I've got a setup using AVAudioEngine with several tone generator nodes, each with a chain of processing nodes, the chains then mixed into the main output. Generator ➡️ Effect ➡️... ➡️ .mainMixerNode ➡️ .outputNode). Generator ➡️ Effect ➡️... ⤴️ ... Generator ➡️ Effect ➡️... ⤴️ The user should be able to mute any chain individually. I've found several potential approaches to muting, but not terribly happy with any of them. Adjust the amplitudes directly in my tone generators. Issue: Consumes CPU even when completely muted. 4 generators adds ~15% cpu, even when all chains are muted. Detach/attach chains that are muted/unmuted. Issue: Causes loud clicking/popping sounds whenever muted/unmuted. Fade mixer output volume while detaching/attaching a chain (just cutting the volume immediately to 0 doesn't get rid of the clicking/popping). Issue: Causes all channels to fade during the transition, so not ideal. The rest of these ideas are variations on making volume control+detatch/attach work for individual chains, since approach #3 worked well. Add an AVAudioMixer to the end of each chain (just for volume control). Issue: Only the mixer on the final chain functions -- the others block all output. Not sure what's going on there. Use matrix mixer (for multi-input volume control). Plus detach/attach to reduce CPU if necessary. Not yet attempted, due to perceived complexity and reports of fragility in order of wiring in. A bunch of effort before I even know if it's going to work. Develop my own fader node to put on the end of each channel. Unlike the tone generator (simple AVSourceNode), developing an effect node seems complex and time consuming. Might not even fix CPU use. I'm not completely averse to the learning curve of either 5 or 6, but would rather get some guidance on best approach before diving in. They both seem likely to take more effort than I'd like for the simple behavior I'm trying to achieve.

Hey everyone, I'm encountering an issue with audio sample rate conversion that I'm hoping someone can help with. Here's the breakdown: Issue Description: I've installed a tap on an input device to convert audio to an optimal sample rate. There's a converter node added on top of this setup. The problem arises when joining Zoom or FaceTime calls—the converter gets deallocated from memory, causing the program to crash. Symptoms: The converter node is being deallocated during video calls. The program crashes entirely when this happens. Traditional methods of monitoring sample rate changes (tracking nominal or actual sample rates) aren't working as expected. The Big Challenge: I can't figure out how to properly monitor sample rate changes. Listeners set up to track these changes don't trigger when the device joins a Zoom or FaceTime call. Please, if anyone has experience with this or knows a solution, I'd really appreciate your help. Thanks in advance! ⁠

We encounter issue with avplayer in case of EXT-X-DISCONTINUITY misalignment between audio and video produced after insertion of gaps. The initial objective is to introduce an EXT-X-DISCONTINUITY in audio playlist after some missing segments (EXT-X-GAP) which durations are aligned to video segments durations, to handle irregular audio durations. Please find below an example of corresponding video and audio playlists: video: #EXTM3U #EXT-X-VERSION:7 #EXT-X-MEDIA-SEQUENCE:872524632 #EXT-X-INDEPENDENT-SEGMENTS #EXT-X-TARGETDURATION:2 #USP-X-TIMESTAMP-MAP:MPEGTS=7096045027,LOCAL=2025-05-09T12:38:32.369100Z #EXT-X-MAP:URI="hls/StreamingBasic-video=979200.m4s" #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:32.369111Z #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524632.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524633.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524634.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524635.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524636.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524637.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524638.m4s #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:46.383111Z #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524639.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524640.m4s audio: EXTM3U #EXT-X-VERSION:7 #EXT-X-MEDIA-SEQUENCE:872524632 #EXT-X-INDEPENDENT-SEGMENTS #EXT-X-TARGETDURATION:2 #USP-X-TIMESTAMP-MAP:MPEGTS=7096045867,LOCAL=2025-05-09T12:38:32.378400Z #EXT-X-MAP:URI="hls/StreamingBasic-audio_99500_eng=98800.m4s" #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:32.378444Z #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524632.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524633.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524634.m4s #EXTINF:1.984, no desc hls/StreamingBasic-audio_99500_eng=98800-872524635.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524636.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-audio_99500_eng=98800-872524637.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-audio_99500_eng=98800-872524638.m4s #EXT-X-DISCONTINUITY #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:46.778444Z #EXTINF:1.6213, no desc hls/StreamingBasic-audio_99500_eng=98800-872524639.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524640.m4s In this case playback is broken with avplayer. Is it conformed to Http Live Streaming? Is it an avplayer bug? What are the guidelines to handle such gaps?

