Dive into the technical aspects of audio on your device, including codecs, format support, and customization options.

Audio Documentation

Posts under Audio subtopic

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Entitlement "com.apple.developer.carplay-driving-task" not allowing audio playback for voice controlled interaction
According to https://developer.apple.com/download/files/CarPlay-Developer-Guide.pdf , apps with entitlement com.apple.developer.carplay-driving-task are allowed to use voice control. In my current implementation the voice recording working fine but the voice response (AVPlayer with category "playback set") does not output any audio. I suspect that it is a entitlement limitation because if I quickly tap to play a music while the voice assistant AVPlayer is "playing", then I can hear the response, but without this trick it stays playing but mute. In parallel I have now requested com.apple.developer.carplay-voice-based-conversation entitlement , but I don't even know if when approved I will be able to use 2 entitlement for the same CarPlay app. Long story short: 1 - Should an app be able to play audio responses when it's CarPlay entitlement is com.apple.developer.carplay-driving-task? 2 - If not, can I combine entitlements com.apple.developer.carplay-driving-task and com.apple.developer.carplay-voice-based-conversation?
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Issues with monitoring and changing WebRTC audio output device in WKWebView
I am developing a VoIP app that uses WebRTC inside a WKWebView. Question 1: How can I monitor which audio output device WebRTC is currently using? I want to display this information in the UI for the user . Question 2: How can I change the current audio output device for WebRTC? I am using a JS Bridge to Objective-C code, attempting to change the audio device with the following code: void set_speaker(int n) { session = [AVAudioSession sharedInstance]; NSError *err = nil; if (n == 1) { [session overrideOutputAudioPort:AVAudioSessionPortOverrideSpeaker error:&err]; } else { [session overrideOutputAudioPort:AVAudioSessionPortOverrideNone error:&err]; } } However, this approach does not work. I am testing on an iPhone with iOS 16.7. Is a higher iOS version required?
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MusicKit developer token returns 401 on all catalog endpoints
My MusicKit developer token returns 401 (empty body) on every Apple Music API catalog endpoint. I've tried two different keys — both fail identically. Setup: Team ID: K79RSBVM9G Key ID: URNQV5UDGB (MusicKit enabled, associated with Media ID media.audio.explore.musickit) Apple Developer Program License Agreement accepted April 14, 2026 Token format (matches docs exactly): Header: {"alg":"ES256","kid":"URNQV5UDGB"} Payload: {"iss":"K79RSBVM9G","iat":,"exp":<now+15777000>} What works: /v1/storefronts/us returns 200 What fails: Every catalog endpoint returns 401 with empty body: /v1/catalog/us/search?types=artists&term=test /v1/catalog/us/artists/5920832 /v1/catalog/us/genres /v1/test The token self-verifies (signature is valid). I've tried with and without typ:"JWT", with the origin claim, and with a manually signed JWT bypassing the jsonwebtoken library. Same 401 every time. What am I missing?
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Is Push to Talk appropriate for a voice-based interactive assistant (not a walkie-talkie app)?
Hello, Looking for guidance from Apple engineers or developers who have used Push to Talk in production I am developing an iOS application called Companion AI / Theo Voice, designed for elderly users. The goal of the app is to provide a simple, voice-first interactive assistant that enables: natural voice interaction (no typing required) daily assistance (reminders, well-being, conversation) bidirectional voice communication (the user can immediately respond by voice) ⸻ How it works The app operates in two main modes: Conversation mode the user opens the app the assistant speaks the user replies naturally by voice Proactive mode in specific useful situations (e.g. medication reminders, check-ins) the app initiates a voice interaction the user can respond immediately ⸻ Important constraints there is no continuous listening the microphone is only active during interactions users can disable proactive interactions frequency is limited and user-controlled ⸻ Question We are considering using the Push to Talk framework in order to: allow the app to be awakened in the background initiate a voice interaction enable immediate voice response from the user Would this usage be considered aligned with the intended use of Push to Talk? Are there any specific recommendations to ensure compliance with App Store Review Guidelines? Thank you very much for your guidance.
