If a Tap or AggregateDevice with the Private property set is created, does it automatically disappear when the process ends? If not, how can I remove the Tap or AggregateDevice before the main process terminates?

Hi, We identified massive amounts of leaked memory with the tvOS 26 standard player user interface as soon as chapters (navigation markers) are involved. Artwork images associated with chapters are not correctly released anymore, leaking memory in chunks of several MiBs. Over time apps will be terminated by the system due to excessive memory consumption. The issue was reported to Apple as tvOS 26 regression: Huge memory leaks associated with navigation marker artworks displayed in the tvOS standard user interface, filed under FB21160665.

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 capturing video stream from GoPro camera (via UDP MPEG-TS packets) but unable to play in iOS app. can someone provide a source code to decode MPEG-TS data to CMSampleBuffer.

I'm working on adding CarPlay support to an audio app and am running into an issue. Occasionally, when a user opens the app from CarPlay while the main app scene is either not connected or is currently in the background, I will receive an error when attempting to activate the audio session. The code below mimics my setup: do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio) try AVAudioSession.sharedInstance().setActive(true) } catch { print(error) // NSOSStatusErrorDomain - 560557684: Session activation failed } That error code maps to AVAudioSession.ErrorCode.cannotInterruptOthers. Once in this state, all subsequent attempts to play different pieces of content will fail. However, things will start working normally if the user opens the app on their phone and tries again from CarPlay (while the app is in the foreground on their phone). I'm not sure why it would behave this way and want to note that I do have the audio background mode capability enabled. Has anyone else encountered this? Are there any workarounds or changes I could make to prevent this from happening?

Fetching the featured artists in a playlist, no longer works in iOS 26.1 beta let detailedPlaylist = try await playlist.with([.tracks, .featuredArtists], preferredSource: .library) Throws error when using .library and using .catalog returns empty array. This works correctly in iOS 26.0 and iOS 18 versions

add a currently playing track endpoint on the apple music api. its kinda wild how apple music goes after spotify without having such a useful endpoint.

Hi there, I recently launched a dj app to the mac app store, and was wondering how I could access raw data to songs from apple music just like how serato, rekordbox, djay, etc. do? Thanks, Gunek

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 have some question about the TTS My device default language is zh-HK. (cantonese) my device is iPhone 16 Pro, IOS 18.6 I create a function speakMandarin I want the device to speak the zh-CN (putonghua) however the device only can speak zh-HK (cantonese). I already set the AVSpeechSynthesisVoice language as zh-CN func speakMandarin(text: String) { print("speakMandarin, \(text)") lastError = nil // Reset error // Stop any ongoing speech before starting new if synthesizer.isSpeaking { synthesizer.stopSpeaking(at: .immediate) } // Configure speech utterance let utterance = AVSpeechUtterance(ssmlRepresentation: text)! utterance.rate = 0.5 // Natural speaking speed utterance.pitchMultiplier = 1.0 utterance.volume = 1.0 utterance.voice = AVSpeechSynthesisVoice(language: "zh-CN") let preferredLanguages = [ "zh-CN" , "zh-TW"] var selectedVoice: AVSpeechSynthesisVoice? for lang in preferredLanguages { if let voice = AVSpeechSynthesisVoice(language: lang) { print(lang) selectedVoice = voice utterance.voice = voice break } } // If no Mandarin voice found, use system default if selectedVoice == nil { selectedVoice = AVSpeechSynthesisVoice(language: nil) lastError = "未偵測到普通話語音包，將使用系統預設語音" print (lastError) } utterance.voice = selectedVoice print(utterance) synthesizer.speak(utterance) } here is my log speakMandarin, <speak>你好！我們來聊聊喜歡的動物吧。你喜歡什麼動物呢？</speak> zh-CN [AVSpeechUtterance 0x1194efb80] String: 你好！我們來聊聊喜歡的動物吧。你喜歡什麼動物呢？ Voice: [AVSpeechSynthesisVoice 0x104ceff90] Language: zh-CN, Name: Tingting, Quality: Default [com.apple.voice.compact.zh-CN.Tingting] Rate: 0.50 Volume: 1.00 Pitch Multiplier: 1.00 Delays: Pre: 0.00(s) Post: 0.00(s)

