I am developing an app that uses MusicKit to play music and then I need to have spoken words played to the user, while ducking the audio coming from MusicKit (application music player) the built in Siri voices are not off sufficient quality so I am using an external service to create an mp3 file and then play this back using AVAudioSession Sample code below the problem I am having is that .duckOthers is not ducking the Application Music Player output Is this a bug or am I doing this wrong? // Configure audio session for system-wide ducking try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, options: [.duckOthers, .mixWithOthers]) try AVAudioSession.sharedInstance().setActive(true) // Set the ducking level to maximum try AVAudioSession.sharedInstance().setPreferredIOBufferDuration(0.005) // Create and configure audio player self.audioPlayer = try AVAudioPlayer(data: audioData) self.audioPlayer?.delegate = self self.audioPlayer?.volume = 1.0 // Ensure full volume for speech self.audioPlayer?.prepareToPlay() // Set the audio player's settings for maximum clarity self.audioPlayer?.enableRate = false self.audioPlayer?.pan = 0.0 // Center the audio self.audioPlayer?.play()

Environment Device: iPhone 16e iOS Version: 18.4.1 - 18.7.1 Framework: AVFoundation (AVAudioEngine) Problem Summary On iPhone 16e (iOS 18.4.1-18.7.1), the installTap callback stops being invoked after resuming from a phone call interruption. This issue is specific to phone call interruptions and does not occur on iPhone 14, iPhone SE 3, or earlier devices. Expected Behavior After a phone call interruption ends and audioEngine.start() is called, the previously installed tap should continue receiving audio buffers. Actual Behavior After resuming from phone call interruption: Tap callback is no longer invoked No audio data is captured No errors are thrown Engine appears to be running normally Note: Normal pause/resume (without phone call interruption) works correctly. Steps to Reproduce Start audio recording on iPhone 16e Receive or make a phone call (triggers AVAudioSession interruption) End the phone call Resume recording with audioEngine.start() Result: Tap callback is not invoked Tested devices: iPhone 16e (iOS 18.4.1-18.7.1): Issue reproduces ✗ iPhone 14 (iOS 18.x): Works correctly ✓ iPhone SE 3 (iOS 18.x): Works correctly ✓ Code Initial Setup (Works) let inputNode = audioEngine.inputNode inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in self.processAudioBuffer(buffer, at: time) } audioEngine.prepare() try audioEngine.start() Interruption Handling NotificationCenter.default.addObserver( forName: AVAudioSession.interruptionNotification, object: AVAudioSession.sharedInstance(), queue: nil ) { notification in guard let userInfo = notification.userInfo, let typeValue = userInfo[AVAudioSessionInterruptionTypeKey] as? UInt, let type = AVAudioSession.InterruptionType(rawValue: typeValue) else { return } if type == .began { self.audioEngine.pause() } else if type == .ended { try? self.audioSession.setActive(true) try? self.audioEngine.start() // Tap callback doesn't work after this on iPhone 16e } } Workaround Full engine restart is required on iPhone 16e: func resumeAfterInterruption() { audioEngine.stop() inputNode.removeTap(onBus: 0) inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in self.processAudioBuffer(buffer, at: time) } audioEngine.prepare() try audioSession.setActive(true) try audioEngine.start() } This works but adds latency and complexity compared to simple resume. Questions Is this expected behavior on iPhone 16e? What is the recommended way to handle phone call interruptions? Why does this only affect iPhone 16e and not iPhone 14 or SE 3? Any guidance would be appreciated!

I've got a setup using AVAudioEngine with several tone generator nodes, each with a chain of processing nodes, the chains then mixed into the main output. Generator ➡️ Effect ➡️... ➡️ .mainMixerNode ➡️ .outputNode). Generator ➡️ Effect ➡️... ⤴️ ... Generator ➡️ Effect ➡️... ⤴️ The user should be able to mute any chain individually. I've found several potential approaches to muting, but not terribly happy with any of them. Adjust the amplitudes directly in my tone generators. Issue: Consumes CPU even when completely muted. 4 generators adds ~15% cpu, even when all chains are muted. Detach/attach chains that are muted/unmuted. Issue: Causes loud clicking/popping sounds whenever muted/unmuted. Fade mixer output volume while detaching/attaching a chain (just cutting the volume immediately to 0 doesn't get rid of the clicking/popping). Issue: Causes all channels to fade during the transition, so not ideal. The rest of these ideas are variations on making volume control+detatch/attach work for individual chains, since approach #3 worked well. Add an AVAudioMixer to the end of each chain (just for volume control). Issue: Only the mixer on the final chain functions -- the others block all output. Not sure what's going on there. Use matrix mixer (for multi-input volume control). Plus detach/attach to reduce CPU if necessary. Not yet attempted, due to perceived complexity and reports of fragility in order of wiring in. A bunch of effort before I even know if it's going to work. Develop my own fader node to put on the end of each channel. Unlike the tone generator (simple AVSourceNode), developing an effect node seems complex and time consuming. Might not even fix CPU use. I'm not completely averse to the learning curve of either 5 or 6, but would rather get some guidance on best approach before diving in. They both seem likely to take more effort than I'd like for the simple behavior I'm trying to achieve.

