Is there a way to permanently disable PHASE SDK logging? It seems to be a lot chattier than Apple's other SDKs. While developing a RealityKit app that uses AudioPlaybackController, I must manually hide the PHASE SDK log output several times each day so I can see my app's log messages. Thank you.

Hi, I've had a new deck installed in my car for about 1.5 weeks. I'm having compatibility issues with my 15PM. It happens both wired and wirelessly, I get the error "Accessory not supported by this device". It used to happen all the time, now it's 50/50. Sometimes it works. I've removed and added Bluetooth multiple times on phone and deck, I bought a belkin usb-c to usb-a cable today and it seems to fix it but the problem comes back. I've changed the setting "FaceID and passcode-allow access when locked-accessories." The car stereo guy reckons it's definitely an issue with the phone not the deck, I'm inclined to believe him since the error states "by this device". Any advice appreciated.

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 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 ?

AVPictureInPictureControllerContentSource *contentSource = [[AVPictureInPictureControllerContentSource alloc] initWithSampleBufferDisplayLayer:self.renderView.sampleBufferDisplayLayer playbackDelegate:self]; AVPictureInPictureController *pictureInPictureController = [[AVPictureInPictureController alloc] initWithContentSource:contentSource]; pictureInPictureController.delegate = self; (void)pictureInPictureController:(AVPictureInPictureController *)pictureInPictureController failedToStartPictureInPictureWithError:(NSError *)error { //error NSError * domain: @"PGPegasusErrorDomain" - code: -1003 0x00000002819fe3a0 } when first start the PiP play, I got the error "//error NSError * domain: @"PGPegasusErrorDomain" - code: -1003 0x00000002819fe3a0", why? and second start is Ok.

Two issues: No matter what I set in try audioSession.setPreferredSampleRate(x) the sample rate on both iOS and macOS is always 48000 when the output goes through the speaker, and 24000 when my Airpods connect to an iPhone/iPad. Now, I'm checking the current output loudness to animate a 3D character, using mixerNode.installTap(onBus: 0, bufferSize: y, format: nil) { [weak self] buffer, time in Task { @MainActor in // calculate rms and animate character accordingly but any buffer size under 4800 is just ignored and the buffers I get are 4800 sized. This is ok, when the sampleRate is currently 48000, as 10 samples per second lead to decent visual results. But when AirPods connect, the samplerate is 24000, which means only 5 samples per second, so the character animation looks lame. My AVAudioEngine setup is the following: audioEngine.connect(playerNode, to: pitchShiftEffect, format: format) audioEngine.connect(pitchShiftEffect, to: mixerNode, format: format) audioEngine.connect(mixerNode, to: audioEngine.outputNode, format: nil) Now, I'd be fine if the outputNode runs at whatever if it needs, as long as my tap would get at least 10 samples per second. PS: Specifying my favorite format in the let format = AVAudioFormat(standardFormatWithSampleRate: 48_000, channels: 2)! mixerNode.installTap(onBus: 0, bufferSize: y, format: format) doesn't change anything either

Environment→ ・Device: iPad 10th generation ・OS：**iOS18.3.2 We're using AVAudioPlayer to play a sound when a button is tapped. In our use case, this button can be tapped very frequently — roughly every 0.1 to 0.2 seconds. Each tap triggers the following function: var audioPlayer: AVAudioPlayer? func soundPlay(resource: String, type: String){ guard let path = Bundle.main.path(forResource: resource, ofType: type) else { return } do { audioPlayer = try AVAudioPlayer(contentsOf: URL(fileURLWithPath: path)) audioPlayer!.delegate = self try audioSession.setCategory(.playback) } catch { return } self.audioPlayer!.play() } The issue is that under high-frequency tapping (especially around 0.1–0.15s intervals), the app occasionally crashes. The crash does not occur every time, but it happens randomly — sometimes within 30 seconds, within 1 minute, or even 3 minutes of continuous tapping. Interestingly, adding a delay of 0.2 seconds between button taps seems to prevent the crash entirely. Delays shorter than 0.2 seconds (e.g.,0.15s,0.18s) still result in occasional crashes. My questions are: **Is this expected behavior from AVAudioPlayer or AVAudioSession? Could this be a known issue or a limitation in AVFoundation? Is there any documentation or guidance on handling frequent sound playback safely?** Any insights or recommendations on how to handle rapid, repeated audio playback more reliably would be appreciated.

I have an AUv3 that passes all validation and can be loaded into Logic Pro without issue. The UI for the plug in can be any aspect ratio but Logic insists on presenting it in a view with a fixed aspect ratio. That is when resizing, both the height and width are resized. I have never managed to work out what it is I need to do specify to Logic to allow the user to resize width or height independently of each other. Can anyone tell me what I need to specify in the AU code that will inform Logic that the view can be resized from any side of the window/panel?

