I started playing which transcription of audio files on macOS today, latest beta of Xcode and latest beta of Tahoe. Transcription itself works really well, but for some reason the majority of the results contain no audioTimeRange. I got 22 single-word results with time ranges, spread out all over total file of 53 minutes. Is there something I can do to improve this? To my understanding, I have followed sample code and instructions very closely, but the SwiftTranscriptionSampleApp and other examples I've seen lead me to believe I should be getting a lot more time ranges than I actually do.

I’m building a music app using Apple Music streaming via ApplicationMusicPlayer. My goal is to decrease the volume of the current song during the last 10 seconds, and when the next track begins, restore the volume to its normal level. I know that ApplicationMusicPlayer doesn’t expose a volume API, and I want to avoid triggering the system volume HUD. ✅ Using Apple Music streaming (not local files) ❓ Is it possible to implement per-track fade-out/fade-in logic with ApplicationMusicPlayer? Appreciate any clarification or official guidance!

After updating to iOS 18.5, we’ve observed that outgoing audio from our app intermittently stops being transmitted during VoIP calls using AVAudioSession configured with .playAndRecord and .voiceChat. The session is set active without errors, and interruptions are handled correctly, yet audio capture suddenly ceases mid-call. This was not observed in earlier iOS versions (≤ 18.4). We’d like to confirm if there have been any recent changes in AVAudioSession, CallKit, or related media handling that could affect audio input behavior during long-running calls. func configureForVoIPCall() throws { try setCategory( .playAndRecord, mode: .voiceChat, options: [.allowBluetooth, .allowBluetoothA2DP, .defaultToSpeaker]) try setActive(true) }

I've just begun to dip my toes into the iOS16 waters. One of the first things that I've attempted is to edit a library playlist using: try await MusicLibrary.shared.edit(targetPlaylist, items: tracksToAdd) Where targetPlaylist is of type MusicItemCollection<MusicKit.Playlist>.Element and tracksToAdd is of type [Track] The targetPlaylist was created, using new iOS16 way, here: let newPlaylist = try await MusicLibrary.shared.createPlaylist(name: name, description: description) tracksToAdd is derived by performing a MusicLibraryRequest on a specific playlist ID, and then doing something like this: if let tracksToAdd = try await playlist.with(.tracks).tracks {    // add tracks to target playlist } My problem is that when I perform attempt the edit, I am faced with a rather sad looking crash. libdispatch.dylib`dispatch_group_leave.cold.1:     0x10b43d62c <+0>:  mov    x8, #0x0     0x10b43d630 <+4>:  stp    x20, x21, [sp, #-0x10]!     0x10b43d634 <+8>:  adrp   x20, 6     0x10b43d638 <+12>: add    x20, x20, #0xfbf          ; "BUG IN CLIENT OF LIBDISPATCH: Unbalanced call to dispatch_group_leave()"     0x10b43d63c <+16>: adrp   x21, 40     0x10b43d640 <+20>: add    x21, x21, #0x260          ; gCRAnnotations     0x10b43d644 <+24>: str    x20, [x21, #0x8]     0x10b43d648 <+28>: str    x8, [x21, #0x38]     0x10b43d64c <+32>: ldp    x20, x21, [sp], #0x10 ->  0x10b43d650 <+36>: brk    #0x1 I assume that I must be doing something wrong, but I frankly have no idea how to troubleshoot this. Any help would be most appreciated. Thanks. @david-apple?

Hi, I’ve been experiencing a strange issue with video playback on my iPhone. While watching videos, the image will suddenly shift it becomes more greyish, then sometimes briefly goes black, and then returns to normal bright quality. This can happen multiple times during a single video. This is not limited to the Photos app. I’ve seen it happen: In the Photos app when playing videos I recorded myself In Snapchat when watching videos sent by others Occasionally in other social media apps as well Additional details: HDR Video is turned off Apple ProRes is turned off Tried both 4K 60fps and 4K 30fps Camera format set to “Most Compatible” Low Power Mode is off Issue happens whether the phone is cool or warm Doesn’t seem related to the video file itself the same file exported to another device looks fine all the way through These exact same videos play back completely normally on my iPad with no brightness or contrast shifts at all I’m currently on the iOS 17 public beta, but this issue was happening before I installed the beta as well, so it’s not beta-specific It almost feels like the display or system is switching between different brightness/contrast profiles mid playback, regardless of the app. Has anyone else experienced this, and is there a way to disable this behavior so the brightness and color stay consistent during video playback? Thanks in advance!

