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.

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

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.

Here is the demo from Apple's site This issues is specific to iOS 18. When running this demo, we are getting new text when we have a gap in speaking, the recognitionTask(with:resultHandler:) provides new text which is only spoken after the gap and not the concatenation of old text and the new spoken text.

Hi, I'm working on an AUv3 project. The app itself displays my icon. However the Auv3 extension does not display any icon in any host app (AUM, Drambo, etc.0). I thought that the extension would inherit the host app icon but that it does not appear to be the case. I tried to add the icon as a 1024x1024 file to the extension target and the update my extension plist file withe a CFBundleIconFile key but no luck either. It must surely be really easy. What am I missing? Thanks in advance for your help!

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.

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?)

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

I am trying to use the new SpeechAnalyzer framework in my Mac app, and am running into an issue for some languages. When I call AssetInstallationRequest.downloadAndInstall() for some languages, it throws an error: Error Domain=SFSpeechErrorDomain Code=1 "transcription.ar asset not found after attempted download." The ".ar" appears to be the language code, which in this case was Arabic. When I call AssetInventory.status(forModules:) before attempting the download, it is giving me a status of "downloading" (perhaps from an earlier attempt?). If this language was completely unsupported, I would expect it to return a status of "unsupported", so I'm not sure what's going on here. For other languages (Polish, for example) SpeechTranscriber.supportedLocale(equivalentTo:) is returning nil, so that seems like a clearly unsupported language. But I can't tell if the languages I'm trying, like Arabic, are supported and something is going wrong, or if this error represents something I can work around. Here's the relevant section of code. The error is thrown from downloadAndInstall(), so I never even get as far as setting up the SpeechAnalyzer itself. private func setUpAnalyzer() async throws { guard let sourceLanguage else { throw Error.languageNotSpecified } guard let locale = await SpeechTranscriber.supportedLocale(equivalentTo: Locale(identifier: sourceLanguage.rawValue)) else { throw Error.unsupportedLanguage } let transcriber = SpeechTranscriber(locale: locale, preset: .progressiveTranscription) self.transcriber = transcriber let reservedLocales = await AssetInventory.reservedLocales if !reservedLocales.contains(locale) && reservedLocales.count == AssetInventory.maximumReservedLocales { if let oldest = reservedLocales.last { await AssetInventory.release(reservedLocale: oldest) } } do { let status = await AssetInventory.status(forModules: [transcriber]) print("status: \(status)") if let installationRequest = try await AssetInventory.assetInstallationRequest(supporting: [transcriber]) { try await installationRequest.downloadAndInstall() } } ...

I am trying to debug the AAX version of my plugin (MIDI effect) on Pro Tools, but I am getting the following error (Mac console) when attempting to load it: dlsym cannot find symbol g_dwILResult in CFBundle etc.. I used Xcode 16.4 to build the plugin. Has anybody come across the same or a similar message? Best, Achillefs Axart Labs

Is there a recommended way on macOS 26 Tahoe to take a CoreAudio AudioObjectID and use it to lookup the underlying USB LocationID? I previously used AudioObjectID to query the corresponding DeviceUID with kAudioDevicePropertyDeviceUID. Then I queried for the IOService matching kIOAudioEngineClassName with property kIOAudioEngineGlobalUniqueIDKey matching DeviceUID, and I loaded kUSBDevicePropertyLocationID from the result. This fails on macOS 26, because the IO Registry for the device has an entry for usbaudiod rather than AppleUSBAudioEngine, and usbaudiod does not include a kIOAudioEngineGlobalUniqueIDKey property (or any other property to map it to a CoreAudio DeviceUID). My use-case here is a piece of audio recording software that allows configuring a set of supported audio devices via USB HID prior to recording. I present the user with a list of CoreAudio devices to use, but without a way to lookup the underlying USB LocationID, I cannot guarantee that the configured device matches the selected device (e.g. if the user plugged in two identical microphones).

