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!}

Hello everyone, I’m new to Swift development and have been working on an audio module that plays a specific sound at regular intervals - similar to a workout timer that signals switching exercises every few minutes. Following AVFoundation documentation, I’m configuring my audio session like this: let session = AVAudioSession.sharedInstance() try session.setCategory( .playback, mode: .default, options: [.interruptSpokenAudioAndMixWithOthers, .duckOthers] ) self.engine.attach(self.player) self.engine.connect(self.player, to: self.engine.outputNode, format: self.audioFormat) try? session.setActive(true) When it’s time to play cues, I schedule playback on a DispatchQueue: // scheduleAudio uses DispatchQueue self.scheduleAudio(at: interval.start) { do { try audio.engine.start() audio.node.play() for sample in interval.samples { audio.node.scheduleBuffer(sample.buffer, at: AVAudioTime(hostTime: sample.hostTime)) } } catch { print("Audio activation failed: \(error)") } } This works perfectly in the foreground. But once the app goes into the background, the scheduled callback runs, yet the audio engine fails to start, resulting in an error with code 561015905. Interestingly, if the app is already playing audio before going to the background, the scheduled sounds continue to play as expected. I have added the required background audio mode to my Info plist file by including the key UIBackgroundModes with the value audio. Is there anything else I should configure? What is the best practice to play periodic audio when the app runs in the background? How do apps like turn-by-turn navigation handle continuous audio playback in the background? Any advice or pointers would be greatly appreciated!

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/.

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!

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

(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?

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’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]

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

Dear Apple Developer Community, I'm encountering a critical issue with the MusicLibrary.shared.createPlaylist() method in MusicKit that's affecting our app's core functionality. Despite implementing all recommended authorization checks, the app consistently freezes for some users when this method is called. What we've already verified before calling createPlaylist(): Network connectivity is properly checked and confirmed Apple Music authorization is explicitly requested via MusicAuthorization.request() User subscription status is verified using MusicSubscription.current.canPlayCatalogContent Despite these precautions, many users report that their app completely freezes when attempting to create a playlist. This is particularly concerning as playlist creation is a core feature of our application. User-reported workarounds (with mixed success): Some users have resolved the issue by restarting their devices or reinstalling our app Others report success after enabling "Sync Library" in Settings → Music Unfortunately, a significant number of users are still experiencing the issue even after trying both solutions above We've reviewed the MusicKit documentation thoroughly and ensured our implementation follows all best practices. Our app correctly handles permissions and uses the async/await pattern as required by the API. Is there a known issue with the createPlaylist() method that might cause it to block indefinitely? Are there additional authorization steps or settings we should be checking before calling this method? Could this be related to how MusicKit communicates with Apple Music servers? Any insights from the developer community or official guidance would be greatly appreciated as this issue is severely impacting our user experience. Thank you for your assistance

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

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

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; }

My Environment: Device: Mac (Apple Silicon, arm64) OS: macOS 15.6.1 Description: I'm developing a music app and have encountered an issue where I cannot update the playbackState in MPNowPlayingInfoCenter after my app loses audio focus to another app. Even though my app correctly calls [MPNowPlayingInfoCenter defaultCenter].playbackState = .paused, the system's Now Playing UI (Control Center, Lock Screen, AirPods controls) does not reflect this change. The UI remains stuck until the app that currently holds audio focus also changes its playback state. I've observed this same behavior in other third-party music apps from the App Store, which suggests it might be a system-level issue. Steps to Reproduce: Use two most popular music apps in Chinese app Store (NeteaseCloud music and QQ music) (let's call them App A and App B): Start playback in App A. Start playback in App B. (App B now has audio focus, and App A is still playing). Attempt to pause App A via the system's Control Center or its own UI. Observed Behavior: App A's audio stream stops, but in the system's Now Playing controls, App A still appears to be playing. The progress bar continues to advance, and the pause button becomes unresponsive. If you then pause App B, the Now Playing UI for App A immediately corrects itself and displays the proper "paused" state. My Questions: Is there a specific procedure required to update MPNowPlayingInfoCenter when an app is not the current "Now Playing" application? Is this a known issue or expected behavior in macOS? Are there any official workarounds or solutions to ensure the UI updates correctly?

1, Devices Apple TV HD tvOS 18.5 (22L572) Mode(A1625) HomePod Gen1 OS18.5 (22L572) 2, Test cases: 2.1 UIViewController + AVPlayerLayer + AVPlayer Result: Can not AirPlay to HomePod Sample code for UIViewController + AVPlayer 2.1 AVPlayerViewController + AVPlayer Result: Can AirPlay to HomePod Sample code for AVPlayerViewController + AVPlayer

When multiple identical songs are added to a playlist, Playlist.Entry.id uses a suffix-based identifier (e.g. songID_0, songID_1, etc.). Removing one entry causes others to shift, changing their .id values. This leads to diffing errors and collection view crashes in SwiftUI or UIKit when entries are updated. Steps to Reproduce: Add the same song to a playlist multiple times. Observe .id.rawValue of entries (e.g. i.SONGID_0, i.SONGID_1). Remove one entry. Fetch playlist again — note the other IDs have shifted. FB18879062

On Apple TV 4K 3rd generation, with tvOS 26 beta 2, when two HomePod 2 are paired to the device, music and movie sources with Dolby Atmos can only be listened to in stereo. dolby atmos not supported

I'm capturing video stream from GoPro camera (via UDP MPEG-TS packets) but unable to play in iOS app. can someone provide a source code to decode MPEG-TS data to CMSampleBuffer.

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).

When I play an m3u8 video using AVPlayer, it can play smoothly at 2x speed. However, when I set it to 3x speed, the playback is not smooth and there is no sound.