Hi, I believe I've found a potential error in the sample code on the documentation page for creating and using a process tap with an aggregate device. The issue is in the section explaining how to add a tap to the aggregate device. I have already filed a Feedback Assistant ticket on this (ID: FB17411663) but haven't heard back for months. Capturing system audio with Core Audio taps The sample code for modifying the kAudioAggregateDevicePropertyTapList incorrectly uses the tapID as the target AudioObjectID when calling AudioObjectSetPropertyData. // (Code to get the list and potentially modify listAsArray) if var listAsArray = list as? [CFString] { // ... (modification logic) ... // Set the list back on the aggregate device. <--- The comment is correct list = listAsArray as CFArray _ = withUnsafeMutablePointer(to: &list) { list in // INCORRECT: This call uses tapID as the target object. AudioObjectSetPropertyData(tapID, &propertyAddress, 0, nil, propertySize, list) } } The kAudioAggregateDevicePropertyTapList is a property that belongs to the aggregate device, not the individual tap. Therefore, to set this property, the AudioObjectSetPropertyData function must target the AudioObjectID of the aggregate device itself. Using tapID as the first argument is logically incorrect for this operation and will not update the aggregate device as intended. Furthermore, the preceding AudioObjectGetPropertyData call to fetch the list also appears to use the incorrect tapID as its target in the sample. The AudioObjectID for both getting and setting this property should be the ID of the aggregate device. _ = AudioObjectGetPropertyData(aggregateDeviceID, &propertyAddress, 0, nil, &propertySize, &list) _ = AudioObjectSetPropertyData(aggregateDeviceID, &propertyAddress, 0, nil, propertySize, newList) Thank you!

hi, Is there an Audio Unit logo I can show on my website? I would love to show that my application is able to host Audio Unit plugins. regards, Joël

I'm streaming mp3 audio data using URLSession/AudioFileStream/AVAudioConverter and getting occasional silent buffers and glitches (little bleeps and whoops as opposed to clicks). The issues are present in an offline test, so this isn't an issue of underruns. Doing some buffering on the input coming from the URLSession (URLSessionDataTask) reduces the glitches/silent buffers to rather infrequent, but they do still happen occasionally. var bufferedData = Data() func parseBytes(data: Data) { bufferedData.append(data) // XXX: this buffering reduces glitching // to rather infrequent. But why? if bufferedData.count > 32768 { bufferedData.withUnsafeBytes { (bytes: UnsafeRawBufferPointer) in guard let baseAddress = bytes.baseAddress else { return } let result = AudioFileStreamParseBytes(audioStream!, UInt32(bufferedData.count), baseAddress, []) if result != noErr { print("❌ error parsing stream: \(result)") } } bufferedData = Data() } } No errors are returned by AudioFileStream or AVAudioConverter. func handlePackets(data: Data, packetDescriptions: [AudioStreamPacketDescription]) { guard let audioConverter else { return } var maxPacketSize: UInt32 = 0 for packetDescription in packetDescriptions { maxPacketSize = max(maxPacketSize, packetDescription.mDataByteSize) if packetDescription.mDataByteSize == 0 { print("EMPTY PACKET") } if Int(packetDescription.mStartOffset) + Int(packetDescription.mDataByteSize) > data.count { print("❌ Invalid packet: offset \(packetDescription.mStartOffset) + size \(packetDescription.mDataByteSize) > data.count \(data.count)") } } let bufferIn = AVAudioCompressedBuffer(format: inFormat!, packetCapacity: AVAudioPacketCount(packetDescriptions.count), maximumPacketSize: Int(maxPacketSize)) bufferIn.byteLength = UInt32(data.count) for i in 0 ..< Int(packetDescriptions.count) { bufferIn.packetDescriptions![i] = packetDescriptions[i] } bufferIn.packetCount = AVAudioPacketCount(packetDescriptions.count) _ = data.withUnsafeBytes { ptr in memcpy(bufferIn.data, ptr.baseAddress, data.count) } if verbose { print("handlePackets: \(data.count) bytes") } // Setup input provider closure var inputProvided = false let inputBlock: AVAudioConverterInputBlock = { packetCount, statusPtr in if !inputProvided { inputProvided = true statusPtr.pointee = .haveData return bufferIn } else { statusPtr.pointee = .noDataNow return nil } } // Loop until converter runs dry or is done while true { let bufferOut = AVAudioPCMBuffer(pcmFormat: outFormat, frameCapacity: 4096)! bufferOut.frameLength = 0 var error: NSError? let status = audioConverter.convert(to: bufferOut, error: &error, withInputFrom: inputBlock) switch status { case .haveData: if verbose { print("✅ convert returned haveData: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(haveData) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } case .inputRanDry: if verbose { print("🔁 convert returned inputRanDry: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(inputRanDry) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } return // wait for next handlePackets case .endOfStream: if verbose { print("✅ convert returned endOfStream") } return case .error: if verbose { print("❌ convert returned error") } if let error = error { print("error converting: \(error.localizedDescription)") } return @unknown default: fatalError() } } }