Hi there, I recently launched a dj app to the mac app store, and was wondering how I could access songs for mixing purposes via Apple Music just like how serato, rekordbox, djay, and other DJ apps do? Thanks, Gunek

Hello, I'm trying to write a shortcut using Toolbox Pro that gets triggered by an accessibility trigger and then favorites the currently playing song. It's working pretty well, but I noticed that for some artists, especially asian ones, it simply doesn't work. While debugging, I noticed that the tool uses the same song ID, artist ID, everything as it should to search for the song and favorite it. However, I noticed that Apple Music treats artists with romanized names as two separate artists! https://music.apple.com/br/artist/王菲/41760704 https://music.apple.com/br/artist/faye-wong/41760704?l=en-GB You can see that the ID is the same (41760704). It seems that, when I search for the artist, the first artist (王菲) returns, so that when I open URLs on the web for the artist I can see a star next to the song name, meaning that it got a like. However, the romanized artist (faye-wong) doesn't have a like on the same song. This is very weird, right?

AVPictureInPictureControllerContentSource *contentSource = [[AVPictureInPictureControllerContentSource alloc] initWithSampleBufferDisplayLayer:self.renderView.sampleBufferDisplayLayer playbackDelegate:self]; AVPictureInPictureController *pictureInPictureController = [[AVPictureInPictureController alloc] initWithContentSource:contentSource]; pictureInPictureController.delegate = self; (void)pictureInPictureController:(AVPictureInPictureController *)pictureInPictureController failedToStartPictureInPictureWithError:(NSError *)error { //error NSError * domain: @"PGPegasusErrorDomain" - code: -1003 0x00000002819fe3a0 } when first start the PiP play, I got the error "//error NSError * domain: @"PGPegasusErrorDomain" - code: -1003 0x00000002819fe3a0", why? and second start is Ok.

