I'm experiencing a significant limitation with MusicKit's Dolby Atmos implementation on macOS and would appreciate clarification on whether this is intended behavior or if there are solutions available. When streaming Dolby Atmos content through MusicKit's ApplicationMusicPlayer, the output is limited to 2-channel stereo, even when: Audio MIDI Setup is configured for 7.1.4 (12-channel) output The same tracks play in full multichannel through the native Apple Music app Dolby Atmos is set to "Automatic" in Apple Music preferences Please let me know if there is anyway to enable this. If not, is this documented anywhere? Thanks!

Hi, I just started to develop audio unit hosting support in my application. Offline rendering seems to work except that I hear no output, but why? I suspect with the player goes something wrong. I connect to CoreAudio in a different location in the code. Here are some error messages I faced so far: 2025-08-14 19:42:04.132930+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:42:04.151171+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:43:08.344530+0200 com.gsequencer.GSequencer[34358:18614927] AUAudioUnit.mm:1417 Cannot set maximumFramesToRender while render resources allocated. 2025-08-14 19:43:08.346583+0200 com.gsequencer.GSequencer[34358:18614927] [avae] AVAEInternal.h:104 [AVAudioSequencer.mm:121:-[AVAudioSequencer(AVAudioSequencer_Player) startAndReturnError:]: (impl->Start()): error -10852 ** (<unknown>:34358): WARNING **: 19:43:08.346: error during audio sequencer start - -10852 I have implemented an AVAudioEngine based AudioUnit host. Here I instantiate player and effect: /* audio engine */ audio_engine = [[AVAudioEngine alloc] init]; fx_audio_unit_audio->audio_engine = (gpointer) audio_engine; av_format = (AVAudioFormat *) fx_audio_unit_audio->av_format; /* av audio player node */ av_audio_player_node = [[AVAudioPlayerNode alloc] init]; /* av audio unit */ av_audio_unit_effect = [[AVAudioUnitEffect alloc] initWithAudioComponentDescription:[((AVAudioUnitComponent *) AGS_AUDIO_UNIT_PLUGIN(base_plugin)->component) audioComponentDescription]]; av_audio_unit = (AVAudioUnit *) av_audio_unit_effect; fx_audio_unit_audio->av_audio_unit = av_audio_unit; /* audio sequencer */ av_audio_sequencer = [[AVAudioSequencer alloc] initWithAudioEngine:audio_engine]; fx_audio_unit_audio->av_audio_sequencer = (gpointer) av_audio_sequencer; /* output node */ [[AVAudioOutputNode alloc] init]; /* audio player and audio unit */ [audio_engine attachNode:av_audio_player_node]; [audio_engine attachNode:av_audio_unit]; [audio_engine connect:av_audio_player_node to:av_audio_unit format:av_format]; [audio_engine connect:av_audio_unit to:[audio_engine outputNode] format:av_format]; ns_error = NULL; [audio_engine enableManualRenderingMode:AVAudioEngineManualRenderingModeOffline format:av_format maximumFrameCount:buffer_size error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("enable manual rendering mode error - %d", [ns_error code]); } ns_error = NULL; [[av_audio_unit AUAudioUnit] allocateRenderResourcesAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("Audio Unit allocate render resources returned error - ErrorCode %d", [ns_error code]); } Then I render in a dedicated thread. ns_error = NULL; [audio_engine startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio engine start - %d", [ns_error code]); } [av_audio_sequencer prepareToPlay]; ns_error = NULL; [av_audio_sequencer startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio sequencer start - %d", [ns_error code]); } [av_audio_player_node play]; while(is_running){ /* pre sync */ /* IO buffers */ av_output_buffer = (AVAudioPCMBuffer *) scope_data->av_output_buffer; av_input_buffer = (AVAudioPCMBuffer *) scope_data->av_input_buffer; /* fill input buffer */ /* schedule av input buffer */ frame_position = 0; // (gint64) ((note_offset * absolute_delay) + delay_counter) * buffer_size; av_audio_player_node = (AVAudioPlayerNode *) fx_audio_unit_audio->av_audio_player_node; AVAudioTime *av_audio_time = [[AVAudioTime alloc] initWithHostTime:frame_position sampleTime:frame_position atRate:((double) samplerate)]; [av_audio_player_node scheduleBuffer:av_input_buffer atTime:av_audio_time options:0 completionHandler:nil]; /* render */ ns_error = NULL; status = [audio_engine renderOffline:AGS_FX_AUDIO_UNIT_AUDIO_FIXED_BUFFER_SIZE toBuffer:av_output_buffer error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("render offline error - %d", [ns_error code]); } } regards, Joël