I am work an app development on an app which request an audio function in background as an alert sound. during debug testing , the function work fine, but once I testing standalone without debugging , The function not work , it will play out the sound when I back to app. does any way to trace the issues ?

On macOS Sequoia, I'm having the hardest time getting this basic audio output to work correctly. I'm compiling in XCode using C99, and when I run this, I get audio for a split second, and then nothing, indefinitely. Any ideas what could be going wrong? Here's a minimum code example to demonstrate: #include &lt;AudioToolbox/AudioToolbox.h&gt; #include &lt;stdint.h&gt; #define RENDER_BUFFER_COUNT 2 #define RENDER_FRAMES_PER_BUFFER 128 // mono linear PCM audio data at 48kHz #define RENDER_SAMPLE_RATE 48000 #define RENDER_CHANNEL_COUNT 1 #define RENDER_BUFFER_BYTE_COUNT (RENDER_FRAMES_PER_BUFFER * RENDER_CHANNEL_COUNT * sizeof(f32)) void RenderAudioSaw(float* outBuffer, uint32_t frameCount, uint32_t channelCount) { static bool isInverted = false; float scalar = isInverted ? -1.f : 1.f; for (uint32_t frame = 0; frame &lt; frameCount; ++frame) { for (uint32_t channel = 0; channel &lt; channelCount; ++channel) { // series of ramps, alternating up and down. outBuffer[frame * channelCount + channel] = 0.1f * scalar * ((float)frame / frameCount); } } isInverted = !isInverted; } AudioStreamBasicDescription coreAudioDesc = { 0 }; AudioQueueRef coreAudioQueue = NULL; AudioQueueBufferRef coreAudioBuffers[RENDER_BUFFER_COUNT] = { NULL }; void coreAudioCallback(void* unused, AudioQueueRef queue, AudioQueueBufferRef buffer) { // 0's here indicate no fancy packet magic AudioQueueEnqueueBuffer(queue, buffer, 0, 0); } int main(void) { const UInt32 BytesPerSample = sizeof(float); coreAudioDesc.mSampleRate = RENDER_SAMPLE_RATE; coreAudioDesc.mFormatID = kAudioFormatLinearPCM; coreAudioDesc.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked; coreAudioDesc.mBytesPerPacket = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mFramesPerPacket = 1; coreAudioDesc.mBytesPerFrame = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mChannelsPerFrame = RENDER_CHANNEL_COUNT; coreAudioDesc.mBitsPerChannel = BytesPerSample * 8; coreAudioQueue = NULL; OSStatus result; // most of the 0 and NULL params here are for compressed sound formats etc. result = AudioQueueNewOutput(&amp;coreAudioDesc, &amp;coreAudioCallback, NULL, 0, 0, 0, &amp;coreAudioQueue); if (result != noErr) { assert(false == "AudioQueueNewOutput failed!"); abort(); } for (int i = 0; i &lt; RENDER_BUFFER_COUNT; ++i) { uint32_t bufferSize = coreAudioDesc.mBytesPerFrame * RENDER_FRAMES_PER_BUFFER; result = AudioQueueAllocateBuffer(coreAudioQueue, bufferSize, &amp;(coreAudioBuffers[i])); if (result != noErr) { assert(false == "AudioQueueAllocateBuffer failed!"); abort(); } } for (int i = 0; i &lt; RENDER_BUFFER_COUNT; ++i) { RenderAudioSaw(coreAudioBuffers[i]-&gt;mAudioData, RENDER_FRAMES_PER_BUFFER, RENDER_CHANNEL_COUNT); coreAudioBuffers[i]-&gt;mAudioDataByteSize = coreAudioBuffers[i]-&gt;mAudioDataBytesCapacity; AudioQueueEnqueueBuffer(coreAudioQueue, coreAudioBuffers[i], 0, 0); } AudioQueueStart(coreAudioQueue, NULL); sleep(10); // some time to hear the audio AudioQueueStop(coreAudioQueue, true); AudioQueueDispose(coreAudioQueue, true); return 0; }

ApplicationMusicPlayer is not available on watchOS but all other platforms. Is there a technical reason for that like battery life? Same goes for SystemMusicPlayer and MPMusicPlayerController. I already filed feedbacks for that.

