Hi all, I'm using Apple Sample Code below to create application using dockkit. "Controlling a DockKit accessory using your camera app" https://developer.apple.com/documentation/dockkit/controlling-a-dockkit-accessory-using-your-camera-app?changes=_8 I used vision hand recognition and put the observation data to dockAccessory.track, but Belkin or Insta360 devices never move on iPhone 16 Pro Max with iOS 18.3. If I use other functions like face search (system tracking) in the app, those work ok. I used Belkin and Insta360 Flow 2 Pro to reproduce the problem. My friend is also saying that the custom tracking feature was working fine on the OS 18 beta, but on recent iOS 18.3 that feature does not work. If I can get the iOS 18.0 beta then we can test that feature. But I cannot revert my iOS from 18.3 to the iOS 18.0 Beta. Regards, TO

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?

I'm using this library: https://github.com/Yummypets/YPImagePicker to capture photos. I've modified it slightly, and I'm using an older version. When testing on my iPhone 16e, ios 26, whenever I take a photo, I get the following two error messages: <<<< FigXPCUtilities >>>> signalled err=-17281 at <>:302 <<<< FigCaptureSourceRemote >>>> Fig assert: "err == 0 " at bail (FigCaptureSourceRemote.m:569) - (err=-17281) These error messages appear, but as far as I can tell, the photo comes through OK, and I can save the data no problem. I've even removed all my handling code to see if it was something I was doing. I don't really want to ship with these errors showing, but I also have no idea what can be causing this error to appear. chatgpt was not helpful diagnosing this. Does anyone know what can cause this error Is there a way I can see the source code to figure out if there's something I'm doing wrong here? It really seems like this is an internal apple error, or else I would have expected more details on the error relating to the code I've written. Any clues would be appreciated!

I have a flutter iOS app that has some simple sound FX for button clicks, swipes, etc. In simulator and on real device the sound works fine, but when i upload the app to testflight (and App store) the sound FX don't play. When I upload the app to my phone via xcode I am using the release profile so I don't see what the difference could be. I have also gone through the archive that i uploaded and verified that the sound files are indeed there. I have other flutter apps that use sound but non since the iOS 26 update. I've tried 3 different flutter sound libraries and all face the same issue. Wondering if anyone else is seeing this issue or if I'm missing a simple permission or something that has changed recently? Thanks in advanced

Hey, There seems to be an inconsistency when capturing a photo using QualityPrioritization.Quality on the iPhone 17 Pro Main wide Lens. If you zoom above "2x" the output image always has "-2.0ev" bias in the meta data and looks underexposued. This does not happen at zoom levels above 2, or if you set the QualityPrioritization to .Balanced. See below: with .Quality with .Balanced This does not happen on the other lenses. I'm using a simple set up and it is consistent across JPEG and ProRAW capture. I have a demo project if that is useful. Thanks, Alex

I am new to Swift and iOS development, and I have a question about video capture performance. Is it possible to capture video at a resolution of 4032×3024 while simultaneously running a vision/ML model on the video stream (e.g., using Vision or CoreML)? I want to know: whether iOS devices support capturing video at that resolution, whether the frame rate drops significantly at that scale, and whether it is practical to run a Vision/ML model in real-time while recording at such a high resolution. If anyone has experience with high-resolution AVCaptureSession setups or combining them with real-time ML processing, I would really appreciate guidance or sample code.

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 - We implemented the code to upload still images. We would like to do the same for Live Photos but we are not sure how to proceed since a Live Photo is composed of 2 resources that we would like to upload in the same job so that we keep the association between the 2 resources. Steps: A Live Photo is captured on the device. Our application is notified for new content in the Photo Library and the asset is queued for upload by our application. The system calls the background upload extension and the Live Photo is prepared for upload but we can schedule a job only for one resource (still photo or paired video) so we can not upload both resources in a single job. 2 - Is there a way to synchronise the application and the extension so that the application would not process the same data or execute the same requests than the extension. For example: our application is working with tokens and we would like to prevent those tokens to be consumed by the application and the extension at the same time. 3 - is there a command in xcode or terminal to start or stop the extension, something similar to what exists for processing tasks (https://developer.apple.com/documentation/backgroundtasks/starting-and-terminating-tasks-during-development).

