How does a third party developer go about supporting the new Enhanced Dialogue option for video apps in tvOS 18? If an app is using the standard AVPlayerViewController, I had assumed it would be a simple-ish matter of building against the tvOS 18 SDK but apparently not, the options don't appear, not even greyed out.

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

AVAudioFormat has no Swift concurrency annotations but the documentation states "Instances of this class are immutable." This made me always assume it was safe to pass AVAudioFormat instances around. Is this the case? If so can it be marked as Sendable? Am I missing something?

As a straightforward example, I've taken Apple's MV-HEVC sample project and added two lines. First, after the AVAssetWriterInput is created: frameInput.performsMultiPassEncodingIfSupported = true Second, after the call to multiviewWriter.startWriting(): print("canPerformMultiplePasses: \(frameInput.canPerformMultiplePasses)") Which prints true. This leads me to believe that the first encoding pass should proceed as-normal (even though I haven't handled the logic for the completion of the first pass, etc.). However, I receive this error when the code attempts to appendTaggedBuffers to the AVAssetWriterInputTaggedPixelBufferGroupAdaptor: Fatal error: Failed to append tagged buffers to multiview output Am I missing a step? Or is the multi-pass encoding only supported for standard sample/pixel buffers (and not tagged buffers)?

Please consider adding the ability to programatically download Premium and Enhanced voices. At the moment it is extremely inconvenient for our users, as they have to navigate to settings themselves to download voices. Our app relies heavily on SpeechSynthesis integration, and it would greatly benefit from this feature. FB16307193

Hello! I've two mics connected to a USB-hub. The USB-hub is then connected to my iPad. Both mics are part of the audio session's list of available inputs. The problem is that regardless of which mic I select in my app (using setPreferredInput() on the audio session), the audio keeps coming from the mic that was last connected to the USB-hub. Anyone that knows if this is a limitation in iPadOS/iOS?

I'm experiencing audio issues while developing for visionOS when playing PCM data through AVAudioPlayerNode. Issue Description: Occasionally, the speaker produces loud popping sounds or distorted noise This occurs during PCM audio playback using AVAudioPlayerNode The issue is intermittent and doesn't happen every time Technical Details: Platform: visionOS Device: vision pro / simulator Audio Framework: AVFoundation Audio Node: AVAudioPlayerNode Audio Format: PCM I would appreciate any insights on: Common causes of audio distortion with AVAudioPlayerNode Recommended best practices for handling PCM playback in visionOS Potential configuration issues that might cause this behavior Has anyone encountered similar issues or found solutions? Any guidance would be greatly helpful. Thank you in advance!

I’m an amateur developer working on a free utility for composers/producers, for which the macOS release needs to create and name RTP-MIDI sessions in Audio MIDI Setup from the command line (so I can ship a small C helper instead of telling users to click through the UI). Here’s what I’ve tried so far, without luck: • Plist hacks: Injecting entries into ~/Library/Audio/MIDI Configurations/*.mcfg works when AMS is closed, but AMS immediately locks and reverts my changes when it’s open. • CoreMIDI C API: I can create virtual ports with MIDISourceCreate, but attempting MIDIObjectGetDataProperty on the apple.midirtp.session plugin always returns err –10836. • Obj-C & Swift: Loading MIDINetworkSession and calling defaultSession, init, setNetworkName: and setting enabled = YES doesn’t produce a new session object in the Network panel. • dlopen/dlsym: I extracted the real CoreMIDI binary out of the dyld shared cache and tried binding _MIDINetworkSessionCreate, _SetName, _SetEnabled, etc., but all the symbols come back null or my tool segfaults. • Plugin registration: I’ve pulled the factory UUID (70C9C5EA-7C65-11D8-B317-000393A34B5A) from /System/Library/Extensions/AppleMIDIRTPDriver.plugin/Contents/Info.plist and called CFPlugInRegisterFactories, but it still never exposes the session-creation calls. At this point I’m convinced I’m either loading the wrong binary or missing one critical step in registering the RTP-MIDI plugin’s private API. Can anyone point me to: The exact path of the dylib or bundle that actually exports the MIDINetworkSessionCreate/MIDINetworkSessionSetName/MIDINetworkSessionSetEnabled symbols? A minimal working snippet (C or Obj-C) that reliably creates and names a Network-MIDI session? Any pointers, sample code, or even ideas about where Apple hides this functionality on macOS 15 would be hugely appreciated. Thanks!