My app uses the AVFoundation to pronounce some words. Running the app from Xcode, either to a simulator or device, I frequently get this crash at start-up: AXSpeech (13): EXC_BAD_ACCESS (code=EXC_I386_GPFLT). It seems to occur randomly, maybe 20%-30% of the time I launch the app. When it does not crash, using audio works as expected. When launched from the device, it never crashes (so far, at least). Here's the code that outputs speech: Declared at the top level of the View struct: @State var synth = AVSpeechSynthesizer() In the View, as part of a Button's action closure: let utterance = AVSpeechUtterance(string: answer) utterance.voice = AVSpeechSynthesisVoice(language: "en_US") synth.speak(utterance) Any idea on how to stop this? It's annoying having to launch the app multiple times to test on a simulator or device.

I have integrated the ShazamKit SDK into my iOS app and would like to implement the same functionality in my Android app. My question is: Can I use the Android version of the ShazamKit SDK for commercial purposes? After extensive research, I could not find any official information regarding the license of the Android version of the ShazamKit SDK. Could you please provide a formal license statement?

I created a virtual audio device to capture system audio with a sample rate of 44.1 kHz. After capturing the audio, I forward it to the hardware sound card using AVAudioEngine, also with a sample rate of 44.1 kHz. However, due to the clock sources being unsynchronized, problems occur after a period of playback. How can I retrieve the clock source of the hardware device and set it for the virtual device?

I have some tried-and-tested code that records and plays back audio via AUHAL which breaks on Tahoe on Intel. The same code works fine on Sequioa and also works on Tahoe on Apple Silicon. To start with something simple, the following code to request access to the Microphone doesn't work as it should: bool RequestMicrophoneAccess () { __block AVAuthorizationStatus status = [AVCaptureDevice authorizationStatusForMediaType: AVMediaTypeAudio]; if (status == AVAuthorizationStatusAuthorized) return true; __block bool done = false; [AVCaptureDevice requestAccessForMediaType: AVMediaTypeAudio completionHandler: ^ (BOOL granted) { status = (granted) ? AVAuthorizationStatusAuthorized : AVAuthorizationStatusDenied; done = true; }]; while (!done) CFRunLoopRunInMode (kCFRunLoopDefaultMode, 2.0, true); return status == AVAuthorizationStatusAuthorized; } On Tahoe on Intel, the code runs to completion but granted is always returned as NO. Tellingly, the popup to ask the user to grant microphone access is never displayed, even though the app is not present in the Privacy pane and never appears there. On Apple Silicon, everything works fine. There are some other problems, but I'm hoping they have a common underlying cause and that the Apple guys can figure out what's wrong from the information in this post. I'd be happy to test any potential fix. Thanks.

Hi, In my project I am using AVFoundation for recording the audio. We are using AVAudioMixerNode class below method to record the audio packet. **func installTap( onBus bus: AVAudioNodeBus, bufferSize: AVAudioFrameCount, format: AVAudioFormat?, block tapBlock: @escaping AVAudioNodeTapBlock ) ** It works perfectly fine. But in production env some small percentage of the user we are facing issue like after recording few packets it stops automatically without stopping the audio engine. Can anyone help here that why this happens? I have also observed for mediaServicesWereResetNotification and added log on receiving this notification but when this issue happens I don't see any occurence of this log. Also is there any callback when the engine stops?

Hi there! We have a suite of AudioUnit v2 plugins that have been shipped for some time as aufx plugins, and we are looking into MIDI-related platform upgrades, so we need a way to update these plugins to request MIDI from Logic (and other AU hosts) but avoid changing our AU type and subtype so we don't break existing sessions. Any ideas on how we can do this?

Does anyone know how to pronounce the sound of a specific instrument when you tap a button on the screen on your iPhone or iPad? Now, in the middle of creating a music learning app, I'm thinking of assigning monotones or chords to the button-like frames on the keyboard and fingerboard on the screen. Can it be achieved with SwiftUI chords alone? Once upon a time, MIDI level 1 I remember that there was a pronunciation function of the instrument, but I don't think about implementing the same function in the current OS. Please lend me your wisdom.

