I might have misunderstood the docs, but is Call Translation going to be available for VOIP applications? Eg in an already connected VOIP call, would it be possible for Call Translations to be enabled on an iOS 26 and Apple Intelligence supported device? I have personally tried it and it doesn’t look like it supported VOIP but would love to confirm this. reference: https://developer.apple.com/documentation/callkit/cxsettranslatingcallaction/

I'm using the new SpeechAnalyzer framework to detect certain commands and want to improve accuracy by giving context. Seems like AnalysisContext is the solution for this, but couldn't find any usage example. So I want to make sure that I'm doing it right or not. let context = AnalysisContext() context.contextualStrings = [ AnalysisContext.ContextualStringsTag("commands"): [ "set speed level", "set jump level", "increase speed", "decrease speed", ... ], AnalysisContext.ContextualStringsTag("vocabulary"): [ "speed", "jump", ... ] ] try await analyzer.setContext(context) With this implementation, it still gives outputs like "Set some speed level", "It's speed level", etc. Also, is it possible to make it expect number after those commands, in order to eliminate results like "set some speed level to" (instead of two).

I am having issues deploying my iOS app, that uses ShazamKit, to get working on a Mac with Apple silicon. When uploading the archive to App Store Connect I do get ITMS-90863: Macs with Apple silicon support issue - The app links with libraries that aren’t present in macOS: /usr/lib/swift/libswiftShazamKit.dylib Is ShazamKit not supported for iOS apps that can run on Macs with Apple silicon? Or is there something I should fix in my setup / deployment?

So experimenting with the new SpeechTranscriber, if I do: let transcriber = SpeechTranscriber( locale: locale, transcriptionOptions: [], reportingOptions: [.volatileResults], attributeOptions: [.audioTimeRange] ) only the final result has audio time ranges, not the volatile results. Is this a performance consideration? If there is no performance problem, it would be nice to have the option to also get speech time ranges for volatile responses. I'm not presenting the volatile text at all in the UI, I was just trying to keep statistics about the non-speech and the speech noise level, this way I can determine when the noise level falls under the noisefloor for a while. The goal here was to finalize the recording automatically, when the noise level indicate that the user has finished speaking.

Since many users like me use Apple Music on Android, the app is almost as feature-rich as iOS. It would be fantastic if the developers could add the new iOS 26 features to the Android app, along with a minor UI change. I know it’s challenging to implement liquid glass on Android hardware or design, but features like auto-mix, pronunciation, and translation could be added. kindly consider this request !!!!

Hi, I am trying to remove the audio controls for my app on the lock screen. Since I use WKWebView, there are 3 audio tags in my html and I play and pause em via JS. However, if I do not play any sound since app launch, there are no audio controls on the lock screen. But if I play one of those 3 files (they are even less then 3 Sec sound effects e.g. for buttons) the audio controls appears on lock screen. Note even when the sounds on pause() or not playing they were listed on the lock screen. What I have tried so far without success MPNowPlayingInfoCenter.default().nowPlayingInfo = [:] and ``try audioSession.setCategory(.playback, mode: .default, options: []) try audioSession.setActive(false, options: .notifyOthersOnDeactivation)`` and UIApplication.shared.endReceivingRemoteControlEvents() Another problem is that the app scales with iOS system settings "display zoom". Is there a way to deny it? It is latest Xcode verion 16.3 and iOS 18. I have no background mode in my Capabilities. Nothing worked so far. Has anyone an idea? Greetings

