Hello, I am getting the following error while attempting to run my LockedCameraCapture compatible app on an iOS 15 device: dyld[434]: Library not loaded: '/System/Library/Frameworks/LockedCameraCapture.framework/LockedCameraCapture' Referenced from: '/private/var/containers/Bundle/Application/.../MyApp.app/MyApp.debug.dylib' Reason: tried: '/System/Library/Frameworks/LockedCameraCapture.framework/LockedCameraCapture' (no such file) Of course iOS 15 doesn't have the library for LockedCameraCapture, but I have had no issue including Lock Screen Widgets (which require iOS 16), so I am not sure why the error is popping up. Thank you!

The operation couldn’t be completed. (CoreMediaErrorDomain error -19156 - The operation couldn’t be completed. (CoreMediaErrorDomain error -19156.

If new photo is added to library and app is not running in foreground or was not opened after the new photo was added but the app is having full access to gallery, can it access, read the new photo - If the app is not specifically a cloud syncing app, can it have this attached function, suppose it is a game app or beauty camera app?

I'm working on a photo app and I want to allow the user to display, edit and delete photos. I can fetch all photos using PHAsset.fetchAssets(with: options). This works as intended. However, I can't seem to find a way to prevent the user from seeing photos from a Shared Library. The PHAssetSourceType only contains typeCloudShared to only show items from a specific album; not library. How can I filter by iCloud Shared Library?

Hi all, I'm working on a custom Metal-based video pipeline using AVCaptureVideoDataOutput, and I've run into an unexpected issue related to exposure. Setup: I'm capturing video frames using kCVPixelFormatType_420YpCbCr8BiPlanarFullRange. In my Metal shader, I: Convert YCbCr (full range Rec.709) to linear Rec.709 RGB. Apply Rec.709 → sRGB gamma encoding. Output to .bgra8Unorm_srgb via MTKView. Everything renders correctly in terms of colorspace math, but the image appears significantly brighter (~+3 stops EV) compared to AVCaptureVideoPreviewLayer and the native iOS Camera app under the same camera exposure settings. What I’ve verified: The color transforms are correct: YCbCr709 to RGB, then linear to sRGB. I'm not applying any tone mapping or aggressive look LUTs yet. Camera exposure is locked using: device.setExposureModeCustom(duration: ..., iso: ...) The same EV (e.g., ISO 50, 1/125s, f/5.6) on my iPhone appears visually 3 stops brighter than on my digital cameras (Sony/Canon etc). To match the look of the preview layer or camera app, I have to simulate a ~–3 EV shift in my custom pipeline. Questions: Is AVCaptureVideoPreviewLayer applying extra tone mapping, digital gain, or contrast shaping (like OOTF etc)? Does the camera ISP expose "hotter" (i.e., with more light) internally for the preview layer than what we get in video frame buffers? Is there a standard way to compensate for this ISP behavior in custom pipelines using AVCaptureVideoDataOutput? Can this be accounted for using metadata (e.g., exposure bias, gain, gamma curve)?

Hi all, with my app ScreenFloat, you can record your screen, along with system- and microphone audio. Those two audio feeds are recorded into separate audio tracks in order to individually remove or edit them later on. Now, these recordings you create with ScreenFloat can be drag-and-dropped to other apps instantly. So far, so good, but some apps, like Slack, or VLC, or even websites like YouTube, do not play back multiple audio tracks, just one. So what I'm trying to do is, on dragging the video recording file out of ScreenFloat, instantly baking together the two individual audio tracks into one, and offering that new file as the drag and drop file, so that all audio is played in the target app. But it's slow. I mean, it's actually quite fast, but for drag and drop, it's slow. My approach is this: "Bake together" the two audio tracks into a one-track m4a audio file using AVMutableAudioMix and AVAssetExportSession Take the video track, add the new audio file as an audio track to it, and render that out using AVAssetExportSession For a quick benchmark, a 3'40'' movie, step 1 takes ~1.7 seconds, and step two adds another ~1.5 seconds, so we're at ~3.2 seconds. That's an eternity for a drag and drop, where the user might cancel if there's no immediate feedback. I could also do it in one step, but then I couldn't use the AV*Passthrough preset, and that makes it take around 32 seconds then, because I assume it touches the video data (which is unnecessary in this case, so I think the two-step approach here is the fastest). So, my question is, is there a faster way? The best idea I can come up with right now is, when initially recording the screen with system- and microphone audio as separate tracks, to also record both of them into a third, muted, "hidden" track I could use later on, basically eliminating the need for step one and just ripping the two single audio tracks out of the movie and only have the video and the "hidden" track (then unmuted), but I'd still have a ~1.5 second delay there. Also, there's the processing and data overhead (basically doubling the movie's audio data). All this would be great for an export operation (where one expects it to take a little time), but for a drag-and-drop operation, it's not ideal. I've discarded the idea of doing a promise file drag, because many apps do not accept those, and I want to keep wide compatibility with all sorts of apps. I'd appreciate any ideas or pointers. Thank you kindly, Matthias

