On some devices, when i select the same media multiple times, the data by` loadFileRepresentation(forTypeIdentifier: completionHandler) ` returned is different(data.count is not equal). environment: * Model: iPhone 12 * Model Number: MGGM3CH/A * iOS Version: 18.3.2 ```Swift // import PhotosUI func picker(_ picker: PHPickerViewController, didFinishPicking results: [PHPickerResult]) { picker.dismiss(animated: true, completion: nil) guard let provider = results.last?.itemProvider else { return } guard provider.hasItemConformingToTypeIdentifier(UTType.movie.identifier) else { return } Task { provider.loadFileRepresentation(forTypeIdentifier: UTType.movie.identifier) { url, error in guard let url = url else { return } if let data = try? Data(contentsOf: url) { print("data count is: \(data.count)") } } } } ``` ps: I also try some other function, eg: ` provide.loadItem(forTypeIdentifier:)`, but not work too.

I am developing an app to stream and download DRM protected HLS videos based on the official “FairPlay Streaming Server SDK”. When I play the downloaded video, it asks the server for .ts or .aac, even though I have passed the path of the downloaded video to AVURLAsset. As a result, playback fails when the device is offline, such as in airplane mode. This behavior depends on the playback time of the video and occurs when trying to download and play a video with a playback time of 19 hours or more. It did not occur for videos with a playback time of 18 hours. The environment we checked is iOS 18.3. The solution at this time is to limit the video playback time to 18 hours, but if possible, we would like to allow download playback of videos longer than 19 hours. Does anyone have any information or know of a solution to this problem, such as if you have experienced this type of event, or if you know that content longer than 19 hours cannot be played offline? // load let path = ".../xxx.movpkg" // Path of the downloaded file videoAsset = AVURLAsset(url: path) playerItem = AVPlayerItem(asset: videoAsset!) player.replaceCurrentItem(with: playerItem) // isPlayableOffline print("videoAsset.assetCache.isPlayableOffline = \(videoAsset.assetCache.isPlayableOffline)") // true

I am working on a project for macOS where I am taking an AVCaptureSession's CVPixelBuffer and I need to convert it into a MTLTexture for rendering. On macOS the pixel format is 2vuy, there does not seem to be a clear format conversion while converting to a metal texture. I have been able to convert it to a texture but the color space seems to be off as it is rendering distorted colors with a double image. I believe 2vuy is a single pane color space and I have tried to account for that, but I am unaware of what is off. I have attached The CVPixelBuffer and The distorted MTLTexture along with a laundry list of errors. On iOS my conversions are fine, it is only the macOS 2vuy pixel format that seems to have issues. My code for the conversion is also attached. If there are any suggestions or guidance on how to properly convert a 2vuy CVPixelBuffer to a MTLTexture I would greatly appreciate it. Many Thanks Conversion_Logs.txt ConversionCode.swift

I’m developing a hybrid app (WebView / Turbo Native) that uses getUserMedia to access the back camera for a PPG/heart rate measurement feature (the user places their finger on the camera). Problem: Even when I specify constraints like: { video: { deviceId: '...', facingMode: { exact: 'environment' }, advanced: [{ zoom: 1.0 }] }, audio: false } On iPhone 15 (iOS 18), iOS unexpectedly switches between the wide, ultra-wide, and telephoto lenses during the measurement. This breaks the heart rate detection, and it forces the user to move their finger in the middle of the measurement. Question: Is there any way, via getUserMedia/WebRTC, to force iOS to use only the wide-angle lens and prevent automatic lens switching? I know that with AVFoundation (Swift) you can pick .builtInWideAngleCamera, but I’m hoping to avoid building a custom native layer and would prefer to stick with WebView/JavaScript if possible to save time and complexity. Any suggestions, workarounds, or updates from Apple would be greatly appreciated! Thanks a lot!

Hi! I am writing a browser extension that allows you to control the playback of media content on a music service website. Unfortunately Safari does not support tracking changes to the audible property in an event tabs.onUpdated. Is there an alternative to this event? I'm looking for a way to track when the automatic inference engine interrupts playback on a music service website. That you.

