Hello, I have an iOS app that is recording audio that is working fine on iPads/iPhones. It asks for microphone permission and after that recording works. I installed the same app on my M3 MacBook via TestFlight since iPad apps are supposed to work without a change that way. The app starts fine and everything, but it never asks for Microphone permission, so I can't record. Do I need to do something to make this happen (this is not macCatalyst, its running the arm64 iPhone binary on macOS) thanks

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

I'm trying to implement airplay into my app. I can successfully playback sound and trigger the airplay selector sheet. If the target device is a Bluetooth only device I can connect with no problem and stream the audio to the Bluetooth device, but if the audio device is a airplay specific device like a HomePod or an Apple TV when I select it, I get a spinning icon, indicating that it is trying to connect, and eventually it times out and stops without connecting. I don't believe it is an AirPlay audio issue because if I go to a different app, for example a podcast app and select my HomePods for output, and then switch back to my app. My audio will correctly stream to the HomePod. Not only that, I have it so that my icon will change color to indicate that it is connected via airplay and it is correctly indicating that it is connected via AirPlay. But I cannot then disconnect it using the Airplay selector. The issue appears to be in the AirPlay selection side, which I have spent several days attempting to troubleshoot mostly using ChatGPT to suggest code different than what I have to maybe work around the issue. Mostly it is focused on the audio player section, but it doesn't seem like that is really the route that is the problem.

I'm developing an iOS app that requires continuous audio recording. Currently, when a phone call comes in, the AVAudioSession is interrupted and recording stops completely during the ringing phase. While I understand recording should stop if the call is answered, my app needs to continue recording while the phone is merely ringing. I've observed that Apple's Voice Memos app maintains recording during incoming call rings. This indicates the hardware and iOS are capable of supporting this functionality. Request Please advise on any available AVAudioSession configurations or APIs that would allow my app to: Continue recording during an incoming call ring Only stop recording if/when the call is actually answered Impact This interruption significantly impacts the user experience and core functionality of my app. Workarounds like asking users to enable airplane mode are impractical and create a poor user experience. Questions Is there an approved way to maintain microphone access during call rings? If not currently possible, could this capability be considered for addition to a future iOS SDK? Are there any interim solutions or best practices Apple recommends for this use case? Thank you for your help. SUPPORT INFORMATION Did someone from Apple ask you to submit a code-level support request? No Do you have a focused test project that demonstrates your issue? Yes, I have a focused test project to submit with my request What code level support issue are you having? Problems with an Apple framework API in my app

Hello Apple Developer Community, I am seeking clarification on the intended display behavior of HLS audio tracks within the iOS 26 (or current beta) native player, specifically concerning the NAME and LANGUAGE attributes of the EXT-X-MEDIA tag. In our HLS manifests, we define alternative audio tracks using EXT-X-MEDIA tags, like so: #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",LANGUAGE="ja",NAME="AUDIO-1",DEFAULT=YES,AUTOSELECT=YES,URI="audio_ja.m3u8" #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",LANGUAGE="ja",NAME="AUDIO-2",URI="audio_en.m3u8" Our observation is that when an audio track is selected and its name is displayed in the native iOS media controls (e.g., Control Center or within a full-screen video player's UI), the value specified in the NAME attribute ("AUDIO-1", "AUDIO-2") does not seem to be used. Instead, the display appears to derive from the LANGUAGE attribute ("ja", "en"), often showing the system's localized string for that language (e.g., "Japanese", "English"). We would like to understand the official or intended behavior regarding this. Is it the expected behavior for the iOS native player to prioritize the LANGUAGE attribute (or its localized equivalent) over the NAME attribute for displaying the selected audio track's label? If this is the intended design, what is the recommended best practice for developers who wish to present a custom, human-readable name for audio tracks (beyond the standard language name) in the native iOS UI? Are there any specific AVPlayer properties or AVMediaSelectionOption considerations that would allow more granular control over this display, or is this entirely managed by the system based on the LANGUAGE attribute? Any insights or official guidance on this behavior in iOS 26 (and potentially previous versions) would be greatly appreciated. Thank you for your time and assistance.

