I’m working with the Push-to-Talk (PTT) framework and observing a consistent delay when starting audio capture. Scenario: A PTT call is already active The AVAudioSession is fully configured I request beginTransmission on the PTT channel I start my Audio Unit for recording (AudioOutputUnitStart) Observed behavior: AudioOutputUnitStart takes ~500 ms This happens whether I start the Audio Unit: after didBeginTransmission, or after AVAudioSession didActivate Comparison: Using the same Audio Unit, same format, and same configuration Without the PTT framework, AudioOutputUnitStart takes ~200 ms Additional notes: I am not modifying or reconfiguring AVAudioSession when requesting beginTransmission The audio session is already set up when the PTT call starts There are no interruptions or route changes at the time of starting the Audio Unit Impact: This extra latency is significant for Push-to-Talk use cases where fast transmit start is critical.

My app - natively iOS but built with the "Designed for iPad" option to run on Mac - does not recognise an attached USB microphone when running on a Mac. This line int32_t items = (int32_t) [[[AVAudioSession sharedInstance] availableInputs] count ]; returns 1, which is the Mac internal mic. On iPad and iPhone it sees both the internal mic and the USB mic. Is this an inherent "Designed for iPad" restriction, and is there some trick I can pull to get the USB microphone to be recognised by the system?

Hello, I'm working on a MusicKit based SwiftUI app. I've integrated AirPlay using the AVRoutePickerView like so: struct UIKitAirPlayPickerView: UIViewRepresentable { func makeUIView(context: Context) -> AVRoutePickerView { let routePickerView = AVRoutePickerView() routePickerView.prioritizesVideoDevices = false return routePickerView } func updateUIView(_ uiView: AVRoutePickerView, context: Context) {} } The AirPlay menu appears as expected, and selecting an AirPlay device functions as expected. I'm currently sending audio from my app to a HomePod. However, the state of the AVRoutePickerView does not reflect the playback state. There is no cover art and it says "Not Playing". When my device is locked, my lock screen shows the album art, metadata and AirPlay routing as expected. My app uses the ApplicationMusicPlayer however I encounter the same behavior using the SystemMusicPlayer. Any guidance on how to troubleshoot this? Is there any other way to integrate the system AirPlay picker into my app, or is this my only option? Thank you for reading.

Is it possible to find IDR frame (CMSampleBuffer) in AVAsset h264 video file?

I am trying to stream audio from local filesystem. For that, I am trying to use an AVAssetResourceLoaderDelegate for an AVURLAsset. However, Content-Length is not known at the start. To overcome this, I tried several methods: Set content length as nil, in the AVAssetResourceLoadingContentInformationRequest Set content length to -1, in the ContentInformationRequest Both of these cause the AVPlayerItem to fail with an error. I also tried setting Content-Length as INT_MAX, and setting a renewalDate = Date(timeIntervalSinceNow: 5). However, that seems to be buggy. Even after updating the Content-Length to the correct value (e.g. X bytes) and finishing that loading request, the resource loader keeps getting requests with requestedOffset = X with dataRequest.requestsAllDataToEndOfResource = true. These requests keep coming indefinitely, and as a result it seems that the next item in the queue does not get played. Also, .AVPlayerItemDidPlayToEndTime notification does not get called. I wanted to check if this is an expected behavior or is there a bug in this implementation. Also, what is the recommended way to stream audio of unknown initial length from local file system? Thanks!

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback, using ApplicationMusicPlayer. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

Hello, I am building an iOS-only, commercial app that uses AVSpeechSynthesizer with system voices, strictly using the APIs provided by Apple. Before distributing the app, I want to ensure that my current implementation does not conflict with the iOS Software License Agreement (SLA) and is aligned with Apple’s intended usage. For a better playback experience (more accurate estimation of utterance duration and smoother skip forward/backward during playback), I currently synthesize speech using: AVSpeechSynthesizer.write(_:toBufferCallback:) Converting the received AVAudioPCMBuffer buffers into audio data Storing the audio inside the app sandbox Playing it back using AVAudioPlayer / AVAudioEngine The cached audio is: Generated fully on-device using system voices Stored only inside the app’s private container Used only for internal playback controls (timeline, seek, skip ±5 seconds) Never shared, exported, uploaded, or exposed outside the app The alternative approaches would be: Keeping the generated audio entirely in memory (RAM) for playback purposes, without writing it to the file system at any point Or using AVSpeechSynthesizer.speak(_:) and playing speech strictly in real time which has a poorer user experience compared to my approach I have reviewed the current iOS Software License Agreement: https://www.apple.com/legal/sla/docs/iOS18_iPadOS18.pdf In particular, section (f) mentions restrictions around System Characters, Live Captions, and Personal Voice, including the following excerpt: “…use … only for your personal, non-commercial use… No other creation or use of the System Characters, Live Captions, or Personal Voice is permitted by this License, including but not limited to the use, reproduction, display, performance, recording, publishing or redistribution in a … commercial context.” I do not see a specific reference in the SLA to system text-to-speech voices used via AVSpeechSynthesizer, and I want to be certain that temporarily caching synthesized speech for internal, non-exported playback is acceptable in a commercial app. My question is: Is caching AVSpeechSynthesizer system-voice output inside the app sandbox for internal playback acceptable, or is Apple’s recommended approach to rely only on real-time playback (speak(_:)) or strictly in-memory buffering without file storage? If this question falls outside DTS technical scope and is instead a policy or licensing matter, I would appreciate guidance on the authoritative Apple documentation or the correct Apple team/contact. Thank you.

