On an iOS 18 phone, I use AVCaptureSession to capture HDR with x420 format. The output CMSampleBuffer is HLG colorspace, the propagated attachments contain kCVImageBufferAmbientViewingEnvironmentKey and kCVImageBufferSceneIlluminationKey. Now I use CAMetalLayer to render the CVPixelBuffer to the screen, but the brightness is brighter than AVSampleBufferDisplayLayer. Here is my code. - (void)_updateColorSpaceIfNeed:(CVPixelBufferRef)pixelBuffer { CAMetalLayer *layer = (CAMetalLayer *)_mtkView.layer; if (![layer isKindOfClass:CAMetalLayer.class]) return; layer.wantsExtendedDynamicRangeContent = YES; CFDataRef ambientViewingEnvironment = (CFDataRef)CVBufferCopyAttachment(pixelBuffer, kCVImageBufferAmbientViewingEnvironmentKey, NULL); NSData *data = (__bridge NSData *)ambientViewingEnvironment; if (ambientViewingEnvironment) CFRelease(ambientViewingEnvironment); CAEDRMetadata *metadata = [CAEDRMetadata HLGMetadataWithAmbientViewingEnvironment:data]; // CAEDRMetadata *metadata = [CAEDRMetadata HLGMetadata]; layer.EDRMetadata = metadata; layer.pixelFormat = MTLPixelFormatRGBA16Float; CGColorSpaceRef colorspace = CGColorSpaceCreateWithName(kCGColorSpaceITUR_2100_HLG); layer.colorspace = colorspace; if (colorspace) CGColorSpaceRelease(colorspace); } Why does the CAEDRMetadata class have "HLGMetadataWithAmbientViewingEnvironment:" and "HLGMetadata" methods, but does not provide the "HLGMetadataWithAmbientViewingEnvironment:sceneIllumination" method? I want to know how kCVImageBufferAmbientViewingEnvironmentKey and kCVImageBufferSceneIlluminationKey affect tone mapping. Is there any documentation I can refer to?

Hi, I'm trying to plan out development of an app and am wondering if it is possible to have user generated content automatically populate into a custom shazamkit catalogue and be able to query this catalogue non-locally? Storing all the submissions locally would obviously not scale.

I've established proper authorization for general music api calls, but when I use that same authorization to retrieve metadata for the latest music feeds (see https://developer.apple.com/documentation/applemusicfeed/requesting-a-feed-export), I get a 401 Unauthorized error. As per the documentation, I'm simply issuing a GET against https://api.media.apple.com/v1/feed/album/latest. Are there different entitlements needed for the Music Feed API?

We are facing a strange issue where a small portion of our large userbase can not start the capture session in our app, as it gets interrupted with the following reason: AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableWithMultipleForegroundApps Our users are all from iPhones, no one is using an iPad. Just to be sure we have set session.isMultitaskingCameraAccessEnabled = true but it does not seem to make any difference. Another weird scenario we are seeing on an even smaller number of users is that the following call: AVCaptureDevice.default(.builtInWideAngleCamera, for: .video, position: .back) returns nil. A quick look at our error reports show this happening on iPhone XR, 13 and 14 models. They should all support this device type. Any help on investigating these issue would be greatly appreciated!

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

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

Hello! I am trying to determine the best approach with AVPlayer for implementing auto-play, that is, playback that automatically starts without user initiation. Ideally this would work for both local and streaming audio. My current approach is using KVO and the status on an AVPlayerItem equal to readyToPlay to do this, but I was wondering if there was a better property or state to use, or, alternatively, whether this use case may already be handled when automaticallyWaitsToMinimizeStalling is true, so that I could simply write: player.replaceCurrentItem(with: AVPlayerItem(url: streamingUrl)) player.rate = 1 or let playerItem = AVPlayerItem(url: streamingUrl) player = AVPlayer(playerItem: playerItem) player.rate = 1 and expect the item to be auto-played when ready. In the context of user-initiated playback, I've typically seen code that makes a button's enabled state contingent on player.currentItem.duration, e.g. in AVFoundationSimplePlayer-iOS. On the other hand, AVAutoWait, which utilizes automaticallyWaitsToMinimizeStalling, does not seem to do this. As a side note, I am not using an AVQueuePlayer.