Hello Apple Developer Community, I am trying to play an HLS stream using the React Native Video player (underneath it's using AvPlayer). I am able to play the stream smoothly, but in some cases the player can not play the stream properly. Behaviour: react-native-video: I am getting the below error. Error details from react-native-video player: Error Code: -12971 Domain: CoreMediaErrorDomain Localised Description: The operation couldn’t be completed. (CoreMediaErrorDomain error -12971.) Target: 2457 The error does not provide a specific failure reason or recovery suggestion, which makes troubleshooting challenging. AvPlayer on native iOS project: Video playback stopped after playing a few seconds. AVPlayer configuration: player.currentItem?.preferredForwardBufferDuration = 1 player.automaticallyWaitsToMinimizeStalling = true N.B.: The same buffer duration is working perfectly for others. Stream properties: video resolution: 1280 x 720 I have attached an overview report generated from MediaStreamValidator. I would appreciate any insights or suggestions on how to address this error. Has anyone in the community experienced a similar issue or have any advice on potential solutions? Thank you for your help!

Hi, In my project I am using AVFoundation for recording the audio. We are using AVAudioMixerNode class below method to record the audio packet. **func installTap( onBus bus: AVAudioNodeBus, bufferSize: AVAudioFrameCount, format: AVAudioFormat?, block tapBlock: @escaping AVAudioNodeTapBlock ) ** It works perfectly fine. But in production env some small percentage of the user we are facing issue like after recording few packets it stops automatically without stopping the audio engine. Can anyone help here that why this happens? I have also observed for mediaServicesWereResetNotification and added log on receiving this notification but when this issue happens I don't see any occurence of this log. Also is there any callback when the engine stops?

I’m implementing a custom playback pipeline using AVSampleBufferAudioRenderer together with AVSampleBufferRenderSynchronizer. hasSufficientMediaDataForReliablePlaybackStart appears to be the intended signal for determining when enough media has been queued to start playback. For local playback, this works well in practice — the property becomes true after a reasonable amount of media is enqueued. However, when the output route is AirPlay, using this property becomes difficult: AirPlay requires significantly more buffered media before the renderer reports sufficient data. The required preroll amount is much larger than for local playback. For short assets, it is possible to enqueue the entire audio track and still never observe hasSufficientMediaDataForReliablePlaybackStart == true. In that situation there is no more media data to enqueue, but the renderer still reports that playback is not ready. Given this behavior, what is the recommended way to determine playback readiness when using AVSampleBufferAudioRenderer with AirPlay?

I developed an educational app that implements audio-video communication through RTC, while using WebView to display course materials during classes. However, some users are experiencing an issue where the audio playback from WebView is very quiet. I've checked that the AVAudioSessionCategory is set by RTC to AVAudioSessionCategoryPlayAndRecord, and the AVAudioSessionCategoryOption also includes AVAudioSessionCategoryOptionMixWithOthers. What could be causing the WebView audio to be suppressed, and how can this be resolved?

Hello! I have been following the UsingAVFoundationToPlayAndPersistHTTPLiveStreams sample code in order to test persisting streams to disk. In addition to support for m3u8, I have noticed in testing that this also seems to work for MP3 Audio, simply by changing the plist entries to point to remote URLs with audio/mpeg content. Is this expected, or are there caveats that I should be aware of? Thanks you!