Thread 5 Crashed: 0 libobjc.A.dylib 0x19af7b038 objc_msgSend + 56 1 CoreFoundation 0x19dfdb618 cow_cleanup + 135 2 CoreFoundation 0x19dfdb6fc -[__NSDictionaryM dealloc] + 147 3 MediaToolbox 0x1b167636c FigRemotePropertyCacheTeardown + 31 4 MediaToolbox 0x1b1c5b648 remoteXPCAsset_Finalize + 107 5 CoreMedia 0x1b1e9166c FigBaseObjectFinalize + 275 6 CoreFoundation 0x19dfcc5ec _CFRelease + 295 7 AVFCore 0x1b1054d64 -[AVFigAssetTrackInspector dealloc] + 151 8 AVFCore 0x1b0f818d8 -[AVAssetTrack dealloc] + 63 9 CoreFoundation 0x19dfdba28 RELEASE_OBJECTS_IN_THE_ARRAY + 115 10 CoreFoundation 0x19dfdb7e0 -[__NSArrayM dealloc] + 147 11 AVFCore 0x1b0f52e04 -[AVURLAsset dealloc] + 167 12 libobjc.A.dylib 0x19af821f8 object_cxxDestructFromClass(objc_object*, objc_class*) + 115 13 libobjc.A.dylib 0x19af7df20 objc_destructInstance_nonnull_realized(objc_object*) + 75 14 libobjc.A.dylib 0x19af7d4a4 _objc_rootDealloc + 71 15 AVFCore 0x1b0fef988 -[AVAssetReaderOutput dealloc] + 415 16 AVFCore 0x1b0ff11ec -[AVAssetReaderTrackOutput dealloc] + 127 17 CoreFoundation 0x19dfe20a4 -[__NSSingleObjectArrayI dealloc] + 63 18 libobjc.A.dylib 0x19af7d3f8 AutoreleasePoolPage::releaseUntil(objc_object**) + 203

3 I am working on an application to get when input audio device is being used. Basically I want to know the application using the microphone (built-in or external) This app runs on macOS. For Mac versions starting from Sonoma I can use this code: int getAudioProcessPID(AudioObjectID process) { pid_t pid; if (@available(macOS 14.0, *)) { constexpr AudioObjectPropertyAddress prop { kAudioProcessPropertyPID, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMain }; UInt32 dataSize = sizeof(pid); OSStatus error = AudioObjectGetPropertyData(process, &prop, 0, nullptr, &dataSize, &pid); if (error != noErr) { return -1; } } else { // Pre sonoma code goes here } return pid; } which works. However, kAudioProcessPropertyPID was added in macOS SDK 14.0. Does anyone know how to achieve the same functionality on previous versions?

Hi everyone, I’m testing audio recording on an iPhone 15 Plus using AVFoundation. Here’s a simplified version of my setup: let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatLinearPCM), AVSampleRateKey: 8000, AVNumberOfChannelsKey: 1, AVLinearPCMBitDepthKey: 16, AVLinearPCMIsFloatKey: false ] audioRecorder = try AVAudioRecorder(url: fileURL, settings: settings) audioRecorder?.record() When I check the recorded file’s sample rate, it logs: Actual sample rate: 8000.0 However, when I inspect the hardware sample rate: try session.setCategory(.playAndRecord, mode: .default) try session.setActive(true) print("Hardware sample rate:", session.sampleRate) I consistently get: `Hardware sample rate: 48000.0 My questions are: Is the iPhone mic actually capturing at 8 kHz, or is it recording at 48 kHz and then downsampling to 8 kHz internally? Is there any way to force the hardware to record natively at 8 kHz? If not, what’s the recommended approach for telephony-quality audio (true 8 kHz) on iOS devices? Thanks in advance for your guidance!

