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

MPVolumeView's showsRouteButton was deprecated (https://aninterestingwebsite.com/documentation/mediaplayer/mpvolumeview/showsroutebutton?language=objc). It's not clear how can we now hide this button without deprecation warning. The documentation is lacking. Please advise. Thank you!

In the AVP project, a selector pops up, only wanting to filter spatial videos. When selecting the material of one of the spatial videos, the selection result returns empty. How can we obtain the video selected by the user and get the path and the URL of the file The code is as follows： PhotosPicker(selection: $selectedItem, matching: .videos) { Text("Choose a spatial photo or video") } func loadTransferable(from imageSelection: PhotosPickerItem) -> Progress { return imageSelection.loadTransferable(type: URL.self) { result in DispatchQueue.main.async { // guard imageSelection == self.imageSelection else { return } print("加载成功的图片集合:(result)") switch result { case .success(let url?): self.selectSpatialVideoURL = url print("获取视频链接:(url)") case .success(nil): break // Handle the success case with an empty value. case .failure(let error): print("spatial错误：(error)") // Handle the failure case with the provided error. } } } }

Hello, Does anyone have a recipe on how to raycast VNFaceLandmarkRegion2D points obtained from a frame's capturedImage? More specifically, how to construct the "from" parameter of the frame's raycastQuery from a VNFaceLandmarkRegion2D point? Do the points need to be flipped vertically? Is there any other transformation that needs to be performed on the points prior to passing them to raycastQuery?

This is an issue with the Insta360 Flow Pro 2. My iOS app uses DockKit to control the gimbal; in particular, my app disables tracking and sends angular velocity commands to control the gimbal's orientation. I only try to modify the yaw (rotation around the vertical axis); never the pitch or yaw. Note that I don't send the gimbal to a particular orientation directly; I modify the velocity. Everything works great for a long period of time: typically for a continuous run of 4-6 hours; in the most recent case, I managed about 36 hours of continous operation before the following problem occurred. I came back to check on the system, and because no visual activity had occurred in the camera's field of view for a while, the phone had commanded the gimbal to rotate back to a yaw angle of 0 degrees. So the phone in the gimbal should have been looking straight ahead (i.e. the 0 degree yaw position), but it was definitely looking off at an angle. I've seen this twice now. The first time, when it should have been looking straight ahead, it was in fact looking 60 degrees off center. This time (caught on video, see below), it was off by 22 degrees from center. Here's the weird part: the gimbal reports this way off center positioning as zero degrees (well close enough to zero, like 0.2 or something that's fine). But, mechanically, the gimbal still knows where zero degrees is: if we double click on the trigger of the Flow Pro 2, which is supposed to reset the gimbal to 0 degrees yaw and pitch, the gimbal responds correctly and reorients to a 0 degree position. However, the yaw values it reports are not zero, but as shown in my video, 22 degrees off axis or so. Power cycling the gimbal and restarting immediately fixes the problem. Also, I switched from my app to the Insta360 app, which caused the phone to flip from landscape to portrait, then when I returned to my app and switched back to landscape, the gimbal now started reporting correct yaw angles. Is there a possibility this is a bug in the DockKit framework? Has anyone seen this? I have a case open with Insta360, but although it's clearly a software issue, it's not clear if it's in Insta360's code or the DockKit layer. Any ideas for how I can get out of this mode? My concern is that the phone is in a tripod about 10' off the floor, and not very accessible. Also, if all goes well, we may have about 50 of these systems running, and having to fix them one by one after a few hours is not good. For a demonstration of this bug, see the following video: https://octoparry.com/offset.MOV Any help greatly appreciated.