Hi all, I have spent a lot of time reading the tech note and watching the WDDC video that introduce the PTTFramework on iOS. I currently have a custom setup where I am using AVAudioEngine to schedule and play buffers that are being streamed through a call. I am looking to use the PTTFramework to allow a user to trigger this push to talk behavior from the lock screen and the various places with the system UI it provides. However I am unsure what the correct behavior is regarding the handling of the audio session. Right now I am using .playback when there is no active voice transmission so that devices such as AirPods can be in AD2P mode where applicable, and then transitioning to .playbackAndRecord category only when the mic input should become active. Following this change in my AVAudioEngine manager I am then manually activating and deactivating the audio session manually when the engine is either playing/recording or idle. In the documentation it states that you should not attempt to activate or deactivate your audio session directly, but allow the framework to handle it. Does that mean that I need to either call the request to transmit delegate function or set an active participant on the channel manager first, and then wait for the didBecomeActive delegate method to trigger before I actually attempt to play or record any audio? (I am using the fullDuplex mode currently.) I noticed that that delegate method will only trigger if the audio session wasn't active before doing one of the above (setting active participant, requesting transmit). Lastly, when using the PTTFramework it also mentions that we get support for PTT devices and I notice on the didBeginTransmittingFrom property we have a handsfreeButton case. Is there any documentation or resources for what is actually supported out of the box for this? I am currently working on handling a lot of the push to talk through bluetooth LE, and wanted to make sure there wasn't overlap with what the system provides. Thank you!

I am trying to debug the AAX version of my plugin (MIDI effect) on Pro Tools, but I am getting the following error (Mac console) when attempting to load it: dlsym cannot find symbol g_dwILResult in CFBundle etc.. I used Xcode 16.4 to build the plugin. Has anybody come across the same or a similar message? Best, Achillefs Axart Labs

I have a memory leak, when using AVAudioPlayer. I managed to narrow down the issue into a very simple app, which code I paste in at the end. The memory leak start immediately when I start playing sound, but only in the emylator. On the real iPhone there is no memory leak. The memory leak on the Simulator looks like this: import SwiftUI import AVFoundation struct ContentView_Audio: View { var sound: AVAudioPlayer? init() { guard let path = Bundle.main.path(forResource: "cd201", ofType: "mp3") else { return } let url = URL(fileURLWithPath: path) do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .default, options: [.mixWithOthers]) } catch { return } do { try AVAudioSession.sharedInstance().setActive(true) } catch { return } do { sound = try AVAudioPlayer(contentsOf: url) } catch { return } } var body: some View { HStack { Button { playSound() } label: { ZStack { Circle() .fill(.mint.opacity(0.3)) .frame(width: 44, height: 44) .shadow(radius: 8) Image(systemName: "play.fill") .resizable() .frame(width: 20, height: 20) } } .padding() Button { stopSound() } label: { ZStack { Circle() .fill(.mint.opacity(0.3)) .frame(width: 44, height: 44) .shadow(radius: 8) Image(systemName: "stop.fill") .resizable() .frame(width: 20, height: 20) } } .padding() } } private func playSound() { guard sound != nil else { return } sound?.volume = 1 // sound?.numberOfLoops = -1 sound?.play() } func stopSound() { sound?.stop() } }

Hi, I am getting into a trap. Please check stack-trace, howto fix this? regards, Joël stack-trace with ExtAudioFileWrite

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?

In MusicKit Web the playback states are provided as numbers. For example the playbackStateDidChange event listener will return: {oldState: 2, state: 3, item:...} When the state changes from playing (2) to paused (3). Those are pretty easy to guess, but I'm having a hard time with some of the others: completed, ended, loading, none, paused, playing, seeking, stalled, stopped, waiting. I cannot find a mapping of states to numbers documented anywhere. I got the above states from an enum in a d.ts file that is often incorrect/incomplete. Can someone help out pointing to the docs or provide a mapping? Thanks.