We are facing a strange issue where a small portion of our large userbase can not start the capture session in our app, as it gets interrupted with the following reason: AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableWithMultipleForegroundApps Our users are all from iPhones, no one is using an iPad. Just to be sure we have set session.isMultitaskingCameraAccessEnabled = true but it does not seem to make any difference. Another weird interruption we are seeing

I've been trying to use AVMIDIControlChangeEvent with a bankSelect message type to change the instrument the sequencer uses on a AVMusicTrack with no luck. I started with the Apple AVAEMixerSample, converting the initial setup/loading and portions dealing with the sequencer to Swift. I got that working and playing the "bluesyRiff" and then modified it to play individual notes. So my createAndSetupSequencer looked like func createAndSetupSequencer() { sequencer = AVAudioSequencer(audioEngine: engine) // guard let midiFileURL = Bundle.main.url(forResource: "bluesyRiff", withExtension: "mid") else { // print (" failed guard trying to get URL for bluesyRiff") // return // } let track = sequencer.createAndAppendTrack() var currTime = 1.0 for i: UInt32 in 0...8 { let newNoteEvent = AVMIDINoteEvent(channel: 0, key: 60+i, velocity: 64, duration: 2.0) track.addEvent(newNoteEvent, at: AVMusicTimeStamp(currTime)) currTime += 2.0 } The notes played, so then I also replaced the gs_instruments sound bank with GeneralUser GS MuseScore v1.442 first by trying guard let soundBankURL = Bundle.main.url(forResource: "GeneralUser GS MuseScore v1.442", withExtension: "sf2") else { return} do { try sampler.loadSoundBankInstrument(at: soundBankURL, program: 0x001C, bankMSB: 0x79, bankLSB: 0x08) } catch{.... } This appears to work, the instrument (8 which is "Funk Guitar") plays. If I change to bankLSB: 0x00 I get the "Palm Muted guitar". So I know that the soundfont has these instruments Stuff goes off the rails when I try to change the instruments in createAndSetupSequencer. Putting let programChange = AVMIDIProgramChangeEvent(channel: 0, programNumber: 0x001C) let bankChange = AVMIDIControlChangeEvent(channel: 0, messageType: AVMIDIControlChangeEvent.MessageType.bankSelect, value: 0x00) track.addEvent(programChange, at: AVMusicTimeStamp(1.0)) track.addEvent(bankChange, at: AVMusicTimeStamp(1.0)) just before my add note loop doesn't produce any change. Loading bankLSB 8 (Funk) in sampler.loadSoundBankInstrument and trying to change with bankSelect 0 (Palm muted) in createAndSetupSequencer results in instrument 8 (Funk) playing not Palm Muted. Loading bankLSB 0 (Palm muted) and trying to change with bankSelect 8 (Funk) doesn't work, 0 (Palm muted) plays I also tried sampler.loadInstrument(at: soundBankURL) and then I always get the first instrument in the sound font file (piano)no matter what values I put in my programChange/bankChange I've also changed the time in the track.addEvent to be 0, 1.0, 3.0 etc to no success The sampler.loadSoundBankInstrument specifies two UInt8 parameters, bankMSB and BankLSB while the AVMIDIControlChangeEvent bankSelect value is UInt32 suggesting it might be some combination of bankMSB and BankLSB. But the documentation makes no mention of what this should look like. I tried various combinations of 0x7908, 0X0879 etc to no avail I will also point out that I am able to successfully execute other control change events For example adding if i == 1 { let portamentoOnEvent = AVMIDIControlChangeEvent(channel: 0, messageType: AVMIDIControlChangeEvent.MessageType.portamento, value: 0xFF) track.addEvent(portamentoOnEvent, at: AVMusicTimeStamp(currTime)) let portamentoRateEvent = AVMIDIControlChangeEvent(channel: 0, messageType: AVMIDIControlChangeEvent.MessageType.portamentoTime, value: 64) track.addEvent(portamentoRateEvent, at: AVMusicTimeStamp(currTime)) } does produce a change in the sound. (As an aside, a definition of what portamento time is, other than "the rate of portamento" would be welcome. is it notes/seconds? freq/minute? beats/hour?) I was able to get the instrument to change in a different program using MusicPlayer and a series of MusicTrackNewMIDIChannelEvent on a track but these operate on a MusicTrack not the AVMusicTrack which the sequencer uses. Has anyone been successful in switching instruments through an AVMIDIControlChangeEvent or have any feedback on how to do this?

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

Hi, I am recording a video at 240 FPS within my application and saving it to the Photos app. The recorded video retains 240 FPS in the Photos app. However, after trimming the video using the Photos app and importing it back into my app, the FPS is reduced to 30 FPS. Steps to Reproduce: Record a video inside the application at 240 FPS. Save the recorded video to the Photos app. Verify that the video retains 240 FPS in the Photos app. Trim the video using the built-in Photos app editor. Import the trimmed video back into the application. The FPS of the imported video is now reduced to 30 FPS. Code Used for Importing Video: I am using the following code to fetch the video from the Photos app: let options: PHVideoRequestOptions = PHVideoRequestOptions() options.version = .current // Using `.original` preserves FPS, but I need `.current` for other changes options.deliveryMode = .highQualityFormat options.isNetworkAccessAllowed = true PHImageManager.default().requestAVAsset(forVideo: self, options: options) { (avAsset, audioMix, info) in if let urlAsset = avAsset as? AVURLAsset { completionHandler(urlAsset.url, self) } else { self.askForOriginal(completionHandler: completionHandler) } } Observations: The original video retains 240 FPS until it is trimmed in the Photos app. After trimming, the FPS automatically drops to 30 FPS when imported back into the app. If I use options.version = .original, the FPS is preserved, but I need .current to apply other modifications. Questions: Is this an expected behavior of PHImageManager when requesting a video with options.version = .current? Is there a way to preserve the original FPS while still using .current? Are there any workarounds to extract the trimmed video without FPS reduction? Any insights or solutions would be greatly appreciated. Thanks in advance!