I'm seeing crashes in _MPRemoteCommandEventDispatch on iOS 26.x devices in 3 apps. According to Bugsnag logs they are: NSInternalInconsistencyException: event dispatch <_MPRemoteCommandEventDispatch: <MPRemoteCommandEvent: 0x11c049500 commandID=THV0 command=<MPRemoteCommand: 0x109ad1ea0 type=Play (0) enabled=YES handlers=[0x109b6a310]> sourceID=(null) ([HostedRoutingSessionDataSource] handleControlSendingCommand<2W5E>)> state:201> deallocated without calling continuation I attached a log from Xcode organizer matching Bugsnag crash. mpr_remote_command_event.crash When I set the brakpoint on the -[_MPRemoteCommandEventDispatch dealloc] I can see it it's hit every time I tap play or pause on locked screen play button. Thread 0 Crashed: 0 libsystem_kernel.dylib 0x00000002370420cc __pthread_kill + 8 (:-1) 1 libsystem_pthread.dylib 0x00000001e975c810 pthread_kill + 268 (pthread.c:1721) 2 libsystem_c.dylib 0x0000000198f8ff64 abort + 124 (abort.c:122) 3 libc++abi.dylib 0x000000018a7cf808 __abort_message + 132 (abort_message.cpp:66) 4 libc++abi.dylib 0x000000018a7be484 demangling_terminate_handler() + 304 (cxa_default_handlers.cpp:76) 5 libobjc.A.dylib 0x000000018a6cff78 _objc_terminate() + 156 (objc-exception.mm:496) 6 xxxxxxxxxxxxxx 0x00000001003a7db8 CPPExceptionTerminate() + 416 (BSG_KSCrashSentry_CPPException.mm:156) 7 libc++abi.dylib 0x000000018a7cebdc std::__terminate(void (*)()) + 16 (cxa_handlers.cpp:59) 8 libc++abi.dylib 0x000000018a7ceb80 std::terminate() + 108 (cxa_handlers.cpp:88) 9 CoreFoundation 0x000000018d7341c4 __CFRunLoopPerCalloutARPEnd + 256 (CFRunLoop.c:769) 10 CoreFoundation 0x000000018d70bb5c __CFRunLoopRun + 1976 (CFRunLoop.c:3179) 11 CoreFoundation 0x000000018d70aa6c _CFRunLoopRunSpecificWithOptions + 532 (CFRunLoop.c:3462) 12 GraphicsServices 0x000000022e31c498 GSEventRunModal + 120 (GSEvent.c:2049) 13 UIKitCore 0x00000001930ceba4 -[UIApplication _run] + 792 (UIApplication.m:3902) 14 UIKitCore 0x0000000193077a78 UIApplicationMain + 336 (UIApplication.m:5577) 15 xxxxxxxxxxxxxx 0x00000001000c0134 main + 308 (main.swift:15) 16 dyld 0x000000018a722e28 start + 7116 (dyldMain.cpp:1477) Is the crash happening when the app is being terminated? Thank you!

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 }

Hi everyone, I am currently on MacOS Tahoe (26.1), and for some weird reason my mac is not connecting via HDMI. To be accurate: it is connecting and the LG TV shows up in the Displays settings, but no image shows up in it, I have no idea why. This used to work as I've tried this cable before with the same exact tv. The cable is a basic Amazon Basics HDMI one. Allow me just to advanced this question a little: usually terminal commands are more advanced recommendations, whereas basic questions like "have you connected it right" are just a waste of time

Hello, I have been running into issues with setting nowPlayingInfo information, specifically updating information for CarPlay and the CPNowPlayingTemplate. When I start playback for an item, I see lock screen information update as expected, along with the CarPlay now playing information. However, the playing items are books with collections of tracks. When I select a new track(chapter) within the book, I set the MPMediaItemPropertyTitle to the new chapter name. This change is reflected correctly on the lock screen, but almost never appears correctly on the CarPlay CPNowPlayingTemplate. The previous chapter title remains set and never updates. I see "Application exceeded audio metadata throttle limit." in the debug console fairly frequently. From that a I figured that I need to minimize updates to the nowPlayingInfo dictionary. What I did: I store the metadata dictionary in a local dictionary and only set values in the main nowPlayingInfo dictionary when they are different from the current value. I kick off the nowPlayingInfo update via a task that initially sleeps for around 2 seconds (not a final value, just for my current testing). If a previous Task is active, it gets cancelled, so that only one update can happen within that time window. Neither of these things have been sufficient. I can switch between different titles entirely and the information updates (including cover art). But when I switch chapters within a title, the MPMediaItemPropertyTitle continues to get dropped. I know the value is getting set, because it updates on the lock screen correctly. In total, I have 12 keys I update for info, though with the above changes, usually 2-4 of them actually get updated with high frequency. I am running out of ideas to satisfy the throttling thresholds to accurately display metadata. I could use some advice. Thanks.