Good day, ladies and gents. I have an application that reads audio from the microphone. I'd like it to also be able to read from the Mac's audio output stream. (A bonus would be if it could detect when the Mac is playing music.) I'd eventually be able to figure it out reading docs, but if someone can give a hint, I'd be very grateful, and would owe you the libation of your choice. Here's the code used to set up the AudioUnit: -(NSString*) configureAU { AudioComponent component = NULL; AudioComponentDescription description; OSStatus err = noErr; UInt32 param; AURenderCallbackStruct callback; if( audioUnit ) { AudioComponentInstanceDispose( audioUnit ); audioUnit = NULL; } // was CloseComponent // Open the AudioOutputUnit description.componentType = kAudioUnitType_Output; description.componentSubType = kAudioUnitSubType_HALOutput; description.componentManufacturer = kAudioUnitManufacturer_Apple; description.componentFlags = 0; description.componentFlagsMask = 0; if( component = AudioComponentFindNext( NULL, &description ) ) { err = AudioComponentInstanceNew( component, &audioUnit ); if( err != noErr ) { audioUnit = NULL; return [ NSString stringWithFormat: @"Couldn't open AudioUnit component (ID=%d)", err] ; } } // Configure the AudioOutputUnit: // You must enable the Audio Unit (AUHAL) for input and output for the same device. // When using AudioUnitSetProperty the 4th parameter in the method refers to an AudioUnitElement. // When using an AudioOutputUnit for input the element will be '1' and the output element will be '0'. param = 1; // Enable input on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &param, sizeof(UInt32) ); chkerr("Couldn't set first EnableIO prop (enable inpjt) (ID=%d)"); param = 0; // Disable output on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(UInt32) ); chkerr("Couldn't set second EnableIO property on the audio unit (disable ootpjt) (ID=%d)"); param = sizeof(AudioDeviceID); // Select the default input device AudioObjectPropertyAddress OutputAddr = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; err = AudioObjectGetPropertyData( kAudioObjectSystemObject, &OutputAddr, 0, NULL, &param, &inputDeviceID ); chkerr("Couldn't get default input device (ID=%d)"); // Set the current device to the default input unit err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDeviceID, sizeof(AudioDeviceID) ); chkerr("Failed to hook up input device to our AudioUnit (ID=%d)"); callback.inputProc = AudioInputProc; // Setup render callback, to be called when the AUHAL has input data callback.inputProcRefCon = self; err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(AURenderCallbackStruct) ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); param = sizeof(AudioStreamBasicDescription); // get hardware device format err = AudioUnitGetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &deviceFormat, &param ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); audioChannels = MAX( deviceFormat.mChannelsPerFrame, 2 ); // Twiddle the format to our liking actualOutputFormat.mChannelsPerFrame = audioChannels; actualOutputFormat.mSampleRate = deviceFormat.mSampleRate; actualOutputFormat.mFormatID = kAudioFormatLinearPCM; actualOutputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved; if( actualOutputFormat.mFormatID == kAudioFormatLinearPCM && audioChannels == 1 ) actualOutputFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved; #if __BIG_ENDIAN__ actualOutputFormat.mFormatFlags |= kAudioFormatFlagIsBigEndian; #endif actualOutputFormat.mBitsPerChannel = sizeof(Float32) * 8; actualOutputFormat.mBytesPerFrame = actualOutputFormat.mBitsPerChannel / 8; actualOutputFormat.mFramesPerPacket = 1; actualOutputFormat.mBytesPerPacket = actualOutputFormat.mBytesPerFrame; // Set the AudioOutputUnit output data format err = AudioUnitSetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &actualOutputFormat, sizeof(AudioStreamBasicDescription)); chkerr("Could not change the stream format of the output device (ID=%d)"); param = sizeof(UInt32); // Get the number of frames in the IO buffer(s) err = AudioUnitGetProperty( audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &audioSamples, &param ); chkerr("Could not determine audio sample size (ID=%d)"); err = AudioUnitInitialize( audioUnit ); // Initialize the AU chkerr("Could not initialize the AudioUnit (ID=%d)"); // Allocate our audio buffers audioBuffer = [self allocateAudioBufferListWithNumChannels: actualOutputFormat.mChannelsPerFrame size: audioSamples * actualOutputFormat.mBytesPerFrame]; if( audioBuffer == NULL ) { [ self cleanUp ]; return [NSString stringWithFormat: @"Could not allocate buffers for recording (ID=%d)", err]; } return nil; } (...again, it would be nice to know if audio output is active and thereby choose the clean output stream over the noisy mic, but that would be a different chunk of code, and my main question may just be a quick edit to this chunk.) Thanks for your attention! ==Dave [p.s. if i get more than one useful answer, can i "Accept" more than one, to spread the credit around?] {pps: of course, the code lines up prettier in a monospaced font!}