Hi Team, We are using AVFoundation to read metadata from a stream and have noticed some delay between when the stream provides metadata and when the app receives it. Could someone from the team advise on ways to reduce this? Thanks

Hi, I’ve been experiencing a strange issue with video playback on my iPhone. While watching videos, the image will suddenly shift it becomes more greyish, then sometimes briefly goes black, and then returns to normal bright quality. This can happen multiple times during a single video. This is not limited to the Photos app. I’ve seen it happen: In the Photos app when playing videos I recorded myself In Snapchat when watching videos sent by others Occasionally in other social media apps as well Additional details: HDR Video is turned off Apple ProRes is turned off Tried both 4K 60fps and 4K 30fps Camera format set to “Most Compatible” Low Power Mode is off Issue happens whether the phone is cool or warm Doesn’t seem related to the video file itself the same file exported to another device looks fine all the way through These exact same videos play back completely normally on my iPad with no brightness or contrast shifts at all I’m currently on the iOS 17 public beta, but this issue was happening before I installed the beta as well, so it’s not beta-specific It almost feels like the display or system is switching between different brightness/contrast profiles mid playback, regardless of the app. Has anyone else experienced this, and is there a way to disable this behavior so the brightness and color stay consistent during video playback? Thanks in advance!

According to the header file the outputVolume properties supported range is 0.0-1.0: /*! @property outputVolume @abstract The mixer's output volume. @discussion This accesses the mixer's output volume (0.0-1.0, inclusive). @property (nonatomic) float outputVolume; However when setting the volume to 2.0 the audio does indeed play louder. Is the header file out of date and if so, what is the supported range for outputVolume? Thanks

Dear Sirs, I'd like to add an icon to my audio driver based on AudioDriverKit. This icon should show up left of my audio device in the audio devices dialog. For an Audio Server Plugin I managed to do this using the property kAudioDevicePropertyIcon and CFBundleCopyResourceURL(...) but how would you do this with AudioDriverKit? Should I use IOUserAudioCustomProperty or IOUserAudioControl and how would I refer to the Bundle? Is there an example available somewhere? Thanks and best regards, Johannes

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’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]

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()

Hi, In my project I am using AVFoundation for recording the audio. We are using AVAudioMixerNode class below method to record the audio packet. **func installTap( onBus bus: AVAudioNodeBus, bufferSize: AVAudioFrameCount, format: AVAudioFormat?, block tapBlock: @escaping AVAudioNodeTapBlock ) ** It works perfectly fine. But in production env some small percentage of the user we are facing issue like after recording few packets it stops automatically without stopping the audio engine. Can anyone help here that why this happens? I have also observed for mediaServicesWereResetNotification and added log on receiving this notification but when this issue happens I don't see any occurence of this log. Also is there any callback when the engine stops?

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?

In Final Cut Pro, keyframes for transform parameters (such as Position, Scale, and Rotation) are automatically set to “Smooth” interpolation. This often results in undesired easing between keyframes, especially when linear motion is required. Currently, we have to manually adjust each keyframe to "Linear" using the Video Animation Editor, which can be time-consuming when working with many keyframes. Would it be possible to add an option to set the default keyframe interpolation to "Linear"—either globally in Preferences or per parameter in the Inspector? This would greatly streamline the animation workflow for many editors. Thank you for considering this request!

My app Balletrax is a music player for people to use while they teach ballet. Used to be you could silence notifications during use, but now the customer seems to have to know how to use Focus mode, remember to turn it on and off, and have to check the notifications one does and doesn't want to use. Is there no way to silence all notifications when the app is in use?

Capturing more than one display is no longer working with macOS Sequoia. We have a product that allows users to capture up to 2 displays/screens. Our application is using gstreamer which in turn is based on AVFoundation. I found a quick way to replicate the issue by just running 2 captures from separate terminals. Assuming display 1 has device index 0, and display 2 has device index 1, here are the steps: install gstreamer with brew install gstreamer Then open 2 terminal windows and launch the following processes: terminal 1 (device-index:0): gst-launch-1.0 avfvideosrc -e device-index=0 capture-screen=true ! queue ! videoscale ! video/x-raw,width=640,height=360 ! videoconvert ! osxvideosink terminal 2 (device-index:1): gst-launch-1.0 avfvideosrc -e device-index=1 capture-screen=true ! queue ! videoscale ! video/x-raw,width=640,height=360 ! videoconvert ! osxvideosink The first process that is launched will show the screen, the second process launched will not. Testing this on macOS Ventura and Sonoma works as expected, showing both screens. I submitted the same issue on Feedback Assistant: FB15900976