Because I want to control the grid size and number of HEIC images myself, I decided to perform HEVC encoding manually and then generate the HEIC image. Previously, I used VTCompressionSession to accomplish this task, and the results were satisfactory. It worked perfectly on iOS 16 through iOS 18 — in other words, it was able to generate correct HEVC encoding, and its CMFormatDescription should also have been correct, since I relied on it to generate the decoderConfig; otherwise, the final image would have decoding issues. However, it can no longer generate a valid HEIC image on a physical device running iOS 26. Interestingly, it still works fine on the iOS 26 simulator — it only fails on real hardware. The abnormal result is that the image becomes completely black, although the image dimensions are still correct. After my troubleshooting, I suspect that the encoding behavior of VTCompressionSession has been modified on iOS 26, which causes the final hvc1 encoding I pass in to be incorrect. I created a VTCompressionSession using the following configuration. var newSession: VTCompressionSession! var status = VTCompressionSessionCreate( allocator: kCFAllocatorDefault, width: Int32(frameSize.width), height: Int32(frameSize.height), codecType: kCMVideoCodecType_HEVC, encoderSpecification: nil, imageBufferAttributes: nil, compressedDataAllocator: nil, outputCallback: nil, refcon: nil, compressionSessionOut: &newSession ) try check(status, VideoToolboxErrorDomain) let properties: [CFString: Any] = [ kVTCompressionPropertyKey_AllowFrameReordering: false, kVTCompressionPropertyKey_AllowTemporalCompression: false, kVTCompressionPropertyKey_RealTime: false, kVTCompressionPropertyKey_MaximizePowerEfficiency: false, kVTCompressionPropertyKey_ProfileLevel: profileLevel, kVTCompressionPropertyKey_Quality: quality.rawValue, ] status = VTSessionSetProperties(newSession, propertyDictionary: properties as CFDictionary) try check(status, VideoToolboxErrorDomain) { VTCompressionSessionInvalidate(newSession) } Then use the following code to encode each Grid of the image. let status = VTCompressionSessionEncodeFrame( session, imageBuffer: buffer, presentationTimeStamp: presentationTimeStamp, duration: frameDuration, frameProperties: nil, infoFlagsOut: nil) { [weak self] status, _, sampleBuffer in try check(status, VideoToolboxErrorDomain) if let sampleBuffer { let encodedImage = try self.encodedImage(from: sampleBuffer) // handle encodedImage } } try check(status, VideoToolboxErrorDomain) If I try to display this abnormal image in the App, my console outputs the following error, so it can be inferred that the issue probably occurred during decoding. createImageBlock:3029: *** ERROR: CGImageBlockCreate {0, 0, 2316, 6176} - data is NULL callDecodeImage:2411: *** ERROR: decodeImageImp failed - NULL _blockArray createImageBlock:3029: *** ERROR: CGImageBlockCreate {0, 0, 2316, 6176} - data is NULL callDecodeImage:2411: *** ERROR: decodeImageImp failed - NULL _blockArray createImageBlock:3029: *** ERROR: CGImageBlockCreate {0, 0, 2316, 6176} - data is NULL callDecodeImage:2411: *** ERROR: decodeImageImp failed - NULL _blockArray It needs to be emphasized again that this code used to work fine in the past, and the issue only occurs on an iOS 26 physical device. I noticed that iOS 26 has introduced many new properties, but I’m not sure whether some of these new properties must be set in the new system, and there’s no information about this in the official documentation.

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.

