New to iOS development and I've been trying to make heads or tails of the documentation. I know there is a difference between the data fields returned from songs from the user library and from the category, but whenever I search on the apple site I can't find a list of each. For example, Im trying to get the releaseDate of a song in my library, but it seems I'll have to cross-query either the catalog entry for the using song.catalogID or the song.irsc but when I try to use them I can't find a cross reference between the two. I'm totally turned around. Also trying to determine if a song in my library has been favorited or not? isFavorited (or something similar) doesn't seem to be a thing. Using this code and trying to find a way to display a solid star if the song has been favorited or an empty one if it's not. Seems like a basic request but I can't find anything on how to do it. I've searched docs, googled, tried. Does apple want us to query the user's Favorited Songs playlist or something? How do I know which playlist that is? I know isFavorited isn't a thing, just using it here so you can see what my intension is: HStack(spacing: 10) { Image(systemName: song.isFavorited ? "star.fill" : "star") .foregroundColor(song.isFavorited ? .yellow : .gray) Image(systemName: "magnifyingglass") }

I work on an iOS app that records video and audio. We've been getting reports for a while from users who are experiencing their video recordings being cut off. After investigating, I found that many users are receiving the AVAudioSessionMediaServicesWereResetNotification (.mediaServicesWereResetNotification) notification while recording. It's associated with the AVFoundationErrorDomain[-11819] error, which seems to indicate that the system audio daemon crashed. We have a handler registered to end the recording, show the user a prompt, and restart our AV sessions. However, from our logs this looks to be happening to hundreds of users every day and it's not an ideal user experience, so I would like to figure out why this is happening and if it's due to something that we're doing wrong. The debug menu option to trigger the audio session reset is not of much use, because it can't be triggered unless you leave the app and go to system settings. So our app can't be recording video when the debug reset is triggered. So far I haven't found a way to reproduced the issue locally, but I can see that it's happening to users from logs. I've found some posts online from developers experiencing similar issues, but none of them seem to directly address our issue. The system error doesn't include a userInfo dictionary, and as far as I can tell it's a system daemon crash so any logs would need to be captured from the OS. Is there any way that I could get more information about what may be causing this error that I may have missed?

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?

Is there any feasible way to get a Core Audio device's system effect status (Voice Isolation, Wide Spectrum)? AVCaptureDevice provides convenience properties for system effects for video devices. I need to get this status for Core Audio input devices.

I have both apple devices, AirPods Pro 3 is up to date and Ultra 3 is on watch os 26.1 latest public beta. Each morning when I would go on my mindfulness app and start a meditation or listen to Apple Music on my watch and AirPods Pro 3, it will play for a few seconds then disconnects. My bluetooth settings on my watch says my AirPods is connected to my watch. I also have removed the tick about connecting automatically to iPhone on the AirPods setting in my iPhone. To fix this I invariably turn off my Apple Watch Ultra 3 and turn it on again. Then the connection becomes stable. I am not sure why I have to do this each morning. It is frustrating. I am not sure why this fix does not last long? Is there something wrong with my AirPods? Has anyone encountered this before?

Hi everyone, I wanted to bring up a question about Core Audio and its potential for future updates or improvements, specifically regarding latency optimization. As someone who relies on Core Audio for real-time audio processing, any enhancements in this area would be incredibly beneficial for professionals in the industry. Does anyone know if Apple has shared any plans or updates regarding Core Audio’s performance, particularly for low-latency applications? I’d appreciate any insights or advice from the community! Thanks so much! Best, Michael

Hello All, It seems that it's "very easy" (😬) to implement a little Swift code inside the prepared AU using Xcode 16.2 on Sequoia 15.1.1 and a Mac Studio M1 Ultra, but my issue is that I finally don't know... where. The documentation says that I've to find the AudioUnitViewController.swift file and then modify the render block : audioUnit.renderBlock = { (numFrames, ioData) in // Process audio here } in the Xcode project automatically generated, but I didn't find such a file... If somebody can help me in showing where is the file to be modified, I'll be very grateful ! Thank you very much. J

I've got a problem with my app where I'm testing it on my own phone. I'm using audio kit to generate tones as part of the app. Everything seems to work fine. Sounds start, Stop, etc. They play when the app is closed and when the phone is locked, so background is working. However, I'm seeing an issue where, even when STOP is pressed and the application exited, if I get a notification such as a text message, the base tone for the app starts to play. If I then open the app, check the Start/Stop button - it says start so that. hasnt' been activated. If I click Start, then a 2nd tone starts. This one stops with the Stop button. However the original tone that was set off by an incoming message carries on playing. Until I go to the Open Apps View on the phone and slide the application upwards. For the life of me, I can't figure out whats happening here.

