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.

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.

On Apple TV 4K 3rd generation, with tvOS 26 beta 2, when two HomePod 2 are paired to the device, music and movie sources with Dolby Atmos can only be listened to in stereo. dolby atmos not supported

Since MacOS 26 Apple Music has inconsitent drops to the Quality of some Tracks indiscrimantly. I don't know if others Expereinced it. It doesn't happen on the Speakers or connected via Bluetooth, but the AUX I/O has it quite often. It is more noticable on Headphones with 48kHz and higher Frequency Bandwidth. Here is the FB18062589

Hello. My app uses AVAudioRecorder to generate recording files, which are consistently only 4kb in size. Most users generate audio files normally, with only a few users experiencing this phenomenon occasionally. After uninstalling and installing the app, it will work normally, but it will reappear after a period of time. I have compared that the problematic audio files generated each time are fixed and cannot be played. Added the audioRecorderDidFinishRecording proxy method, which shows that the recording was completed normally. The user also reported that the recording is normal, but there is a problem with the generated file. How should I handle this issue? Look forward to your reply. - (void)startRecordWithOrderID:(NSString *)orderID { AVAudioSession *audioSession = [AVAudioSession sharedInstance]; [audioSession setCategory:AVAudioSessionCategoryRecord error:nil]; [audioSession setActive:YES error:nil]; NSMutableDictionary *settings = [[NSMutableDictionary alloc] init]; [settings setObject:[NSNumber numberWithFloat: 8000.0] forKey:AVSampleRateKey]; [settings setObject:[NSNumber numberWithInt: kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [settings setObject:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [settings setObject:[NSNumber numberWithInt: 1] forKey:AVNumberOfChannelsKey]; [settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; NSString *path = [WDUtility createDirInDocument:@"audios" withOrderID:orderID withPathExtension:@"wav"]; NSURL *tmpFile = [NSURL fileURLWithPath:path]; recorder = [[AVAudioRecorder alloc] initWithURL:tmpFile settings:settings error:nil]; [recorder setDelegate:self]; [recorder prepareToRecord]; [recorder record]; }

I am working on an app which plays audio - https://youtu.be/VbAfUk_eYl0?si=nJg5ayy2faWE78-g - and one of the features is, on restart, if you had paused playback of a file at the time the app was previously shut down (or were playing one at the time of shutdown), the paused state and position in the file is restored exactly as it was, on restart. The functionality works. However, it seems impossible to get the "now playing" information in iOS into the right state to reflect that via the MediaPlayer API. On restart, handlers are attached to the play/pause/togglePlayPause actions on MPRemoteCommandCenter.shared(), and the map of media info is updated on MPNowPlayingInfoCenter.default().nowPlayingInfo. What happens is that iOS's media view shows the audio as playing and offers a pause button - even though the play action is enabled and the pause action is disabled. Once playback has been initiated (my workaround is to have the pause action toggle the play state, since otherwise you wouldn't be able to initiate playback from controls in a car without initiating it once from a device first). I've created a simplified white-noise-player demo to illustrate the problem - simply build and deploy it, and then start the app, lock your device and look at the playback controls on the lock screen. It will show a pause button - same behavior I've described. https://github.com/timboudreau/ios-play-pause-demo I've tried a few things to narrow down the source of the issue - for example, thinking that not MPNowPlayingInfoPropertyPlaybackProgress and MPMediaItemPropertyPlaybackDuration might be the culprit (since the system interpolates elapsed time and it's recommended to update those properties infrequently) on startup might do the trick, but the result is the same, just without a duration or progress shown. What governs this behavior, and is there some way to explicitly tell the media player API your current state is paused?

It sounds simple but searching for the name "Favorite Songs" is a non-starter because it's called different names in different countries, even if I specify "&l=en_us" on the query. So is there another property, relationship or combination thereof which I can use to tell me when I've found the right playlist? Properties I've looked at so far: canEdit: will always be false so narrows things down a little inFavorites: not helpful as it depends on whether the user has favourite the favourites playlist, so not relevant hasCatalog: seems always true so again may narrow things down a bit isPublic: doesn't help Adding the catalog relationship doesn't seem to show anything immediately useful either. Can anyone help? Ideally I'd like to see this as a "kind" or "type" as it has different properties to other playlists, but frankly I'll take anything at this point.