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Bug: Channels erroneously populated when sending audio from an iPhone to a linux gadget audio device.
I have a device which is using linux gadget audio to receive audio input via USB, exposing 24 capture channels. This device works well with Mac, Windows, and Android phones. However, when sending audio from an iPhone (both USB-C iPhones and lightning iPhones using an official Apple lightning -> usb adaptor) I am seeing strange behaviour. Audio which is sent from the iPhone to any one of inputs 12, 19, 20, 21, or 22 appears in all of those channels, rather than only the channel to which audio is routed. I have confirmed on my linux device that these channels are not being erroneously populated by the software running on that device; the issue is visible in audio recorded directly from the gadget using arecord, meaning it is present in the audio being sent from the iPhone. I have confirmed that the gadget channel mask is correct for 24 channel audio (0xFFFFFF). As said above, audio routed to this device from any non-iPhone device (Mac, Windows, Android) works fine. The only sensible conclusion seems to be that the iPhone is populating the additional channels erroneously due to some bug in CoreAudio's handling of gadget audio devices. I would appreciate any insight on this from Apple developers, or from anyone else who has come across this issue and found a workaround.
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How to use the SpeechDetector Module
I am trying to use SpeechDetector Module in Speech framework along with SpeechTranscriber. and it is giving me an error Cannot convert value of type 'SpeechDetector' to expected element type 'Array.ArrayLiteralElement' (aka 'any SpeechModule') Below is how I am using it let speechDetector = Speech.SpeechDetector() let transcriber = SpeechTranscriber(locale: Locale.current, transcriptionOptions: [], reportingOptions: [.volatileResults], attributeOptions: [.audioTimeRange]) speechAnalyzer = try SpeechAnalyzer(modules: [transcriber,speechDetector])
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SpeechAnalyzer speech to text wwdc sample app
I am using the sample app from: https://developer.apple.com/videos/play/wwdc2025/277/?time=763 I installed this on an Iphone 15 Pro with iOS 26 beta 1. I was able to get good transcription with it. The app did crash sometimes when transcribing and I was going to post here with the details. I then installed iOS beta 2 and uninstalled the sample app. Now every time I try to run the sample app on the 15 Pro I get this message: SpeechAnalyzer: Input loop ending with error: Error Domain=SFSpeechErrorDomain Code=10 "Cannot use modules with unallocated locales [en_US (fixed en_US)]" UserInfo={NSLocalizedDescription=Cannot use modules with unallocated locales [en_US (fixed en_US)]} I can't continue our our work towards using SpeechAnalyzer now with this error. I have set breakpoints on all the catch handlers and it doesn't catch this error. My phone region is "United States"
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tvOS: Background audio + local caching works on Simulator but stops on real Apple TV device
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!
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tvOS: Background audio + local caching works on Simulator but stops on real Apple TV device
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!
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AVAudioSession : Audio issues when recording the screen in an app that changes IOBufferDuration on iOS 26.
Among Japanese end users, audio issues during screen recording—primarily in game applications—have become a topic of discussion. We have confirmed that the trigger for this issue is highly likely to be related to changes to IOBufferDuration. When using setPreferredIOBufferDuration and the IOBufferDuration is set to a value smaller than the default, audio problems occur in the recorded screen capture video. Audio playback is performed using AudioUnit (RemoteIO). https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc This issue was not observed on iOS 18, and it appears to have started occurring after upgrading to iOS 26. We provide an audio middleware solution, and we had incorporated changes to IOBufferDuration into our product to achieve low-latency audio playback. As a result, developers using our product as well as their end users are being affected by this issue. We kindly request that this issue be investigated and addressed in a future update. “This document has been translated by AI. The original text is included below for reference.” 日本のエンドユーザー間で主にゲームアプリケーションにおける画面収録時の音声の問題が話題になっています。 こちらの症状のトリガーが、IOBufferDurationの変更によるものである可能性が高いことを確認しました。 setPreferredIOBufferDurationを使用し、IOBufferDurationがデフォルトより小さい状態の時、画面収録された動画の音声に問題が発生することをしています。 音声の再生にはAudioUnit(RemoteIO)を使用しています。 https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc iOS 18ではこのような問題は確認されておらず、iOS26になってから問題が発生しているようです。 私たちはオーディオミドルウェアを提供しており、低遅延の再生のためにIOBufferDurationの変更を製品に組み込んでいました。 そのため、弊社製品をご利用いただいている開発者およびエンドユーザーの皆様がこの不具合の影響を受けています。 こちらの不具合の調査及び修正対応を検討いただけますでしょうか。
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SpeechAnalyzer error "asset not found after attempted download" for certain languages
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() } } ...
