I’m currently developing an iOS metronome app using DispatchSourceTimer as the timer. The interval is set very small, around 50 milliseconds, and I’m using CFAbsoluteTimeGetCurrent to calculate the elapsed time to ensure the beat is played within a ±0.003-second margin. The problem is that once the app goes to the background, the timing becomes unstable—it slows down noticeably, then recovers after 1–2 seconds. When coming back to the foreground, it suddenly speeds up, and again, it takes 1–2 seconds to return to normal. It feels like the app is randomly “powering off” and then “overclocking.” It’s super frustrating. I’ve noticed that some metronome apps in the App Store have similar issues, but there’s one called “Professional Metronome” that’s rock solid with no such problems. What kind of magic are they using? Any experts out there who can help? Thanks in advance! P.S. I’ve already enabled background audio permissions. The professional metronome that has no issues: https://link.zhihu.com/?target=https%3A//apps.apple.com/cn/app/pro-metronome-%25E4%25B8%2593%25E4%25B8%259A%25E8%258A%2582%25E6%258B%258D%25E5%2599%25A8/id477960671

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 all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!

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]

I have spent a long time refactoring lots of older Swift code to compile without error in Swift 6. The app is a v3 audio unit host and audio unit. Having installed Sonoma and XCode 16 I compile the code using Swift 6 and it compiles and runs without any warnings or errors. My host will load my AU no problem. LOGIC PRO is still the ONLY audio unit host that will load native Mac V3 audio units and so I like to test my code using Logic. In Sonoma with XCode 16... My AU passes the most stringent AUVAL tests both in terminal and Logic pro. If I compile the AU source in Swift 5 Logic will see the AU, load it and run it without problems. But when I compile the AU in Swift 6 Logic sees the AU, will scan it and verify it passes the tests but will not load the AU. In XCode I see a log message that a "helper application failed to run" but the debugger never connects to the AU and I don't think Logic even gets as far as instantiating the AU. So... what is causing this? I'm stumped.. Developing AUv3 is a brain-aching maze of undocumented hurdles and I'm hoping someone might have found a solution for this one. Meanwhile I guess my only option is to continue using the Swift 5 compiler. (appending a little note just to mention that all the DSP code is written in C/C++, Swift is used mainly for the user interface and also does some offline thready work )

Dear Apple, I am currently working on Mental Health related research project supported by South Korea Government funding. In addition to SensorKit access, we are working on the data from microphone. Is there any contact point aside SensorKit access application to discuss the possibility of research data collection from restricted participant samples?

I'm trying to implement Ambisonic B-Format audio playback on Vision Pro with head tracking. So far audio plays, head tracking works, and the sound appears to be stereo. The problem is that it is not a proper binaural playback when compared to playing back the audiofile with a DAW. Has anyone successfully implemented B-Format playback on Vision Pro? Any suggestions on my current implementation: func playAmbiAudioForum() async { do { try AVAudioSession.sharedInstance().setCategory(.playback) try AVAudioSession.sharedInstance().setActive(true) // AudioFile laoding/preperation guard let testFileURL = Bundle.main.url(forResource: "audiofile", withExtension: "wav") else { print("Test file not found") return } let audioFile = try AVAudioFile(forReading: testFileURL) let audioFileFormat = audioFile.fileFormat // create AVAudioFormat with Ambisonics B Format guard let layout = AVAudioChannelLayout(layoutTag: kAudioChannelLayoutTag_Ambisonic_B_Format) else { print("layout failed") return } let format = AVAudioFormat( commonFormat: audioFile.processingFormat.commonFormat, sampleRate: audioFile.fileFormat.sampleRate, interleaved: false, channelLayout: layout ) // write audiofile to buffer guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: UInt32(audioFile.length)) else { print("buffer failed") return } try audioFile.read(into: buffer) playerNode.renderingAlgorithm = .HRTF // connecting nodes audioEngine.attach(playerNode) audioEngine.connect(playerNode, to: audioEngine.outputNode, format: format) audioEngine.prepare() playerNode.scheduleBuffer(buffer, at: nil) { print("File finished playing") } try audioEngine.start() playerNode.play() } catch { print("Setup error:", error) } }

Dear Sirs, I’ve written a virtual audio driver based on AudioDriverKit and running as dext in my MacOS app. Sometimes when waking up from a sleep state the recording side of my driver extension seems to hang and I don’t see any calls to my io_operation callback. Then the recording app like a DAW seems to hang when trying to start a recording. This doesn’t happen after short sleep states or after a complete new start of my MacBook. I already opened a case in Feedback-Assistant on 5th of May (FB17503622) which also includes a sysdiagnose and a ktrace but I didn't get any feedback so far. Meanwhile some of our customers are getting angry and I'd like to know if there's anything I could do to fix this problem on my side. We’re not sure whether this worked in previous MacOS versions, we think we didn’t observe this before 15.3.1 but at least since 15.3.1. we’ve seen this problem. Best regards, Johannes

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?