I am using Apple's original Lightning Digital AV-adapter (Lightning-to-HDMI dongle) to connect my iPhone to an external display via a HDMI cable. I need to synchronize rendering with the external display's refresh rate, so I create a new CADisplayLink tied to the external display's UIScreen: UIScreen.screens[externalDisplayIdx].displayLink(withTarget:, selector:). The callback is being called regularly, but with increasing delay relative to the CADisplayLink.timestamp, so the next time the callback is called, I have less and less time to draw the next frame (see the snippet below). Assuming 60 FPS, the value of secondsTillDeadline starts at an arbitrary value in the range of approx -0.0001 to 0.0166667, and then it slowly decreases towards zero (and for a brief period it goes into small negative numbers). Once it reaches zero, it flips back to 0.0166667 and continues to decrease again. This cycle repeats indefinitely. Changing the external display's resolution (UIScreen's mode) or the CADisplayLink's preferredFrameRateRange to a lower FPS does not seem to have any effect on the temporal drifting (even the rate of change seem to be the same). When I create a new CADisplayLink for the iPhone's main screen, the value of secondsTillDeadline is stable, it does not drift and it is very close to 0.0166667, as expected. Is this drift caused by the external monitor or by Apple's Lightning-to-HDMI dongle ...or is the problem somewhere else? Can the drifting be stopped? func onDisplayLinkUpdate(displayLink: CADisplayLink) { // Gradually decreases from 0.01667 to -0.0001, then flips back to 0.01667 and continues to decrease let secondsTillDeadline = displayLink.targetTimestamp - CACurrentMediaTime() }

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

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 encountering errors while using AVAudioEngine with voice processing enabled (setVoiceProcessingEnabled(true)) in scenarios where the input and output audio devices are not the same. This issue arises specifically with mismatched devices, preventing the application from functioning as expected. Works: Paired devices (e.g., MacBook Pro mic → MacBook Pro speakers) Fails: Mismatched devices (e.g., AirPods mic → MacBook Pro speakers) When using paired input and output devices: The setup works as expected. Example: MacBook Pro microphone → MacBook Pro speakers. When using mismatched devices: AVAudioEngine setup fails during aggregate device construction. Example: AirPods microphone → MacBook Pro speakers. Error logs indicate a channel count mismatch. Here are the partial logs. Due to the content limit, I cannot post the entire logs. AUVPAggregate.cpp:1000 client-side input and output formats do not match (err=-10875) AUVPAggregate.cpp:1036 err=-10875 AVAEInternal.h:109 [AVAudioEngineGraph.mm:1344:Initialize: (err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)): error -10875 AggregateDevice.mm:329 Failed expectation of constructed aggregate (312): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (312): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (336): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (336): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AudioHardware-mac-imp.cpp:3484 AudioDeviceSetProperty: no device with given ID AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (348): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (348): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) Is it possible to use voice processing with different input/output devices? If yes, are there any specific configurations required to handle mismatched devices? How can we resolve channel count mismatch errors during aggregate device construction? Are there settings or API adjustments to enforce compatibility between input/output devices? Are there any workarounds or alternative approaches to achieve voice processing functionality with mismatched devices? For instance, can we force an intermediate channel configuration or downmix input/output formats?

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:

Hey there, I just upgraded to Mac OS Tahoe ,son an apple MacBook Pro 2019 16inch. am using IntellijIDEA and Flutter to develop a mobile app which I test on the simulator app running iOS 18.4 . the issue: when I start the simulator app. ( while in the loading phase and in the operation phase as well ), the audio from an already open YouTube tab on safari (this happens on chrome browser as well). the sound glitches and becomes Noise. a fix I found online is to kill the audio deamon on Mac OS, This works using the command: "sudo killall coreaudiod" this kills the audio process, (while the emulator is operational), then the macOS restarts the audio deamon then the audio works fine alongside with the simulator being open. I just want to ask is there a permanent fix for this? is Apple working on a fix for this in the upcoming update?

I am unable to find any clearcut documentation on configuring AVCaptureSession pipeline to capture video with proResRAW codec type, which is 16 bit format. Is it supported only with AVCaptureMovieFileOutput or one can have AVCaptureVideoDataOutput emitting 16-bit sample buffers that can be vended to AVAssetWriter?

Hi, I'm currently developping an AVB hardware device, and I'm currently stuck because because the apple AVB stack is throwing me errors without much informations. Is there any way to have more information about these assertions and why they are happening ? Furtermore is there any documentation on theAppleAVBAudio module ? It would be very handy Here are the logs shown in the console: Filtering the log data using "process == "coreaudiod"" Timestamp Thread Type Activity PID TTL 2025-12-05 15:44:27.087043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.087545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.088043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.088546+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.089043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.089545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.090043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.090545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.091043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.091545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.092044+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.092544+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.093044+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.093552+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.094050+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.094543+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533