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SpeechAnalyzer > AnalysisContext lack of documentation
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).
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CarPlay outputs no audio
I have an application that includes custom artwork for the album cover and text details setup with the MPRemoteCommandCenter.shared() reference. I need the user to have a full featured "now playing" display to see all of this. My experience is that cannot find a set of parameters for AVAudioSession.setCategory() that route audio successfully, and display the full featured now playing deck. If I use .playAndRecord, the audio I send out plays out on the radio. But, the now-playing deck is empty and nothing I do with the command center seems to change that. If I instead use .playback, I cannot use .defaultToSpeaker option which is the only way I've found to cause the "now-playing" navigation button to appear so that the full featured deck will display. But, of course setCategory() fails with an error about .defaultToSpeaker only available with .playAndRecord, so some default or intermediate state is entered and I see the full featured deck, but no audio goes out to the radio. What combination is supposed to be used here and is this more likely a problem with thread use (@MainActor) and/or some ordering of operations that I've overlooked?
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AVAudioFile.read extremely slow after seeking in FLAC and MP3 files
I'm developing an audio player app that uses AVAudio​File to read PCM data from various formats. I'm experiencing severe performance issues when seeking in FLAC, while other compressed formats (M4A/AAC) work correctly. I don't intend to use them in my app, but I also tested mp3 files just by curiosity and they also have this issue. Environment: macOS 26 (Tahoe) Xcode 26.3 Apple Silicon (M1) The issue: After setting AVAudio​File​.frame​Position to a position mid-file, the subsequent call to AVAudio​File​.read(into​:frame​Count:) blocks for an unreasonable amount of time for FLAC and MP3 files. The delay scales linearly with the seek target, seeking near the beginning is fast, seeking toward the end is proportionally slower, which suggests the decoder is decoding linearly from the beginning of the file rather than using any seek index. (My app deals with “images” of Audio CDs ripped as a single long audio file.) The issue is particularly severe when reading files from an SMB network share (server on Ethernet, client on Wi-Fi with the access point ~2 meters away in line of sight). Quick Benchmark results: I tested with the same 75-minute audio content (16-bit/44.1 kHz stereo, 200,502,708 frames) encoded in five formats, seeking to the midpoint. Over SMB (Local Network, Server on Ethernet, Client on WiFi): Format | Seek + Read Time ----------|------------------ WAV | 0.007 s AIFF | 0.009 s Apple | 0.015 s Lossless | MP3 | 9.2 s FLAC | 30.2 s Locally (MacBook Air M1 SSD) : Format | Seek + Read Time ----------|------------------ WAV | 0.0005 s AIFF | 0.0004 s Apple | 0.0011 s Lossless | MP3 | 0.1958 s FLAC | 0.7528 s WAV, AIFF, and M4A all seek virtually instantly (< 15 ms). MP3 and FLAC exhibit linear-time behavior, with FLAC being the worst affected. Note that M4A (AAC) is also a compressed format that requires decoding after seeking, yet it completes in 15 ms. This rules out any inherent limitation of compressed formats, the MP4 container's packet index (stts/stco) is clearly being used for fast random access. Both MP3 (Xing/LAME TOC) and FLAC (SEEKTABLE metadata block) have their own seek mechanisms that should provide similar performance. Minimal CLI tool to reproduce: import Foundation guard CommandLine.arguments.count > 1 else { print("Usage: FLACSpeed <audio-file-path>") exit(1) } let path = CommandLine.arguments[1] let fileURL = URL(fileURLWithPath: path) do { let file = try AVAudioFile(forReading: fileURL) let format = file.processingFormat let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: 8192)! let totalFrames = file.length let seekTarget = totalFrames / 2 print("File: \(fileURL.lastPathComponent)") print("Format: \(format)") print("Total frames: \(totalFrames)") print("Seeking to frame: \(seekTarget)") file.framePosition = seekTarget let start = CFAbsoluteTimeGetCurrent() try file.read(into: buffer, frameCount: 8192) let elapsed = CFAbsoluteTimeGetCurrent() - start print("Read after seek took \(elapsed) seconds") } catch { print("Error: \(error.localizedDescription)") exit(1) } Expected behavior: AVAudio​File​.read(into​:frame​Count:) after setting frame​Position should use the available seek mechanisms in FLAC and MP3 files for fast random access, as it already does for M4A (AAC). Even accounting for the fact that seek tables provide approximate (not sample-precise) positioning, the "jump to nearest index point + decode forward" approach should complete in milliseconds, not seconds. Workaround: For FLAC, I've worked around this by using libFLAC directly, which provides instant seeking via FLAC__stream​_decoder​_seek​_absolute(). libFLAC Performance: For comparison, libFLAC's FLAC__stream​_decoder​_seek​_absolute() performs the same seek + read on the same FLAC file in around 0.015, using the FLAC seek table to jump to the nearest preceding seek point, then decoding forward a small number of frames to the exact target sample.
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Does the OS has dedicated volume levels for each AVAudioSessionCategory.
We have an VoiP application, our application can be configured to amplify the PCM samples before feeding it to the Player to achieve volume gain at the receiver. In order to support this, We follow as below. If User has configured this Gain Settings within application, Application applies the amplification for the samples to introduce the gain. Application will also set the AVAudioSessionCategory to AVAudioSessionCategoryPlayback Provided the User has chosen the output to Speaker. This settings was working for us but we see there is a difference in behaviour w.r.t Volume Level System Settings between OS 26.3.1 and OS 26.4 When user has chosen earpiece as Output, then we will set the AVAudioSessionCategory to AVAudioSessionCategoryPlayAndRecord. User would have set the volume level to minimum. When user will change the output to Speaker, then we will set the AVAudioSessionCategory to AVAudioSessionCategoryPlayback. The expectation is, the volume level should be of AVAudioSessionCategoryPlayback what was set earlier instead we are seeing the volume level stays as minimum which was set to AVAudioSessionCategoryPlayAndRecord Could you please explain about this inconsistency w.r.t Volume level.
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CarPlay: Voice Conversational Entitlement Details
With the Voice Conversational Entitlement, can a CarPlay app establish a turn-based audio interface that operates in two modes: Speaking mode: Audio Session configured for playback Buffered audio Listening mode: Switch Audio Session to .record or .playAndRecord Activate SFSpeechRecognizer And continue toggling back and forth. The app should listen for responses to questions or other audio cues, and assuming those answers are correct (based on analysis of results from SFSpeechRecognizer), continue this pattern of mode 1 and 2 alternating. This appears to be a valid use of this entitlement. Does this also require the Audio App Entitlement, or is the Voice Conversational Entitlement sufficient? Are there other obstacles to this type of app that I'm not seeing? Or perhaps this is technically possible, but unlikely to pass app store review?
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Android MusicKit canSetRadioLikeState and setRadioLikeState
The Android MusicKit documentation documents two functions that are not actually exposed/added to the SDK. https://developer.apple.com/musickit/android/com/apple/android/music/playback/controller/MediaPlayerController.html#canSetRadioLikeState-- https://developer.apple.com/musickit/android/com/apple/android/music/playback/controller/MediaPlayerController.html#setRadioLikeState-int- Is the documentation stale or is the SDK out of date?
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Is iTunesTagging no longer support?
I'm currently trying to develope ipod control function on IVI for vehicle. From previous experience I remember we need to implement iTunetagging, but since I can't find it in Accessory Firmware Specification R46, I'm wondering whether iTunesTagging is no longer support. Thanks in advance for you support!