Hi all, I'm working on an audio visualizer app that plays files from the user's music library utilizing MediaPlayer and AVAudioEngine. I'm working on getting the music library functionality working before the visualizer aspect. After setting up the engine for file playback, my app inexplicably crashes with an EXC_BREAKPOINT with code = 1. Usually this means I'm unwrapping a nil value, but I think I'm handling the optionals correctly with guard statements. I'm not able to pinpoint where it's crashing. I think it's either in the play function or the setupAudioEngine function. I removed the processAudioBuffer function and my code still crashes the same way, so it's not that. The device that I'm testing this on is running iOS 26 beta 3, although my app is designed for iOS 18 and above. After commenting out code, it seems that the app crashes at the scheduleFile call in the play function, but I'm not fully sure. Here is the setupAudioEngine function: private func setupAudioEngine() { do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .default) try AVAudioSession.sharedInstance().setActive(true) } catch { print("Audio session error: \(error)") } engine.attach(playerNode) engine.attach(analyzer) engine.connect(playerNode, to: analyzer, format: nil) engine.connect(analyzer, to: engine.mainMixerNode, format: nil) analyzer.installTap(onBus: 0, bufferSize: 1024, format: nil) { [weak self] buffer, _ in self?.processAudioBuffer(buffer) } } Here is the play function: func play(_ mediaItem: MPMediaItem) { guard let assetURL = mediaItem.assetURL else { print("No asset URL for media item") return } stop() do { audioFile = try AVAudioFile(forReading: assetURL) guard let audioFile else { print("Failed to create audio file") return } duration = Double(audioFile.length) / audioFile.fileFormat.sampleRate if !engine.isRunning { try engine.start() } playerNode.scheduleFile(audioFile, at: nil) playerNode.play() DispatchQueue.main.async { [weak self] in self?.isPlaying = true self?.startDisplayLink() } } catch { print("Error playing audio: \(error)") DispatchQueue.main.async { [weak self] in self?.isPlaying = false self?.stopDisplayLink() } } } Here is a link to my test project if you want to try it out for yourself: https://github.com/aabagdi/VisualMan-example Thanks!

Sequoia 15.4.1 (24E263) XCode: 16.3 (16E140) Logic Pro: 11.2.1 I’ve been developing a complex audio unit for Mac OS that works perfectly well in its own bespoke host app and is now well into its beta testing stage. It did take some effort to get it to work well in Logic Pro however and all was fine and working well until: The AU part is an empty app extension with a framework containing its code. The framework contains Swift code for the UI and C code for the DSP parts. When the framework is compiled using the Swift 5 compiler the AU will run in Logic with no problems. (I should also mention that AU passes the most strict auval tests). But… when the framework is compiled with Swift 6 Logic Pro cannot load it. Logic displays a message saying the audio unit could not be loaded and to contact the developer. My own host app loads the AU perfectly well with the Swift 6 version, so I know there’s nothing wrong with the audio unit. I cannot find any differences in any of the built output files except, of course, the actual binary code in the framework. I’ve worked for hours on this and cannot find a solution other than to build the framework in Swift 5. (I worked hard to get all the async code updated and working with Swift 6! so I feel a little cheated!) What is happening? Is this a bug in Logic? Is this a bug in Swift 6 compiler/linker? I’m at the Duh! hands in the air, tearing out hair stage! ( once again!)

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.

I'm trying to implement airplay into my app. I can successfully playback sound and trigger the airplay selector sheet. If the target device is a Bluetooth only device I can connect with no problem and stream the audio to the Bluetooth device, but if the audio device is a airplay specific device like a HomePod or an Apple TV when I select it, I get a spinning icon, indicating that it is trying to connect, and eventually it times out and stops without connecting. I don't believe it is an AirPlay audio issue because if I go to a different app, for example a podcast app and select my HomePods for output, and then switch back to my app. My audio will correctly stream to the HomePod. Not only that, I have it so that my icon will change color to indicate that it is connected via airplay and it is correctly indicating that it is connected via AirPlay. But I cannot then disconnect it using the Airplay selector. The issue appears to be in the AirPlay selection side, which I have spent several days attempting to troubleshoot mostly using ChatGPT to suggest code different than what I have to maybe work around the issue. Mostly it is focused on the audio player section, but it doesn't seem like that is really the route that is the problem.