Hi, I am a newbie here. We have been given a task to build a robotic vision system to capture an immersive video in a hazed environment, which will later be played on Apple Vision Pro. I am thinking of starting with 2 or 4 basic CMOS camera sensors, such as IMX378, AR0144, or VD66GY, and designing an FPGA-based circuit to synchronously capture and store raw frame-by-frame data. Some frame initial processing such as demosaicing and filtering can also be done by the FPGA. Then, I would use software for post-processing to convert the data into a compatible video format for Apple Vision Pro. Will this idea work? I can handle the raw data capture, but I’m unsure if this approach is feasible and what post-processing software I should use. Thanks a lot for your suggestions! Charlie

Our Final Cut Pro workflow extension built with ProExtensionHost framework uses an advanced NSPasteboardItemDataProvider system with multi-version FCPXML support (1.9, 1.10, 1.13) and proper relative path UIDs for Motion templates. We've implemented clip wrapper approach with placeholder assets and elements containing effects to enable direct timeline drag functionality. However, drag and drop from our Final Cut Pro workflow extension directly to timeline is still not working despite proper element structure in our FCPXML. Our implementation creates valid clip elements with effects applied, but Final Cut Pro timeline doesn't accept them during drag operations from our ProExtensionHost-based workflow extension. Steps to Reproduce: Create Final Cut Pro workflow extension using ProExtensionHost framework with NSPasteboardItemDataProvider implementation Generate FCPXML with proper element structure: Expected Result: Clip should be accepted by timeline and effect applied from workflow extension Actual Result: Timeline rejects drag operation from ProExtensionHost-based workflow extension Question: Are there additional requirements or ProExtensionHost API calls needed beyond standard NSPasteboardItemDataProvider for Final Cut Pro workflow extension timeline drag functionality?

I'm using AVFoundation to make a multi-track editor app, which can insert multiple track and clip, including scale some clip to change the speed of the clip, (also I'm not sure whether AVFoundation the best choice for me) but after making the scale with scaleTimeRange API, there is some short noise sound in play back. Also, sometimes it's fine when play AVMutableCompostion using AVPlayer with AVPlayerItem, but after exporting with AVAssetReader, will catch some short noise sounds in result file.... Not sure why. Here is the example project, which can build and run directly. https://github.com/luckysmg/daily_images/raw/refs/heads/main/TestDemo.zip

I am work an app development on an app which request an audio function in background as an alert sound. during debug testing , the function work fine, but once I testing standalone without debugging , The function not work , it will play out the sound when I back to app. does any way to trace the issues ?

I'm reaching out regarding a recurring issue I'm experiencing with MusicKit developer tokens. I'm using a valid .p8 private key to sign JWTs for Apple MusicKit integration. Each token I generate includes the appropriate claims (iss, iat, exp) and is signed with the ES256 algorithm, with an expiration date set approximately 6 months ahead. Everything works as expected immediately after generating the token. However, after a few days, the same JWT (still well within its expiration period) suddenly begins returning invalid/unauthorized responses when used in Postman and other API clients. Importantly: I did not delete or revoke the .p8 key during this time. I verified the JWT contains valid claims and a proper structure. The issue consistently resolves only when I create a new .p8 file and regenerate a fresh JWT with it—after which the cycle repeats. This issue occurs even when the environment and app identifiers remain unchanged. I would greatly appreciate it if you could help me understand: Why these tokens become invalid after a few days, despite having a long exp value and an unchanged key. Whether there's any automatic revocation or timeout policy on .p8 keys that could explain this behavior. If there's a better way to maintain long-lived developer tokens without requiring new .p8 key generation every few days. Thank you for your help and clarification on this issue. Best regards, Liad Altif