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Entitlement "com.apple.developer.carplay-driving-task" not allowing audio playback for voice controlled interaction
According to https://developer.apple.com/download/files/CarPlay-Developer-Guide.pdf , apps with entitlement com.apple.developer.carplay-driving-task are allowed to use voice control. In my current implementation the voice recording working fine but the voice response (AVPlayer with category "playback set") does not output any audio. I suspect that it is a entitlement limitation because if I quickly tap to play a music while the voice assistant AVPlayer is "playing", then I can hear the response, but without this trick it stays playing but mute. In parallel I have now requested com.apple.developer.carplay-voice-based-conversation entitlement , but I don't even know if when approved I will be able to use 2 entitlement for the same CarPlay app. Long story short: 1 - Should an app be able to play audio responses when it's CarPlay entitlement is com.apple.developer.carplay-driving-task? 2 - If not, can I combine entitlements com.apple.developer.carplay-driving-task and com.apple.developer.carplay-voice-based-conversation?
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Radio stations unable to play on Android with MusicKit SDK
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
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Issues with monitoring and changing WebRTC audio output device in WKWebView
I am developing a VoIP app that uses WebRTC inside a WKWebView. Question 1: How can I monitor which audio output device WebRTC is currently using? I want to display this information in the UI for the user . Question 2: How can I change the current audio output device for WebRTC? I am using a JS Bridge to Objective-C code, attempting to change the audio device with the following code: void set_speaker(int n) { session = [AVAudioSession sharedInstance]; NSError *err = nil; if (n == 1) { [session overrideOutputAudioPort:AVAudioSessionPortOverrideSpeaker error:&err]; } else { [session overrideOutputAudioPort:AVAudioSessionPortOverrideNone error:&err]; } } However, this approach does not work. I am testing on an iPhone with iOS 16.7. Is a higher iOS version required?
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MusicKit developer token returns 401 on all catalog endpoints
My MusicKit developer token returns 401 (empty body) on every Apple Music API catalog endpoint. I've tried two different keys — both fail identically. Setup: Team ID: K79RSBVM9G Key ID: URNQV5UDGB (MusicKit enabled, associated with Media ID media.audio.explore.musickit) Apple Developer Program License Agreement accepted April 14, 2026 Token format (matches docs exactly): Header: {"alg":"ES256","kid":"URNQV5UDGB"} Payload: {"iss":"K79RSBVM9G","iat":,"exp":<now+15777000>} What works: /v1/storefronts/us returns 200 What fails: Every catalog endpoint returns 401 with empty body: /v1/catalog/us/search?types=artists&term=test /v1/catalog/us/artists/5920832 /v1/catalog/us/genres /v1/test The token self-verifies (signature is valid). I've tried with and without typ:"JWT", with the origin claim, and with a manually signed JWT bypassing the jsonwebtoken library. Same 401 every time. What am I missing?
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Is Push to Talk appropriate for a voice-based interactive assistant (not a walkie-talkie app)?
Hello, Looking for guidance from Apple engineers or developers who have used Push to Talk in production I am developing an iOS application called Companion AI / Theo Voice, designed for elderly users. The goal of the app is to provide a simple, voice-first interactive assistant that enables: natural voice interaction (no typing required) daily assistance (reminders, well-being, conversation) bidirectional voice communication (the user can immediately respond by voice) ⸻ How it works The app operates in two main modes: Conversation mode the user opens the app the assistant speaks the user replies naturally by voice Proactive mode in specific useful situations (e.g. medication reminders, check-ins) the app initiates a voice interaction the user can respond immediately ⸻ Important constraints there is no continuous listening the microphone is only active during interactions users can disable proactive interactions frequency is limited and user-controlled ⸻ Question We are considering using the Push to Talk framework in order to: allow the app to be awakened in the background initiate a voice interaction enable immediate voice response from the user Would this usage be considered aligned with the intended use of Push to Talk? Are there any specific recommendations to ensure compliance with App Store Review Guidelines? Thank you very much for your guidance.
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Bug: Channels erroneously populated when sending audio from an iPhone to a linux gadget audio device.