This is my native module code implementation I'm getting base64 encoded string from server and passing this to my native module of pcm player to play audio App.tsx PcmPlayer.writeChunk(e.data); PcmPlayer.swift import AVFoundation @objc(PcmPlayer) class PcmPlayer: RCTEventEmitter { private var engine: AVAudioEngine? private var playerNode: AVAudioPlayerNode? private var format: AVAudioFormat? private var bufferQueue = [Data]() private var isPlaying = false private var hasEnded = false private var scheduledBufferCount = 0 private let minBufferBytes = 50000 private let pcmQueue = DispatchQueue(label: "pcm.queue") override init() { super.init() } override func supportedEvents() -> [String]! { return ["onStatus", "onMessage"] } @objc(initPlayer:channels:bitsPerSample:) func initPlayer(_ sampleRate: NSNumber, channels: NSNumber, bitsPerSample: NSNumber) { pcmQueue.async { self.stopInternal() let session = AVAudioSession.sharedInstance() do { try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true, options: .notifyOthersOnDeactivation) try session.setMode(.default) print("🔈 Audio session active. Output route:", session.currentRoute.outputs) } catch { print("❌ Audio session setup failed:", error) return } self.engine = AVAudioEngine() self.playerNode = AVAudioPlayerNode() guard let engine = self.engine, let playerNode = self.playerNode else { print("❌ Engine or playerNode is nil") return } engine.attach(playerNode) self.format = AVAudioFormat(commonFormat: .pcmFormatFloat32, sampleRate: sampleRate.doubleValue, channels: AVAudioChannelCount(channels.uintValue), interleaved: false) guard let format = self.format else { print("❌ Failed to create AVAudioFormat") return } engine.connect(playerNode, to: engine.mainMixerNode, format: format) do { try engine.start() playerNode.play() engine.mainMixerNode.outputVolume = 1.0 print("✅ AVAudioEngine started with format:", format) } catch { print("❌ Engine start failed:", error) } self.hasEnded = false } } @objc(writeChunk:) func writeChunk(_ base64Pcm: String) { pcmQueue.async { guard base64Pcm.count >= 10 else { print("⚠️ Skipping short base64 string") return } var padded = base64Pcm let mod4 = base64Pcm.count % 4 if mod4 > 0 { padded += String(repeating: "=", count: 4 - mod4) } guard let data = Data(base64Encoded: padded, options: .ignoreUnknownCharacters) else { print("❌ Failed to decode base64") return } self.bufferQueue.append(data) print("📥 Received PCM chunk (\(data.count) bytes)") print("📥 writeChunk called. isPlaying=\(self.isPlaying), bufferQueue.count=\(self.bufferQueue.count)") if !self.isPlaying { self.isPlaying = true self.waitForBufferAndStartPlayback() } else if self.scheduledBufferCount == 0 { self.isPlaying = true self.waitForBufferAndStartPlayback() } } } private func waitForBufferAndStartPlayback() { DispatchQueue.global().async { while self.queueSize() < self.minBufferBytes && !self.hasEnded { Thread.sleep(forTimeInterval: 0.01) } self.writeLoop() } } private func writeLoop() { DispatchQueue.global().async { writeLoop: while self.isPlaying { if self.bufferQueue.isEmpty { for _ in 0..<100 { Thread.sleep(forTimeInterval: 0.01) if !self.bufferQueue.isEmpty { break } } if self.bufferQueue.isEmpty { print("🔇 No more data to play after waiting") self.isPlaying = false break writeLoop } } var data: Data? self.pcmQueue.sync { if !self.bufferQueue.isEmpty { data = self.bufferQueue.removeFirst() } } guard let chunk = data else { print("⚠️ No data to process") continue } if let buffer = self.pcmBufferFromData(chunk) { self.scheduledBufferCount += 1 self.playerNode?.scheduleBuffer(buffer, completionHandler: { self.pcmQueue.async { self.scheduledBufferCount -= 1 if self.bufferQueue.isEmpty && self.scheduledBufferCount == 0 { print("ℹ️ Playback idle - waiting for more data") self.isPlaying = false } } }) } } } } private func pcmBufferFromData(_ data: Data) -> AVAudioPCMBuffer? { guard let format = self.format else { return nil } let frameCount = UInt32(data.count / 2) guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: frameCount) else { print("❌ Failed to create AVAudioPCMBuffer") return nil } buffer.frameLength = frameCount guard let floatChannelData = buffer.floatChannelData?[0] else { print("❌ floatChannelData is nil") return nil } data.withUnsafeBytes { (rawBuffer: UnsafeRawBufferPointer) in let int16Buffer = rawBuffer.bindMemory(to: Int16.self) let count = min(int16Buffer.count, Int(frameCount)) for i in 0..<count { floatChannelData[i] = Float32(int16Buffer[i]) / Float32(Int16.max) } } return buffer } @objc(stopPlayer) func stopPlayer() { pcmQueue.async { self.stopInternal() } } private func stopInternal() { print("🛑 stopInternal called") self.playerNode?.stop() self.engine?.stop() self.engine?.reset() self.playerNode = nil self.engine = nil self.format = nil self.bufferQueue.removeAll() self.isPlaying = false self.hasEnded = true self.scheduledBufferCount = 0 } @objc(canWrite:rejecter:) func canWrite(_ resolve: @escaping RCTPromiseResolveBlock, rejecter reject: RCTPromiseRejectBlock) { pcmQueue.async { resolve(self.bufferQueue.count < 20) } } @objc(flushPlayer:rejecter:) func flushPlayer(_ resolve: @escaping RCTPromiseResolveBlock, rejecter reject: RCTPromiseRejectBlock) { pcmQueue.async { self.bufferQueue.removeAll() resolve(nil) } } @objc static override func requiresMainQueueSetup() -> Bool { return false } private func queueSize() -> Int { return pcmQueue.sync { return self.bufferQueue.reduce(0) { $0 + $1.count } } } } I couldn't able to hear any audio via my real iOS device also it is working fine on emulator.

Hey Swift community! I'm exploring building a macOS app that needs to monitor what's currently playing in music apps like Spotify and Apple Music (track info, playback position, play/pause state). I'm trying to figure out the most efficient architecture before diving in. The Goal: Monitor playback state across multiple music players to react to changes in real-time, ideally with minimal CPU overhead since this would run continuously in the background. Approaches I'm Considering AppleScript / ScriptingBridge Distributed Notifications Native Frameworks (Apple Music only) What's the recommended way to do this on macOS? Are distributed notifications reliable enough to avoid polling entirely? Is there a performance difference between AppleScript and ScriptingBridge for IPC? For Apple Music specifically, should I use MusicKit, MediaPlayer, or stick with AppleScript? Are there other approaches I'm missing?