Bug Report: ScreenCaptureKit System Audio Capture Crashes with EXC_BAD_ACCESS Summary When using ScreenCaptureKit to capture system audio for extended periods, the application crashes with EXC_BAD_ACCESS in Swift's error handling runtime. The crash occurs in swift_getErrorValue when trying to process an error from the SCStream delegate method didStopWithError. This appears to be a framework-level issue in ScreenCaptureKit or its underlying ReplayKit implementation. Environment macOS Sonoma 14.6.1 Swift 5.8 ScreenCaptureKit framework Detailed Description Our application captures system audio using ScreenCaptureKit's audio capture capabilities. After successfully capturing for several minutes (typically after 3-4 segments of 60-second recordings), the application crashes with an EXC_BAD_ACCESS error. The crash happens when the Swift runtime attempts to process an error in the SCStreamDelegate.stream(_:didStopWithError:) method. The crash consistently occurs in swift_getErrorValue when attempting to access the class of what appears to be a null object. This suggests that the error being passed from the system framework to our delegate method is malformed or contains invalid memory. Steps to Reproduce Create an SCStream with audio capture enabled Add audio output to the stream Start capture and write audio data to disk Allow the capture to run for several minutes (3-5 minutes typically triggers the issue) The app will crash with EXC_BAD_ACCESS in swift_getErrorValue Code Sample func stream(_ stream: SCStream, didStopWithError error: Error) { print("Stream stopped with error: \(error)") // Crash occurs before this line executes } func stream(_ stream: SCStream, didOutputSampleBuffer sampleBuffer: CMSampleBuffer, of type: SCStreamOutputType) { guard type == .audio, sampleBuffer.isValid else { return } // Process audio data... } Expected Behavior The error should be properly propagated to the delegate method, allowing for graceful error handling and recovery. Actual Behavior The application crashes with EXC_BAD_ACCESS when the Swift runtime attempts to process the error in swift_getErrorValue. Crash Log Details Thread #35, queue = 'com.apple.NSXPCConnection.m-user.com.apple.replayd', stop reason = EXC_BAD_ACCESS (code=1, address=0x0) frame #0: 0x0000000194c3088c libswiftCore.dylib`swift::_swift_getClass(void const*) + 8 frame #1: 0x0000000194c30104 libswiftCore.dylib`swift_getErrorValue + 40 frame #2: 0x00000001057fba30 shadow`NewScreenCaptureService.stream(stream=0x0000600002de6700, error=Swift.Error @ 0x000000016b7b5e30) at NEW+ScreenCaptureService.swift:365:15 frame #3: 0x00000001057fc050 shadow`@objc NewScreenCaptureService.stream(_:didStopWithError:) at <compiler-generated>:0 frame #4: 0x0000000219ec5ca0 ScreenCaptureKit`-[SCStreamManager stream:didStopWithError:] + 456 frame #5: 0x00000001ca68a5cc ReplayKit`-[RPScreenRecorder stream:didStopWithError:] + 84 frame #6: 0x00000001ca696ff8 ReplayKit`-[RPDaemonProxy stream:didStopWithError:] + 224 Printing description of stream._streamQueue: error: ObjectiveC.id:4294967281:18: note: 'id' has been explicitly marked unavailable here public typealias id = AnyObject ^ error: /var/folders/v4/3xg1hmp93gjd8_xlzmryf_wm0000gn/T/expr23-dfa421..cpp:1:65: 'id' is unavailable in Swift: 'id' is not available in Swift; use 'Any' Swift._DebuggerSupport.stringForPrintObject(Swift.UnsafePointer<id>(bitPattern: 0x104ae08c0)!.pointee) ^~ ObjectiveC.id:2:18: note: 'id' has been explicitly marked unavailable here public typealias id = AnyObject ^ warning: /var/folders/v4/3xg1hmp93gjd8_xlzmryf_wm0000gn/T/expr23-dfa421..cpp:5:7: initialization of variable '$__lldb_error_result' was never used; consider replacing with assignment to '_' or removing it var $__lldb_error_result = __lldb_tmp_error ~~~~^~~~~~~~~~~~~~~~~~~~ _ Before the crash, we observed this error message in the console: [ERROR] *****SCStream*****RemoteAudioQueueOperationHandlerWithError:1015 Error received from the remote queue -16665 Additional Context The issue occurs consistently after approximately 3-4 successful audio segment recordings of 60 seconds each Commenting out custom segment rotation logic does not prevent the crash The crash involves XPC communication with Apple's ReplayKit daemon The error appears to be corrupted or malformed when crossing the XPC boundary Workarounds Attempted Added proper thread safety for all published properties using DispatchQueue.main.async Implemented more robust error handling in the delegate methods None of these approaches prevented the crash since it occurs at the Swift runtime level before our code executes. Impact This issue prevents reliable long-duration audio capture using ScreenCaptureKit. This bug significantly limits the usefulness of ScreenCaptureKit for any application requiring continuous system audio capture for more than a few minutes. Perhaps this issue might be related to a macOS bug where the system dialog indicates that the screen is being shared, even though nothing is actually being shared. Moreover, when attempting to stop sharing, nothing happens.