Bug Report: ScreenCaptureKit System Audio Capture Crashes with EXC_BAD_ACCESS Summary When using ScreenCaptureKit to capture system audio for extended periods, the application crashes with EXC_BAD_ACCESS in Swift's error handling runtime. The crash occurs in swift_getErrorValue when trying to process an error from the SCStream delegate method didStopWithError. This appears to be a framework-level issue in ScreenCaptureKit or its underlying ReplayKit implementation. Environment macOS Sonoma 14.6.1 Swift 5.8 ScreenCaptureKit framework Detailed Description Our application captures system audio using ScreenCaptureKit's audio capture capabilities. After successfully capturing for several minutes (typically after 3-4 segments of 60-second recordings), the application crashes with an EXC_BAD_ACCESS error. The crash happens when the Swift runtime attempts to process an error in the SCStreamDelegate.stream(_:didStopWithError:) method. The crash consistently occurs in swift_getErrorValue when attempting to access the class of what appears to be a null object. This suggests that the error being passed from the system framework to our delegate method is malformed or contains invalid memory. Steps to Reproduce Create an SCStream with audio capture enabled Add audio output to the stream Start capture and write audio data to disk Allow the capture to run for several minutes (3-5 minutes typically triggers the issue) The app will crash with EXC_BAD_ACCESS in swift_getErrorValue Code Sample func stream(_ stream: SCStream, didStopWithError error: Error) { print("Stream stopped with error: \(error)") // Crash occurs before this line executes } func stream(_ stream: SCStream, didOutputSampleBuffer sampleBuffer: CMSampleBuffer, of type: SCStreamOutputType) { guard type == .audio, sampleBuffer.isValid else { return } // Process audio data... } Expected Behavior The error should be properly propagated to the delegate method, allowing for graceful error handling and recovery. Actual Behavior The application crashes with EXC_BAD_ACCESS when the Swift runtime attempts to process the error in swift_getErrorValue. Crash Log Details Thread #35, queue = 'com.apple.NSXPCConnection.m-user.com.apple.replayd', stop reason = EXC_BAD_ACCESS (code=1, address=0x0) frame #0: 0x0000000194c3088c libswiftCore.dylib`swift::_swift_getClass(void const*) + 8 frame #1: 0x0000000194c30104 libswiftCore.dylib`swift_getErrorValue + 40 frame #2: 0x00000001057fba30 shadow`NewScreenCaptureService.stream(stream=0x0000600002de6700, error=Swift.Error @ 0x000000016b7b5e30) at NEW+ScreenCaptureService.swift:365:15 frame #3: 0x00000001057fc050 shadow`@objc NewScreenCaptureService.stream(_:didStopWithError:) at <compiler-generated>:0 frame #4: 0x0000000219ec5ca0 ScreenCaptureKit`-[SCStreamManager stream:didStopWithError:] + 456 frame #5: 0x00000001ca68a5cc ReplayKit`-[RPScreenRecorder stream:didStopWithError:] + 84 frame #6: 0x00000001ca696ff8 ReplayKit`-[RPDaemonProxy stream:didStopWithError:] + 224 Printing description of stream._streamQueue: error: ObjectiveC.id:4294967281:18: note: 'id' has been explicitly marked unavailable here public typealias id = AnyObject ^ error: /var/folders/v4/3xg1hmp93gjd8_xlzmryf_wm0000gn/T/expr23-dfa421..cpp:1:65: 'id' is unavailable in Swift: 'id' is not available in Swift; use 'Any' Swift._DebuggerSupport.stringForPrintObject(Swift.UnsafePointer<id>(bitPattern: 0x104ae08c0)!.pointee) ^~ ObjectiveC.id:2:18: note: 'id' has been explicitly marked unavailable here public typealias id = AnyObject ^ warning: /var/folders/v4/3xg1hmp93gjd8_xlzmryf_wm0000gn/T/expr23-dfa421..cpp:5:7: initialization of variable '$__lldb_error_result' was never used; consider replacing with assignment to '_' or removing it var $__lldb_error_result = __lldb_tmp_error ~~~~^~~~~~~~~~~~~~~~~~~~ _ Before the crash, we observed this error message in the console: [ERROR] *****SCStream*****RemoteAudioQueueOperationHandlerWithError:1015 Error received from the remote queue -16665 Additional Context The issue occurs consistently after approximately 3-4 successful audio segment recordings of 60 seconds each Commenting out custom segment rotation logic does not prevent the crash The crash involves XPC communication with Apple's ReplayKit daemon The error appears to be corrupted or malformed when crossing the XPC boundary Workarounds Attempted Added proper thread safety for all published properties using DispatchQueue.main.async Implemented more robust error handling in the delegate methods None of these approaches prevented the crash since it occurs at the Swift runtime level before our code executes. Impact This issue prevents reliable long-duration audio capture using ScreenCaptureKit. This bug significantly limits the usefulness of ScreenCaptureKit for any application requiring continuous system audio capture for more than a few minutes. Perhaps this issue might be related to a macOS bug where the system dialog indicates that the screen is being shared, even though nothing is actually being shared. Moreover, when attempting to stop sharing, nothing happens.