I'm using Picture-in-Picture (PiP) mode in my native iOS application, which is similar to Google Meet, using the VideoSDK.live Native iOS SDK. The SDK has built-in support for PiP and it's working fine for the most part. However, I'm running into a problem: When the app enters PiP mode, the local camera (self-video) of the participant freezes or stops. I want to fix this and achieve the same smooth behavior seen in apps like Google Meet and WhatsApp, where the local camera keeps working even in PiP mode. I've been trying to find documentation or examples on how to achieve this but haven't had much luck. I came across a few mentions that using the camera in the background may require special entitlements from Apple (like in the entitlements file). Most of the official documentation says background camera access is restricted due to Apple’s privacy policies. So my questions are: Has anyone here successfully implemented background camera access in PiP mode on iOS? If yes, what permissions or entitlements are required? How do apps like WhatsApp and Google Meet achieve this functionality on iPhones? Any help, advice, or pointers would be really appreciated!

In Final Cut Pro, keyframes for transform parameters (such as Position, Scale, and Rotation) are automatically set to “Smooth” interpolation. This often results in undesired easing between keyframes, especially when linear motion is required. Currently, we have to manually adjust each keyframe to "Linear" using the Video Animation Editor, which can be time-consuming when working with many keyframes. Would it be possible to add an option to set the default keyframe interpolation to "Linear"—either globally in Preferences or per parameter in the Inspector? This would greatly streamline the animation workflow for many editors. Thank you for considering this request!