The presentation "create audio drivers with DriverKit" from WWDC 2021 demonstrates how to use a dext to implement a virtual audio driver. It also says " If a virtual audio driver or device is all that is needed, the audio server plug-in driver model should continue to be used". Indeed, in AudioDriverKit/AudioDriverKitTypes.h, there is no IOUserAudioTransportType Virtual, although CoreAudio/AudioHardwareBase.h includes kAudioDeviceTransportTypeVirtual. For one of our products, we require virtual devices to implement a software loopback "cable". We've implemented this using the "traditional" HAL plugin, and as a proof-of-concept, also using a dext. In the dext, I tried setting the transport type to 'virt', which seems to only have the effect of changing the icon shown in Audio Midi Setup. HAL plugins require an installer, and the installer has to kill coreaudiod in a post-install script. You have to turn off SIP to debug them. Just like AudioDriverKit drivers, they are out-of-process and run in a process not owned by the hosting app. Our HAL plugin's interface is property based; we had to write a lot of boiler-plate code to implement required properties. Writing an AudioDriverKit driver is in most respects easier - a lot of the scaffolding is implemented in the base driver, which we only alter where required. Debugging and installation is much easier. The dext works just fine, as far as we can ascertain, just as well as a HAL plugin. So, my question is - is the advice to use a HAL plugin for a virtual device still correct in 2025? And if so, what's the objection? We'd really prefer to ship the AudioDriverKit virtual audio device.

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. 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: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. 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? 3. 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.

I've been generating new Audio Unit Extension apps with Xcode 16 (and newer), and although they generally work initially, it is easy (although I'm not sure how to do it reliably) to cause the app to no longer be able to instantiate the audiounit. Generally the call to AVAudioUnit.findComponent fails and SimplePlayEngine hits the fatalError("Failed to find component with type...") In the most recent project, merely adding files to the extension (without making any use of them) caused it to go off the rails. If I "Archive" the app+plugin, there is no audio unit extension in the bundle. If I switch to the audiounit extension and build it it's fine. If I look at the build folder in Library/Developer/Xcode/project_folder the extension_name.appex is there. Any ideas? If I can coax an unmodified audio unit extension project to exhibit this behavior I'll attach it here. Right now what I have has code I don't want to share.

I am unable to access the Int32 error from the errors that CoreAudio throws in Swift type AudioHardwareError. This is critical. There is no way to access the errors or even create an AudioHardwareError to test for errors. do { _ = try AudioHardwareDevice(id: 0).streams // will throw } catch { if let error = error as? AudioHardwareError { // cast to AudioHardwareError print(error) // prints error code but not the errorDescription } } How can get reliably get the error.Int32? Or create a AudioHardwareError with an error constant? There is no way for me to handle these error with code or run tests without knowing what the error is. On top of that, by default the error localizedDescription does not contain the errorDescription unless I extend AudioHardwareError with CustomStringConvertible. extension AudioHardwareError: @retroactive CustomStringConvertible { public var description: String { return self.localizedDescription } }

Hi, for the implementation of an audio player with signed URL's, I need to be able to set an authorization header to the request for an AVURLAsset. This works but not on Airplay when trying to stream multiple songs in a queue. For each item I do: let headerFields: [String: String] = ["Authorization": getIdToken()!] super.init(url: url, options: ["AVURLAssetHTTPHeaderFieldsKey": headerFields]) But only the first 2 songs in the queue actually get this authorization header sent along, somehow it is removed for subsequent songs. Any ideas on how I can fix this? thanks, Thomas

Hello. My team and I think we have an issue where our app is asked to gracefully shutdown with a following SIGTERM. As we’ve learned, this is normally not an issue. However, it seems to also be happening while our app (an audio streamer) is actively playing in the background. From our perspective, starting playback is indicating strong user intent. We understand that there can be extreme circumstances where the background audio needs to be killed, but should it be considered part of normal operation? We hope that’s not the case. All we see in the logs is the graceful shutdown request. We can say with high certainty that it’s happening though, as we know that playback is running within 0.5 seconds of the crash, without any other tracked user interaction. Can you verify if this is intended behavior, and if there’s something we can do about it from our end. From our logs it doesn’t look to be related to either memory usage within the app, or the system as a whole. Best, John

Hey folks, I'm running into an odd issue suddenly with an app that had a working MusicKit integration before. I'm using ApplicationMusicPlayer to play Apple Music albums and songs. I'm testing on a physical device, signed in to Apple ID, and with a valid subscription. Apple Music via the first-party app works entirely fine on this device. Attempting to play back any content at all gives the log: <ICUserIdentityStoreACAccountBackend: 0x1070bf3e0> Failed to initialize primary apple account, error=Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store} [ICUserIdentityStore] - initializing account histories with activeAccountDSID = nil, activeLockerAccountDSID = nil, timestamp = 14605951908 [ICUserIdentityStore] Failed to fetch local store account with error: Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store}. The album artwork, track names, etc, all appear in the control center playback controls, but the music doesn't play. Trying to trigger playback with control center just results in it skipping to the next track, which doesn't play either. This exact code used to work. I have the MusicKit service selected in Apple Connect. Since this isn't entitlement-based, I'm not sure how else to check that I'm set up correctly. I've tried deleting/reinstalling the app, restarting the device, cleaning/rebuilding, and deleting DerivedData, to no avail. Any help? Running Xcode 16.4 (16F6), testing on iOS 18.5 (22F76)