Is there a recommended way on macOS 26 Tahoe to take a CoreAudio AudioObjectID and use it to lookup the underlying USB LocationID? I previously used AudioObjectID to query the corresponding DeviceUID with kAudioDevicePropertyDeviceUID. Then I queried for the IOService matching kIOAudioEngineClassName with property kIOAudioEngineGlobalUniqueIDKey matching DeviceUID, and I loaded kUSBDevicePropertyLocationID from the result. This fails on macOS 26, because the IO Registry for the device has an entry for usbaudiod rather than AppleUSBAudioEngine, and usbaudiod does not include a kIOAudioEngineGlobalUniqueIDKey property (or any other property to map it to a CoreAudio DeviceUID). My use-case here is a piece of audio recording software that allows configuring a set of supported audio devices via USB HID prior to recording. I present the user with a list of CoreAudio devices to use, but without a way to lookup the underlying USB LocationID, I cannot guarantee that the configured device matches the selected device (e.g. if the user plugged in two identical microphones).

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.

Hello everyone, I'm implementing the new AVInputPickerInteraction API on iOS 26 to allow users to select their microphone from a custom settings menu before recording. The implementation seems correct, but I'm encountering a strange issue where the input selection immediately reverts to the previous device. The Situation: The picker is presented correctly via a manual call to .present(). I can see all available inputs (e.g., "iPhone Microphone" and "AirPods"). The current input is "iPhone Microphone". I tap on "AirPods". The UI updates to show "AirPods" as selected for a fraction of a second, then immediately jumps back to "iPhone Microphone". The same thing happens in reverse. It seems like the system is automatically reverting the audio route change requested by the picker. My Implementation: My setup follows the standard pattern discussed in the WWDC sessions. Setup Code: This setup is performed once before the user can trigger the picker. @available(iOS 26.0, *) var inputPickerInteraction: AVInputPickerInteraction? // Note: The AVAudioSession is configured to .playAndRecord // and set to active elsewhere in the code before this setup is called. if #available(iOS 26.0, *) { // Setup the picker let picker = AVInputPickerInteraction() self.inputPickerInteraction = picker self.view.addInteraction(picker) // Added to establish context } Presentation Code: When a user selects "Change Input" from my custom settings menu, I call .present() on the main thread. // In a delegate method from a custom menu if #available(iOS 26.0, *) { DispatchQueue.main.async { self.inputPickerInteraction?.present(animated: true) } } What I've already checked: The AVAudioSession is active and its category is .playAndRecord. The inputPickerInteraction object is not nil. The .present() method is being called on the main thread. The picker is added to a view using view.addInteraction() in the setup phase. I've reviewed my code to ensure there is no other logic that could be manually resetting the AVAudioSession's preferred input. Has anyone else experienced this behavior? I suspect this might be a bug in the new API, but I want to make sure I'm not missing a crucial step in managing the AVAudioSession state. Any insights or potential workarounds would be greatly appreciated. Thank you.