I have a device which is using linux gadget audio to receive audio input via USB, exposing 24 capture channels. This device works well with Mac, Windows, and Android phones. However, when sending audio from an iPhone (both USB-C iPhones and lightning iPhones using an official Apple lightning -> usb adaptor) I am seeing strange behaviour. Audio which is sent from the iPhone to any one of inputs 12, 19, 20, 21, or 22 appears in all of those channels, rather than only the channel to which audio is routed. I have confirmed on my linux device that these channels are not being erroneously populated by the software running on that device; the issue is visible in audio recorded directly from the gadget using arecord, meaning it is present in the audio being sent from the iPhone. I have confirmed that the gadget channel mask is correct for 24 channel audio (0xFFFFFF). As said above, audio routed to this device from any non-iPhone device (Mac, Windows, Android) works fine. The only sensible conclusion seems to be that the iPhone is populating the additional channels erroneously due to some bug in CoreAudio's handling of gadget audio devices. I would appreciate any insight on this from Apple developers, or from anyone else who has come across this issue and found a workaround.
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How to use the SpeechDetector Module
I am trying to use SpeechDetector Module in Speech framework along with SpeechTranscriber. and it is giving me an error Cannot convert value of type 'SpeechDetector' to expected element type 'Array.ArrayLiteralElement' (aka 'any SpeechModule') Below is how I am using it let speechDetector = Speech.SpeechDetector() let transcriber = SpeechTranscriber(locale: Locale.current, transcriptionOptions: [], reportingOptions: [.volatileResults], attributeOptions: [.audioTimeRange]) speechAnalyzer = try SpeechAnalyzer(modules: [transcriber,speechDetector])
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SpeechAnalyzer speech to text wwdc sample app
I am using the sample app from: https://developer.apple.com/videos/play/wwdc2025/277/?time=763 I installed this on an Iphone 15 Pro with iOS 26 beta 1. I was able to get good transcription with it. The app did crash sometimes when transcribing and I was going to post here with the details. I then installed iOS beta 2 and uninstalled the sample app. Now every time I try to run the sample app on the 15 Pro I get this message: SpeechAnalyzer: Input loop ending with error: Error Domain=SFSpeechErrorDomain Code=10 "Cannot use modules with unallocated locales [en_US (fixed en_US)]" UserInfo={NSLocalizedDescription=Cannot use modules with unallocated locales [en_US (fixed en_US)]} I can't continue our our work towards using SpeechAnalyzer now with this error. I have set breakpoints on all the catch handlers and it doesn't catch this error. My phone region is "United States"
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tvOS: Background audio + local caching works on Simulator but stops on real Apple TV device
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!
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tvOS: Background audio + local caching works on Simulator but stops on real Apple TV device
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!
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AVAudioSession : Audio issues when recording the screen in an app that changes IOBufferDuration on iOS 26.
Among Japanese end users, audio issues during screen recording—primarily in game applications—have become a topic of discussion. We have confirmed that the trigger for this issue is highly likely to be related to changes to IOBufferDuration. When using setPreferredIOBufferDuration and the IOBufferDuration is set to a value smaller than the default, audio problems occur in the recorded screen capture video. Audio playback is performed using AudioUnit (RemoteIO). https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc This issue was not observed on iOS 18, and it appears to have started occurring after upgrading to iOS 26. We provide an audio middleware solution, and we had incorporated changes to IOBufferDuration into our product to achieve low-latency audio playback. As a result, developers using our product as well as their end users are being affected by this issue. We kindly request that this issue be investigated and addressed in a future update. “This document has been translated by AI. The original text is included below for reference.” 日本のエンドユーザー間で主にゲームアプリケーションにおける画面収録時の音声の問題が話題になっています。 こちらの症状のトリガーが、IOBufferDurationの変更によるものである可能性が高いことを確認しました。 setPreferredIOBufferDurationを使用し、IOBufferDurationがデフォルトより小さい状態の時、画面収録された動画の音声に問題が発生することをしています。 音声の再生にはAudioUnit(RemoteIO)を使用しています。 https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc iOS 18ではこのような問題は確認されておらず、iOS26になってから問題が発生しているようです。 私たちはオーディオミドルウェアを提供しており、低遅延の再生のためにIOBufferDurationの変更を製品に組み込んでいました。 そのため、弊社製品をご利用いただいている開発者およびエンドユーザーの皆様がこの不具合の影響を受けています。 こちらの不具合の調査及び修正対応を検討いただけますでしょうか。
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After upgrade to iOS 26.4, averagePowerLevel and peakHoldLevel are stuck -120
We have an application that capture audio and video. App captures audio PCM on internal or external microphone and displays audio level on the screen. App was working fine for many years but after iOS 26.4 upgrade, averagePowerLevel and peakHoldLevel are stuck to -120 values. Any suggestion?