I am developing an app that plays HLS audio. When using AVPlayerItem with AVURLAsset, can AVAssetResourceLoaderDelegate correctly handle HLS segments? My goal is to use AVAssetResourceLoaderDelegate to add authentication HTTP headers when accessing HLS .m3u8 and .ts files. I can successfully download the files, but playback fails with errors. Specifically, I am observing the following cases: A. AVAssetResourceLoaderDelegate is canceled, and CoreMediaErrorDomain -12881 occurs In NSURLConnectionDataDelegate’s didReceiveResponse method, set contentInformationRequest In didReceiveData, call dataRequest respondWithData resourceLoader didCancelLoadingRequest is called CoreMediaErrorDomain -12881 occurs B. CoreMediaErrorDomain -12881 occurs In NSURLConnectionDataDelegate’s didReceiveResponse method, set contentInformationRequest In connection didReceiveData, buffer all received data until the end In connectionDidFinishLoading, pass the buffered data to respondWithData Call loadingRequest finishLoading CoreMediaErrorDomain -12881 occurs In both cases, dataRequest.requestsAllDataToEndOfResource is YES. For this use case, I am not using AVURLAssetHTTPHeaderFieldsKey because I need to apply the most up-to-date authentication data at the moment each file is accessed. I would appreciate any advice or suggestions you might have. Thank you in advance!

Hello. My app uses AVAudioRecorder to generate recording files, which are consistently only 4kb in size. Most users generate audio files normally, with only a few users experiencing this phenomenon occasionally. After uninstalling and installing the app, it will work normally, but it will reappear after a period of time. I have compared that the problematic audio files generated each time are fixed and cannot be played. Added the audioRecorderDidFinishRecording proxy method, which shows that the recording was completed normally. The user also reported that the recording is normal, but there is a problem with the generated file. How should I handle this issue? Look forward to your reply. - (void)startRecordWithOrderID:(NSString *)orderID { AVAudioSession *audioSession = [AVAudioSession sharedInstance]; [audioSession setCategory:AVAudioSessionCategoryRecord error:nil]; [audioSession setActive:YES error:nil]; NSMutableDictionary *settings = [[NSMutableDictionary alloc] init]; [settings setObject:[NSNumber numberWithFloat: 8000.0] forKey:AVSampleRateKey]; [settings setObject:[NSNumber numberWithInt: kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [settings setObject:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [settings setObject:[NSNumber numberWithInt: 1] forKey:AVNumberOfChannelsKey]; [settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; NSString *path = [WDUtility createDirInDocument:@"audios" withOrderID:orderID withPathExtension:@"wav"]; NSURL *tmpFile = [NSURL fileURLWithPath:path]; recorder = [[AVAudioRecorder alloc] initWithURL:tmpFile settings:settings error:nil]; [recorder setDelegate:self]; [recorder prepareToRecord]; [recorder record]; }

Use case: When SharePlay -ing a fully immersive 3D scene (e.g. a virtual stage), I would like to shine lights on specific Personas, so they show up brighter when someone in the scene is recording the feed (think a camera person in the scene wearing Vision Pro). Note: This spotlight effect only needs to render in the camera person's headset and does NOT need to be journaled or shared. Before I dive into this, my technical question: Can environmental and/or scene lighting affect Persona brightness in a SharePlay? If not, is there a way to programmatically make Personas "brighter" when recording? My screen recordings always seem to turn out darker than what's rendered in environment, and manually adjusting the contrast tends to blow out the details in a Persona's face (especially in visionOS 26).

