavior occurs with a completely fresh install of Catalina as well as Big Sur with the stock Music app on a 2019 MacBook Pro 16 (full specs below). The Console logs show coreaudiod got an error from a kernel trap, a "USB Sound assertion" in AppleUSBAudio/AppleUSBAudio-401.4/KEXT/AppleUSBAudioDevice.cpp at line 6644, and the Music app "skipping cycle due to overload." I've added a short snippet from Console logs around the time of the audio skip/drop out. The more complete logs are at this gist: https://gist.github.com/djflux/08d9007e2146884e6df1741770de5105 I've also opened a Feedback Assistant ticket (FB9037528): https://feedbackassistant.apple.com/feedback/9037528 Does anyone know what could be causing this issue? Thanks for any help. Cheers, Flux aka Andy. Hardware Overview:  Model Name: MacBook Pro  Model Identifier: MacBookPro16,1  Processor Name: 8-Core Intel Core i9  Processor Speed: 2.4 GHz  Number of Processors: 1  Total Number of Cores: 8  L2 Cache (per Core): 256 KB  L3 Cache: 16 MB  Hyper-Threading Technology: Enabled  Memory: 64 GB  System Firmware Version: 1554.80.3.0.0 (iBridge: 18.16.14347.0.0,0) System Software Overview: System Version: macOS 11.2.3 (20D91)  Kernel Version: Darwin 20.3.0  Boot Volume: Macintosh HD  Boot Mode: Normal  Computer Name: mycomputername  User Name: myusername  Secure Virtual Memory: Enabled  System Integrity Protection: Enabled USB interface: Denon DJ DS1 Snippet of Console logs error 21:07:04.848721-0500 coreaudiod HALS_IOA1Engine::EndWriting: got an error from the kernel trap, Error: 0xE00002D7 default 21:07:04.848855-0500 Music HALC_ProxyIOContext::IOWorkLoop: skipping cycle due to overload default 21:07:04.857903-0500 kernel USB Sound assertion (Resetting engine due to error returned in Read Handler) in /AppleInternal/BuildRoot/Library/Caches/com.apple.xbs/Sources/AppleUSBAudio/AppleUSBAudio-401.4/KEXT/AppleUSBAudioDevice.cpp at line 6644 ... default 21:07:05.102746-0500 coreaudiod Audio IO Overload inputs: 'private' outputs: 'private' cause: 'Unknown' prewarming: no recovering: no default 21:07:05.102926-0500 coreaudiod   CAReportingClient.mm:508  message {   HostApplicationDisplayID = "com.apple.Music";   cause = Unknown;   deadline = 2615019;   "input_device_source_list" = Unknown;   "input_device_transport_list" = USB;   "input_device_uid_list" = "AppleUSBAudioEngine:Denon DJ:DS1:000:1,2";   "io_buffer_size" = 512;   "io_cycle" = 1;   "is_prewarming" = 0;   "is_recovering" = 0;   "issue_type" = overload;   lateness = "-535";   "output_device_source_list" = Unknown;   "output_device_transport_list" = USB;   "output_device_uid_list" = "AppleUSBAudioEngine:Denon DJ:DS1:000:1,2"; }: (null)

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?

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.