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SpeechAnalyzer error "asset not found after attempted download" for certain languages
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() } } ...
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SpeechAnalyzer > AnalysisContext lack of documentation
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).
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CarPlay outputs no audio
I have an application that includes custom artwork for the album cover and text details setup with the MPRemoteCommandCenter.shared() reference. I need the user to have a full featured "now playing" display to see all of this. My experience is that cannot find a set of parameters for AVAudioSession.setCategory() that route audio successfully, and display the full featured now playing deck. If I use .playAndRecord, the audio I send out plays out on the radio. But, the now-playing deck is empty and nothing I do with the command center seems to change that. If I instead use .playback, I cannot use .defaultToSpeaker option which is the only way I've found to cause the "now-playing" navigation button to appear so that the full featured deck will display. But, of course setCategory() fails with an error about .defaultToSpeaker only available with .playAndRecord, so some default or intermediate state is entered and I see the full featured deck, but no audio goes out to the radio. What combination is supposed to be used here and is this more likely a problem with thread use (@MainActor) and/or some ordering of operations that I've overlooked?
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AVAudioFile.read extremely slow after seeking in FLAC and MP3 files
I'm developing an audio player app that uses AVAudio​File to read PCM data from various formats. I'm experiencing severe performance issues when seeking in FLAC, while other compressed formats (M4A/AAC) work correctly. I don't intend to use them in my app, but I also tested mp3 files just by curiosity and they also have this issue. Environment: macOS 26 (Tahoe) Xcode 26.3 Apple Silicon (M1) The issue: After setting AVAudio​File​.frame​Position to a position mid-file, the subsequent call to AVAudio​File​.read(into​:frame​Count:) blocks for an unreasonable amount of time for FLAC and MP3 files. The delay scales linearly with the seek target, seeking near the beginning is fast, seeking toward the end is proportionally slower, which suggests the decoder is decoding linearly from the beginning of the file rather than using any seek index. (My app deals with “images” of Audio CDs ripped as a single long audio file.) The issue is particularly severe when reading files from an SMB network share (server on Ethernet, client on Wi-Fi with the access point ~2 meters away in line of sight). Quick Benchmark results: I tested with the same 75-minute audio content (16-bit/44.1 kHz stereo, 200,502,708 frames) encoded in five formats, seeking to the midpoint. Over SMB (Local Network, Server on Ethernet, Client on WiFi): Format | Seek + Read Time ----------|------------------ WAV | 0.007 s AIFF | 0.009 s Apple | 0.015 s Lossless | MP3 | 9.2 s FLAC | 30.2 s Locally (MacBook Air M1 SSD) : Format | Seek + Read Time ----------|------------------ WAV | 0.0005 s AIFF | 0.0004 s Apple | 0.0011 s Lossless | MP3 | 0.1958 s FLAC | 0.7528 s WAV, AIFF, and M4A all seek virtually instantly (< 15 ms). MP3 and FLAC exhibit linear-time behavior, with FLAC being the worst affected. Note that M4A (AAC) is also a compressed format that requires decoding after seeking, yet it completes in 15 ms. This rules out any inherent limitation of compressed formats, the MP4 container's packet index (stts/stco) is clearly being used for fast random access. Both MP3 (Xing/LAME TOC) and FLAC (SEEKTABLE metadata block) have their own seek mechanisms that should provide similar performance. Minimal CLI tool to reproduce: import Foundation guard CommandLine.arguments.count > 1 else { print("Usage: FLACSpeed <audio-file-path>") exit(1) } let path = CommandLine.arguments[1] let fileURL = URL(fileURLWithPath: path) do { let file = try AVAudioFile(forReading: fileURL) let format = file.processingFormat let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: 8192)! let totalFrames = file.length let seekTarget = totalFrames / 2 print("File: \(fileURL.lastPathComponent)") print("Format: \(format)") print("Total frames: \(totalFrames)") print("Seeking