I neet to take pcm data from aac data, but this api has fossy me deeply.

I’m facing a strange audio routing issue that seems specific to iPhone 14 Pro / Pro Max. I’m using LiveKit (WebRTC) in a React Native app, which uses AVAudioSession internally for audio capture (VoIP / call-style usage). 🔍 What’s happening: I’m using an external USB microphone. On these devices: iPhone 11 → ✅ USB mic works iPhone 13 → ✅ USB mic works iPhone 17 Pro → ✅ USB mic works iPhone 14 Pro Max → ❌ USB mic does NOT work On iPhone 14 Pro Max: The same USB mic: ✅ Works in Voice Memos ✅ Works in Instagram Live ❌ Does NOT appear as an input option in my app ❌ Does NOT work in WhatsApp / Instagram calls Also: In my app on iPhone 14 Pro Max, iOS does not show the audio input selector UI On iPhone 17 Pro, the same app and same build does show the selector and the USB mic works ⚙️ My audio session config ( LiveKit ): await AudioSession.setAppleAudioConfiguration({ audioCategory: 'playAndRecord', audioMode: 'default', audioCategoryOptions: ['allowBluetooth', 'defaultToSpeaker'], }); await AudioSession.startAudioSession(); ❓ My questions: Is this a known limitation or behavior specific to iPhone 14 Pro / Pro Max? Does iPhone 14 Pro have different audio routing rules for call / VoIP mode compared to other devices? Why does the same USB mic work in recording apps (Voice Memos, Instagram Live) but not in call-style apps (LiveKit, WhatsApp, Instagram call)? Is there any documented difference in AVAudioSession behavior on iPhone 14 Pro regarding external USB audio inputs?

Feature Request: Long-Lived Access to Personal Apple Music Data Use Case Summary I'm developing a personal portfolio website (using Nuxt) and want to display information from my own Apple Music library - showcasing personal playlists, recently played tracks, or a read-only "now playing" widget. This is purely for personal use on my website and doesn't require other users to log in. With Spotify's API, implementing this was straightforward thanks to automatic token refresh. I want a similarly seamless integration with Apple Music. Challenge with MusicKit and Music User Tokens Apple Music API requirements Apple's Music API requires a valid Music User Token (MUT) for requests involving personal library data. Beyond the Apple Developer Token, you must obtain a user-specific token via MusicKit authentication to access your own library playlists, play history, or current playback status. Token expiration and manual renewal Music User Tokens expire after approximately 6 months without any mechanism to automatically refresh or renew them - unlike typical OAuth flows that provide refresh tokens. Apple's guidance suggests the device (e.g., iPhone) is responsible for obtaining new user tokens when old ones expire. This works for interactive apps on Apple devices but fails in server-side or long-lived web contexts like a personal website widget. Impact on personal projects Displaying Apple Music data on a public-facing site becomes difficult. I would need to periodically re-authenticate through the MusicKit JS flow every few months just to keep a widget alive. Embedding credentials in a public site is insecure, and manual token refreshing is cumbersome and easy to forget. Comparison to Spotify's Token Model Spotify's API offers a developer-friendly authentication model. Their OAuth flow provides a Refresh Token that applications can use to obtain new access tokens automatically without requiring user re-authorization. This means a personal app can maintain continuous access to a user's Spotify data for extended periods until access is revoked. When building a similar feature with Spotify, this automatic token renewal was crucial. I could safely store the refresh token on my server and have my app periodically update the access token. Many developers have created public-facing widgets showing currently playing tracks on blogs or GitHub profiles using this model. Unfortunately, Apple Music's API lacks an equivalent capability, putting it at a disadvantage for personal projects. Proposed Solutions I request Apple's consideration for one of these enhancements: Provide a mechanism to refresh or extend a Music User Token programmatically for server-side applications. This could be an OAuth-style refresh token issued alongside the MUT, or a dedicated endpoint to exchange an expired MUT for a new one. This would enable renewal without a full user re-auth/login each time. Allow developers to access their own Apple Music library data with just the long-lived Developer Token. Apple could permit GET requests to personal library endpoints using the Developer Token alone, or a special token tied to the developer's Apple ID. This access would be read-only - no ability to modify the library, purely for retrieving data. It could be an opt-in feature in the Apple Developer account settings. Either solution would significantly improve the developer experience for Apple Music API in personal projects. Security and Privacy Considerations This request is not about accessing others' data or creating privacy loopholes - it's about empowering an Apple Music subscriber to access their own information more conveniently. The proposed options respect privacy principles: The data accessed is only what the user already has access to - their own playlists, library items, or playback status. An automatic token refresh can be designed securely (revocable tokens bound to a single account with no increase in permissions). Read-only developer token access could be restricted to non-sensitive data and require explicit opt-in. Conclusion I request an improvement to Apple Music's developer experience through either (1) an automatic Music User Token refresh mechanism, or (2) a provision for read-only personal library access using a Developer Token. This would bring Apple Music integration capabilities closer to parity with services like Spotify for personal projects. I ask Apple's Developer Relations and the Apple Music API team to consider this feature request. If there are existing best practices or workarounds with current APIs, I would appreciate guidance. I invite feedback from Apple or other developers. Are there known patterns for maintaining an Apple Music user token for server-side applications, or any plans to support non-interactive use cases? Any advice is welcome. Thank you for your consideration. I look forward to integrating Apple Music into my personal site as smoothly as with other services, and believe many developers would benefit from this added flexibility. Sources: User Authentication for MusicKit - Requirements for Music User Tokens StackOverflow: Do Apple Music User Tokens expire? - Confirmation of 6-month expiration MetaBrainz GSoC Blog - Documentation of MusicKit authentication limitations Apple Developer Forums - Information on token renewal behavior Spotify for Developers - Documentation on refresh token mechanism