Environment Windows 11 [edition/build]: [e.g., 23H2, 22631.x] Apple Music for Windows version: [e.g., 1.x.x from Microsoft Store] Library folder: C:\Users<user>\Music\Apple Music\Apple Music Library.musiclibrary Summary I need a supported way to programmatically enumerate the local Apple Music library on Windows (track file paths, playlists, etc.) for reconciliation with the on-disk Media folder. On macOS this used to be straightforward via scripting/export; on Windows I can’t find an equivalent. What I’m seeing in the library bundle Library.musicdb → not SQLite. First 4 bytes: 68 66 6D 61 ("hfma"). Library Preferences.musicdb → also starts with "hfma". artwork.sqlite → SQLite but appears to be artwork cache only (no track file paths). Extras.itdb → has SQLite format 3 header but (from a quick scan) not seeing track locations. Genius.itdb → not a SQLite database on this machine. What I’ve tried Attempted to open Library.musicdb with SQLite providers → error: “file is not a database.” Binary/string scans (ASCII, UTF-16LE/BE, null-stripped) of Library.musicdb → did not reveal file paths or obvious plist/XML/JSON blobs. The Windows Apple Music UI doesn’t appear to expose “Export Library / Export Playlist” like legacy iTunes did, and I can’t find a public API for local library enumeration on Windows. What I’m trying to accomplish Read local track entries (absolute or relative paths), detect broken links, and reconcile against the Media folder. A read-only solution is fine; I do not need to modify the library. Questions for Apple Is the Library.musicdb file format documented anywhere, or is there a supported SDK/API to enumerate the local library on Windows? Is there a supported export mechanism (CLI, UI, or API) on Windows Apple Music to dump the local library and/or playlists (XML/CSV/JSON)? Is there a Windows-specific equivalent to the old iTunes COM automation or any MusicKit surface that can return local library items (not streaming catalog) and their file locations? If none of the above exist today, is there a recommended workaround from Apple for library reconciliation on Windows (e.g., documented support for importing M3U/M3U8 to rebuild the local library from disk)? Are there any plans/timeline for adding Windows feature parity with iTunes/Music on macOS for exporting or scripting the local library? Why this matters For large personal libraries, users occasionally end up with orphaned files on disk or broken links in the app. Without an export or API, it’s difficult to audit and fix at scale on Windows. Reference details (in case it helps triage) Library.musicdb header bytes: 68-66-6D-61-A0-00-00-00-10-26-34-00-15-00-01-00 (ASCII shows hfma…). artwork.sqlite is readable but doesn’t contain track file paths (appears limited to artwork). I can supply a minimal repro tool and logs if that’s helpful. Feature request (if no current API) Add an official Export Library/Playlists action on Windows Apple Music, or Provide a read-only Windows API (or schema doc) that surfaces track file locations and playlist membership from the local library. Thanks in advance for any guidance or pointers to docs I might have missed.

Hi, I am creating an app that can include videos or images in it's data. While @Attribute(.externalStorage) helps with images, with AVAssets I actually would like access to the URL behind that data. (as it would be stupid to load and then save the data again just to have a URL) One key component is to keep all of this clean enough so that I can use (private) CloudKit syncing with the resulting model. All the best Christoph

Environment→ ・Device: iPad 10th generation ・OS：**iOS18.3.2 We're using AVAudioPlayer to play a sound when a button is tapped. In our use case, this button can be tapped very frequently — roughly every 0.1 to 0.2 seconds. Each tap triggers the following function: var audioPlayer: AVAudioPlayer? func soundPlay(resource: String, type: String){ guard let path = Bundle.main.path(forResource: resource, ofType: type) else { return } do { audioPlayer = try AVAudioPlayer(contentsOf: URL(fileURLWithPath: path)) audioPlayer!.delegate = self try audioSession.setCategory(.playback) } catch { return } self.audioPlayer!.play() } The issue is that under high-frequency tapping (especially around 0.1–0.15s intervals), the app occasionally crashes. The crash does not occur every time, but it happens randomly — sometimes within 30 seconds, within 1 minute, or even 3 minutes of continuous tapping. Interestingly, adding a delay of 0.2 seconds between button taps seems to prevent the crash entirely. Delays shorter than 0.2 seconds (e.g.,0.15s,0.18s) still result in occasional crashes. My questions are: **Is this expected behavior from AVAudioPlayer or AVAudioSession? Could this be a known issue or a limitation in AVFoundation? Is there any documentation or guidance on handling frequent sound playback safely?** Any insights or recommendations on how to handle rapid, repeated audio playback more reliably would be appreciated.