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.

hello, I'm using VideoTololbox VTFrameRateConversionConfiguration to perform frame interpolation: https://developer.apple.com/documentation/videotoolbox/vtframerateconversionconfiguration?language=objc ,when using 640x480 vidoe input, I got error: Error ! Invalid configuration [VEEspressoModel] build failure : flow_adaptation_feature_extractor_rev2.espresso.net. Configuration: landscape640x480 [EpsressoModel] Cannot load Net file flow_adaptation_feature_extractor_rev2.espresso.net. Configuration: landscape640x480 Error: failed to create FRCFlowAdaptationFeatureExtractor for usage 8 Failed to switch (0x12c40e140) [usage:8, 1/4 flow:0, adaptation layer:1, twoStage:0, revision:2, flow size (320x240)]. Could not init FlowAdaptation initFlowAdaptationWithError fail tried 2048x1080 is ok.

Hello, I am wondering if it is possible to have audio from my AirPods be sent to my speech to text service and at the same time have the built in mic audio input be sent to recording a video? I ask because I want my users to be able to say "CAPTURE" and I start recording a video (with audio from the built in mic) and then when the user says "STOP" I stop the recording.

Hi all, I've been working on some personal programming projects and have gotten into using the Apple Music API. I'm currently looking to get a list of recent songs using the /v1/me/recent/played/tracks endpoint and it's working well. However, I know there are some songs I've listened to multiple times in a row, and those are not showing up as unique tracks when querying this endpoint. I'm only seeing a list of the different songs I've listened to lately, not a true list of the most recent plays on my account. Is this intended behavior or am I going about something incorrectly here? My query is using that endpoint & specifying the types to be only [songs]. Thanks in advance for any ideas or insight.

Hi team, In the Apple Music Feed datasets, we've noticed some unexpected values in the song and album tables. The primaryartists column from either song or album may contain a "non-default" artist name such as the katakana name shown in the example below: select id, name, namedefault, primaryartists from amf_song where id = '1698723329' id | name | namedefault | primaryartists ---------------------------------------- 1698723329 | {default=California} | California | [{id=1264818718, name=チャペル・ローン}] select * from amf_artist where id = '1264818718' id | name | namedefault | namepronunciation | ---------------------------------------------- 1264818718 | {default=Chappell Roan, ja=チャペル・ローン} | Chappell Roan | {ja=チャペルローン} | Shouldn't the primaryartists column be showing the namedefault instead of the Japanese language version? When can we expect this bug to resolved? Thanks,

Hello, As far as I know and in all of my testing there is no way for a user or a developer to change the frame rate of the video output on iPadOS. If you connect an iPad via a USB Hub or a USB to HDMI Adaptor and then connect it to an external monitor it will output at 59.94fps. I have a video app where a user monitors live video at 25fps and 30fps, they often output to an external display and there are times when the external display will stutter due to the mismatch in frame rate, ie. using 25fps and outputting at 59.94fps. I thought it was impossible to change the video output frame rate, then in V3.1 of the Blackmagic Camera App I saw an interesting change in their release notes: ‘Support for HDMI Monitoring at Sensor Rate and Resolution’ This means there is some way to modify it, not sure if this is done via a Private API that Apple has allowed Blackmagic to use. If so, how can we access this or is there a way to enable this that is undocumented? Thanks!

PHPhotoLibrary.authorizationStatus(for: .readWrite) == .authorized Iinfo.plist Privacy - Photo Library Usage Description set I check authorization before attempting to get the photoPickerItem.itemIdentifier, but every time the return value from itemIdentifier is nil. Seems I missing some permissions, but unsure why the system is still keeping _shouldExposeItemIdentifier set to false.

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

I develop a application with an uvc camera, this camera is a webcam, I use the AVFoundation library ,but when I run the code "[self.mCaptureSession startRunning]" ,I can not get the buffer, I already set the delegate, any answer will help.