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.

iPadOS 18.3 beta 3 (22D5055b) fixed the issue for me and my 7th generation iPad.

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’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.

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.

使用AVSpeechUtterance实现iOS语音播报，选择语言为简体中文“zh-CN”，读取中文“袆”（hui 第一声)错误，读成了“祎”（yi 第一声），希望能优化。

I ran 5.1 audio tests in both YouTube and Apple Music, and I noticed that when sound is supposed to play from the rear or front surround speakers, it’s also duplicated in the front left and right channels. I’m absolutely sure the issue is with the Apple TV, because I played the same video directly through my TV’s native system, and the channel separation was correct. Everything used to work perfectly before, so this must be a software issue. I’m currently on tvOS 26 Developer Beta 5, but I’m certain the problem also existed on the stable tvOS 18.5. I’ve already reset and updated my Apple TV, and I also tried switching the audio format to forced Dolby Atmos 5.1. On the forums, I mostly see complaints about Dolby Atmos not working at all — in my case, everything technically works, but not the way it’s supposed to.

Hi. I am working on an audio app for iOS. I have added the CPNowPlayingPlaybackRateButton to my CPNowPlayingTemplate. When the button is clicked, my handler changes the rate in the AVPlayer and updates the MPNowPlayingInfoCenter to the new rate, for example, 2.0. Throughout, the Carplay button always displays "0x". I am wondering how to get this UI to accurately reflect the playback rate the user has selected, as always displaying 0x is a poor user experience. You may suggest MPChangePlaybackRateCommand is relevant here, but I have not been able to get that to work either, and judging by posts online, not many other people have either. I have made a post about that here: https://developer.apple.com/forums/thread/773099 Is this a known Apple bug? Is there a way to get the UI to accurately reflect the playback rate of my audio? Kind regards.

Issue Description When playing certain MIDI files using AVMIDIPlayer, the initial volume settings for individual tracks are being ignored during the first playback. This results in all tracks playing at the same volume level, regardless of their specified volume settings in the MIDI file. Steps to Reproduce Load a MIDI file that contains different volume settings for multiple tracks Start playback using AVMIDIPlayer Observe that all tracks play at the same volume level, ignoring their individual volume settings Current Behavior All tracks play at the same volume level during initial playback Track volume settings specified in the MIDI file are not being respected This behavior consistently occurs on first playback of affected MIDI files Expected Behavior Each track should play at its specified volume level from the beginning Volume settings in the MIDI file should be respected from the first playback Workaround I discovered that the correct volume settings can be restored by: Starting playback of the MIDI file Setting the currentPosition property to (current time - 1 second) After this operation, all tracks play at their intended volume levels However, this is not an ideal solution as it requires manual intervention and may affect the playback experience. Questions Is there a way to ensure the track volume settings are respected during the initial playback? Is this a known issue with AVMIDIPlayer? Are there any configuration settings or alternative approaches that could resolve this issue? Technical Details iOS Version: 18.1.1 (22B91) Xcode Version: 16.1 (16B40)

Hello All, It seems that it's "very easy" (😬) to implement a little Swift code inside the prepared AU using Xcode 16.2 on Sequoia 15.1.1 and a Mac Studio M1 Ultra, but my issue is that I finally don't know... where. The documentation says that I've to find the AudioUnitViewController.swift file and then modify the render block : audioUnit.renderBlock = { (numFrames, ioData) in // Process audio here } in the Xcode project automatically generated, but I didn't find such a file... If somebody can help me in showing where is the file to be modified, I'll be very grateful ! Thank you very much. J

I bought two "Apple USB-C to Headphone Jack Adapters". Upon closer inspection, they seems to be of different generations: The one with product ID 0x110a on top is working fine. The one with product ID 0x110b has two issues: There is a short but loud click noise on the headphone when I connect it to the iPad. When I play audio using AVAudioPlayer the first half of a second or so is cut off. Here's how I'm playing the audio: audioPlayer = try AVAudioPlayer(contentsOf: url) audioPlayer?.delegate = self audioPlayer?.prepareToPlay() audioPlayer?.play() Is this a known issue? Am I doing something wrong?