Context: I am currently developing an app using the Push-to-Talk (PTT) framework. I have reviewed both the PTT framework documentation and the CallKit demo project to better understand how to properly manage audio session activation and AVAudioEngine setup. I am not activating the audio session manually. The audio session configuration is handled in the incomingPushResult or didBeginTransmitting callbacks from the PTChannelManagerDelegate. I am using a single AVAudioEngine instance for both input and playback. The engine is started in the didActivate callback from the PTChannelManagerDelegate. When I receive a push in full duplex mode, I set the active participant to the user who is speaking. Issue When I attempt to talk while the other participant is already speaking, my input tap on the input node takes a few seconds to return valid PCM audio data. Initially, it returns an empty PCM audio block. Details: The audio session is already active and configured with .playAndRecord. The input tap is already installed when the engine is started. When I talk from a neutral state (no one is speaking), the system plays the standard "microphone activation" tone, which covers this initial delay. However, this does not happen when I am already receiving audio. Assumptions / Current Setup Because the audio session is active in play and record, I assumed that microphone input would be available immediately, even while receiving audio. However, there seems to be a delay before valid input is delivered to the tap, only occurring when switching from a receive state to simultaneously talking. Questions Is this expected behavior when using the PTT framework in full duplex mode with a shared AVAudioEngine? Should I be restarting or reconfiguring the engine or audio session when beginning to talk while receiving audio? Is there a recommended pattern for managing microphone readiness in this scenario to avoid the initial empty PCM buffer? Would using separate engines for input and output improve responsiveness? I would like to confirm the correct approach to handling simultaneous talk and receive in full duplex mode using PTT framework and AVAudioEngine. Specifically, I need guidance on ensuring the microphone is ready to capture audio immediately without the delay seen in my current implementation. Relevant Code Snippets Engine Setup func setup() { let input = audioEngine.inputNode do { try input.setVoiceProcessingEnabled(true) } catch { print("Could not enable voice processing \(error)") return } input.isVoiceProcessingAGCEnabled = false let output = audioEngine.outputNode let mainMixer = audioEngine.mainMixerNode audioEngine.connect(pttPlayerNode, to: mainMixer, format: outputFormat) audioEngine.connect(beepNode, to: mainMixer, format: outputFormat) audioEngine.connect(mainMixer, to: output, format: outputFormat) // Initialize converters converter = AVAudioConverter(from: inputFormat, to: outputFormat)! f32ToInt16Converter = AVAudioConverter(from: outputFormat, to: inputFormat)! audioEngine.prepare() } Input Tap Installation func installTap() { guard AudioHandler.shared.checkMicrophonePermission() else { print("Microphone not granted for recording") return } guard !isInputTapped else { print("[AudioEngine] Input is already tapped!") return } let input = audioEngine.inputNode let microphoneFormat = input.inputFormat(forBus: 0) let microphoneDownsampler = AVAudioConverter(from: microphoneFormat, to: outputFormat)! let desiredFormat = outputFormat let inputFramesNeeded = AVAudioFrameCount((Double(OpusCodec.DECODED_PACKET_NUM_SAMPLES) * microphoneFormat.sampleRate) / desiredFormat.sampleRate) input.installTap(onBus: 0, bufferSize: inputFramesNeeded, format: input.inputFormat(forBus: 0)) { [weak self] buffer, when in guard let self = self else { return } // Output buffer: 1920 frames at 16kHz guard let outputBuffer = AVAudioPCMBuffer(pcmFormat: desiredFormat, frameCapacity: AVAudioFrameCount(OpusCodec.DECODED_PACKET_NUM_SAMPLES)) else { return } outputBuffer.frameLength = outputBuffer.frameCapacity let inputBlock: AVAudioConverterInputBlock = { inNumPackets, outStatus in outStatus.pointee = .haveData return buffer } var error: NSError? let converterResult = microphoneDownsampler.convert(to: outputBuffer, error: &error, withInputFrom: inputBlock) if converterResult != .haveData { DebugLogger.shared.print("Downsample error \(converterResult)") } else { self.handleDownsampledBuffer(outputBuffer) } } isInputTapped = true }

We are seeing logs were on iOS devices we see some keyframes request. but on safari browser don’t see any request like this. could you please explain what is it. /d8ceb9244ff889b42b82eb807327531-c27dbcb10e0bbf3cde6c-1/d8ceb9244ff88e9b42b82eb807327531-c27dbcb10e0bbf3cde6c-1/keyframes/hls/.

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, Basically, I am reading and writing an asset. To simplify, I am just reading the asset and rewriting it into an output video without any modifications. However, I want to add a fade-out effect to the last three seconds of the output video. I don’t know how to do this. So far, before adding the CMSampleBuffer to the output video, I tried reducing its volume using an extension on CMSampleBuffer. In the extension, I passed 0.4 for testing, aiming to reduce the video's overall volume by 60%. My question is: How can I directly adjust the volume of a CMSampleBuffer? Here is the extension: extension CMSampleBuffer { func adjustVolume(by factor: Float) -> CMSampleBuffer? { guard let blockBuffer = CMSampleBufferGetDataBuffer(self) else { return nil } var length = 0 var dataPointer: UnsafeMutablePointer<Int8>? guard CMBlockBufferGetDataPointer(blockBuffer, atOffset: 0, lengthAtOffsetOut: nil, totalLengthOut: &length, dataPointerOut: &dataPointer) == kCMBlockBufferNoErr else { return nil } guard let dataPointer = dataPointer else { return nil } let sampleCount = length / MemoryLayout<Int16>.size dataPointer.withMemoryRebound(to: Int16.self, capacity: sampleCount) { pointer in for i in 0..<sampleCount { let sample = Float(pointer[i]) pointer[i] = Int16(sample * factor) } } return self } }

Can i use iokit usb lib to disable build-in camera?