Hello everyone, I'm implementing the new AVInputPickerInteraction API on iOS 26 to allow users to select their microphone from a custom settings menu before recording. The implementation seems correct, but I'm encountering a strange issue where the input selection immediately reverts to the previous device. The Situation: The picker is presented correctly via a manual call to .present(). I can see all available inputs (e.g., "iPhone Microphone" and "AirPods"). The current input is "iPhone Microphone". I tap on "AirPods". The UI updates to show "AirPods" as selected for a fraction of a second, then immediately jumps back to "iPhone Microphone". The same thing happens in reverse. It seems like the system is automatically reverting the audio route change requested by the picker. My Implementation: My setup follows the standard pattern discussed in the WWDC sessions. Setup Code: This setup is performed once before the user can trigger the picker. @available(iOS 26.0, *) var inputPickerInteraction: AVInputPickerInteraction? // Note: The AVAudioSession is configured to .playAndRecord // and set to active elsewhere in the code before this setup is called. if #available(iOS 26.0, *) { // Setup the picker let picker = AVInputPickerInteraction() self.inputPickerInteraction = picker self.view.addInteraction(picker) // Added to establish context } Presentation Code: When a user selects "Change Input" from my custom settings menu, I call .present() on the main thread. // In a delegate method from a custom menu if #available(iOS 26.0, *) { DispatchQueue.main.async { self.inputPickerInteraction?.present(animated: true) } } What I've already checked: The AVAudioSession is active and its category is .playAndRecord. The inputPickerInteraction object is not nil. The .present() method is being called on the main thread. The picker is added to a view using view.addInteraction() in the setup phase. I've reviewed my code to ensure there is no other logic that could be manually resetting the AVAudioSession's preferred input. Has anyone else experienced this behavior? I suspect this might be a bug in the new API, but I want to make sure I'm not missing a crucial step in managing the AVAudioSession state. Any insights or potential workarounds would be greatly appreciated. Thank you.

Using the official SwiftTranscriptionSampleApp from WWDC 2025, speech transcription takes 14+ seconds from audio input to first result, making it unusable for real-time applications. Environment iOS: 26.0 Beta Xcode: Beta 5 Device: iPhone 16 pro Sample App: Official Apple SwiftTranscriptionSampleApp from WWDC 2025 Configuration Tested Locale: en-US (properly allocated with AssetInventory.allocate(locale:)) and es-ES Setup: All optimizations applied (preheating, high priority, model retention) I started testing in my own app to replace SFSpeech API and include speech detection but after long fights with documentation (this part is quite terrible TBH) I tested the example (https://developer.apple.com/documentation/speech/bringing-advanced-speech-to-text-capabilities-to-your-app) and saw same results. I added some logs to check the specific time: 🎙️ [20:30:41.532] ✅ Analyzer started successfully - ready to receive audio! 🎙️ [20:30:41.532] Listening for transcription results... 🎙️ [20:30:56.342] 🚀 FIRST TRANSCRIPTION RESULT after 14.810s: 'Hello' (isFinal: false) Questions Is this expected performance for iOS 26 Beta, because old SFSpeech is far faster? Are there additional optimization steps for SpeechTranscriber? Should we expect significant performance improvements in later betas?

Using an iPhone Pro 12 running iOS 26.0.1, with AirPods Pro 3. Camera app does capture video with what seems to be "Studio Quality Recording". Am trying to replicate that SQR with my own Camera like app, and while I can pull audio in from the APP3 mic, and my video capture app is recording a 48,000Hz high-bitrate video, the audio still sounds non-SQR. I'm seeing bluetoothA2DP , bluetoothLE , bluetoothHFP as portType, and not sure if SQR depends on one of those? Is there sample code demonstrating a SQR capture? Nevermind video and camera, just audio even? Also, I don't understand what SQR is doing between the APP3 and the iPhone. What codec is that? What bitrate is that? If I capture video using Capture and inspect the audio stream I see mono 74.14 kbit/s MPEG-4 AAC, 48000 Hz. But I assume that's been recompressed and not really giving me any insight into the APP3 H2 transmission?

My audio app shows a control bar at the bottom of the window. The controls show nicely, but there is a black "slab" appearing behind the inline controls, the same size as the playerView. Setting the player view background color does nothing: playerView.wantsLayer = true playerView.layer?.backgroundColor = NSColor.clear.cgColor How can I clear the background? If I use .floating controlsStyle, I don't get the background "slab".