I have a PCM audio buffer (AVAudioPCMFormatInt16). When I try to play it using AVPlayerNode / AVAudioEngine an exception is thrown: "[[busArray objectAtIndexedSubscript:(NSUInteger)element] setFormat:format error:&nsErr]: returned false, error Error Domain=NSOSStatusErrorDomain Code=-10868 (related thread https://forums.developer.apple.com/forums/thread/700497?answerId=780530022#780530022) If I convert the buffer to AVAudioPCMFormatFloat32 playback works. My questions are: Does AVAudioEngine / AVPlayerNode require AVAudioPCMBuffer to be in the Float32 format? Is there a way I can configure it to accept another format instead for my application? If 1 is YES is this documented anywhere? If 1 is YES is this required format subject to change at any point? Thanks! I was looking to watch the "AVAudioEngine in Practice" session video from WWDC 2014 but I can't find it anywhere (https://forums.developer.apple.com/forums/thread/747008).

iPhoneやiPadにおいて、画面上のボタンなどをタップした際に、特定の楽器音を発音させる方法をご存知の方いらっしゃいませんか？ 現在音楽学習アプリを作成途中で、画面上の鍵盤や指板のボタン状のframeに、単音又は和音を割当て発音させる事を考えております SwiftUIのcodeのみで実現できないでしょうか 嘗て、MIDIのlevel1の楽器の発音機能があった様に記憶していますが、現在のOS上では同様の機能を実装してないように思えます 皆様のお知恵をお貸しください

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'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.

I have a simple AVAudioEngine graph as follows: AVAudioPlayerNode -> AVAudioUnitEQ -> AVAudioUnitTimePitch -> AVAudioUnitReverb -> Main mixer node of AVAudioEngine. I noticed that whenever I have AVAudioUnitTimePitch or AVAudioUnitVarispeed in the graph, I noticed a very distinct crackling/popping sound in my Airpods Pro 2 when starting up the engine and playing the AVAudioPlayerNode and unable to find the reason why this is happening. When I remove the node, the crackling completely goes away. How do I fix this problem since i need the user to be able to control the pitch and rate of the audio during playback. import AVKit @Observable @MainActor class AudioEngineManager { nonisolated private let engine = AVAudioEngine() private let playerNode = AVAudioPlayerNode() private let reverb = AVAudioUnitReverb() private let pitch = AVAudioUnitTimePitch() private let eq = AVAudioUnitEQ(numberOfBands: 10) private var audioFile: AVAudioFile? private var fadePlayPauseTask: Task<Void, Error>? private var playPauseCurrentFadeTime: Double = 0 init() { setupAudioEngine() } private func setupAudioEngine() { guard let url = Bundle.main.url(forResource: "Song name goes here", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: url) } catch { print("Failed to load audio file: \(error)") return } reverb.loadFactoryPreset(.mediumHall) reverb.wetDryMix = 50 pitch.pitch = 0 // Increase pitch by 500 cents (5 semitones) engine.attach(playerNode) engine.attach(pitch) engine.attach(reverb) engine.attach(eq) // Connect: player -> pitch -> reverb -> output engine.connect(playerNode, to: eq, format: audioFile?.processingFormat) engine.connect(eq, to: pitch, format: audioFile?.processingFormat) engine.connect(pitch, to: reverb, format: audioFile?.processingFormat) engine.connect(reverb, to: engine.mainMixerNode, format: audioFile?.processingFormat) } func prepare() { guard let audioFile else { return } playerNode.scheduleFile(audioFile, at: nil) } func play() { DispatchQueue.global().async { [weak self] in guard let self else { return } engine.prepare() try? engine.start() DispatchQueue.main.async { [weak self] in guard let self else { return } playerNode.play() fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: true) // Ramp up volume until 1 is reached if volume >= 1 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 1 } } } } func pause() { fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: false) // Ramp down volume until 0 is reached if volume <= 0 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 0 playerNode.pause() // Shut down engine once ramp down completes DispatchQueue.global().async { [weak self] in guard let self else { return } engine.pause() } } } private func updateVolume(for x: Double, rising: Bool) -> Float { if rising { // Fade in return Float(pow(x, 2) * (3.0 - 2.0 * (x))) } else { // Fade out return Float(1 - (pow(x, 2) * (3.0 - 2.0 * (x)))) } } func setPitch(_ value: Float) { pitch.pitch = value } func setReverbMix(_ value: Float) { reverb.wetDryMix = value } } struct ContentView: View { @State private var audioManager = AudioEngineManager() @State private var pitch: Float = 0 @State private var reverb: Float = 0 var body: some View { VStack(spacing: 20) { Text("🎵 Audio Player with Reverb & Pitch") .font(.title2) HStack { Button("Prepare") { audioManager.prepare() } Button("Play") { audioManager.play() } .padding() .background(Color.green) .foregroundColor(.white) .cornerRadius(10) Button("Pause") { audioManager.pause() } .padding() .background(Color.red) .foregroundColor(.white) .cornerRadius(10) } VStack { Text("Pitch: \(Int(pitch)) cents") Slider(value: $pitch, in: -2400...2400, step: 100) { _ in audioManager.setPitch(pitch) } } VStack { Text("Reverb Mix: \(Int(reverb))%") Slider(value: $reverb, in: 0...100, step: 1) { _ in audioManager.setReverbMix(reverb) } } } .padding() } }