I'm capturing video stream from GoPro camera (I demux UDP MPEG-TS packets) and create CMSampleBuffers from them, this works fine when I display them using CMSampleBufferLayer. However when I dump them to disk using AVAssetWriter and then playback it with AVPlayer, AVPlayer has problems with scrubbing, it also cannot render previous frames, it needs to go back to key frames. Also thumbnails generated with AVAssetImageGenerator are mostly distorted and green, even though I set the requestedTimeToleranceAfter longer than the key frames frequency. When I re-encode saved video once again with AVAssetExportSession and play it back then I can scrub the video just fine. Is it because re-transcoding adds additional metadata to enable generating frames when rewinding the video and scrubbing? If so is there a way to achieve it with AVAssetWriter without much time penalty? I need the dump/save operation to be very fast. I also considered the following: Instead of de-muxing video and creating CMSampleBuffers, maybe I could directly dump the stream to disk and somehow add moov atoms with timing information. Would this approach work? If so where I can find information how to do it? Thank you!

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 have sent in a feedback report (FB18222398) but I have no idea if anyone has looked at it. I know from past experiences that Apple devs do look at these forums. This applies to each of the betas, 1, 2 and 3. I have created a new Personal Voice with each beta. I create a personal voice in English. When it's done processing, I tap Preview and it says in English what is expected. But after some time, an hour or a day, the language of the voice file changes languages and no longer works properly. If I press Preview it is no longer intelligible. I have a text to speech app and initially the created voice works but then when the language of the file changes, it no longer works. I have run an app on my iphone through Xcode that prints to the console the voices installed on the device with the language. Currently this is the voice file: Voice Identifier: com.apple.speech.personalvoice.AAA9C6F2-9125-475F-BA2F-22C63274991D Language: es-MX and on a second device the same personal voice is in a different language: Voice Identifier: com.apple.speech.personalvoice.AAA9C6F2-9125-475F-BA2F-22C63274991D Language: zh-CN Although, a previous personal voice file that listed as Spanish-Mexican played in English with a Spanish accent or when playing Spanish text, it sounded almost perfect. This current personal voice doesn't do that, and is unintelligible. Previous attempts have converted to Chinese. I hope someone can look into this.

When I use the ColorSync Utility to convert Display P3 color (1, 0, 0) to an XYZ color, the result is (0.5151, 0.2412, -0.0011). I expected that result because that is identical to the red colorant tristimulus value in the Display P3.icc file.When I use the CGColor converted method to do the same, the XYZ color is approximately (0.5151, 0.2412, 0.0). Note that the third element is 0.0 whereas it is -0.0011 when using the ColorSync Utility. I have printed out the Z component to 16 digits of precision, and Z is all 0s. It appears that the CGColor converted method is clamping the result from 0 to 1.My questions are:1. Which conversion is correct? The ColorSync utility or the CGColor converted method?2. I am not a color specialist, but I thought that the XYZ components should never be negative. If so, is the colorant tristimulus value in the Display P3.icc file wrong?3. Because CGColor clamped the Z component to 0, the XYZ color cannot be converted back exactly or closely to the Display P3 color (1, 0, 0). I would have expected to be able to go back and forth between the two color spaces when starting from a valid P3 Display color especially since the XYZ color space completely encompasses the P3 Display color space. Is that not true?4. Is (1, 0, 0) an invalid Display P3 color? If so, I can understand the peculiar results. I'm not sure how I would know if a Display P3 color is valid or not. (I only know that the component values must be from 0 to 1.) I think it is valid because Apple uses that color in the UIColor API Reference in an example.

I have a crash related to playing video in AVPlayerViewController and AVQueuePlayer. I download the video locally from the network and then initialize it using AVAsset and AVPlayerItem. Can't reproduce locally, but crashes occur from firebase crashlytics only for users starting with iOS 18.4.0 with this trace: Crashed: com.apple.avkit.playerControllerBackgroundQueue 0 libobjc.A.dylib 0x1458 objc_retain + 16 1 libobjc.A.dylib 0x1458 objc_retain_x0 + 16 2 AVKit 0x12afdc __77-[AVPlayerController currentEnabledAssetTrackForMediaType:completionHandler:]_block_invoke + 108 3 libdispatch.dylib 0x1aac _dispatch_call_block_and_release + 32 4 libdispatch.dylib 0x1b584 _dispatch_client_callout + 16 5 libdispatch.dylib 0x6560 _dispatch_continuation_pop + 596 6 libdispatch.dylib 0x5bd4 _dispatch_async_redirect_invoke + 580 7 libdispatch.dylib 0x13db0 _dispatch_root_queue_drain + 364 8 libdispatch.dylib 0x1454c _dispatch_worker_thread2 + 156 9 libsystem_pthread.dylib 0x4624 _pthread_wqthread + 232 10 libsystem_pthread.dylib 0x19f8 start_wqthread + 8