Hello! We stumbled upon a problem with our karaoke app where user on iPhone 16e/iOS 18.5 has problem with mic capture, other users cannot hear him. The mic capture is working fine on 17.5, 16.8. Maybe there is something else we need when configuring AVAudioSession for iOS 18.5? Currently it's set up like this: override func viewDidLoad() { super.viewDidLoad() UIApplication.shared.isIdleTimerDisabled = true mRoomId = appDelegate.getRoomId() let audioSession = AVAudioSession.sharedInstance() try! audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: [.defaultToSpeaker]) try! audioSession.setPreferredSampleRate(48000) try! audioSession.setActive(true, options: []) }

Two issues: No matter what I set in try audioSession.setPreferredSampleRate(x) the sample rate on both iOS and macOS is always 48000 when the output goes through the speaker, and 24000 when my Airpods connect to an iPhone/iPad. Now, I'm checking the current output loudness to animate a 3D character, using mixerNode.installTap(onBus: 0, bufferSize: y, format: nil) { [weak self] buffer, time in Task { @MainActor in // calculate rms and animate character accordingly but any buffer size under 4800 is just ignored and the buffers I get are 4800 sized. This is ok, when the sampleRate is currently 48000, as 10 samples per second lead to decent visual results. But when AirPods connect, the samplerate is 24000, which means only 5 samples per second, so the character animation looks lame. My AVAudioEngine setup is the following: audioEngine.connect(playerNode, to: pitchShiftEffect, format: format) audioEngine.connect(pitchShiftEffect, to: mixerNode, format: format) audioEngine.connect(mixerNode, to: audioEngine.outputNode, format: nil) Now, I'd be fine if the outputNode runs at whatever if it needs, as long as my tap would get at least 10 samples per second. PS: Specifying my favorite format in the let format = AVAudioFormat(standardFormatWithSampleRate: 48_000, channels: 2)! mixerNode.installTap(onBus: 0, bufferSize: y, format: format) doesn't change anything either

We are developing an apple music app on phone, the developed web works fine on chrome, but when i load it on webivew on my phone, i can't play the first song, We doubt that the drm init, key exchange, session creation was on the music.play() function, while we trigger the play, the drm or session was not ok for play a real song, so it got an error so we may wanna know: what about the realative process of drm, key, session, etc in the play() function? are there some state detect function to show weather the drm is ok？

Hi I'm new to the forum, I'm planning an app just for Apple watch, I would like to use bluetooth audio in background, how can I do it? The messages I send via bluetooth stop as soon as the watch display turns off. Thank you! Nax