Hello, I'm trying to receive parquet files using the example that provided in documentation. I've done all required steps but receive constantly error 500 with "Upstream Service Error". By looking into the issues list, seems this error exists for months. Is it possible to get it working?

I am currently developing an HEVC player using VideoToolbox on an iOS device. I have successfully created an HEVC decoder that receives HEVC streams from our custom image capture and encoding device, and it can decode and display images properly. However, when my image capture and encoding device configures the encoder to output HEVC streams with ​​fragmented NALUs​​ (i.e., an I-frame or P-frame is split and stored across multiple slice NALUs), the iOS decoder can be initialized successfully but fails to decode and output images. ​​Can VideoToolbox properly decode HEVC bitstreams when a single frame is split into multiple slice NALUs?​​ Key Observations: ​​1. Single-NALU frames​​ work fine. ​​2. Multi-NALU frames​​ (sliced I/P-frames) cause decoding failure. 3. The decoder session is created successfully (VTDecompressionSessionCreate returns no error).

I am creating an app that decodes H.265 elementary streams on iOS. I use VideoToolBox to decode from H.265 to NV12. The decoded data is enqueued in the CMSampleBufferDisplayLayer as a CMSampleBuffer. However, nothing is displayed in the VideoPlayerView. It remains black. The decoding in VideoToolBox is successful. I confirmed this by saving the NV12 data in the CMSampleBuffer to a file and displaying it using a tool. Why is nothing displayed in the VideoPlayerView? I can provide other source code as well. // // ContentView.swift // H265Decoder // // Created by Kohshin Tokunaga on 2025/02/15. // import SwiftUI struct ContentView: View { var body: some View { VStack { Text("H.265 Player (temp.h265)") .font(.headline) VideoPlayerView() .frame(width: 360, height: 640) // Adjust or make it responsive for iOS } .padding() } } #Preview { ContentView() } // // VideoPlayerView.swift // H265Decoder // // Created by Kohshin Tokunaga on 2025/02/15. // import SwiftUI import AVFoundation struct VideoPlayerView: UIViewRepresentable { // Return an H265Player as the coordinator, and start playback there. func makeCoordinator() -> H265Player { H265Player() } func makeUIView(context: Context) -> UIView { let uiView = UIView(frame: .zero) // Base layer for attaching sublayers uiView.backgroundColor = .black // Screen background color (for iOS) // Create the display layer and add it to uiView.layer let displayLayer = context.coordinator.displayLayer displayLayer.frame = uiView.bounds displayLayer.backgroundColor = UIColor.clear.cgColor uiView.layer.addSublayer(displayLayer) // Start playback context.coordinator.startPlayback() return uiView } func updateUIView(_ uiView: UIView, context: Context) { // Reset the frame of the AVSampleBufferDisplayLayer when the view's size changes. let displayLayer = context.coordinator.displayLayer displayLayer.frame = uiView.layer.bounds // Optionally update the layer's background color, etc. uiView.backgroundColor = .black displayLayer.backgroundColor = UIColor.clear.cgColor // Flush transactions if necessary CATransaction.flush() } } // // H265Player.swift // H265Decoder // // Created by Kohshin Tokunaga on 2025/02/15. // import Foundation import AVFoundation import CoreMedia class H265Player: NSObject, VideoDecoderDelegate { let displayLayer = AVSampleBufferDisplayLayer() private var decoder: H265Decoder? override init() { super.init() // Initial configuration for the display layer displayLayer.videoGravity = .resizeAspect // Initialize the decoder (delegate = self) decoder = H265Decoder(delegate: self) // For simple playback, set isBaseline to true decoder?.isBaseline = true } func startPlayback() { // Load the file "cars_320x240.h265" guard let url = Bundle.main.url(forResource: "temp2", withExtension: "h265") else { print("File not found") return } do { let data = try Data(contentsOf: url) // Set FPS and video size as needed let packet = VideoPacket(data: data, type: .h265, fps: 30, videoSize: CGSize(width: 1080, height: 1920)) // Decode as a single packet decoder?.decodeOnePacket(packet) } catch { print("Failed to load file: \(error)") } } // MARK: - VideoDecoderDelegate func decodeOutput(video: CMSampleBuffer) { // When decoding is complete, send the output to AVSampleBufferDisplayLayer displayLayer.enqueue(video) } func decodeOutput(error: DecodeError) { print("Decoding error: \(error)") } }