I want the audio session to always use the built-in microphone. However, when using the setPreferredInput() method like in this example private func enableBuiltInMic() { // Get the shared audio session. let session = AVAudioSession.sharedInstance() // Find the built-in microphone input. guard let availableInputs = session.availableInputs, let builtInMicInput = availableInputs.first(where: { $0.portType == .builtInMic }) else { print("The device must have a built-in microphone.") return } // Make the built-in microphone input the preferred input. do { try session.setPreferredInput(builtInMicInput) } catch { print("Unable to set the built-in mic as the preferred input.") } } and calling that function once in the initializer, the audio session still switches to the external microphone once one is plugged in. The session's preferredInput is nil again at that point, even if the built-in microphone is still listed in availableInputs. So, why is the preferredInput suddenly reset? when would be the appropriate time to set the preferredInput again? Observing the session’s availableInputs did not work and setting the preferredInput again in the routeChangeNotification handler seems a bad choice as it’s already a bit too late then.

Hello, Has anyone else experienced variations in the accuracy of the playbackTime value? After a few seconds of playback, the reported time adjusts by a fraction of a second, making it difficult to calculate the actual playbackTime of the audio. This can be recreated by playing a song in MusicKit, recording the start time of the audio, playing for at least 10-20 seconds, and then comparing the playbackTime value to one calculated using the start time of the audio. In my experience this jump occurs after about 10 seconds of playback. Any help would be appreciated. Thanks!

My app want Converting iphone12 HDR Video to SDR，to edit。 follow the doc Apple-HDR-Convert. My code setting the pixBuffAttributes        [pixBuffAttributes setObject:(id)(kCVImageBufferYCbCrMatrix_ITU_R_709_2) forKey:(id)kCVImageBufferYCbCrMatrixKey];       [pixBuffAttributes setObject:(id)(kCVImageBufferColorPrimaries_ITU_R_709_2) forKey:(id)kCVImageBufferColorPrimariesKey];       [pixBuffAttributes setObject:(id)kCVImageBufferTransferFunction_ITU_R_709_2 forKey:(id)kCVImageBufferTransferFunctionKey];       playerItemOutput = [[AVPlayerItemVideoOutput alloc] initWithPixelBufferAttributes:pixBuffAttributes]; but I get the playerItemOutput's output buffer   CFTypeRef colorAttachments = CVBufferGetAttachment(pixelBuffer, kCVImageBufferYCbCrMatrixKey, NULL);     CFTypeRef colorPrimaries = CVBufferGetAttachment(pixelBuffer, kCVImageBufferColorPrimariesKey, NULL);     CFTypeRef colorTransFunc = CVBufferGetAttachment(pixelBuffer, kCVImageBufferTransferFunctionKey, NULL);      NSLog(@"colorAttachments = %@", colorAttachments);     NSLog(@"colorPrimaries = %@", colorPrimaries);     NSLog(@"colorTransFunc = %@", colorTransFunc); log output: colorAttachments = ITU_R_2020 colorPrimaries = ITU_R_2020 colorTransFunc = ITU_R_2100_HLG pixBuffAttributes setting output format invalid，please help！

I am developing an app with transcription and I am exploring ways to improve the transcription from the SpeechAnalyzer/Transcriber for technical terms. SFSpeech... recognition had the capability of being augmented by contextualStrings. Does something similar exist for SpeechAnalyzer/Transcriber? If so please point me towards the documentation and any sample code that may exist for this. If there are other options, please let me know.