to frame: \(seekTarget)") file.framePosition = seekTarget let start = CFAbsoluteTimeGetCurrent() try file.read(into: buffer, frameCount: 8192) let elapsed = CFAbsoluteTimeGetCurrent() - start print("Read after seek took \(elapsed) seconds") } catch { print("Error: \(error.localizedDescription)") exit(1) } Expected behavior: AVAudio​File​.read(into​:frame​Count:) after setting frame​Position should use the available seek mechanisms in FLAC and MP3 files for fast random access, as it already does for M4A (AAC). Even accounting for the fact that seek tables provide approximate (not sample-precise) positioning, the "jump to nearest index point + decode forward" approach should complete in milliseconds, not seconds. Workaround: For FLAC, I've worked around this by using libFLAC directly, which provides instant seeking via FLAC__stream​_decoder​_seek​_absolute(). libFLAC Performance: For comparison, libFLAC's FLAC__stream​_decoder​_seek​_absolute() performs the same seek + read on the same FLAC file in around 0.015, using the FLAC seek table to jump to the nearest preceding seek point, then decoding forward a small number of frames to the exact target sample.
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Does the OS has dedicated volume levels for each AVAudioSessionCategory.
We have an VoiP application, our application can be configured to amplify the PCM samples before feeding it to the Player to achieve volume gain at the receiver. In order to support this, We follow as below. If User has configured this Gain Settings within application, Application applies the amplification for the samples to introduce the gain. Application will also set the AVAudioSessionCategory to AVAudioSessionCategoryPlayback Provided the User has chosen the output to Speaker. This settings was working for us but we see there is a difference in behaviour w.r.t Volume Level System Settings between OS 26.3.1 and OS 26.4 When user has chosen earpiece as Output, then we will set the AVAudioSessionCategory to AVAudioSessionCategoryPlayAndRecord. User would have set the volume level to minimum. When user will change the output to Speaker, then we will set the AVAudioSessionCategory to AVAudioSessionCategoryPlayback. The expectation is, the volume level should be of AVAudioSessionCategoryPlayback what was set earlier instead we are seeing the volume level stays as minimum which was set to AVAudioSessionCategoryPlayAndRecord Could you please explain about this inconsistency w.r.t Volume level.
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CarPlay: Voice Conversational Entitlement Details
With the Voice Conversational Entitlement, can a CarPlay app establish a turn-based audio interface that operates in two modes: Speaking mode: Audio Session configured for playback Buffered audio Listening mode: Switch Audio Session to .record or .playAndRecord Activate SFSpeechRecognizer And continue toggling back and forth. The app should listen for responses to questions or other audio cues, and assuming those answers are correct (based on analysis of results from SFSpeechRecognizer), continue this pattern of mode 1 and 2 alternating. This appears to be a valid use of this entitlement. Does this also require the Audio App Entitlement, or is the Voice Conversational Entitlement sufficient? Are there other obstacles to this type of app that I'm not seeing? Or perhaps this is technically possible, but unlikely to pass app store review?
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Android MusicKit canSetRadioLikeState and setRadioLikeState
The Android MusicKit documentation documents two functions that are not actually exposed/added to the SDK. https://developer.apple.com/musickit/android/com/apple/android/music/playback/controller/MediaPlayerController.html#canSetRadioLikeState-- https://developer.apple.com/musickit/android/com/apple/android/music/playback/controller/MediaPlayerController.html#setRadioLikeState-int- Is the documentation stale or is the SDK out of date?
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Is iTunesTagging no longer support?
I'm currently trying to develope ipod control function on IVI for vehicle. From previous experience I remember we need to implement iTunetagging, but since I can't find it in Accessory Firmware Specification R46, I'm wondering whether iTunesTagging is no longer support. Thanks in advance for you support!
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