Hi All I have some problem when I using the IOS 18.4.1 I have iphone16 pro and ipad Air, both are updated to IOS 18.4.1 I tried to following sample code. However, when I run the app around 30 seconds to 1 minutes, the application would be crashed When I using another Ipad with IOS 17, it would not have the same problem. https://developer.apple.com/documentation/createml/creating-an-action-classifier-model https://developer.apple.com/documentation/createml/detecting_human_actions_in_a_live_video_feed#overview%29,

I created a virtual audio device to capture system audio with a sample rate of 44.1 kHz. After capturing the audio, I forward it to the hardware sound card using AVAudioEngine, also with a sample rate of 44.1 kHz. However, due to the clock sources being unsynchronized, problems occur after a period of playback. How can I retrieve the clock source of the hardware device and set it for the virtual device?

iOS 26 added smoothness to CIRoundedRectangleGenerator, for use with CIFilter.roundedRectangleGenerator. What should the smoothness value be to achieve the same corner curve as CALayerCornerCurve.continuous? Does it need to be calculated based on the extent size, if so, how?

PLATFORM AND VERSION :iOS 18.5 I wanted to bring to your attention a critical issue some of our production users are experiencing with the CoinOut app. Specifically, users are encountering a problem when attempting to capture photos of receipts using the app's customized camera feature. The camera, which utilizes AVCaptureVideoPreviewLayer and AVCaptureDevice, occasionally fails to load the preview, resulting in a black screen instead of the expected camera view. This camera blackout issue is significantly impacting the user experience as it prevents them from snapping photos of their receipts, which is a core functionality of the CoinOut app. Any help/suggestion to this issue would be greatly appreciated. STEPS TO REPRODUCE Open the app and click on camera icon. It will display camera to capture photo. Camera shows black for few production user's. class ViewController: UIViewController { @IBOutlet private weak var captureButton: UIButton! private var fillLayer: CAShapeLayer! private var previewLayer : AVCaptureVideoPreviewLayer! private var output: AVCapturePhotoOutput! private var device: AVCaptureDevice! private var session : AVCaptureSession! private var highResolutionEnabled: Bool = false private let sessionQueue = DispatchQueue(label: "session queue") override func viewDidLoad() { super.viewDidLoad() setupCamera() customiseUI() } @IBAction func startCamera(sender: UIButton) { didTapTakePhoto() } private func setupCamera() { let session = AVCaptureSession() session.sessionPreset = AVCaptureSession.Preset.high previewLayer = AVCaptureVideoPreviewLayer(session: session) output = AVCapturePhotoOutput() device = AVCaptureDevice.default(.builtInWideAngleCamera, for: AVMediaType.video, position: .back) if let device = self.device{ do{ let input = try AVCaptureDeviceInput(device: device) if session.canAddInput(input){ session.addInput(input)} else { print("\(#fileID):\(#function):\(#line) : Session Input addition failed") } if session.canAddOutput(output){ output.isHighResolutionCaptureEnabled = self.highResolutionEnabled session.addOutput(output) } else { print("\(#fileID):\(#function):\(#line) : Session Input high resolution failed") } previewLayer.videoGravity = .resizeAspectFill previewLayer.session = session sessionQueue.async { session.startRunning() } self.session = session