I’m building a standalone Apple Watch metronome app for running. My goal is for these 3 apps to work at the same time: Runna owns the workout session Spotify plays music my app plays a metronome click in the background So far this is what I've found: Using HKWorkout​Session in my metronome app works well with Spotify, but conflicts with Runna and other workout apps, so I removed that. Using watchOS background audio with longFormAudio allows my app run in the background, and it can coexist with Runna. However, it seems to conflict with Spotify playback, and one app tends to stop the other. Is there any supported watchOS audio/background configuration that allows all 3 at once? More specifically this is what I need: another app owns HKWorkout​Session Spotify keeps playing my app keeps generating metronome clicks in the background Or is this simply not supported by current watchOS session/background rules? My metronome uses AVAudio​Engine / AVAudio​Player​Node with generated click audio. Thank you!

The AVB AVnu MILAN Convention has a groweing Population. Many big companies (Cisco, Meyer Sound, d&b Audio, l‘acoustics, Presonus, digico etc.) implements the AVB AVnu Milan Standards. Is there a plan on the Apple side to also implement AVnu Milan on top of the AVB Protocol? The advantage for Apple Sound would be a great Integration in the professionell Audio market and a more stable intergration on top of the AVB protocol. The atdecc work, but Not that stable.

I have a bunch of Audio Unit v3 plugins that are approaching release, and I was considering using subscription-model pricing, as I have done in a soon to be released iOS app. However, whether this is possible or not is not at all obvious. Specifically: The plugin can, depending on the host app, be loaded in-process or out-of-process - yes, I know, Logic Pro and Garage Band will not load a plug-in in-process anymore, but I am not going to rule that out for other audio apps and force on them the overhead of IPC (I spent two solid weeks deciphering the process to actually make it possible for an AUv3 to run in-process - see this - https://github.com/timboudreau/audio_unit_rust_demo - example with notes) Depending on how it is loaded, the value of Bundle.main.bundleIdentifier will vary. If I use the StoreKit API, will that return product results for my bundle identifier when being called as a library from a foreign application? I would expect it would be a major security hole if random apps could query about purchases of other random apps, so I assume not. Even if I restricted the plugins to running out-of-process, I have to set up the in-app purchases on the app store for the App container's ID, not the extension's ID, and the extension is what run - the outer app that is what you purchase is just a toy demo that exists solely to register the audio unit. I have similar questions with regard to MetricKit, which I would similarly like to use, but which may be running inside some random app. If there were some sort of signed token, or similar mechanism, that could be bundled or acquired by the running plugin extension that could be used to ensure both StoreKit and MetricKit operate under the assumption that purchases and metrics should be accessed as if called from the container app, that would be very helpful. This is the difference between having a one-and-done sales model and something that provides ongoing revenue to maintain these products - I am a one-person shop - if I price these products where they would need to be to pay the bills assuming a single sale per customer ever, the price will be too high for anyone to want to try products from a small vendor they've never heard of. So, being able to do a free trial period and then subscription is the difference between this being a viable business or not.

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