Since iOS/iPadOs/tvOS 18 then we have run into a new problem with streaming of FairPlay encrypted video. On the affected streams then the audio plays perfectly but the video freezes for periods of a few seconds, so it will freeze for 5s or so, then be OK for a few seconds then freeze again. It is entirely reproducible when all the following are true the video streams were produced by a particular encoder (or particular settings, not sure on that) the video must be encrypted device is running some variety of iOS 18 (or iPadOS or tvOS) the device is an affected device Known devices are AppleTV 4K 2nd Gen iPad Pro 11" 1st and 2nd gen Devices known not to show the problem are all other AppleTV models iPhone 13 Pro and 16 Pro If we stream the same content, but unencrypted, then it plays perfectly, or if you play the encrypted stream on, say, tvOS 17. When the freezing occurs then we can see in the console logs repeating blocks of lines like the following default 18:08:46.578582+0000 videocodecd AppleAVD: AppleAVDDecodeFrameResponse(): Frame# 5771 DecodeFrame failed with error 0x0000013c default 18:08:46.578756+0000 videocodecd AppleAVD: AppleAVDDecodeFrameInternal(): failed - error: 316 default 18:08:46.579018+0000 videocodecd AppleAVD: AppleAVDDecodeFrameInternal(): avdDec - Frame# 5771, DecodeFrame failed with error: 0x13c default 18:08:46.579169+0000 videocodecd AppleAVD: AppleAVDDisplayCallback(): Asking fig to drop frame # 5771 with err -12909 - internalStatus: 315 also more relevant looking lines: default 18:17:39.122019+0000 kernel AppleAVD: avdOutbox0ISR(): FRM DONE (cid: 2.0, fno: 10970, codecT: 1) FAILED!! default 18:17:39.122155+0000 videocodecd AppleAVD: AppleAVDDisplayCallback(): Asking fig to drop frame # 10970 with err -12909 - internalStatus: 315 default 18:17:39.122221+0000 kernel AppleAVD: ## client[ 2.0] @ frm 10970, errStatus: 0x10 default 18:17:39.122338+0000 kernel AppleAVD: decodeFailIdentify(): VP error bit 4 has EP3B0 error default 18:17:39.122401+0000 kernel AppleAVD: processHWResponse(): clientID 2.0 frameNumber 10970 error 315, offsetIndex 10, isHwErr 1 So it would seem to me that one of the following must be happening: When these particular HLS files are encrypted then the data is being corrupted in some way that played back on iOS 17 and earlier but now won't on 18+, or There's a regression in iOS 18 that means that this particular format of video data is corrupted on decryption If anyone has seen similar behaviour, or has any ideas how to identify which of the two scenarios it is, please say. Unfortunately we don't have control of the servers so can't make changes there unless we can identify they are definitely the cause of the problem. Thanks, Simon.

I use AVPlayer to play HLS video successfully on macOS Sonoma, but I encountered this error on macOS Sequoia. Please help me: Error Domain=AVFoundationErrorDomain Code=-11833 ‘Cannot Decode’ UserInfo={NSUnderlyingError=0x600001e57330 {Error Domain=CoreMediaErrorDomain Code=-12906 ‘(null)’}, NSLocalizedFailureReason=The decoder required for this media cannot be found., AVErrorMediaTypeKey=vide, NSLocalizedDescription=Cannot Decode} Thanks!

Hello, I have an iOS app that is recording audio that is working fine on iPads/iPhones. It asks for microphone permission and after that recording works. I installed the same app on my M3 MacBook via TestFlight since iPad apps are supposed to work without a change that way. The app starts fine and everything, but it never asks for Microphone permission, so I can't record. Do I need to do something to make this happen (this is not macCatalyst, its running the arm64 iPhone binary on macOS) thanks

Hi, our CourAudio server plugin utilizes the SystemConfiguration.framework to store and restore specific shared system wide settings. While our application can authenticate to utilize the SystemConfiguration.framework to gain write access to the shared configuration settings the CoreAudio server plugin obviously can't have any user interaction and therefor does not authenticate. Is it possible to authenticate the CoreAudio server plugin to gain write permissions? Are there any entitlements or other means that would allow this? Thanks!