I'm experiencing an unexpected behavior with AVURLAsset and cookies. When setting cookies through AVURLAssetHTTPCookiesKey option, they seem to be sent only on the initial request but not on retry attempts. Here's my current implementation: let cookieProperties: [HTTPCookiePropertyKey: Any] = [ .name: "sessionCookie", .value: "testValue", .domain: url.host ?? "", .path: "/", .secure: true ] if let cookie = HTTPCookie(properties: cookieProperties) { let asset = AVURLAsset(url: url, options: [ AVURLAssetHTTPCookiesKey: [cookie], ]) } According to the documentation, AVURLAssetHTTPCookiesKey should apply the cookies to all requests made by this asset. However, when the initial request fails and AVPlayer retries, the cookies are not included in subsequent requests. Only when I store the cookie with HTTPCookieStorage.shared.setCookie, then it persists. Questions: Is this the expected behavior? If not, what could be causing the cookies to not persist for retry attempts? Is using HTTPCookieStorage.shared the recommended approach instead? Environment: iOS 16+ Using AVPlayer with AVURLAsset Streaming HLS content Any insights would be greatly appreciated.

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)

In an m3u8 manifest, audio EXT-X-MEDIA tags usually contain CHANNELS tag containing the audio channels count like so: #EXT-X-MEDIA:TYPE=AUDIO,URI="audio_clear_eng_stereo.m3u8",GROUP-ID="default-audio-group",LANGUAGE="en",NAME="stream_5",AUTOSELECT=YES,CHANNELS="2" Is it possible to get this info from AVPlayer, AVMediaSelectionOption or some related API?

I have an app under development - demo here - https://youtu.be/VbAfUk_eYl0?si=s6EDBx-4G6P_QbZO - which is sort of an audio player for airdropped files - something useful to musicians who dump work in progress to their phone, make notes, revise and update. I've been testing my handling of audio session interruption notifications, but seems to be a lot of inconsistency in how, when and why iOS delivers them, and I'm wondering if there is some rhyme or reason to it that I'm just not detecting. For example, I am playing a song in my app. Switch to Apple Music and start playing a song there. My app gets an interruption began notification - this is consistent. Switch back to my app, and about half the time, I will get an interruption ended notification (coupled often with a blast of the tail of whatever audio buffer was partially played when the interruption started, even though the engine was stopped - and followed by call to my AVAudioPlayerNodeCompletionCallback - is there some way to avoid this?). Half the time I don't get an interruption ended notification; my app can (as expected) end the interruption by activating the AVAudioSession and playing something. I have not been able to determine any pattern to this behavior, other than that if my app started playing using AVAudioPlayerNode.scheduleSegment rather than scheduleFile I think the notification will be consistently delivered on app activation rather than when I activate the session programmatically. I would like my app to behave deterministically, and would appreciate any help in deciphering what causes the inconsistent behavior in notifications from iOS.

We have a Push To Talk application which allow user to record video and audio. When user is recording a video using AVCaptureSession and receive's an Push To Talk call, from moment the Push To Talk call is received the audio in the video which is being captured is stopped while the video capture is still in progress. Here after the PTT call is completed, we have tried restarting the audio session, there are no errors that are getting printed but we still don't see the audio getting restarted in video capture. We have also tried to add a new input for AVCaptureSession we are receiving error that is resulting in video capture stopping, error mentioned below: [OS-PLT] [CameraManager] Movie file finished with error: Error Domain=AVFoundationErrorDomain Code=-11818 "Recording Stopped" UserInfo={AVErrorRecordingSuccessfullyFinishedKey=true, NSLocalizedDescription=Recording Stopped, NSLocalizedRecoverySuggestion=Stop any other actions using the recording device and try again., AVErrorRecordingFailureDomainKey=1, NSUnderlyingError=0x3026bff60 {Error Domain=NSOSStatusErrorDomain Code=-16414 "(null)"}}, success We have also raised a Feedback Ticket on same: https://feedbackassistant.apple.com/feedback/16050598

I am using an AVAudioPlayer to play a "tick" sound once per second in a SwiftUI app. When running the app on an iPhone 16 (18.2.1) the tick sounds increase in volume after a few seconds. This does not happen in the simulator nor on an iPhone SE 2020 (18.1.1).