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've got a setup using AVAudioEngine with several tone generator nodes, each with a chain of processing nodes, the chains then mixed into the main output. Generator ➡️ Effect ➡️... ➡️ .mainMixerNode ➡️ .outputNode). Generator ➡️ Effect ➡️... ⤴️ ... Generator ➡️ Effect ➡️... ⤴️ The user should be able to mute any chain individually. I've found several potential approaches to muting, but not terribly happy with any of them. Adjust the amplitudes directly in my tone generators. Issue: Consumes CPU even when completely muted. 4 generators adds ~15% cpu, even when all chains are muted. Detach/attach chains that are muted/unmuted. Issue: Causes loud clicking/popping sounds whenever muted/unmuted. Fade mixer output volume while detaching/attaching a chain (just cutting the volume immediately to 0 doesn't get rid of the clicking/popping). Issue: Causes all channels to fade during the transition, so not ideal. The rest of these ideas are variations on making volume control+detatch/attach work for individual chains, since approach #3 worked well. Add an AVAudioMixer to the end of each chain (just for volume control). Issue: Only the mixer on the final chain functions -- the others block all output. Not sure what's going on there. Use matrix mixer (for multi-input volume control). Plus detach/attach to reduce CPU if necessary. Not yet attempted, due to perceived complexity and reports of fragility in order of wiring in. A bunch of effort before I even know if it's going to work. Develop my own fader node to put on the end of each channel. Unlike the tone generator (simple AVSourceNode), developing an effect node seems complex and time consuming. Might not even fix CPU use. I'm not completely averse to the learning curve of either 5 or 6, but would rather get some guidance on best approach before diving in. They both seem likely to take more effort than I'd like for the simple behavior I'm trying to achieve.

I have an AUv3 plugin which uses an FFT - which requires n samples before it can produce any output - so, depending on the relation between the host's buffer size and the FFT window size, it may receive a several buffers of samples, producing no output, and then dumping out what it has once a sufficient number of samples have been received. This means that output is produced in fits and starts, in batches that match the FFT size (modulo oversampling) - e.g. if being fed buffers of 256 samples with an fft size of 1024, the output buffer sizes will be 0 for the first 3 buffers, and upon the fourth, the first 256 processed samples are returned and the remaining 768 cached; the next three buffers will return the remaining cached samples while processing and buffering subsequent ones, and so forth. The internal mechanics of that I have solved, caching output if the current output buffer is too small, and so forth - so it all works as advertised, and the plugin reports its latency correctly. And when run as an app in demo-mode, playback works as expected. In the plugin's render block, it captures the number of frames written, and if it is less than the number of frames passed in, adjusts the mDataByteSize of the output buffers to match the actual quantity of data being returned: unsigned int framesWritten = (unsigned int) processHelper->processWithEvents(inAudioBufferList, outAudioBufferList, timestamp, frameCount, realtimeEventListHead); if (framesWritten < frameCount) { for (UInt32 i = 0; i < outAudioBufferList->mNumberBuffers; ++i) { outAudioBufferList->mBuffers[i].mDataByteSize = framesWritten * 4; // assume 4 byte floats } } However, there are a couple of serious issues: auval -v fails it with - Render Test at 64 frames, sample rate: 22050 Hz ERROR: Output Buffer Size does not match requested When connected to Logic Pro, it appears that mDataByteSize is ignored, and the entire allocated buffer is read - audio has sections of silence snipped into it which corresponds the number of empty buffers being returned If I set Logic's buffer size to 1024 and use a 1024 sample FFT window, the plugin works correctly - but of course a plugin cannot dictate buffer size, and `1024 is too small a window size to be useful for anything but filtering very high frequencies This seems like it has to be a solvable problem, and most likely the issue is in how my code reports the number of usable samples in the returned buffer. So, what is the correct way for a plugin to report that it has no samples to return, but will, uh, real soon now? I know I could convert this plugin to be one that does offline rendering of the entire input, but this is real-time processing, just with a fixed amount of latency, so that should not be necessary.

I wrote a Swift macOS app to control a PCI audio device. The code switches between the default output and input channels. As soon as I launch the Audio-Midi Setup utility, channel switching stops working. The driver properties allow switching, but the system doesn't respond. I have to delete the contents of /Library/Preferences/Audio and reset Core Audio. What am I missing? func setDefaultChannelsOutput() { guard let deviceID = getDeviceIDByName(deviceName: "PCI-424") else { return } let selectedIndex = DefaultChannelsOutput.indexOfSelectedItem if selectedIndex < 0 || selectedIndex >= 24 { return } let channel1 = UInt32(selectedIndex * 2 + 1) let channel2 = UInt32(selectedIndex * 2 + 2) var channels: [UInt32] = [channel1, channel2] var propertyAddress = AudioObjectPropertyAddress( mSelector: kAudioDevicePropertyPreferredChannelsForStereo, mScope: kAudioDevicePropertyScopeOutput, mElement: kAudioObjectPropertyElementWildcard ) let dataSize = UInt32(MemoryLayout<UInt32>.size * channels.count) let status = AudioObjectSetPropertyData(deviceID, &propertyAddress, 0, nil, dataSize, &channels) if status != noErr { print("Error setting default output channels: \(status)") } }

Session player regions populate blank, with no sound media when tracks or regions are created.