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

Hello, I'm developing an app that displays a photo library using UICollectionView and PHCachingImageManager. I'd like to achieve a user experience similar to the native iOS Photos app, where low-quality images are shown quickly while scrolling, and higher-quality images are loaded for visible cells once scrolling stops. I'm currently using the following approach: While Scrolling: I'm using the UICollectionViewDataSourcePrefetching protocol. In the prefetchItemsAt method, I call startCachingImages with low-quality options to cache images in advance. After Scrolling Stops: In the scrollViewDidEndDecelerating method, I intend to load high-quality images for the currently visible cells. I have a few questions regarding this approach: What is the best practice for managing both low-quality and high-quality images efficiently with PHCachingImageManager? Is it correct to call startCachingImages with fastFormat options and then call it again with highQualityFormat in scrollViewDidEndDecelerating? How can I minimize the delay when a low-quality image is replaced by a high-quality one? Are there any additional strategies to help pre-load high-quality images more effectively?

In my app I use AVAssetReaderTrackOutput to extract PCM audio from a user-provided video or audio file and display it as a waveform. Recently a user reported that the waveform is not in sync with his video, and after receiving the video I noticed that the waveform is in fact double as long as the video duration, i.e. it shows the audio in slow-motion, so to speak. Until now I was using CMFormatDescription.audioStreamBasicDescription.mSampleRate which for this particular user video returns 22'050. But in this case it seems that this value is wrong... because the audio file has two audio channels with different sample rates, as returned by CMFormatDescription.audioFormatList.map({ $0.mASBD.mSampleRate }) The first channel has a sample rate of 44'100, the second one 22'050. If I use the first sample rate, the waveform is perfectly in sync with the video. The problem is given by the fact that the ratio between the audio data length and the sample rate multiplied by the audio duration is 8, double the ratio for the first audio file (4). In the code below this ratio is given by Double(length) / (sampleRate * asset.duration.seconds) When commenting out the line with the sampleRate variable definition in the code below and uncommenting the following line, the ratios for both audio files are 4, which is the expected result. I would expect audioStreamBasicDescription to return the correct sample rate, i.e. the one used by AVAssetReaderTrackOutput, which (I think) somehow merges the stereo tracks. The documentation is sparse, and in particular it’s not documented whether the lower or higher sample rate is used; in this case, it seems like the higher one is used, but audioStreamBasicDescription for some reason returns the lower one. Does anybody know why this is the case or how I should extract the sample rate of the produced PCM audio data? Should I always take the higher one? I created FB19620455. let openPanel = NSOpenPanel() openPanel.allowedContentTypes = [.audiovisualContent] openPanel.runModal() let url = openPanel.urls[0] let asset = AVURLAsset(url: url) let assetTrack = asset.tracks(withMediaType: .audio)[0] let assetReader = try! AVAssetReader(asset: asset) let readerOutput = AVAssetReaderTrackOutput(track: assetTrack, outputSettings: [AVFormatIDKey: Int(kAudioFormatLinearPCM), AVLinearPCMBitDepthKey: 16, AVLinearPCMIsBigEndianKey: false, AVLinearPCMIsFloatKey: false, AVLinearPCMIsNonInterleaved: false]) readerOutput.alwaysCopiesSampleData = false assetReader.add(readerOutput) let formatDescriptions = assetTrack.formatDescriptions as! [CMFormatDescription] let sampleRate = formatDescriptions[0].audioStreamBasicDescription!.mSampleRate //let sampleRate = formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate }).max()! print(formatDescriptions[0].audioStreamBasicDescription!.mSampleRate) print(formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate })) if !assetReader.startReading() { preconditionFailure() } var length = 0 while assetReader.status == .reading { guard let sampleBuffer = readerOutput.copyNextSampleBuffer(), let blockBuffer = sampleBuffer.dataBuffer else { break } length += blockBuffer.dataLength } print(Double(length) / (sampleRate * asset.duration.seconds))