I'm trying to write 16-bit interleaved 2-channel data captured from a LiveSwitch audio source to a AVAudioFile. The buffer and file formats match but I get a bad parameter error from the API. Does this API not support the specified format or is there some other issue? Here is the debugger output. (lldb) po audioFile.url ▿ file:///private/var/mobile/Containers/Data/Application/1EB14379-0CF2-41B6-B742-4C9A80728DB3/tmp/Heart%20Sounds%201 - _url : file:///private/var/mobile/Containers/Data/Application/1EB14379-0CF2-41B6-B742-4C9A80728DB3/tmp/Heart%20Sounds%201 - _parseInfo : nil - _baseParseInfo : nil (lldb) po error Error Domain=com.apple.coreaudio.avfaudio Code=-50 "(null)" UserInfo={failed call=ExtAudioFileWrite(_impl->_extAudioFile, buffer.frameLength, buffer.audioBufferList)} (lldb) po buffer.format <AVAudioFormat 0x302a12b20: 2 ch, 44100 Hz, Int16, interleaved> (lldb) po audioFile.fileFormat <AVAudioFormat 0x302a515e0: 2 ch, 44100 Hz, Int16, interleaved> (lldb) po buffer.frameLength 882 (lldb) po buffer.audioBufferList ▿ 0x0000000300941e60 - pointerValue : 12894608992 This code handles the details of converting the Live Switch frame into an AVAudioPCMBuffer. extension FMLiveSwitchAudioFrame { func convertedToPCMBuffer() -> AVAudioPCMBuffer { Self.convertToAVAudioPCMBuffer(from: self)! } static func convertToAVAudioPCMBuffer(from frame: FMLiveSwitchAudioFrame) -> AVAudioPCMBuffer? { // Retrieve the audio buffer and format details from the FMLiveSwitchAudioFrame guard let buffer = frame.buffer(), let format = buffer.format() as? FMLiveSwitchAudioFormat else { return nil } // Extract PCM format details from FMLiveSwitchAudioFormat let sampleRate = Double(format.clockRate()) let channelCount = AVAudioChannelCount(format.channelCount()) // Determine bytes per sample based on bit depth let bitsPerSample = 16 let bytesPerSample = bitsPerSample / 8 let bytesPerFrame = bytesPerSample * Int(channelCount) let frameLength = AVAudioFrameCount(Int(buffer.dataBuffer().length()) / bytesPerFrame) // Create an AVAudioFormat from the FMLiveSwitchAudioFormat guard let avAudioFormat = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: sampleRate, channels: channelCount, interleaved: true) else { return nil } // Create an AudioBufferList to wrap the existing buffer let audioBufferList = UnsafeMutablePointer<AudioBufferList>.allocate(capacity: 1) audioBufferList.pointee.mNumberBuffers = 1 audioBufferList.pointee.mBuffers.mNumberChannels = channelCount audioBufferList.pointee.mBuffers.mDataByteSize = UInt32(buffer.dataBuffer().length()) audioBufferList.pointee.mBuffers.mData = buffer.dataBuffer().data().mutableBytes // Directly use LiveSwitch buffer // Transfer ownership of the buffer to AVAudioPCMBuffer let pcmBuffer = AVAudioPCMBuffer(pcmFormat: avAudioFormat, bufferListNoCopy: audioBufferList) /* { buffer in // Ensure the buffer is freed when AVAudioPCMBuffer is deallocated buffer.deallocate() // Only call this if LiveSwitch allows manual deallocation } */ pcmBuffer?.frameLength = frameLength return pcmBuffer } } This is the handler that is invoked with every frame in order to convert it for use with AVAudioFile and optionally update a scrolling signal display on the screen. private func onRaisedFrame(obj: Any!) -> Void { // Bail out early if no one is interested in the data. guard isMonitoring else { return } // Convert LS frame to AVAudioPCMBuffer (no-copy) let frame = obj as! FMLiveSwitchAudioFrame let buffer = frame.convertedToPCMBuffer() // Hand subscribers a reference to the buffer for rendering to display. bufferPublisher?.send(buffer) // If we have and output file, store the data there, as well. guard let audioFile = self.audioFile else { return } do { try audioFile.write(from: buffer) // FIXME: This call is throwing error -50 } catch { FMLiveSwitchLog.error(withMessage: "Failed to write buffer to audio file at \(audioFile.url): \(error)") self.audioFile = nil } } This is how the audio file is being setup. static var recordingFormat: AVAudioFormat = { AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44_100, channels: 2, interleaved: true)! }() let audioFile = try AVAudioFile(forWriting: outputURL, settings: Self.recordingFormat.settings)

I'm working on a media app that would like to be able to tell if the TV connected to tvOS is running at 59.94hz or 60.00hz, so it can optimize a video stream. It looks like the best I can currently do is to check if the user has Match Content Rate enabled, and based on that, when calling displayManager.preferredDisplayCriteria to change video modes, I could guess which rate their TV might be in. It's not very ideal, because not all TVs support both of these rates, and my request for 59.94 might end up as 60 and vice versa. I dug around and can't find any available method in UIScreen to get this info. The odd thing is, the data is right there in currentMode when I look in the debugger, but it seems to be in a private or undocumented class. Is there any way to get at it?