Hello Apple Developer Community, I am seeking clarification on the intended display behavior of HLS audio tracks within the iOS 26 (or current beta) native player, specifically concerning the NAME and LANGUAGE attributes of the EXT-X-MEDIA tag. In our HLS manifests, we define alternative audio tracks using EXT-X-MEDIA tags, like so: #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",LANGUAGE="ja",NAME="AUDIO-1",DEFAULT=YES,AUTOSELECT=YES,URI="audio_ja.m3u8" #EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="audio",LANGUAGE="ja",NAME="AUDIO-2",URI="audio_en.m3u8" Our observation is that when an audio track is selected and its name is displayed in the native iOS media controls (e.g., Control Center or within a full-screen video player's UI), the value specified in the NAME attribute ("AUDIO-1", "AUDIO-2") does not seem to be used. Instead, the display appears to derive from the LANGUAGE attribute ("ja", "en"), often showing the system's localized string for that language (e.g., "Japanese", "English"). We would like to understand the official or intended behavior regarding this. Is it the expected behavior for the iOS native player to prioritize the LANGUAGE attribute (or its localized equivalent) over the NAME attribute for displaying the selected audio track's label? If this is the intended design, what is the recommended best practice for developers who wish to present a custom, human-readable name for audio tracks (beyond the standard language name) in the native iOS UI? Are there any specific AVPlayer properties or AVMediaSelectionOption considerations that would allow more granular control over this display, or is this entirely managed by the system based on the LANGUAGE attribute? Any insights or official guidance on this behavior in iOS 26 (and potentially previous versions) would be greatly appreciated. Thank you for your time and assistance.

So, I've been wondering how fast a an offline STT -> ML Prompt -> TTS roundtrip would be. Interestingly, for many tests, the SpeechTranscriber (STT) takes the bulk of the time, compared to generating a FoundationModel response and creating the Audio using TTS. E.g. InteractionStatistics: - listeningStarted: 21:24:23 4480 2423 - timeTillFirstAboveNoiseFloor: 01.794 - timeTillLastNoiseAboveFloor: 02.383 - timeTillFirstSpeechDetected: 02.399 - timeTillTranscriptFinalized: 04.510 - timeTillFirstMLModelResponse: 04.938 - timeTillMLModelResponse: 05.379 - timeTillTTSStarted: 04.962 - timeTillTTSFinished: 11.016 - speechLength: 06.054 - timeToResponse: 02.578 - transcript: This is a test. - mlModelResponse: Sure! I'm ready to help with your test. What do you need help with? Here, between my audio input ending and the Text-2-Speech starting top play (using AVSpeechUtterance) the total response time was 2.5s. Of that time, it took the SpeechAnalyzer 2.1s to get the transcript finalized, FoundationModel only took 0.4s to respond (and TTS started playing nearly instantly). I'm already using reportingOptions: [.volatileResults, .fastResults] so it's probably as fast as possible right now? I'm just surprised the STT takes so much longer compared to the other parts (all being CoreML based, aren't they?)