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 have created an app where you can speak using SFSpeechRecognizer and it will recognize you speech into text, translate it and then return it back using speech synthesis. All locales for SFSpeechRecognizer and switching between them work fine when the app is in the foreground but after I turn off my screen(the app is still running I just turned off the screen) and try to create new recognitionTask it it receives this error inside the recognition task: User denied access to speech recognition. The weird thing about this is it only happens with some languages. The error happens with Croatian or Hungarian locale for speech recognition but doesn't with English or Spanish locale.

Hi there! We have a suite of AudioUnit v2 plugins that have been shipped for some time as aufx plugins, and we are looking into MIDI-related platform upgrades, so we need a way to update these plugins to request MIDI from Logic (and other AU hosts) but avoid changing our AU type and subtype so we don't break existing sessions. Any ideas on how we can do this?

I am trying to stream audio from local filesystem. For that, I am trying to use an AVAssetResourceLoaderDelegate for an AVURLAsset. However, Content-Length is not known at the start. To overcome this, I tried several methods: Set content length as nil, in the AVAssetResourceLoadingContentInformationRequest Set content length to -1, in the ContentInformationRequest Both of these cause the AVPlayerItem to fail with an error. I also tried setting Content-Length as INT_MAX, and setting a renewalDate = Date(timeIntervalSinceNow: 5). However, that seems to be buggy. Even after updating the Content-Length to the correct value (e.g. X bytes) and finishing that loading request, the resource loader keeps getting requests with requestedOffset = X with dataRequest.requestsAllDataToEndOfResource = true. These requests keep coming indefinitely, and as a result it seems that the next item in the queue does not get played. Also, .AVPlayerItemDidPlayToEndTime notification does not get called. I wanted to check if this is an expected behavior or is there a bug in this implementation. Also, what is the recommended way to stream audio of unknown initial length from local file system? Thanks!

I've been working with MusicKit without enrolling for a developer account and I haven't run into any issues until I noticed nil on the SysteyMusicPlayer item for some songs and I don't understand why. Are some songs blocked from the framework? Or is it somehow a limitation to not having registered MusicKit to the bundle ID? I am planning on using MusicKit properly in prod and this is just a test app for a package I'm working on. These are the nil songs which I got from the Discovery Station: https://music.apple.com/ca/album/okay/950816298?i=950816304 https://music.apple.com/ca/album/youre-so-cool/1670485433?i=1670485446

I prefer to use the album fetched from the library instead of the catalog since this is faster. If doing so, how can I check if all tracks of an album are added to the library. In this case I'd like to fetch the catalog version or throw an error (for example when offline). Using .with(.tracks) on the library album fetches the tracks added to the library. The trackCount property is referring to the tracks that can be fetched from the library. The isComplete property is always nil when fetching from the library. One possible way is checking the trackNumber and discCount properties. However this only detects that not all tracks of an album are added to the library if there is a song not added ahead of one that is. I'd like to be able to handle this edge case as well. Is there currently a way to do this? I'd prefer to not rely on the apple music catalog for this since this is supposed to work offline as well. Fetching and storing all trackIDs when connected and later comparing against these would work, but this would potentially mean storing tens of thousands of track ids. Thank you

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。

t has been quite some time since I requested the Apple FPS package, yet I haven’t received it. I haven’t received any email either. Is there a developer support inquiry center where I can check the status of the process? Alternatively, could you share approximately how long it took for you to receive a response email?

Hi all, we try migrate project to Swift 6 Project use AVPlayer in MainActor Selection audio and subtitiles not work Task { @MainActor in let group = try await item.asset.loadMediaSelectionGroup(for: AVMediaCharacteristic.audible) get error: Non-sendable type 'AVMediaSelectionGroup?' returned by implicitly asynchronous call to nonisolated function cannot cross actor boundary and second example `if #available(iOS 15.0, *) { player?.currentItem?.asset.loadMediaSelectionGroup(for: AVMediaCharacteristic.audible, completionHandler: { group, error in if error != nil { return } if let groupWrp = group { DispatchQueue.main.async { self.setupAudio(groupWrp, audio: audioLang) } } }) }` get error: Sending 'groupWrp' risks causing data races