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

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

I am writing an iOS app to present a slide show of assets in a Photo album, in a random order, including videos and live photos. I have got it all working quite nicely but for a Live Photo, I need to know what effect is selected (Live, Loop, Bounce, Long Exposure, Live Off) to display the image correctly. I can't find any mention of getting this information in the documentation. Anyone know how to do this? Thanks in advance. Adrian. (Xcode 16.1 iOS 18.0)

(Note: this is part 3 of a 3 part posting. See Part 1 or Part 2) At WWDC25 we launched a new type of Lab event for the developer community - Group Labs. A Group Lab is a panel Q&A designed for a large audience of developers. Group Labs are a unique opportunity for the community to submit questions directly to a panel of Apple engineers and designers. Here are the highlights from the WWDC25 Group Lab for Camera & Photos. WWDC25 Camera & Photos group lab ran for one hour at 6 PM PST on Tuesday June 10th, 2025 Question 24 What’s the best approach for optimizing barcode scanning using AVFoundation or Vision in low-light or angled scenarios Turn on flash in low-light scenarios Lower framerate to improve exposure and reduce noise Wait until the capture is in focus/notify your user that they need to get closer Question 25 Recent iPhone models introduced macro mode which automatically switch between lenses to take into account of the focal distance difference. Is there official API to implement this, or should I implement them myself using LiDAR values. Using builtInTripleCamera and builtInDualWideCamera will automatically switch to macro when available Question 26 Is there a way to quickly create a thumbnail after the user selects an image with PhotosPicker? File provider API Additional questions from the WWDC25 in-person labs that occurred later in the WWDC week Question 1 When should I build my custom photo picker instead of using the system one? Always start with the system picker -> try embeddable customization APIs -> fallback to custom picker for very special needs Question 2 I'm building a new camera app for pros and I want to give my users the most un-processed image possible, and the most control over the capture as possible. How can I do that with AVCapture? If stills, Brief Bayer RAW capture overview, or Pro RAW if you want Apple's processing and dynamic range If video, talk about prores LOG. Custom exposure settings are available throguh the apis maybe global/local tonemapping discussion?

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/.

I’m an amateur developer working on a free utility for composers/producers, for which the macOS release needs to create and name RTP-MIDI sessions in Audio MIDI Setup from the command line (so I can ship a small C helper instead of telling users to click through the UI). Here’s what I’ve tried so far, without luck: • Plist hacks: Injecting entries into ~/Library/Audio/MIDI Configurations/*.mcfg works when AMS is closed, but AMS immediately locks and reverts my changes when it’s open. • CoreMIDI C API: I can create virtual ports with MIDISourceCreate, but attempting MIDIObjectGetDataProperty on the apple.midirtp.session plugin always returns err –10836. • Obj-C & Swift: Loading MIDINetworkSession and calling defaultSession, init, setNetworkName: and setting enabled = YES doesn’t produce a new session object in the Network panel. • dlopen/dlsym: I extracted the real CoreMIDI binary out of the dyld shared cache and tried binding _MIDINetworkSessionCreate, _SetName, _SetEnabled, etc., but all the symbols come back null or my tool segfaults. • Plugin registration: I’ve pulled the factory UUID (70C9C5EA-7C65-11D8-B317-000393A34B5A) from /System/Library/Extensions/AppleMIDIRTPDriver.plugin/Contents/Info.plist and called CFPlugInRegisterFactories, but it still never exposes the session-creation calls. At this point I’m convinced I’m either loading the wrong binary or missing one critical step in registering the RTP-MIDI plugin’s private API. Can anyone point me to: The exact path of the dylib or bundle that actually exports the MIDINetworkSessionCreate/MIDINetworkSessionSetName/MIDINetworkSessionSetEnabled symbols? A minimal working snippet (C or Obj-C) that reliably creates and names a Network-MIDI session? Any pointers, sample code, or even ideas about where Apple hides this functionality on macOS 15 would be hugely appreciated. Thanks!

I am developing an app that uses MusicKit to play music and then I need to have spoken words played to the user, while ducking the audio coming from MusicKit (application music player) the built in Siri voices are not off sufficient quality so I am using an external service to create an mp3 file and then play this back using AVAudioSession Sample code below the problem I am having is that .duckOthers is not ducking the Application Music Player output Is this a bug or am I doing this wrong? // Configure audio session for system-wide ducking try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, options: [.duckOthers, .mixWithOthers]) try AVAudioSession.sharedInstance().setActive(true) // Set the ducking level to maximum try AVAudioSession.sharedInstance().setPreferredIOBufferDuration(0.005) // Create and configure audio player self.audioPlayer = try AVAudioPlayer(data: audioData) self.audioPlayer?.delegate = self self.audioPlayer?.volume = 1.0 // Ensure full volume for speech self.audioPlayer?.prepareToPlay() // Set the audio player's settings for maximum clarity self.audioPlayer?.enableRate = false self.audioPlayer?.pan = 0.0 // Center the audio self.audioPlayer?.play()