On macOS Sequoia, I'm having the hardest time getting this basic audio output to work correctly. I'm compiling in XCode using C99, and when I run this, I get audio for a split second, and then nothing, indefinitely. Any ideas what could be going wrong? Here's a minimum code example to demonstrate: #include <AudioToolbox/AudioToolbox.h> #include <stdint.h> #define RENDER_BUFFER_COUNT 2 #define RENDER_FRAMES_PER_BUFFER 128 // mono linear PCM audio data at 48kHz #define RENDER_SAMPLE_RATE 48000 #define RENDER_CHANNEL_COUNT 1 #define RENDER_BUFFER_BYTE_COUNT (RENDER_FRAMES_PER_BUFFER * RENDER_CHANNEL_COUNT * sizeof(f32)) void RenderAudioSaw(float* outBuffer, uint32_t frameCount, uint32_t channelCount) { static bool isInverted = false; float scalar = isInverted ? -1.f : 1.f; for (uint32_t frame = 0; frame < frameCount; ++frame) { for (uint32_t channel = 0; channel < channelCount; ++channel) { // series of ramps, alternating up and down. outBuffer[frame * channelCount + channel] = 0.1f * scalar * ((float)frame / frameCount); } } isInverted = !isInverted; } AudioStreamBasicDescription coreAudioDesc = { 0 }; AudioQueueRef coreAudioQueue = NULL; AudioQueueBufferRef coreAudioBuffers[RENDER_BUFFER_COUNT] = { NULL }; void coreAudioCallback(void* unused, AudioQueueRef queue, AudioQueueBufferRef buffer) { // 0's here indicate no fancy packet magic AudioQueueEnqueueBuffer(queue, buffer, 0, 0); } int main(void) { const UInt32 BytesPerSample = sizeof(float); coreAudioDesc.mSampleRate = RENDER_SAMPLE_RATE; coreAudioDesc.mFormatID = kAudioFormatLinearPCM; coreAudioDesc.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked; coreAudioDesc.mBytesPerPacket = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mFramesPerPacket = 1; coreAudioDesc.mBytesPerFrame = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mChannelsPerFrame = RENDER_CHANNEL_COUNT; coreAudioDesc.mBitsPerChannel = BytesPerSample * 8; coreAudioQueue = NULL; OSStatus result; // most of the 0 and NULL params here are for compressed sound formats etc. result = AudioQueueNewOutput(&coreAudioDesc, &coreAudioCallback, NULL, 0, 0, 0, &coreAudioQueue); if (result != noErr) { assert(false == "AudioQueueNewOutput failed!"); abort(); } for (int i = 0; i < RENDER_BUFFER_COUNT; ++i) { uint32_t bufferSize = coreAudioDesc.mBytesPerFrame * RENDER_FRAMES_PER_BUFFER; result = AudioQueueAllocateBuffer(coreAudioQueue, bufferSize, &(coreAudioBuffers[i])); if (result != noErr) { assert(false == "AudioQueueAllocateBuffer failed!"); abort(); } } for (int i = 0; i < RENDER_BUFFER_COUNT; ++i) { RenderAudioSaw(coreAudioBuffers[i]->mAudioData, RENDER_FRAMES_PER_BUFFER, RENDER_CHANNEL_COUNT); coreAudioBuffers[i]->mAudioDataByteSize = coreAudioBuffers[i]->mAudioDataBytesCapacity; AudioQueueEnqueueBuffer(coreAudioQueue, coreAudioBuffers[i], 0, 0); } AudioQueueStart(coreAudioQueue, NULL); sleep(10); // some time to hear the audio AudioQueueStop(coreAudioQueue, true); AudioQueueDispose(coreAudioQueue, true); return 0; }

I'm writing some camera functionality that uses AVCaptureVideoDataOutput. I've set it up so that it calls my AVCaptureVideoDataOutputSampleBufferDelegate on a background thread, by making my own dispatch_queue and configuring the AVCaptureVideoDataOutput. My question is then, if I configure my AVCaptureSession differently, or even stop it altogether, is this guaranteed to flush all pending jobs on my background thread? For example, does [AVCaptureSession stopRunning] imply a blocking call until all pending frame-callbacks are done? I have a more practical example below, showing how I am accessing something from the foreground thread from the background thread, but I wonder when/how it's safe to clean up that resource. I have setup similar to the following: // Foreground thread logic dispatch_queue_t queue = dispatch_queue_create("qt_avf_camera_queue", nullptr); AVCaptureSession *captureSession = [[AVCaptureSession alloc] init]; setupInputDevice(captureSession); // Connects the AVCaptureDevice... // Store some arbitrary data to be attached to the frame, stored on the foreground thread FrameMetaData frameMetaData = ...; MySampleBufferDelegate *sampleBufferDelegate = [MySampleBufferDelegate alloc]; // Capture frameMetaData by reference in lambda [sampleBufferDelegate setFrameMetaDataGetter: [&frameMetaData]() { return &frameMetaData; }]; AVCaptureVideoDataOutput *captureVideoDataOutput = [[AVCaptureVideoDataOutput alloc] init]; [captureVideoDataOutput setSampleBufferDelegate:sampleBufferDelegate queue:queue]; [captureSession addOutput:captureVideoDataOutput]; [captureSession startRunning]; [captureSession stopRunning]; // Is it now safe to destroy frameMetaData, or do we need manual barrier? And then in MySampleBufferDelegate: - (void)captureOutput:(AVCaptureOutput *)captureOutput didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection *)connection { // Invokes the callback set above FrameMetaData *frameMetaData = frameMetaDataGetter(); emitSampleBuffer(sampleBuffer, frameMetaData); }