Hi, I'm facing an issuer with audio worklet in safari. This issue is clearly an iOS bug (it doesn't occur on iPad or Mac) Here's the minimal reproduction: Go to https://googlechromelabs.github.io/web-audio-samples/audio-worklet/basic/hello-audio-worklet/ Press start Audio will not be playing Open YouTube on another tab and start any video Audio from the worklet will start playing Is this a known issue? Any plans to address that? Any workaround available?

I am trying to get access to raw audio samples from mic. I've written a simple example application that writes the values to a text file. Below is my sample application. All the input samples from the buffers connected to the input tap is zero. What am I doing wrong? I did add the Privacy - Microphone Usage Description key to my application target properties and I am allowing microphone access when the application launches. I do find it strange that I have to provide permission every time even though in Settings > Privacy, my application is listed as one of the applications allowed to access the microphone. class AudioRecorder { private let audioEngine = AVAudioEngine() private var fileHandle: FileHandle? func startRecording() { let inputNode = audioEngine.inputNode let audioFormat: AVAudioFormat #if os(iOS) let hardwareSampleRate = AVAudioSession.sharedInstance().sampleRate audioFormat = AVAudioFormat(standardFormatWithSampleRate: hardwareSampleRate, channels: 1)! #elseif os(macOS) audioFormat = inputNode.inputFormat(forBus: 0) // Use input node's current format #endif setupTextFile() inputNode.installTap(onBus: 0, bufferSize: 1024, format: audioFormat) { [weak self] buffer, _ in self!.processAudioBuffer(buffer: buffer) } do { try audioEngine.start() print("Recording started with format: \(audioFormat)") } catch { print("Failed to start audio engine: \(error.localizedDescription)") } } func stopRecording() { audioEngine.stop() audioEngine.inputNode.removeTap(onBus: 0) print("Recording stopped.") } private func setupTextFile() { let tempDir = FileManager.default.temporaryDirectory let textFileURL = tempDir.appendingPathComponent("audioData.txt") FileManager.default.createFile(atPath: textFileURL.path, contents: nil, attributes: nil) fileHandle = try? FileHandle(forWritingTo: textFileURL) } private func processAudioBuffer(buffer: AVAudioPCMBuffer) { guard let channelData = buffer.floatChannelData else { return } let channelSamples = channelData[0] let frameLength = Int(buffer.frameLength) var textData = "" var allZero = true for i in 0..<frameLength { let sample = channelSamples[i] if sample != 0 { allZero = false } textData += "\(sample)\n" } if allZero { print("Got \(frameLength) worth of audio data on \(buffer.stride) channels. All data is zero.") } else { print("Got \(frameLength) worth of audio data on \(buffer.stride) channels.") } // Write to file if let data = textData.data(using: .utf8) { fileHandle!.write(data) } } }

After an Album, Playlist, or collection of songs have been added to the ApplicationMusicPlayer queue, clearing the queue can be easily accomplished with: ApplicationMusicPlayer.shared.queue.entries = [] This transitions the player to a paused state with an empty queue. After queueing a Station, the same code cannot be used to clear the queue. Instead, it causes the queue to be refilled with a current and next MusicItem from the Station. What's the correct way to detect that the ApplicationMusicPlayer is in the state where it's being refilled by a Station and clear it? I've tried the following approaches with no luck: # Reinitialize queue ApplicationMusicPlayer.shared.queue = ApplicationMusicPlayer.Queue() # Create empty Queue let songs: [Song] = [] let emptyQueue = ApplicationMusicPlayer.Queue(for: songs) ApplicationMusicPlayer.shared.queue = emptyQueue

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 get personal recommendations MusicItemCollection using this code: func getRecommendations() async throws -> MusicItemCollection<MusicPersonalRecommendation> { let request = MusicPersonalRecommendationsRequest() let response = try await request.response() let recommendations = response.recommendations return recommendations } However, all recommendations contain no more than 12 MusicItem's, while the Music.app application provides much more for some recommendations, for example, for the You recently listened recommendation, the Music.app application displays 40 items. Each recommendation has an items property that contains a collection of musical items MusicItemCollection<MusicPersonalRecommendation.Item>, the hasNextBatch property for these collections is always false. I expected that for some collections loading of new items would be available. Please tell me if I'm doing something wrong or is this a MusicKit bug? Thank you!