One thing I've noticed on tvOS 26 is that if you try to set the AVPlayerViewController customInfoViewControllers property while the Content Tabs are on screen, your app will crash. *** Terminating app due to uncaught exception 'UIViewControllerHierarchyInconsistency', reason: 'trying to add child view controller that is already presented: <AVInfoPanelViewController: 0x1030cdc00>' *** First throw call stack: (0x18a7167bc 0x189a77510 0x18a7166a8 0x1ab425658 0x1b2ee9d54 0x1b2efcd60 0x1b2eaf3f0 0x1080f744c 0x107e021a8 0x107e01b3c 0x18de41c14 0x18de41ba8 0x18de48d28 0x18ad9e358 0x101fac5f0 0x101fc6228 0x101fe7278 0x101fbc6fc 0x101fbc63c 0x18a67a2e0 0x18a679418 0x18a673b34 0x1937e4d5c 0x1abb36588 0x1abb3ae80 0x1aae9dec4 0x108610174 0x1086100e4 0x108615140 0x189abd4d0) I've logged a feedback (FB19554461) but it's getting awfully late in the dev cycle. So I've been trying to think of a workaround. The problem is that customInfoViewControllers is pretty declarative in nature. There are no properties or delegate methods I am aware of that let me know when they are displaying or not. One trick I came up with was seeing if my custom info view controller's view was "visible" or not - I put that in quotes because it turns out it can be visible even when I think it's not, as when the transport bar is scrolled to the top my custom VC still has its top pixels showing, so it gets a viewDidAppear call. So, I tried to see if my view controllers view is completely visible, ie based on the results of the GGRect contains method. And that works! But the problem is it only accounts for my own custom info view controllers, and not the standard one that Apple provides. I can't think of a way at all to know whether that is showing. Any ideas?

After Picture-in-Picture is opened on the camera interface, the custom UI cannot be displayed. Is there any way to make the custom UI visible? If custom UI cannot be displayed, how do teleprompter-type apps in the store manage to show custom teleprompter text within the camera?

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!

I'm using Picture-in-Picture (PiP) mode in my native iOS application, which is similar to Google Meet, using the VideoSDK.live Native iOS SDK. The SDK has built-in support for PiP and it's working fine for the most part. However, I'm running into a problem: When the app enters PiP mode, the local camera (self-video) of the participant freezes or stops. I want to fix this and achieve the same smooth behavior seen in apps like Google Meet and WhatsApp, where the local camera keeps working even in PiP mode. I've been trying to find documentation or examples on how to achieve this but haven't had much luck. I came across a few mentions that using the camera in the background may require special entitlements from Apple (like in the entitlements file). Most of the official documentation says background camera access is restricted due to Apple’s privacy policies. So my questions are: Has anyone here successfully implemented background camera access in PiP mode on iOS? If yes, what permissions or entitlements are required? How do apps like WhatsApp and Google Meet achieve this functionality on iPhones? Any help, advice, or pointers would be really appreciated!

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!

I was testing audio playback from YouTube in Safari, and the sound was clipping heavily. At first, I thought it might be due to the poor quality of my small sound system. However, when I took a screenshot and the screenshot sound effect itself produced a loud clipping noise, it became clear that this is not a mechanical problem with my speakers, nor an issue specific to YouTube or Safari. This appears to be a system-wide audio issue in macOS Tahoe 26 - Beta 5.