In iOS 26 (Developer Beta), the AVCaptureMetadataOutputObjectsDelegate no longer receives callbacks when metadataOutput.metadataObjectTypes = [.face] is set. On earlier iOS versions the issue does not occur. Interestingly, face detection works if I set the sessionPreset to .medium, but not with .high — except on the iPhone 16 Pro Max, where it works regardless.

I'm developing the VisionOS app. I want to know how to play spatial audio in addition to RealityKit? If it's iOS or macOS, how to play spatial audio in addition to RealityKit?

The presentation "create audio drivers with DriverKit" from WWDC 2021 demonstrates how to use a dext to implement a virtual audio driver. It also says " If a virtual audio driver or device is all that is needed, the audio server plug-in driver model should continue to be used". Indeed, in AudioDriverKit/AudioDriverKitTypes.h, there is no IOUserAudioTransportType Virtual, although CoreAudio/AudioHardwareBase.h includes kAudioDeviceTransportTypeVirtual. For one of our products, we require virtual devices to implement a software loopback "cable". We've implemented this using the "traditional" HAL plugin, and as a proof-of-concept, also using a dext. In the dext, I tried setting the transport type to 'virt', which seems to only have the effect of changing the icon shown in Audio Midi Setup. HAL plugins require an installer, and the installer has to kill coreaudiod in a post-install script. You have to turn off SIP to debug them. Just like AudioDriverKit drivers, they are out-of-process and run in a process not owned by the hosting app. Our HAL plugin's interface is property based; we had to write a lot of boiler-plate code to implement required properties. Writing an AudioDriverKit driver is in most respects easier - a lot of the scaffolding is implemented in the base driver, which we only alter where required. Debugging and installation is much easier. The dext works just fine, as far as we can ascertain, just as well as a HAL plugin. So, my question is - is the advice to use a HAL plugin for a virtual device still correct in 2025? And if so, what's the objection? We'd really prefer to ship the AudioDriverKit virtual audio device.

Hi everyone, I noticed that Apple recently added a few new beta sample codes related to video encoding: Encoding video for low-latency conferencing Encoding video for live streaming While experimenting with H.264 encoding, I came across some questions regarding certain configurations: When I enable kVTVideoEncoderSpecification_EnableLowLatencyRateControl, is it still possible to use kVTCompressionPropertyKey_VariableBitRate? In my tests, I get an error. It also seems that kVTVideoEncoderSpecification_EnableLowLatencyRateControl cannot be used together with kVTCompressionPropertyKey_ConstantBitRate when encoding H264. Is that expected? When using kVTCompressionPropertyKey_ConstantBitRate with kVTCompressionPropertyKey_MaxKeyFrameInterval set to 2, the encoder outputs only keyframes, and the frame size keeps increasing, which doesn’t seem like the intended behavior. Regarding the following code from the sample: let byteLimit = (Double(bitrate) / 8) * 1.5 as CFNumber let secLimit = Double(1.0) as CFNumber let limitsArray = [ byteLimit, secLimit ] as CFArray // Each 1 second limit byte as bitrate err = VTSessionSetProperty(session, key: kVTCompressionPropertyKey_DataRateLimits, value: limitsArray) This DataRateLimits setting doesn’t seem to have any effect in my tests. Whether I set it or not, the values remain unchanged. Since the documentation on developer.apple.com/documentation doesn’t clearly explain these cases, I’d like to ask if anyone has insights or recommendations about the proper usage of these settings. Thanks in advance!

I am trying to debug the AAX version of my plugin (MIDI effect) on Pro Tools, but I am getting the following error (Mac console) when attempting to load it: dlsym cannot find symbol g_dwILResult in CFBundle etc.. I used Xcode 16.4 to build the plugin. Has anybody come across the same or a similar message? Best, Achillefs Axart Labs