Hi, I just started to develop audio unit hosting support in my application. Offline rendering seems to work except that I hear no output, but why? I suspect with the player goes something wrong. I connect to CoreAudio in a different location in the code. Here are some error messages I faced so far: 2025-08-14 19:42:04.132930+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:42:04.151171+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:43:08.344530+0200 com.gsequencer.GSequencer[34358:18614927] AUAudioUnit.mm:1417 Cannot set maximumFramesToRender while render resources allocated. 2025-08-14 19:43:08.346583+0200 com.gsequencer.GSequencer[34358:18614927] [avae] AVAEInternal.h:104 [AVAudioSequencer.mm:121:-[AVAudioSequencer(AVAudioSequencer_Player) startAndReturnError:]: (impl->Start()): error -10852 ** (<unknown>:34358): WARNING **: 19:43:08.346: error during audio sequencer start - -10852 I have implemented an AVAudioEngine based AudioUnit host. Here I instantiate player and effect: /* audio engine */ audio_engine = [[AVAudioEngine alloc] init]; fx_audio_unit_audio->audio_engine = (gpointer) audio_engine; av_format = (AVAudioFormat *) fx_audio_unit_audio->av_format; /* av audio player node */ av_audio_player_node = [[AVAudioPlayerNode alloc] init]; /* av audio unit */ av_audio_unit_effect = [[AVAudioUnitEffect alloc] initWithAudioComponentDescription:[((AVAudioUnitComponent *) AGS_AUDIO_UNIT_PLUGIN(base_plugin)->component) audioComponentDescription]]; av_audio_unit = (AVAudioUnit *) av_audio_unit_effect; fx_audio_unit_audio->av_audio_unit = av_audio_unit; /* audio sequencer */ av_audio_sequencer = [[AVAudioSequencer alloc] initWithAudioEngine:audio_engine]; fx_audio_unit_audio->av_audio_sequencer = (gpointer) av_audio_sequencer; /* output node */ [[AVAudioOutputNode alloc] init]; /* audio player and audio unit */ [audio_engine attachNode:av_audio_player_node]; [audio_engine attachNode:av_audio_unit]; [audio_engine connect:av_audio_player_node to:av_audio_unit format:av_format]; [audio_engine connect:av_audio_unit to:[audio_engine outputNode] format:av_format]; ns_error = NULL; [audio_engine enableManualRenderingMode:AVAudioEngineManualRenderingModeOffline format:av_format maximumFrameCount:buffer_size error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("enable manual rendering mode error - %d", [ns_error code]); } ns_error = NULL; [[av_audio_unit AUAudioUnit] allocateRenderResourcesAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("Audio Unit allocate render resources returned error - ErrorCode %d", [ns_error code]); } Then I render in a dedicated thread. ns_error = NULL; [audio_engine startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio engine start - %d", [ns_error code]); } [av_audio_sequencer prepareToPlay]; ns_error = NULL; [av_audio_sequencer startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio sequencer start - %d", [ns_error code]); } [av_audio_player_node play]; while(is_running){ /* pre sync */ /* IO buffers */ av_output_buffer = (AVAudioPCMBuffer *) scope_data->av_output_buffer; av_input_buffer = (AVAudioPCMBuffer *) scope_data->av_input_buffer; /* fill input buffer */ /* schedule av input buffer */ frame_position = 0; // (gint64) ((note_offset * absolute_delay) + delay_counter) * buffer_size; av_audio_player_node = (AVAudioPlayerNode *) fx_audio_unit_audio->av_audio_player_node; AVAudioTime *av_audio_time = [[AVAudioTime alloc] initWithHostTime:frame_position sampleTime:frame_position atRate:((double) samplerate)]; [av_audio_player_node scheduleBuffer:av_input_buffer atTime:av_audio_time options:0 completionHandler:nil]; /* render */ ns_error = NULL; status = [audio_engine renderOffline:AGS_FX_AUDIO_UNIT_AUDIO_FIXED_BUFFER_SIZE toBuffer:av_output_buffer error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("render offline error - %d", [ns_error code]); } } regards, Joël