Hi folks - I'm having trouble finding specific documentation about Audio Unit MIDI plugins - as in MIDI -only. Any suggestions welcome as searches aren't returning much. (too niche? user error?)

Currently, I have successfully used ChannelMap to map hardware input channels and obtained audio data from the hardware device's MIC and OTG inputs. Additionally, I have used ChannelMap to map output channels to freely feed data for playback to each output channel. However, I now have a problem. I have a hardware device that only has output channels (no input channels), and the system has set this hardware device as the default playback device. In this case, how can I obtain the audio data being played to the output channels for modification?

I’m developing a macOS audio monitoring app using AVAudioEngine, and I’ve run into a critical issue on macOS 26 beta where AVFoundation fails to detect any input devices, and AVAudioEngine.start() throws the familiar error 10877. FB#: FB19024508 Strange Behavior: AVAudioEngine.inputNode shows no channels or input format on bus 0. AVAudioEngine.start() fails with -10877 (AudioUnit connection error). AVCaptureDevice.DiscoverySession returns zero audio devices. Microphone permission is granted (authorized), and the app is properly signed and sandboxed with com.apple.security.device.audio-input. However, CoreAudio HAL does detect all input/output devices: Using AudioObjectGetPropertyDataSize and AudioObjectGetPropertyData with kAudioHardwarePropertyDevices, I can enumerate 14+ devices, including AirPods, USB DACs, and BlackHole. This suggests the lower-level audio stack is functional. I have tried: Resetting CoreAudio with sudo killall coreaudiod Rebuilding and re-signing the app Clearing TCC with tccutil reset Microphone Running on Apple Silicon and testing Rosetta/native detection via sysctl.proc_translated Using a fallback mechanism that logs device info from HAL and rotates logs for submission via Feedback Assistant I have submitted logs and a reproducible test case via Feedback Assitant : FB#: FB19024508]

I'm encountering numerous crashes involving the com.apple.coreaudio.AQClient thread on our application. The crash details are as follows: #10 com.apple.coreaudio.AQClient SIGSEGV SEGV_ACCERR 0 libobjc.A.dylib _objc_msgSend + 44 1 AudioToolbox ClientMessageHandler::PropertyChanged(unsigned int) + 872 2 AudioToolbox ClientAudioQueue::FetchAndDeliverPendingCallbacks(unsigned int) + 924 3 AudioToolbox __XCallbackNotificationsAvailable + 212 4 libAudioToolboxUtility.dylib _mshMIGPerform + 260 5 CoreFoundation ___CFRUNLOOP_IS_CALLING_OUT_TO_A_SOURCE1_PERFORM_FUNCTION__ + 56 6 CoreFoundation ___CFRunLoopDoSource1 + 596 7 CoreFoundation ___CFRunLoopRun + 2392 8 CoreFoundation _CFRunLoopRunSpecific + 572 9 AudioToolbox CADeprecated::GenericRunLoopThread::Entry(void*) + 156 10 libAudioToolboxUtility.dylib CADeprecated::CAPThread::Entry(CADeprecated::CAPThread*) + 88 11 libsystem_pthread.dylib __pthread_start + 116 All these crashes occur on system versions below iOS/iPadOS 17, primarily when the device's available RAM is low. What steps can I take to resolve this issue? Any insights would be greatly appreciated!

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, 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?

Since iOS 18, the system setting “Allow Audio Playback” (enabled by default) allows third-party app audio to continue playing while the user is recording video with the Camera app. This has created a problem for the app I’m developing. ➡️ The problem: My app plays continuous audio in both foreground and background states. If the user starts recording video using the iOS Camera app, the app’s audio — still playing in the background — gets captured in the video — obviously an unintended behavior. Yes, the user could stop the app manually before starting the video recording, but that can’t be guaranteed. As a developer, I need a way to stop the app’s audio before the video recording begins. So far, I haven’t found a reliable way to detect when video recording starts if ‘Allow Audio Playback’ is ON. ➡️ What I’ve tried: — AVAudioSession.interruptionNotification → doesn’t fire — devicesChangedEventStream → not triggered I don’t want to request mic permission (app doesn’t use mic). also, disabling the app from playing audio in the background isn’t an option as it is a crucial part of the user experience ➡️ What I need: A reliable, supported way to detect when the Camera app begins video recording, without requiring mic access — so I can stop audio and avoid unintentional overlap with the user’s recordings. Any official guidance, workarounds, or AVFoundation techniques would be greatly appreciated. Thanks.

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.