On an iPhone running iOS 26 beta 5, url(for: FilePath("subdir/asset.mov")) most always throws this error: The URL for “subdir/asset.mov” couldn’t be retrieved: “asset.mov” couldn’t be copied to “subdir” because an item with the same name already exists. Yet, contents(at: FilePath("subdir/asset.mov")) always returns Data for a playable AVMovie. How can I avoid this url(for:) error? The asset pack in question is downloaded. The error persists even after pack deletion, redownload, relaunch, and combinations of that. // Assets repo root subdir.aar subdir/asset.mov subdir/asset_thumb.heic subdir/Manifest.json // Manifest.json { "assetPackID": "subdir", "downloadPolicy": { "onDemand": {} }, "fileSelectors": [ { "directory": "subdir", }, ], "platforms": [ "iOS", "visionOS" ] } xcrun ba-package subdir/Manifest.json -o subdir.aar xcrun ba-serve --host 192.168.0.10 -p 443 subdir.aar

After updating to iOS 18.5, we’ve observed that outgoing audio from our app intermittently stops being transmitted during VoIP calls using AVAudioSession configured with .playAndRecord and .voiceChat. The session is set active without errors, and interruptions are handled correctly, yet audio capture suddenly ceases mid-call. This was not observed in earlier iOS versions (≤ 18.4). We’d like to confirm if there have been any recent changes in AVAudioSession, CallKit, or related media handling that could affect audio input behavior during long-running calls. func configureForVoIPCall() throws { try setCategory( .playAndRecord, mode: .voiceChat, options: [.allowBluetooth, .allowBluetoothA2DP, .defaultToSpeaker]) try setActive(true) }

Since iOS 18, the system setting “Allow Audio Playback” (enabled by default) allows third-party app audio to continue playing while the user is recording video with the Camera app. This has created a problem for the app I’m developing. ➡️ The problem: My app plays continuous audio in both foreground and background states. If the user starts recording video using the iOS Camera app, the app’s audio — still playing in the background — gets captured in the video — obviously an unintended behavior. Yes, the user could stop the app manually before starting the video recording, but that can’t be guaranteed. As a developer, I need a way to stop the app’s audio before the video recording begins. So far, I haven’t found a reliable way to detect when video recording starts if ‘Allow Audio Playback’ is ON. ➡️ What I’ve tried: — AVAudioSession.interruptionNotification → doesn’t fire — devicesChangedEventStream → not triggered I don’t want to request mic permission (app doesn’t use mic). also, disabling the app from playing audio in the background isn’t an option as it is a crucial part of the user experience ➡️ What I need: A reliable, supported way to detect when the Camera app begins video recording, without requiring mic access — so I can stop audio and avoid unintentional overlap with the user’s recordings. Any official guidance, workarounds, or AVFoundation techniques would be greatly appreciated. Thanks.

Using the official SwiftTranscriptionSampleApp from WWDC 2025, speech transcription takes 14+ seconds from audio input to first result, making it unusable for real-time applications. Environment iOS: 26.0 Beta Xcode: Beta 5 Device: iPhone 16 pro Sample App: Official Apple SwiftTranscriptionSampleApp from WWDC 2025 Configuration Tested Locale: en-US (properly allocated with AssetInventory.allocate(locale:)) and es-ES Setup: All optimizations applied (preheating, high priority, model retention) I started testing in my own app to replace SFSpeech API and include speech detection but after long fights with documentation (this part is quite terrible TBH) I tested the example (https://developer.apple.com/documentation/speech/bringing-advanced-speech-to-text-capabilities-to-your-app) and saw same results. I added some logs to check the specific time: 🎙️ [20:30:41.532] ✅ Analyzer started successfully - ready to receive audio! 🎙️ [20:30:41.532] Listening for transcription results... 🎙️ [20:30:56.342] 🚀 FIRST TRANSCRIPTION RESULT after 14.810s: 'Hello' (isFinal: false) Questions Is this expected performance for iOS 26 Beta, because old SFSpeech is far faster? Are there additional optimization steps for SpeechTranscriber? Should we expect significant performance improvements in later betas?