It's been well over a year since Apple added favoriting of artists back to Apple Music (the little star icon on an artist page), but yet I still haven't seen a way to get this data from an authenticated user from Music API. I was expecting to hear something about this during the WWDC, but there have been no announcements that I've caught. Has anyone else heard anything? People assume when they provide access to their Apple Music account that we can actually get to the data in their Apple Music account, and we end up looking a little dumb not being able to get this core data.

Hi, I'm sending an API request to: https://api.music.apple.com/v1/me/library/playlists?limit=$limit&offset=$offset To list all of the users library playlists, however the resulting objects do not contain the playlist artwork in the JSON. I've tried adding the extend and include attributes as well but to no avail. A partial example of the response: {"id": PLAYLIST_ID, "type": "library-playlists", "href": "/v1/me/library/playlists/PLAYLIST_ID", "attributes": {"lastModifiedDate": "2024-09-18T20:18:24Z", "canEdit": true, "name": "Afro Party Anthems", "isPublic": false, "description": {"standard": "Definitive African party starters"}, "hasCatalog": false, "dateAdded": "2022-03-10T18:30:56Z", "playParams": {"id": PLAYLIST_ID, "kind": "playlist", "isLibrary": true}}, "relationships": {"catalog": {"href": "/v1/me/library/playlists/PLAYLIST_ID/catalog", "data": []}}} Is there a way to get the artwork URL without sending a request for each playlist? And if not can this be fixed?

I'm reaching out regarding a recurring issue I'm experiencing with MusicKit developer tokens. I'm using a valid .p8 private key to sign JWTs for Apple MusicKit integration. Each token I generate includes the appropriate claims (iss, iat, exp) and is signed with the ES256 algorithm, with an expiration date set approximately 6 months ahead. Everything works as expected immediately after generating the token. However, after a few days, the same JWT (still well within its expiration period) suddenly begins returning invalid/unauthorized responses when used in Postman and other API clients. Importantly: I did not delete or revoke the .p8 key during this time. I verified the JWT contains valid claims and a proper structure. The issue consistently resolves only when I create a new .p8 file and regenerate a fresh JWT with it—after which the cycle repeats. This issue occurs even when the environment and app identifiers remain unchanged. I would greatly appreciate it if you could help me understand: Why these tokens become invalid after a few days, despite having a long exp value and an unchanged key. Whether there's any automatic revocation or timeout policy on .p8 keys that could explain this behavior. If there's a better way to maintain long-lived developer tokens without requiring new .p8 key generation every few days. Thank you for your help and clarification on this issue. Best regards, Liad Altif

I’m building a teleprompter-style app that relies on Picture in Picture. PiP starts correctly on device. Everything works — until another app (e.g. TikTok / Instagram) starts active video recording. When camera capture begins in the foreground app, iOS terminates my PiP session. Some teleprompter apps appear to keep PiP active while recording in other apps, so I’m trying to understand the recommended architectural pattern for this scenario. Is there a documented approach or best practice to keep PiP stable during third-party camera capture? Looking specifically for guidance on the correct AVKit / AVAudioSession configuration for this use case.