I've got a problem with my app where I'm testing it on my own phone. I'm using audio kit to generate tones as part of the app. Everything seems to work fine. Sounds start, Stop, etc. They play when the app is closed and when the phone is locked, so background is working. However, I'm seeing an issue where, even when STOP is pressed and the application exited, if I get a notification such as a text message, the base tone for the app starts to play. If I then open the app, check the Start/Stop button - it says start so that. hasnt' been activated. If I click Start, then a 2nd tone starts. This one stops with the Stop button. However the original tone that was set off by an incoming message carries on playing. Until I go to the Open Apps View on the phone and slide the application upwards. For the life of me, I can't figure out whats happening here.

Hi, I am getting into a trap. Please check stack-trace, howto fix this? regards, Joël stack-trace with ExtAudioFileWrite

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

AVAudioSessionCategoryOptionAllowBluetooth is marked as deprecated in iOS 8 in iOS 26 beta 5 when this option was not deprecated in iOS 18.6. I think this is a mistake and the deprecation is in iOS 26. Am I right? It seems that the substitute for this option is "AVAudioSessionCategoryOptionAllowBluetoothHFP". The documentation does not make clear if the behaviour is exactly the same or if any difference should be expected... Has anyone used this option in iOS 26? Should I expect any difference with the current behaviour of "AVAudioSessionCategoryOptionAllowBluetooth"? Thank you.

Radio stations are currently not supported by the MusicKit SDK for Android. The SDK has not been updated for years now. It lacks pretty big features of Apple Music

Hello, I’m new here. I'm developing an iOS app and I’d like to know whether it is possible to detect if a phone call is being recorded by another app running in the background. I’ve already reviewed the documentation for CallKit and AVAudioSession, but I couldn’t find anything related. My expectation was that iOS might provide some callback or API to indicate if a call is being recorded (third-party apps), but so far I haven’t found a way. My questions are: Does iOS expose any API to detect if a call is being recorded? If not, is there any indirect, Apple's policy compliant method (e.g., microphone usage events) that can be relied upon? Or is this something that iOS explicitly prevents for privacyreasons? Expecting solutions that align with Apple’s policies and would be accepted under the App Store Review Guidelines. Thanks in advance for any guidance.

Description: I’m developing a tvOS app using SwiftUI where we play background audio (music) in the Welcome screen, with support for offline playback via local caching. 🔹 Feature Overview App fetches audio metadata from API Starts streaming audio (HLS .m3u8) immediately In parallel, downloads the raw audio file (.mp3) Once download completes: Switches playback from streaming → local file On next launch (offline mode), app plays audio from local storage 🔹 Issue This flow works perfectly on the Simulator, but on a real Apple TV device: Audio plays for a few seconds (2–5 sec) and then stops Especially after switching from streaming → local file No explicit AVPlayer error is logged Playback sometimes stops after UI updates or periodic API refresh 🔹 Implementation Details Using AVPlayer with AVPlayerItem Background audio controlled via a shared manager (singleton) Files stored locally using FileManager (currently using .cachesDirectory) Switching playback using: player.replaceCurrentItem(with: AVPlayerItem(url: localURL)) player.play() 🔹 Observations Works reliably on Simulator On device: Playback stops silently Seems related to lifecycle, buffering, or file access No issues when continuously streaming (without switching to local) 🔹 Questions Is there any limitation or known issue with AVPlayer when switching from streaming (HLS) to local file playback on tvOS? Are there specific requirements for playing locally cached media files on tvOS (e.g., file location, permissions, or sandbox behavior)? What is the recommended storage location and size limit for cached media files on tvOS? We understand tvOS has limited persistent storage Is .cachesDirectory the correct approach for this use case? Are there known differences in AVPlayer behavior between Simulator and real Apple TV devices (especially regarding buffering or lifecycle)? What is the recommended approach for implementing offline background audio on tvOS apps? 🔹 Goal We want to implement a reliable system where: Audio streams initially Seamlessly switches to local file after download Continues playing without interruption Supports offline playback on subsequent launches Any guidance or best practices would be greatly appreciated. Thank you!