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

I'm seeking to a specific sync frame in a video file (HEVC, recorded on iPad). When I feed the buffers from that sync frame on to VTDecompressionSession it consistently drops the 2.,3.,4. buffer with a kVTVideoDecoderReferenceMissingErr (or no error but no buffer on the simulator). If I feed all the buffers from the penultimate sync frame prior to the desired frame the buffers come out fine but that would just create a massive overhead to always do it. Tried multiple OS versions, devices etc. Seems a consistent problem. Here's a sample project with the offending video (disregard memory handling etc): https://github.com/marcuseckert/vtSample I've filed a radar FB18228296 but would appreciate any feedback on circumventing or at least detecting this behavior prior to decoding.

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

I prefer to use the album fetched from the library instead of the catalog since this is faster. If doing so, how can I check if all tracks of an album are added to the library. In this case I'd like to fetch the catalog version or throw an error (for example when offline). Using .with(.tracks) on the library album fetches the tracks added to the library. The trackCount property is referring to the tracks that can be fetched from the library. The isComplete property is always nil when fetching from the library. One possible way is checking the trackNumber and discCount properties. However this only detects that not all tracks of an album are added to the library if there is a song not added ahead of one that is. I'd like to be able to handle this edge case as well. Is there currently a way to do this? I'd prefer to not rely on the apple music catalog for this since this is supposed to work offline as well. Fetching and storing all trackIDs when connected and later comparing against these would work, but this would potentially mean storing tens of thousands of track ids. Thank you

Hey, Quick question. I noticed that Adobe's new app, Project Indigo, allows you to open the app using the Camera Control button. However, when your device is locked it just shows this screen: Would this normally be approved by the Appstore approval process? I ask because I would like to do something similar with my camera app. I know that this is not the best user experience, but my apps UI is not built in Swift and I don't have the resources to build the UI again. At least this way the user experience would be improved from what it is now, where users cannot even launch the app. I get many requests per week about this feature and would love to improve the UX for my users, even if it's not the best possible. Thanks, Alex

Hello Apple Developers, I’m reaching out to the community with a concept that I truly believe could be a natural fit for the Apple ecosystem: A privacy-focused, iOS-exclusive dating app designed to enhance connections between Apple users while staying true to Apple’s commitment to security and user privacy. The idea is to create an iOS-only dating platform that fosters relationships between users who are part of the Apple ecosystem. The app would integrate seamlessly with Apple’s services (iMessage, FaceTime, Siri, etc.) and provide a premium user experience, where privacy is a priority. Apple users already prefer to communicate using Apple services (iMessage, FaceTime). A dating app designed specifically for iOS users would deepen this ecosystem lock-in, making it easier for Apple customers to connect within a trusted space. Apple is already known for its privacy focus, and an iOS-exclusive dating app would build upon that reputation. It would ensure secure, private interactions, minimizing the risks associated with data sharing in most dating apps today. The app could integrate directly with features like iCloud, Apple Pay (for date-night bookings), and Siri (for matchmaking suggestions), offering users a truly native iOS experience. While the app would remain free to use, here are a few potential monetization methods: Bundling with Apple One/iCloud+ for premium matchmaking features. Apple Pay-based date-night deals with local partners. I’d love to hear your thoughts on whether Apple might be open to this idea. Would there be any challenges from a technical or business perspective in creating a dating app exclusively for iOS users? I’m looking forward to hearing from you all, and thank you for your time and insights. Yours Truly, CapNKirk P.S. This is an idea. But I do not care who uses, implements, or executes this idea. I just want to see Apple take advantage of it.

I am getting high error rates from the Apple Music API. This has been happening for months now, and it is quite frustrating. It is a mix of 404, 504, and random 500 errors. I hit these endpoints all of the time, so it is not like I am hitting a resource that doesn't exist. Why is this happening? Is this a known issue that is getting worked on?

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

Hi All. I'm working on Single-Sign-On feature in my application to let customers sign into their TV Provider. I need to add Video Subscriber SSO entitlement (com.apple.developer.video-subscriber-single-sign-on) to the app, but I found out that it's a special entitlement, need to contact Apple to enable it for my Apple account. On https://developer.apple.com/account I navigated to Support -> Contact Us -> Development and Technical -> Entitlements and ask in the email about missing entitlement (ticket ID 102478794279). The support team couldn't help me, they redirected me to the operations team. I've been waiting for a few months now but they inform me to keep waiting. Is there a better way to contact Apple and get Video Subscriber SSO entitlement in an efficient way?