Hello, I have an existing AUv3 instrument plugin. In the plug in, users can access files (audio files, song projects) via a UIDocumentPickerViewController In Logic Pro, (and some other hosts, but not all), the document picker is unable to receive touches, while a keyboard case is attached to the iPad. Removing the case (this is an Apple brand iPad case) allows the interactions to resume and allows me to pick files in the usual way. One of my users reports this non-responsive behavior occurs even after disconnecting their keyboard. I have fiddled with entitlements all day, and have determined that is not the issue, since the keyboard disconnection appears to fix it every time for me. Here is my, very boilerplate, presentation code : guard let type = UTType("com.my.type") else { return } let fileBrowser = UIDocumentPickerViewController(forOpeningContentTypes: [type]) fileBrowser.overrideUserInterfaceStyle = .dark fileBrowser.delegate = self fileBrowser.directoryURL = myFileFolderURL() self.present(fileBrowser, animated: true) {

Hi, I'm developing a SwiftUI app using RealityKit and ARKit for an AR measuring feature. I’ve noticed that after navigating away from my AR view and performing extensive cleanup (including removing all anchors/entities, pausing the ARSession, and nil-ing out all references), memory usage remains elevated and sometimes grows with repeated AR sessions. Each time I enter and exit the AR view, memory increases The memory does not return to the baseline after cleanup, even though all custom objects are deallocated. Are there best practices beyond what I’ve described to ensure all ARKit/RealityKit resources are released after an AR session?

Hi Apple Music API / MusicKit / MediaPlayer Team, Similar to the currentPlaybackRate keeps the same pitch, it would be great to have a currentPlaybackPitch parameter as well. Alternatively, adding a preservesPitch parameter would also work. I see that iOS 26 AutoMix on Apple Music currently does pitch shifting during music transitions, so maybe this is something that could be exposed on the later betas of iOS 26? Main feature request we get is to have simple pitch changes to Apple Music we play through our app. Is this being considered?

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

I have an app (currently in development stage) which needs to use ffmpeg, so I tried searching how to embed ffmpeg in apple apps and found this article https://doc.qt.io/qt-6/qtmultimedia-building-ffmpeg-ios.html It is working correctly for iOS but not for macOS ( I have made changes macOS specific using chatgpt and traditional web searching) Drive link for the file and instructions which I'm following: https://drive.google.com/drive/folders/11wqlvb8SU2thMSfII4_Xm3Kc2fPSCZed?usp=share_link Please can someone from apple or in general help me to figure out what I'm doing wrong?

I’m running HomePod OS 26 on two HomePod minis and OS 18.6 on main HomePod (original) I’ve enabled Crossfade in the Home app. I’m playing Apple Music directly in the HomePod mini. Crossfade just doesn’t work on any HomePod. I can understand it not working on the HomePod - but why isn’t it working on the minis running OS 26? I’ve tried disabling and enabling Crossfade, rebooting HomePods etc but nothing?!

The documentation for the Apple Music API indicates that the genreNames field for a given artist (see https://developer.apple.com/documentation/applemusicapi/artists/attributes-data.dictionary) is an array of strings. However, it only appears as though you return ONE SINGLE GENRE per Artist, regardless of how many genres might be attached to that artist's albums. Am I missing something? Is there an artist where multiple genres may be returned, or is this a bug in the documentation?

I have some question about the TTS My device default language is zh-HK. (cantonese) my device is iPhone 16 Pro, IOS 18.6 I create a function speakMandarin I want the device to speak the zh-CN (putonghua) however the device only can speak zh-HK (cantonese). I already set the AVSpeechSynthesisVoice language as zh-CN func speakMandarin(text: String) { print("speakMandarin, \(text)") lastError = nil // Reset error // Stop any ongoing speech before starting new if synthesizer.isSpeaking { synthesizer.stopSpeaking(at: .immediate) } // Configure speech utterance let utterance = AVSpeechUtterance(ssmlRepresentation: text)! utterance.rate = 0.5 // Natural speaking speed utterance.pitchMultiplier = 1.0 utterance.volume = 1.0 utterance.voice = AVSpeechSynthesisVoice(language: "zh-CN") let preferredLanguages = [ "zh-CN" , "zh-TW"] var selectedVoice: AVSpeechSynthesisVoice? for lang in preferredLanguages { if let voice = AVSpeechSynthesisVoice(language: lang) { print(lang) selectedVoice = voice utterance.voice = voice break } } // If no Mandarin voice found, use system default if selectedVoice == nil { selectedVoice = AVSpeechSynthesisVoice(language: nil) lastError = "未偵測到普通話語音包，將使用系統預設語音" print (lastError) } utterance.voice = selectedVoice print(utterance) synthesizer.speak(utterance) } here is my log speakMandarin, <speak>你好！我們來聊聊喜歡的動物吧。你喜歡什麼動物呢？</speak> zh-CN [AVSpeechUtterance 0x1194efb80] String: 你好！我們來聊聊喜歡的動物吧。你喜歡什麼動物呢？ Voice: [AVSpeechSynthesisVoice 0x104ceff90] Language: zh-CN, Name: Tingting, Quality: Default [com.apple.voice.compact.zh-CN.Tingting] Rate: 0.50 Volume: 1.00 Pitch Multiplier: 1.00 Delays: Pre: 0.00(s) Post: 0.00(s)

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.