My current app implements a custom video player, based on a AVSampleBufferRenderSynchronizer synchronising two renderers: an AVSampleBufferDisplayLayer receiving decoded CVPixelBuffer-based video CMSampleBuffers, and an AVSampleBufferAudioRenderer receiving decoded lpcm-based audio CMSampleBuffers. The AVSampleBufferRenderSynchronizer is started when the first image (in presentation order) is decoded and enqueued, using avSynchronizer.setRate(_ rate: Float, time: CMTime), with rate = 1 and time the presentation timestamp of the first decoded image. Presentation timestamps of video and audio sample buffers are consistent, and on most streams, the audio and video are correctly synchronized. However on some network streams, on iOS, the audio and video aren't synchronized, with a time difference that seems to increase with time. On the other hand, with the same player code and network streams on macOS, the synchronization always works fine. This reminds me of something I've read, about cases where an AVSampleBufferRenderSynchronizer could not synchronize audio and video, causing them to run with independent and potentially drifting clocks, but I cannot find it again. So, any help / hints on this sync problem will be greatly appreciated! :)

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

After upgrading to iOS 18.4, I'm no longer able to establish an AirPlay v1 connection to an audio system. The symptom is that the AirPlay route picker just spins when trying to connect to an audio system. It eventually gives up. I tested this on an iPhone 14, connecting to a HomePod, AirPort express, AppleTV and a Wiim Pro. If I try connecting with AirPlay v2, ex: using Apple Music, the connection succeeds and audio can be played. I'm the developer of an app that plays audio over AirPlay while also recording. My app has to use AirPlay v1 because AvAudioSession doesn't allow the policy .longFormAudio when the category is .playAndRecord. This issue is a real pain as it means my app is suddenly broken for many thousands of users. Is anyone else seeing this issue? Any suggestions for a workaround?

Hello, We are developing a real-time speech recognition application and are utilizing AVAudioEngine with voice processing enabled on the input node. However, we have observed that enabling this mode interferes with the built-in iOS screen recording feature - specifically, the recorded video does not capture any audio when this mode is active. Since we want users to be able to record their experience within our app, this issue significantly impacts our functionality. Is there a known workaround or recommended approach to ensure that both voice processing and screen recording can function simultaneously? Any guidance would be greatly appreciated. Thank you!

I'm getting this error when I launch my application on the iPhone 14 Pro via Xcode. Everything builds OK. I"m using the audio kit plugin and Sound Pipe Audiokit. The error starts as soon as I start the app and will carry on repeatedly. I have background processing turned on as I'd like the sounds to play when the phone is locked via the headphones. I can't find anything online about this error. None of my catches are printing anything in the logs either. So I don't know if this is just something that pops up repeatedly or whether there is something fundamentally wrong. private func setupAudioSession() { do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: [.mixWithOthers]) try session.setActive(true, options: .notifyOthersOnDeactivation) } catch { errorMessage = "Failed to set up audio session: (error.localizedDescription)" print(errorMessage ?? "") } } // MARK: - Background Task Handling private func setupBackgroundTaskHandling() { // Handle app entering background notificationObservers.append( NotificationCenter.default.addObserver( forName: UIApplication.didEnterBackgroundNotification, object: nil, queue: .main, using: { [weak self] _ in // Safely unwrap self guard let self = self else { return } self.handleBackgroundTransition() } ) ) I'm not sure if this is the code causing the issue. Any help would be gratefully appreciated. This is my first app I'm working on .

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’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 able to get text to speech to audio file using the following code for iOS 12 iPhone 8 to create a car file: audioFile = try AVAudioFile( forWriting: saveToURL, settings: pcmBuffer.format.settings, commonFormat: .pcmFormatInt16, interleaved: false) where pcmBuffer.format.settings is: [AVAudioFileTypeKey: kAudioFileMP3Type, AVSampleRateKey: 48000, AVEncoderBitRateKey: 128000, AVNumberOfChannelsKey: 2, AVFormatIDKey: kAudioFormatLinearPCM] However, this code does not work when I run the app in iOS 18 on iPhone 13 Pro Max. The audio file is created, but it doesn't sound right. It has a lot of static and it seems the speech is very low pitch. Can anyone give me a hint or an answer?