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’m a normie: not a developer at all. My idea might be super dumb. Would it be possible to please let us have a button in iphone photos that when toggled allows us to save certain chosen raw images to an iphone block chain, AND have Apple authenticate they are native photos, marked the milisecond they were taken, that they are native and no AI was used on those images? That might go a long way toward restoring trust in truth in photos again. We could also have the same thing for AI. Marked notification in the data on AI photos that can't be erased. Sorry if this is already underway and I'm just a normal person and therefore don't know it. I just want to be able to trust things again. 🤷🏽‍♀️

On iPhone 16 Pro Max (not tested other devices) there's a noticeable jump in the framing of the preview video when you record in the iOS AVCam Sample App. The same jump in camera framing can be observed by switching to the front facing camera and then back to the rear one. It looks roughly consistent with switching between the 0.5x and 1x camera (but not quite a match for the same viewable area in the Camera app) - and it's only when it's initially loaded, once recording is started it retains the 'closer' image no matter how many times it's stopped/started thereafter. I'm relatively new to Swift and haven't done anything with the camera before, so odd 'buggy' behaviour in the sample code isn't helping me understand it! :-) Is there any way to fix this?

Hey, I've noticed that in some scenarios photo data can be corrupted from the cameras on iPhone 17 Pro. The requirements are: The zoom level is greater than 2 times the base zoom, so 2x for the wide lens, and 8x for the telephoto. QualityPrioritization is set to .Quality. If set to .Balanced the images look as expected. The scene is well lit. I haven't managed to work out if there's an ISO cut off, but in darker scenes the images look as expected. The scene does not contain any objects or texture, e.g. a blank white screen, a blue sky, up close against a bright wall. Here is an example: This is really weird behavior. I have opened a ticket here: https://feedbackassistant.apple.com/feedback/22092908 There's also a repo here if anyone would like to try it: https://github.com/alexfoxy/CameraQualityTest. Thanks, Alex

I am developing an iOS camera app that can record video directly to external storage connected to an iPhone. To detect whether an external USB storage device is connected and to obtain its URL, I am considering using AVExternalStorageDeviceDiscoverySession. However, when checking support using AVExternalStorageDeviceDiscoverySession.isSupported, I observe that it returns true only on Pro model iPhones, and false on non-Pro models in my environment. I have reviewed Apple’s official documentation, but I could not find any clear description of the supported devices or requirements (for example, whether this API is limited to Pro models or requires specific hardware capabilities). I would appreciate any information regarding the following points: ①The actual requirements for AVExternalStorageDeviceDiscoverySession to be supported Device limitations (Pro vs non-Pro models) Hardware requirements (USB controller, external recording capability, etc.) iOS version dependencies ②Whether support for non-Pro models is planned in the future Tested environments iPhone 16 Pro (iOS 18.7.1) → isSupported == true iPhone 16e (iOS 26.2) → isSupported == false iPhone 17 (iOS 26.2) → isSupported == false iPhone Air (iOS 26.2) → isSupported == false If anyone has observed similar behavior or has official information from Apple regarding this API, I would greatly appreciate your insights.

Hi everyone, I'm developing a camera application that requires precise, predictable control over the focus system. I'm encountering unexpected behavior with face-driven autofocus in continuous autofocus mode. Issue: When using AVCaptureDevice.FocusMode.continuousAutoFocus, the system continues to prioritize faces for focus even after attempting to disable face-driven autofocus with: device.automaticallyAdjustsFaceDrivenAutoFocusEnabled = false device.isFaceDrivenAutoFocusEnabled = false Observations: The behavior is inconsistent across different scenes. In well-lit/properly exposed scenes: focus persistently locks onto faces, ignoring my configuration. . In underexposed scenes: the intended focus behavior is more consistently respected.

Hello, i can successfully match music using shazamkit on Apple using SwiftUI, a simple app that let user to load an audio file and exctracts the relative match, while i am unable to match music using shamzamkit on Android. I am trying to make the same simple app but i cannot match music as i get MATCH_ATTEMPT_FAILED every time i try to. I don't know what i am doing wrong but the shazam part in the kotlin Android code is in this method : suspend fun processAudioFileInBackground( filePath: String, developerTokenProvider: DeveloperTokenProvider ) = withContext(Dispatchers.IO) { val bufferSize = 1024 * 1024 val audioFile = FileInputStream(filePath) val byteBuffer = ByteBuffer.allocate(bufferSize) byteBuffer.order(ByteOrder.LITTLE_ENDIAN) var bytesRead: Int while (audioFile.read(byteBuffer.array()).also { bytesRead = it } != -1) { val signatureGenerator = (ShazamKit.createSignatureGenerator(AudioSampleRateInHz.SAMPLE_RATE_44100) as ShazamKitResult.Success).data signatureGenerator.append(byteBuffer.array(), bytesRead, System.currentTimeMillis()) val signature = signatureGenerator.generateSignature() println("Signature: ${signature.durationInMs}") val catalog = ShazamKit.createShazamCatalog(developerTokenProvider, Locale.ENGLISH) val session = (ShazamKit.createSession(catalog) as ShazamKitResult.Success).data val matchResult = session.match(signature) println("MatchResult : $matchResult") setMatchResult(matchResult) byteBuffer.clear() } audioFile.close() } I noticed that changing Locale in catalog creation results in different result as i get NoMatch without exception. Can you please help me with this? Do i need to create a custom catalog?