Hello, I've discovered a buffer initialization bug in AVAudioUnitSampler that happens when loading presets with multiple zones referencing different regions in the same audio file (monolith/concatenated samples approach). Almost all zones output silence (i.e. zeros) at the beginning of playback instead of starting with actual audio data. The Problem Setup: Single audio file (monolith) containing multiple concatenated samples Multiple zones in an .aupreset, each with different sample start and sample end values pointing to different regions of the same file All zones load successfully without errors Expected Behavior: All zones should play their respective audio regions immediately from the first sample. Actual Behavior: Last zone in the zone list: Works perfectly - plays audio immediately All other zones: Output [0, 0, 0, 0, ..., _audio_data] instead of [real_audio_data] The number of zeros varies from event to event for each zone. It can be a couple of samples (<30) up to several buffers. After the initial zeros, the correct audio plays normally, so there is no shift in audio playback, just missing samples at the beginning. Minimal Reproduction 1. Create Test Monolith Audio File Create a single Wav file with 3 concatenated 1-second samples (44.1kHz): Sample 1: frames 0-44099 (constant amplitude 0.3) Sample 2: frames 44100-88199 (constant amplitude 0.6) Sample 3: frames 88200-132299 (constant amplitude 0.9) 2. Create Test Preset Create an .aupreset with 3 zones all referencing the same file: Pseudo code <Zone array> <zone 1> start : 0, end: 44099, note: 60, waveform: ref_to_monolith.wav; <zone 2> start sample: 44100, note: 62, end sample: 88199, waveform: ref_to_monolith.wav; <zone 3> start sample: 88200, note: 64, end sample: 132299, waveform: ref_to_monolith.wav; </Zone array> 3. Load and Test // Load preset into AVAudioUnitSampler let sampler = AVAudioUnitSampler() try sampler.loadAudioFiles(from: presetURL) // Play each zone (MIDI notes C4=60, D4=62, E4=64) sampler.startNote(60, withVelocity: 64, onChannel: 0) // Zone 1 sampler.startNote(62, withVelocity: 64, onChannel: 0) // Zone 2 sampler.startNote(64, withVelocity: 64, onChannel: 0) // Zone 3 4. Observed Result Zone 1 (C4): [0, 0, 0, ..., 0.3, 0.3, 0.3] ❌ Zeros at beginning Zone 2 (D4): [0, 0, 0, ..., 0.6, 0.6, 0.6] ❌ Zeros at beginning Zone 3 (E4): [0.9, 0.9, 0.9, ...] ✅ Works correctly (last zone) What I've Extensively Tested What DOES Work Separate files per zone: Each zone references its own individual audio file All zones play correctly without zeros Problem: Not viable for iOS apps with 500+ sample libraries due to file handle limitations What DOESN'T Work (All Tested) 1. Different Audio Formats: CAF (Float32 PCM, Int16 PCM, both interleaved and non-interleaved) M4A (AAC compressed) WAV (uncompressed) SF2 (SoundFont2) Bug persists across all formats 2. CAF Region Chunks: Created CAF files with embedded region chunks defining zone boundaries Set zones with no sampleStart/sampleEnd in preset (nil values) AVAudioUnitSampler completely ignores CAF region metadata Bug persists 3. Unique Waveform IDs: Gave each zone a unique waveform ID (268435456, 268435457, 268435458) Each ID has its own file reference entry (all pointing to same physical file) Hypothesized this might trigger separate buffer initialization Bug persists - no improvement 4. Different Sample Rates: Tested: 44.1kHz, 48kHz, 96kHz Bug occurs at all sample rates 5. Mono vs Stereo: Bug occurs with both mono and stereo files Environment macOS: Sonoma 14.x (tested across multiple minor versions) iOS: Tested on iOS 17.x with same results Xcode: 16.x Frameworks: AVFoundation, AudioToolbox Reproducibility: 100% reproducible with setup described above Impact & Use Case This bug severely impacts professional music applications that need: Small file sizes: Monolith files allow sharing compressed audio data (AAC/M4A) iOS file handle limits: Opening 400+ individual sample files is not viable on iOS Performance: Single file loading is much faster than hundreds of individual files Standard industry practice: Monolith/concatenated samples are used by EXS24, Kontakt, and most professional samplers Current Impact: Cannot use monolith files with AVAudioUnitSampler on iOS Forced to choose between: unusable audio (zeros at start) OR hitting iOS file limits No viable workaround exists Root Cause Hypothesis The bug appears to be in AVAudioUnitSampler's internal buffer initialization when: Multiple zones share the same source audio file Each zone specifies different sampleStart/sampleEnd offsets Key observation: The last zone in the zone array always works correctly. This is NOT related to: File permissions or security-scoped resources (separate files work fine) Audio codec issues (happens with uncompressed PCM too) Preset parsing (preset loads correctly, all zones are valid) Questions Is this a known issue? I couldn't find any documentation, bug reports, or discussions about this. Is there ANY workaround that allows monolith files to work with AVAudioUnitSampler? Alternative APIs? Is there a different API or approach for iOS that properly supports monolith sample files?