I'm using an AVCaptureSession to send video and audio samples to an AVAssetWriter. When I play back the resultant video, sometimes there is a significant lag between the audio compared with the video, so they're just not in sync. But sometimes they are, with the same code. If I look at the very first presentation time stamps of the buffers being sent to the delegate, via func captureOutput(_: AVCaptureOutput, didOutput sampleBuffer: CMSampleBuffer, from connection: AVCaptureConnection) I see something like this: Adding audio samples for pts time 227711.0855328798, Adding video samples for pts time 227710.778785374 That is, the clock for audio vs video is behind: the first audio sample I receive is at 11.08 something, while the video video sample is earlier in time, at 10.778 something. The times are the presentation time stamps of the buffer, and the outputPresentationTimeStamp is the exact same number. It feels like "video" vs the "audio" clock are just mismatched. This doesn't always happen: sometimes they're synced. Sometimes they're not. Any ideas? The device I'm recording is a webcam, on iPadOS, connected via the usb-c port.

I use htttps://api.music.apple.com/v1/me/library/playlists/${playlistId}/tracks to add tracks to a playlist I created. How do I DELETE tracks from the playlist? The documentation does not mention a method for this. I have tried calling DELETE methods in various combinations but nothing seems to work. Is this possible?

Hello, I'm working on a Flutter app targeting both Android and iOS, where I implemented ShazamKit. In order to achieve that, I first tried with the flutter_shazam_kit package, but since it's not maintained anymore, I forked it here, and tried to update it to meet the Google Play Store requirements, as you can see here: https://github.com/mregnauld/flutter_shazam_kit/tree/fix-16k Unfortunately, after trying everything, my app still doesn't meet the (not so) new 16 KB native library alignment. Also, I'm 100% sure it comes from that because the error message disappears if I remove that package from my app. So after investigating, it seems that the problem comes from the ShazamKit for Android (that you can find here: https://aninterestingwebsite.com/download/all/?q=Android%20ShazamKit), and especially the .so files in the .aar file. Is there anything I can do to fix that, or should I wait before the ShazamKit team fix that? I'm totally stuck with that so any help is highly appreciated. Thanks.

Dear Apple Developer Team, On iOS 26, the contents of PDF pages appear to be swapped. Could you please advise if there is a workaround or a planned fix for this issue? Steps to Reproduce: Download the attached PDF on iOS 26. Open the PDF in the Files app. Tap the PDF to view it in Quick Look. Navigate to page 5. Expected Result: The page number displayed at the bottom should be 5. Actual Result: The page number displayed at the bottom is 4. Issue: This is not limited to page 5—multiple page contents appear to be swapped. I have also submitted feedback via Feedback Assistant (FB20743531) on October 20. Best regards, Yoshihito Suezawa

Hi all, I'm trying to diagnose and resolve an issue with stuttering video playback using the standard AVPlayer. The video in question is a 4K, 39-second file in *.mov format, being played on an iOS device. It's served via a local HTTP server that proxies requests to a backend to fetch and process the content. The project uses end-to-end encrypted storage, which necessitates the proxy for handling data processing. While playback in offline scenarios is smooth, we are encountering issues with smooth playback during streaming. The same video streams smoothly on other platforms using the same connection, so network limitations are not a factor. On iOS, playback is consistently choppy, with pauses every 1-3 seconds. The video does not appear to buffer adequately for smooth playback. One particularly curious aspect is the seemingly random pattern of Content-Range requests made by the AVPlayer when streaming the video. Below is an example of the range requests:

Hi, I have just implemented an Audio Unit v3 host. AgsAudioUnitPlugin *audio_unit_plugin; AVAudioUnitComponentManager *audio_unit_component_manager; NSArray<AVAudioUnitComponent *> *av_component_arr; AudioComponentDescription description; guint i, i_stop; if(!AGS_AUDIO_UNIT_MANAGER(audio_unit_manager)){ return; } audio_unit_component_manager = [AVAudioUnitComponentManager sharedAudioUnitComponentManager]; /* effects */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_Effect; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } /* instruments */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_MusicDevice; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } But this doesn't show me Audio Unit v2 plugins, why? regards, Joël

mac os 系统版本：26.0 (25A354) Xcode版本：Version 26.0 (17A324) 项目编译报错 `SwiftExplicitDependencyCompileModuleFromInterface arm64 /Users/zhz/Library/Developer/Xcode/DerivedData/ModuleCache.noindex/AssetsLibrary-HTIJ05N58KN3.swiftmodule /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS26.0.sdk/usr/lib/swift/AssetsLibrary.swiftmodule/arm64e-apple-ios.swiftinterface:10:25: error: 'ALAssetsLibrary' is unavailable in iOS: Use PHPhotoLibrary from the Photos framework instead 8 | public import _StringProcessing 9 | public import _SwiftConcurrencyShims 10 | extension AssetsLibrary.ALAssetsLibrary { | `- error: 'ALAssetsLibrary' is unavailable in iOS: Use PHPhotoLibrary from the Photos framework instead 11 | #if compiler(>=5.3) && $NonescapableTypes 12 | @available(iOS, introduced: 9.0, deprecated: 9.0, obsoleted: 26.0) /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS26.0.sdk/System/Library/Frameworks/AssetsLibrary.framework/Headers/ALAssetsLibrary.h:80:12: note: 'ALAssetsLibrary' was obsoleted in iOS 26.0 78 | 79 | OS_EXPORT AL_DEPRECATED(4, "Use PHPhotoLibrary from the Photos framework instead") 80 | @interface ALAssetsLibrary : NSObject { | `- note: 'ALAssetsLibrary' was obsoleted in iOS 26.0 81 | @package 82 | id _internal; /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS26.0.sdk/usr/lib/swift/AssetsLibrary.swiftmodule/arm64e-apple-ios.swiftinterface:1:1: error: failed to build module 'AssetsLibrary'; this SDK is not supported by the compiler (the SDK is built with 'Apple Swift version 6.2 effective-5.10 (swiftlang-6.2.0.17.14 clang-1700.3.17.1)', while this compiler is 'Apple Swift version 6.2 effective-5.10 (swiftlang-6.2.0.19.9 clang-1700.3.19.1)'). Please select a toolchain which matches the SDK.

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

In iOS 26, AVSpeechSynthesizer read Mandarin into Cantonese pronunciation. No matter how you set the language, and change the settings of my phone system, it doesn't work. let utterance = AVSpeechUtterance(string: "你好啊") //let voice = AVSpeechSynthesisVoice(language: "zh-CN") // not work let voice = AVSpeechSynthesisVoice(language: "zh-Hans") // not work too utterance.voice = voice et synth = AVSpeechSynthesizer() synth.speak(utterance)

I’m using the shared instance of AVAudioSession. After activating it with .setActive(true), I observe the outputVolume, and it correctly reports the device’s volume. However, after deactivating the session using .setActive(false), changing the volume, and then reactivating it again, the outputVolume returns the previous volume (before deactivation), not the current device volume. The correct volume is only reported after the user manually changes it again using physical buttons or Control Center, which triggers the observer. What I need is a way to retrieve the actual current device volume immediately after reactivating the audio session, even on the second and subsequent activations. Disabling and re-enabling the audio session is essential to how my application functions. I’ve tested this behavior with my colleagues, and the issue is consistently reproducible on iOS 18.0.1, iOS 18.1, iOS 18.3, iOS 18.5 and iOS 18.6.2. On devices running iOS 17.6.1 and iOS 16.0.3, outputVolume correctly reflects the current volume immediately after calling .setActive(true) multiple times.