self.session.accessibilityElementIsFocused() try device.lockForConfiguration() if device.isWhiteBalanceModeSupported(AVCaptureDevice.WhiteBalanceMode.autoWhiteBalance) { device.whiteBalanceMode = .autoWhiteBalance } else { print("\(#fileID):\(#function):\(#line) : isWhiteBalanceModeSupported no supported") } if device.isWhiteBalanceModeSupported(AVCaptureDevice.WhiteBalanceMode.continuousAutoWhiteBalance) { device.whiteBalanceMode = .continuousAutoWhiteBalance } else { print("\(#fileID):\(#function):\(#line) : isWhiteBalanceModeSupported no supported") } if device.isFocusModeSupported(.continuousAutoFocus) { device.focusMode = .continuousAutoFocus} else if device.isFocusModeSupported(.autoFocus) { device.focusMode = .autoFocus } device.unlockForConfiguration() } catch { print("\(#fileID):\(#function):\(#line) : \(error.localizedDescription)") } } else { print("\(#fileID):\(#function):\(#line) : Device found as nil") } } private func customiseUI() { let path = UIBezierPath(roundedRect: CGRect(x: 0, y: 0, width: self.view.bounds.width, height: self.view.bounds.height), cornerRadius: 0) let rectangleWidth = view.frame.width - (view.frame.width * 0.16) let x = (view.frame.width - rectangleWidth) / 2 let rectangleHeight = view.frame.height - (view.frame.height * 0.16) let y = (view.frame.height - rectangleHeight) / 2 let roundRect = UIBezierPath(roundedRect: CGRect(x: x, y: y, width: rectangleWidth, height: rectangleHeight), byRoundingCorners:.allCorners, cornerRadii: CGSize(width: 0, height: 0)) roundRect.move(to: CGPoint(x: self.view.center.x , y: self.view.center.y)) path.append(roundRect) path.usesEvenOddFillRule = true fillLayer = CAShapeLayer() fillLayer.path = path.cgPath fillLayer.fillRule = .evenOdd fillLayer.opacity = 0.4 previewLayer.addSublayer(fillLayer) previewLayer.frame = view.bounds view.layer.addSublayer(previewLayer) view.bringSubviewToFront(captureButton) } private func didTapTakePhoto() { let settings = self.getSettings(camera: self.device) if device.isAdjustingFocus { do { try device.lockForConfiguration() device.focusMode = .continuousAutoFocus device.unlockForConfiguration() device.addObserver(self, forKeyPath: "adjustingFocus", options: [.new], context: nil) } catch { print(error) } } else { output.capturePhoto(with: settings, delegate: self) } } func getSettings(camera: AVCaptureDevice) -> AVCapturePhotoSettings { var settings = AVCapturePhotoSettings() if let rawFormat = output.availableRawPhotoPixelFormatTypes.first { settings = AVCapturePhotoSettings(rawPixelFormatType: OSType(rawFormat)) } settings.isHighResolutionPhotoEnabled = self.highResolutionEnabled let previewPixelType = settings.availablePreviewPhotoPixelFormatTypes.first! let previewFormat = [kCVPixelBufferPixelFormatTypeKey as String: previewPixelType] as [String : Any] settings.previewPhotoFormat = previewFormat return settings } } extension ViewController: AVCapturePhotoCaptureDelegate { func photoOutput(_ output: AVCapturePhotoOutput, willCapturePhotoFor resolvedSettings: AVCaptureResolvedPhotoSettings) { AudioServicesDisposeSystemSoundID(1108) } func photoOutput(_ output: AVCapturePhotoOutput, didFinishProcessingPhoto photo: AVCapturePhoto, error: Error?) { guard let data = photo.fileDataRepresentation() else { return } let image = UIImage(data: data)! showImage(cropped: image) } func showImage(cropped: UIImage) { let vc = self.storyboard?.instantiateViewController(withIdentifier: "ImagePreviewViewController") as? ImagePreviewViewController vc?.captured = cropped self.present(vc!, animated: true) } }```