I found that the aggregated device correctly obtains input channels in the standard microphone mode. However, in voice isolation mode, it only retrieves channels from the first sub-device in the aggregated device's list. If I want to properly obtain channel information in voice isolation mode, how should I do it?

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

Hi, when a CoreMIDI driver controls physical HW it is probably quite commune to have to control the amount of MIDI data received from the system. What comes to mind is to just delay returning control of the MIDIDriverInterface::Send() callback to the calling process. While the application trying to send MIDI really stalls until the callback returns it seems only to be a side effect of a generally stalled CoreMIDI server. Between the callbacks the application can send as much MIDI data as it wants to CoreMIDI, it's buffering seems to be endless... However the HW might not be able to play out all the data. It seems there is no way to indicate an overflow/full buffer situation back the application/CoreMIDI. How is this supposed to work? Thanks, any hints or pointers are highly appreciated! Hagen.

I developed an educational app that implements audio-video communication through RTC, while using WebView to display course materials during classes. However, some users are experiencing an issue where the audio playback from WebView is very quiet. I've checked that the AVAudioSessionCategory is set by RTC to AVAudioSessionCategoryPlayAndRecord, and the AVAudioSessionCategoryOption also includes AVAudioSessionCategoryOptionMixWithOthers. What could be causing the WebView audio to be suppressed, and how can this be resolved?

When trying to edit some Live Photos, calling PHLivePhotoEditingContext.saveLivePhoto results in the following error: Error Domain=AVFoundationErrorDomain Code=-11800 "The operation could not be completed" UserInfo={NSLocalizedFailureReason=An unknown error occurred (-12815), NSLocalizedDescription=The operation could not be completed, NSUnderlyingError=0x300d05380 {Error Domain=NSOSStatusErrorDomain Code=-12815 "(null)"}} I was able to replicate it on my device by taking a new Live Photo. Not sure what's wrong with that one specifically, not all Live Photos replicate the issue. I've submitted FB15880825 with a sysdiagnose and a Photos Diagnostics as well. Any ideas what's going on here? It's impacting multiple customers. Thanks!

I’m developing a macOS audio monitoring app using AVAudioEngine, and I’ve run into a critical issue on macOS 26 beta where AVFoundation fails to detect any input devices, and AVAudioEngine.start() throws the familiar error 10877. FB#: FB19024508 Strange Behavior: AVAudioEngine.inputNode shows no channels or input format on bus 0. AVAudioEngine.start() fails with -10877 (AudioUnit connection error). AVCaptureDevice.DiscoverySession returns zero audio devices. Microphone permission is granted (authorized), and the app is properly signed and sandboxed with com.apple.security.device.audio-input. However, CoreAudio HAL does detect all input/output devices: Using AudioObjectGetPropertyDataSize and AudioObjectGetPropertyData with kAudioHardwarePropertyDevices, I can enumerate 14+ devices, including AirPods, USB DACs, and BlackHole. This suggests the lower-level audio stack is functional. I have tried: Resetting CoreAudio with sudo killall coreaudiod Rebuilding and re-signing the app Clearing TCC with tccutil reset Microphone Running on Apple Silicon and testing Rosetta/native detection via sysctl.proc_translated Using a fallback mechanism that logs device info from HAL and rotates logs for submission via Feedback Assistant I have submitted logs and a reproducible test case via Feedback Assitant : FB#: FB19024508]

SBS ViewPacking add a half a frame to the opposite eye. Meaning if you look all the way right you can see an extra half frame with left eye and vice versa. OU doesn't work at all, the preview just doesn't show a thumbnail and the video doesn't play. Any hints on how to fix this? I submitted a bug report but haven't heard anything.

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

ApplicationMusicPlayer is not available on watchOS but all other platforms. Is there a technical reason for that like battery life? Same goes for SystemMusicPlayer and MPMusicPlayerController. I already filed feedbacks for that.

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!}

Device: iPhone 16 Pro Max OS: iOS 26.0.1 AVCaptureDevice.DeviceType: builtInUltraWideCamera avtiveFormat: <AVCaptureDeviceFormat: 0x10ffb9ac0 'vide'/'420v' 1440x1080, { 1- 60 fps}, photo dims:{1440x1080,2016x1512}, fov:101.022, gdc fov:103.625, binned, max zoom:94.50 (upscales @1.40), system zoom range:1.0-3.0, AF System:1, ISO:15.0-3600.0, SS:0.000023-1.000000, system exposure bias range:-2.0-2.0, supports multicam, supports CS RoI, supports Smart Style, supports Smudge Detection> API: device.isFocusModeSupported(.continuousAutoFocus) == true setting: device.focusMode = .continuousAutoFocus setting is ok, but it's not working actually with continuousAutoFocus