Hello, I've discovered a buffer initialization bug in AVAudioUnitSampler that happens when loading presets with multiple zones referencing different regions in the same audio file (monolith/concatenated samples approach). Almost all zones output silence (i.e. zeros) at the beginning of playback instead of starting with actual audio data. The Problem Setup: Single audio file (monolith) containing multiple concatenated samples Multiple zones in an .aupreset, each with different sample start and sample end values pointing to different regions of the same file All zones load successfully without errors Expected Behavior: All zones should play their respective audio regions immediately from the first sample. Actual Behavior: Last zone in the zone list: Works perfectly - plays audio immediately All other zones: Output [0, 0, 0, 0, ..., _audio_data] instead of [real_audio_data] The number of zeros varies from event to event for each zone. It can be a couple of samples (<30) up to several buffers. After the initial zeros, the correct audio plays normally, so there is no shift in audio playback, just missing samples at the beginning. Minimal Reproduction 1. Create Test Monolith Audio File Create a single Wav file with 3 concatenated 1-second samples (44.1kHz): Sample 1: frames 0-44099 (constant amplitude 0.3) Sample 2: frames 44100-88199 (constant amplitude 0.6) Sample 3: frames 88200-132299 (constant amplitude 0.9) 2. Create Test Preset Create an .aupreset with 3 zones all referencing the same file: Pseudo code <Zone array> <zone 1> start : 0, end: 44099, note: 60, waveform: ref_to_monolith.wav; <zone 2> start sample: 44100, note: 62, end sample: 88199, waveform: ref_to_monolith.wav; <zone 3> start sample: 88200, note: 64, end sample: 132299, waveform: ref_to_monolith.wav; </Zone array> 3. Load and Test // Load preset into AVAudioUnitSampler let sampler = AVAudioUnitSampler() try sampler.loadAudioFiles(from: presetURL) // Play each zone (MIDI notes C4=60, D4=62, E4=64) sampler.startNote(60, withVelocity: 64, onChannel: 0) // Zone 1 sampler.startNote(62, withVelocity: 64, onChannel: 0) // Zone 2 sampler.startNote(64, withVelocity: 64, onChannel: 0) // Zone 3 4. Observed Result Zone 1 (C4): [0, 0, 0, ..., 0.3, 0.3, 0.3] ❌ Zeros at beginning Zone 2 (D4): [0, 0, 0, ..., 0.6, 0.6, 0.6] ❌ Zeros at beginning Zone 3 (E4): [0.9, 0.9, 0.9, ...] ✅ Works correctly (last zone) What I've Extensively Tested What DOES Work Separate files per zone: Each zone references its own individual audio file All zones play correctly without zeros Problem: Not viable for iOS apps with 500+ sample libraries due to file handle limitations What DOESN'T Work (All Tested) 1. Different Audio Formats: CAF (Float32 PCM, Int16 PCM, both interleaved and non-interleaved) M4A (AAC compressed) WAV (uncompressed) SF2 (SoundFont2) Bug persists across all formats 2. CAF Region Chunks: Created CAF files with embedded region chunks defining zone boundaries Set zones with no sampleStart/sampleEnd in preset (nil values) AVAudioUnitSampler completely ignores CAF region metadata Bug persists 3. Unique Waveform IDs: Gave each zone a unique waveform ID (268435456, 268435457, 268435458) Each ID has its own file reference entry (all pointing to same physical file) Hypothesized this might trigger separate buffer initialization Bug persists - no improvement 4. Different Sample Rates: Tested: 44.1kHz, 48kHz, 96kHz Bug occurs at all sample rates 5. Mono vs Stereo: Bug occurs with both mono and stereo files Environment macOS: Sonoma 14.x (tested across multiple minor versions) iOS: Tested on iOS 17.x with same results Xcode: 16.x Frameworks: AVFoundation, AudioToolbox Reproducibility: 100% reproducible with setup described above Impact & Use Case This bug severely impacts professional music applications that need: Small file sizes: Monolith files allow sharing compressed audio data (AAC/M4A) iOS file handle limits: Opening 400+ individual sample files is not viable on iOS Performance: Single file loading is much faster than hundreds of individual files Standard industry practice: Monolith/concatenated samples are used by EXS24, Kontakt, and most professional samplers Current Impact: Cannot use monolith files with AVAudioUnitSampler on iOS Forced to choose between: unusable audio (zeros at start) OR hitting iOS file limits No viable workaround exists Root Cause Hypothesis The bug appears to be in AVAudioUnitSampler's internal buffer initialization when: Multiple zones share the same source audio file Each zone specifies different sampleStart/sampleEnd offsets Key observation: The last zone in the zone array always works correctly. This is NOT related to: File permissions or security-scoped resources (separate files work fine) Audio codec issues (happens with uncompressed PCM too) Preset parsing (preset loads correctly, all zones are valid) Questions Is this a known issue? I couldn't find any documentation, bug reports, or discussions about this. Is there ANY workaround that allows monolith files to work with AVAudioUnitSampler? Alternative APIs? Is there a different API or approach for iOS that properly supports monolith sample files?