Hi, I am looking for a good way to play sounds at a high frequency. At the moment I am using the AVAudioEngine, and create a couple AVAudioPlayerNode and for each sound I need to play I create a AVAudioPCMBuffer. When the app needs to play a sound, I get the correct AVAudioPCMBuffer for the sound and use the first available AVAudioPlayerNode and feed it to the buffer. The timing for a metronome app has to be very precise because if it's of by about 16ms the user can hear that it is not playing had the right interval. For low speeds this is working without any problems, but at high speeds it is getting worse. Maybe anyone has an idea on how I can improve my method. Its a Plugin for Flutter. import AVFoundation class FastSoundPlayer { private var audioPlayers: [SoundPlayer?] = [] private var sounds: [String: Sound] = [:] private var engine = AVAudioEngine() let session = AVAudioSession.sharedInstance() init() { do { try session.setCategory(AVAudioSession.Category.playback, mode: AVAudioSession.Mode.default, options: [AVAudioSession.CategoryOptions.mixWithOthers]) try session.setActive(true) createSoundPlayers(count: 20) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } // Selector method to handle applicationDidBecomeActiveNotification func applicationDidBecomeActive() { // Reinitialize AVAudioEngine and reattach all nodes do { engine.reset() objc_sync_enter(audioPlayers) audioPlayers.removeAll() createSoundPlayers(count: 20) objc_sync_exit(audioPlayers) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } func createSoundPlayers(count: Int) { for _ in 0..<count { let player = SoundPlayer() engine.attach(player.player) engine.connect(player.player, to: engine.mainMixerNode, format: nil) audioPlayers.append(player) } } func load(sound: Data, name: String) { let sound = Sound(soundData: sound) sounds[name] = sound } func play(name: String) { if !engine.isRunning { applicationDidBecomeActive() } guard let sound = sounds[name] else { print("Sound not found") return } if let player = getAvailablePlayer() { player.play(sound: sound) } } func getAvailablePlayer() -> SoundPlayer? { for player in audioPlayers { if !player!.isPlaying { return player } } return nil } } class SoundPlayer { let player = AVAudioPlayerNode() var isPlaying = false init() { player.volume = 1.0 } func play(sound: Sound) { player.scheduleBuffer(sound.sound!, at: nil, options: .interrupts, completionCallbackType: .dataPlayedBack) { _ in self.complete() } if (player.engine != nil && player.engine!.isRunning) { player.play() isPlaying = true } } func complete() { isPlaying = false } } class Sound { var sound: AVAudioPCMBuffer? init(soundData: Data) { do { let temporaryURL = FileManager.default.temporaryDirectory.appendingPathComponent("tempSound.wav") try soundData.write(to: temporaryURL) // Create AVAudioFile from the temporary file URL let audioFile = try AVAudioFile(forReading: temporaryURL) // Define the format for the PCM buffer (44100Hz, stereo) let format = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44100, channels: 2, interleaved: false) // Create AVAudioPCMBuffer guard let pcmBuffer = AVAudioPCMBuffer(pcmFormat: format!, frameCapacity: AVAudioFrameCount(audioFile.length)) else { // Failed to create PCM buffer self.sound = nil return } // Read audio file into PCM buffer try audioFile.read(into: pcmBuffer) // Assign the created AVAudioPCMBuffer to the sound property self.sound = pcmBuffer } catch { print("Error loading sound file: \(error.localizedDescription)") self.sound = nil } } } Thanks!

I have the new iOS 26 SpeechTranscriber working in my application. The issue I am facing is how to determine if the device I am running on supports SpeechTranscriber. I was able to create code that tests if the device supports transcription but it takes a bit of time to run and thus the results are not available when the app launches. What I am looking for is a list of what iOS 26 devices it doesn't run on. I think its safe to assume any new devices will support it so if we can just have a list of what devices that can run iOS 26 and not able to do transcription it would be much faster for the app. I have determined it doesn't work on a SE 2nd Gen, it works on iPhone 12, SE 3rd Gen, iPhone 14 Pro, 15 Pro. As the SpeechTranscriber doesn't work in the simulator I can't determine that way. I have checked the docs and it doesn't list the devices it doesn't work on.

using iOS 26.2; Airpods 4 Long press stem to launch Siri Speak "Record Voice Memo" -> Recording starts Recording in progress... Long press stem to launch Siri -> Nothing happens. To stop recording need use phone. is this intended behaviour? i would like to be able to stop recording with Siri I am able to launch Siri from phone while recording, but point is to keep phone in pocket and start/stop recordings only via Airpods.