who can help me to make app for iOS with XCode

The introduction of PHBackgroundResourceUploadExtension is a welcome addition in iOS 26.1. I wonder however, how to attach a debugger and actually get the system to call the process() method of the extension. I tried to run the extension both inside photos app (and also the main app for testing), but when I take a photo or add photos to the library (saving), the process() method does never get called. Any hints would be appreciated to debug the PHBackgroundResourceUploadExtension during development.

We are planning to develop an app that connects to a UVC camera to capture and display video via AVFoundation. Could you please advise on which iPhone models support UVC cameras?

In my app I use AVAssetReaderTrackOutput to extract PCM audio from a user-provided video or audio file and display it as a waveform. Recently a user reported that the waveform is not in sync with his video, and after receiving the video I noticed that the waveform is in fact double as long as the video duration, i.e. it shows the audio in slow-motion, so to speak. Until now I was using CMFormatDescription.audioStreamBasicDescription.mSampleRate which for this particular user video returns 22'050. But in this case it seems that this value is wrong... because the audio file has two audio channels with different sample rates, as returned by CMFormatDescription.audioFormatList.map({ $0.mASBD.mSampleRate }) The first channel has a sample rate of 44'100, the second one 22'050. If I use the first sample rate, the waveform is perfectly in sync with the video. The problem is given by the fact that the ratio between the audio data length and the sample rate multiplied by the audio duration is 8, double the ratio for the first audio file (4). In the code below this ratio is given by Double(length) / (sampleRate * asset.duration.seconds) When commenting out the line with the sampleRate variable definition in the code below and uncommenting the following line, the ratios for both audio files are 4, which is the expected result. I would expect audioStreamBasicDescription to return the correct sample rate, i.e. the one used by AVAssetReaderTrackOutput, which (I think) somehow merges the stereo tracks. The documentation is sparse, and in particular it’s not documented whether the lower or higher sample rate is used; in this case, it seems like the higher one is used, but audioStreamBasicDescription for some reason returns the lower one. Does anybody know why this is the case or how I should extract the sample rate of the produced PCM audio data? Should I always take the higher one? I created FB19620455. let openPanel = NSOpenPanel() openPanel.allowedContentTypes = [.audiovisualContent] openPanel.runModal() let url = openPanel.urls[0] let asset = AVURLAsset(url: url) let assetTrack = asset.tracks(withMediaType: .audio)[0] let assetReader = try! AVAssetReader(asset: asset) let readerOutput = AVAssetReaderTrackOutput(track: assetTrack, outputSettings: [AVFormatIDKey: Int(kAudioFormatLinearPCM), AVLinearPCMBitDepthKey: 16, AVLinearPCMIsBigEndianKey: false, AVLinearPCMIsFloatKey: false, AVLinearPCMIsNonInterleaved: false]) readerOutput.alwaysCopiesSampleData = false assetReader.add(readerOutput) let formatDescriptions = assetTrack.formatDescriptions as! [CMFormatDescription] let sampleRate = formatDescriptions[0].audioStreamBasicDescription!.mSampleRate //let sampleRate = formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate }).max()! print(formatDescriptions[0].audioStreamBasicDescription!.mSampleRate) print(formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate })) if !assetReader.startReading() { preconditionFailure() } var length = 0 while assetReader.status == .reading { guard let sampleBuffer = readerOutput.copyNextSampleBuffer(), let blockBuffer = sampleBuffer.dataBuffer else { break } length += blockBuffer.dataLength } print(Double(length) / (sampleRate * asset.duration.seconds))

Hello, We are trying to use the new Background Upload Extension to improve uploads of assets (Photos, Live Photos, Videos) in the background in our application. 1-The assets have finished uploading, but I'm unable to retrieve successful records using PHAssetResourceUploadJob.fetchJobs(action: .acknowledge, options: nil). When will the successful records be returned? 2-How to retrieve the system's pending tasks ? We want to cancel tasks handed over to the system when returning to the main app to avoid duplicate resource uploads. 3-When we set UploadJobExtensionEnabled = true, will tasks handed over to the system still execute after returning to the main app? Do we need to set UploadJobExtensionEnabled = false upon returning to the main app? If we set UploadJobExtensionEnabled = false, will previously submitted upload tasks be cleared?