I wrote a Swift macOS app to control a PCI audio device. The code switches between the default output and input channels. As soon as I launch the Audio-Midi Setup utility, channel switching stops working. The driver properties allow switching, but the system doesn't respond. I have to delete the contents of /Library/Preferences/Audio and reset Core Audio. What am I missing? func setDefaultChannelsOutput() { guard let deviceID = getDeviceIDByName(deviceName: "PCI-424") else { return } let selectedIndex = DefaultChannelsOutput.indexOfSelectedItem if selectedIndex < 0 || selectedIndex >= 24 { return } let channel1 = UInt32(selectedIndex * 2 + 1) let channel2 = UInt32(selectedIndex * 2 + 2) var channels: [UInt32] = [channel1, channel2] var propertyAddress = AudioObjectPropertyAddress( mSelector: kAudioDevicePropertyPreferredChannelsForStereo, mScope: kAudioDevicePropertyScopeOutput, mElement: kAudioObjectPropertyElementWildcard ) let dataSize = UInt32(MemoryLayout<UInt32>.size * channels.count) let status = AudioObjectSetPropertyData(deviceID, &propertyAddress, 0, nil, dataSize, &channels) if status != noErr { print("Error setting default output channels: \(status)") } }

We’re developing an AVFoundation-based video recording app (4K @ 60 fps required for biomechanical analysis). On most devices this works perfectly (iPhone 12/14/15/16 non-Pro models), but on several iPhone Pro models (12 Pro, 13 Pro, 14 Pro, 15 Pro/Pro Max), we consistently get 4K 30 fps recordings—even when the device should support 4K 60 fps on the wide-angle camera. What we observe We configure the session for .hd4K3840x2160. We iterate through AVCaptureDevice.formats and select formats that: have 3840×2160 resolution support ≥60 fps (videoSupportedFrameRateRanges) On some Pro devices, this format search returns no results, even though: The Camera app records 4K60 fine. External references list the wide camera as 4K60 capable. The fallback becomes the device's default 4K30 format, so final files are 3840×2160 @ 30 fps. This happens immediately on app launch (not after heating), so not thermal-related. What we’ve tried Force selecting .builtInWideAngleCamera instead of dual/triple cameras. Disabling HDR (videoHDREnabled = false). Disabling low-light boost. Allowing 59.94 fps formats (in case exact 60.0 isn’t exposed). Logging all videoSupportedFrameRateRanges per format. What we’re seeing in logs On affected Pro devices, the capture device reports only 4K formats with maxFrameRate ≈ 30 fps, despite the hardware being able to do 4K60. Main question Has anyone encountered cases where 4K60 formats are available in the Camera app but not exposed through AVFoundation, especially on Pro models or multi-camera devices? Could HEVC/HDR capability or multi-camera constraints be preventing certain formats from appearing? Are there known conditions where 4K60 formats are hidden unless specific device configuration is applied? Any guidance on reliably locking 4K60 on iPhone Pro models via AVFoundation would be hugely appreciated.

I am developing an iOS application that supports screen mirroring to Google TV (or Chromecast with Google TV). My goal is to mirror the iPhone/iPad screen in real time to a Google TV device. What I Have Tried So Far I have explored multiple approaches but haven't found a direct way to achieve low-latency screen mirroring. Here are some of my findings: Google Cast SDK: Google Cast SDK is primarily designed for casting media (videos, images, audio) rather than real-time mirroring. It supports custom receiver applications, but there are no direct APIs for full screen mirroring. Casting a recorded video is possible, but it introduces latency and is not real-time. ReplayKit for Screen Capture: RPScreenRecorder.shared().startCapture(handler: ...) allows capturing the iPhone screen as a video stream. However, sending this stream to Google TV in real time is a challenge. I could potentially encode the video as HLS and stream it, but the delay is significant. RTSP/UDP Streaming: Some third-party libraries support RTSP/UDP streaming for real-time screen sharing. Google TV does not natively support RTSP, making this approach difficult. My Questions: Is it possible to achieve real-time screen mirroring on Google TV using Google Cast SDK? Does Google TV support WebRTC or any low-latency streaming protocol that can be used from iOS? Are there any alternative approaches to mirror an iOS screen to Google TV with minimal latency? I would appreciate any guidance, code examples, or references to relevant documentation.