Hi team, With regards to Call (Live) Translations on VOIP: Is it possible to invoke live translations within the app? (without going into the Call System UI) Is it possible to navigate users from app to Call System UI via an API? (and also invoking the new live translations directly) Will Apple support more languages apart from the current ones? (Currently I see 4 supported languages)

I ran 5.1 audio tests in both YouTube and Apple Music, and I noticed that when sound is supposed to play from the rear or front surround speakers, it’s also duplicated in the front left and right channels. I’m absolutely sure the issue is with the Apple TV, because I played the same video directly through my TV’s native system, and the channel separation was correct. Everything used to work perfectly before, so this must be a software issue. I’m currently on tvOS 26 Developer Beta 5, but I’m certain the problem also existed on the stable tvOS 18.5. I’ve already reset and updated my Apple TV, and I also tried switching the audio format to forced Dolby Atmos 5.1. On the forums, I mostly see complaints about Dolby Atmos not working at all — in my case, everything technically works, but not the way it’s supposed to.

I started playing which transcription of audio files on macOS today, latest beta of Xcode and latest beta of Tahoe. Transcription itself works really well, but for some reason the majority of the results contain no audioTimeRange. I got 22 single-word results with time ranges, spread out all over total file of 53 minutes. Is there something I can do to improve this? To my understanding, I have followed sample code and instructions very closely, but the SwiftTranscriptionSampleApp and other examples I've seen lead me to believe I should be getting a lot more time ranges than I actually do.

Dear Sirs, I’ve written a virtual audio driver based on AudioDriverKit and running as dext in my MacOS app. Sometimes when waking up from a sleep state the recording side of my driver extension seems to hang and I don’t see any calls to my io_operation callback. Then the recording app like a DAW seems to hang when trying to start a recording. This doesn’t happen after short sleep states or after a complete new start of my MacBook. I already opened a case in Feedback-Assistant on 5th of May (FB17503622) which also includes a sysdiagnose and a ktrace but I didn't get any feedback so far. Meanwhile some of our customers are getting angry and I'd like to know if there's anything I could do to fix this problem on my side. We’re not sure whether this worked in previous MacOS versions, we think we didn’t observe this before 15.3.1 but at least since 15.3.1. we’ve seen this problem. Best regards, Johannes

Hi all, I'm working on a custom Metal-based video pipeline using AVCaptureVideoDataOutput, and I've run into an unexpected issue related to exposure. Setup: I'm capturing video frames using kCVPixelFormatType_420YpCbCr8BiPlanarFullRange. In my Metal shader, I: Convert YCbCr (full range Rec.709) to linear Rec.709 RGB. Apply Rec.709 → sRGB gamma encoding. Output to .bgra8Unorm_srgb via MTKView. Everything renders correctly in terms of colorspace math, but the image appears significantly brighter (~+3 stops EV) compared to AVCaptureVideoPreviewLayer and the native iOS Camera app under the same camera exposure settings. What I’ve verified: The color transforms are correct: YCbCr709 to RGB, then linear to sRGB. I'm not applying any tone mapping or aggressive look LUTs yet. Camera exposure is locked using: device.setExposureModeCustom(duration: ..., iso: ...) The same EV (e.g., ISO 50, 1/125s, f/5.6) on my iPhone appears visually 3 stops brighter than on my digital cameras (Sony/Canon etc). To match the look of the preview layer or camera app, I have to simulate a ~–3 EV shift in my custom pipeline. Questions: Is AVCaptureVideoPreviewLayer applying extra tone mapping, digital gain, or contrast shaping (like OOTF etc)? Does the camera ISP expose "hotter" (i.e., with more light) internally for the preview layer than what we get in video frame buffers? Is there a standard way to compensate for this ISP behavior in custom pipelines using AVCaptureVideoDataOutput? Can this be accounted for using metadata (e.g., exposure bias, gain, gamma curve)?