Hello there, I'm trying to implement feature which uses AirPlay with Apple TV. I want to disconnect from the device programmatically when something happens. Under something I mean a situation when a user wants to stop broadcasting (for example close the PiP window on his phone). I use this snippet: try audioSession.setCategory(.playAndRecord, options: .defaultToSpeaker) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) It works fine sometimes but not always (it works on iOS 18 but it doesn't on iOS 17 or ). So I thought it's a bug and create a ticker to feedback assistant (FB21220013). The support told me write a post on the forum.

When using AVSampleBufferDisplayLayer to play uncompressed H.264 and H.265 video with B-frames more than 7, frame drops occur. The more B-frames there are, the more noticeable the frame drops become, for example 15 bframes. Use FFmpeg to transcode a video file with visible timestamps and frame numbers (x264 or x265 ): ffmpeg -i test.mp4 -vf "drawtext=fontsize=45:text=%{pts} %{n}:y=400" -c:v libx264 -x264-params "bframes=15:b-adapt=0" -crf 30 -y x264_bf15.mp4 ffmpeg -i test.mp4 -vf "drawtext=fontsize=45:text=%{pts} %{n}:y=400" -c:v libx265 -x265-params "bframes=15:b-adapt=0" -crf 30 -y x265_bf15.mp4 Use the demo player from this repository to reproduce the issue: https://github.com/msfrms/CustomPlayer frame drops can be observed. And following log can be found in devices console. mediaserverd <<<< IQ-CA >>>> piqca_gmstats_dump: FIQCA(0x1266f4000) recent frames: enqueued: 184, displayed: 138, dropped: 42, flushed: 0, evicted: 3, >16ms late: 2 PS. I was using iphone11 iOS14.6, to replay this issue. May I ask why frame drops occur in this case? Is there any configuration or API usage change that could help fix the frame drop issue? Many thanks!

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

Add RPSystemBroadcastPickerView to the app, After clicking, no method of SampleHandler is triggered

the problem is when using HLS live stream with AVPlayer on iOS/ tvOS the player chooses first highest bandwidth then slowly steps down to lowest (within 1-3min) and eventually steps up again then repeats to step down. the AVPlayer error log sends events: errorStatusCode: -12888, errorDomain: Optional("CoreMediaErrorDomain"), errorComment: Optional("The operation couldn't be completed. (CoreMediaErrorDomain error -12888 - Playlist File unchanged for longer than 1.5 * target duration we use standard segments in CMAF format, 2sec duration #EXTM3U #EXT-X-VERSION:6 #EXT-X-TARGETDURATION:2 #EXT-X-MEDIA-SEQUENCE:147065903 #EXT-X-MAP:URI="video_1_4660000_init.mp4?device_profile=cmaf_cbcs_verimatrix_cei%26seg_size=2%26cmaf=2" #EXT-X-PROGRAM-DATE-TIME:2025-04-30T12:51:07 #EXTINF:2.000, video_1_4660000_t17460174670001555.mp4?device_profile=cmaf_cbcs_verimatrix_cei%26seg_size=2%26cmaf=2 #EXTINF:2.000, video_1_4660000_t17460174690001555.mp4?device_profile=cmaf_cbcs_verimatrix_cei%26seg_size=2%26cmaf=2 #EXTINF:2.000, video_1_4660000_t17460174710001555.mp4?device_profile=cmaf_cbcs_verimatrix_cei%26seg_size=2%26cmaf=2 when using 6sec segments the player stays stable at highest bandwidth. is there a way to avoid this error? in AVPlayer or HLS configuration?