I have a simple AVAudioEngine graph as follows: AVAudioPlayerNode -> AVAudioUnitEQ -> AVAudioUnitTimePitch -> AVAudioUnitReverb -> Main mixer node of AVAudioEngine. I noticed that whenever I have AVAudioUnitTimePitch or AVAudioUnitVarispeed in the graph, I noticed a very distinct crackling/popping sound in my Airpods Pro 2 when starting up the engine and playing the AVAudioPlayerNode and unable to find the reason why this is happening. When I remove the node, the crackling completely goes away. How do I fix this problem since i need the user to be able to control the pitch and rate of the audio during playback. import AVKit @Observable @MainActor class AudioEngineManager { nonisolated private let engine = AVAudioEngine() private let playerNode = AVAudioPlayerNode() private let reverb = AVAudioUnitReverb() private let pitch = AVAudioUnitTimePitch() private let eq = AVAudioUnitEQ(numberOfBands: 10) private var audioFile: AVAudioFile? private var fadePlayPauseTask: Task<Void, Error>? private var playPauseCurrentFadeTime: Double = 0 init() { setupAudioEngine() } private func setupAudioEngine() { guard let url = Bundle.main.url(forResource: "Song name goes here", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: url) } catch { print("Failed to load audio file: \(error)") return } reverb.loadFactoryPreset(.mediumHall) reverb.wetDryMix = 50 pitch.pitch = 0 // Increase pitch by 500 cents (5 semitones) engine.attach(playerNode) engine.attach(pitch) engine.attach(reverb) engine.attach(eq) // Connect: player -> pitch -> reverb -> output engine.connect(playerNode, to: eq, format: audioFile?.processingFormat) engine.connect(eq, to: pitch, format: audioFile?.processingFormat) engine.connect(pitch, to: reverb, format: audioFile?.processingFormat) engine.connect(reverb, to: engine.mainMixerNode, format: audioFile?.processingFormat) } func prepare() { guard let audioFile else { return } playerNode.scheduleFile(audioFile, at: nil) } func play() { DispatchQueue.global().async { [weak self] in guard let self else { return } engine.prepare() try? engine.start() DispatchQueue.main.async { [weak self] in guard let self else { return } playerNode.play() fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: true) // Ramp up volume until 1 is reached if volume >= 1 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 1 } } } } func pause() { fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: false) // Ramp down volume until 0 is reached if volume <= 0 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 0 playerNode.pause() // Shut down engine once ramp down completes DispatchQueue.global().async { [weak self] in guard let self else { return } engine.pause() } } } private func updateVolume(for x: Double, rising: Bool) -> Float { if rising { // Fade in return Float(pow(x, 2) * (3.0 - 2.0 * (x))) } else { // Fade out return Float(1 - (pow(x, 2) * (3.0 - 2.0 * (x)))) } } func setPitch(_ value: Float) { pitch.pitch = value } func setReverbMix(_ value: Float) { reverb.wetDryMix = value } } struct ContentView: View { @State private var audioManager = AudioEngineManager() @State private var pitch: Float = 0 @State private var reverb: Float = 0 var body: some View { VStack(spacing: 20) { Text("🎵 Audio Player with Reverb & Pitch") .font(.title2) HStack { Button("Prepare") { audioManager.prepare() } Button("Play") { audioManager.play() } .padding() .background(Color.green) .foregroundColor(.white) .cornerRadius(10) Button("Pause") { audioManager.pause() } .padding() .background(Color.red) .foregroundColor(.white) .cornerRadius(10) } VStack { Text("Pitch: \(Int(pitch)) cents") Slider(value: $pitch, in: -2400...2400, step: 100) { _ in audioManager.setPitch(pitch) } } VStack { Text("Reverb Mix: \(Int(reverb))%") Slider(value: $reverb, in: 0...100, step: 1) { _ in audioManager.setReverbMix(reverb) } } } .padding() } }

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?

Hi everyone, I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms. Problem: When the app is recording audio and an interruption occurs: I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began). On .ended, I check for .shouldResume and call audioRecorder?.record() again. The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder. Repro: Start a recording with AVAudioRecorder Simulate a system interruption (e.g., incoming call) Resume recording after the interruption Stop and inspect the output audio file Expected: Full audio (before and after interruption) should be saved. Actual: Only the audio after interruption is saved; the earlier part is missing Notes: According to the documentation, calling .record() after .pause() should resume recording into the same file. I confirmed that the file URL does not change, and I do not recreate the recorder instance. No error is thrown by the system during this process. This behavior happens consistently when the app is interrupted and resumed. Question: Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen? Thanks in advance!

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?

Songs can be unavailable (greyed out) in Apple Music. How can I check if a song is unavailable via the MusicKit framework? Obviously the playback will fail with MPMusicPlayerControllerErrorDomain Code=6 "Failed to prepare to play" but how can I know that in advance? I need to check the availability of hundreds of albums and therefore initiating a playback for each of them is not an option. Things I have tried: Checking if the release date property is set to a future date. This filters out all future releases but doesn't solve the problem for already released songs. Checking if the duration is 0. This does not work since the duration of unavailable songs does not have to be 0. Initiating a playback and checking for the "Failed to prepare to play" error. This is not suitable for a huge amount of Albums. I couldn't find a solution yet but somehow other third-party-apps are able ignore/don't shows these albums. I believe the Apple Music app is only displaying albums where at least one song is available. I am using this function to fetch all albums of an artist. private func fetchAlbumsFor(_ artist: Artist) async throws -> [Album] { let artistWithAlbums = try await artist.with(.albums) var allAlbums = [Album]() guard var currentBadge = artistWithAlbums.albums else { return [] } allAlbums.append(contentsOf: currentBadge) while currentBadge.hasNextBatch { if let nextBatch = try await currentBadge.nextBatch() { currentBadge = nextBatch allAlbums.append(contentsOf: nextBatch) } else { break } } return allAlbums } Here is an example album where I am unable to detect its unavailability (at least in Germany): https://music.apple.com/de/album/die-haferhorde-immer-den-n%C3%BCstern-nach-h%C3%B6rspiel-zu-band-3/1755774804 Furthermore I was unable to navigate to this album via the Apple Music app directly. Thanks for any help Edit: Apparently this album is not included in an apple music subscription but can be bought seperately. The question remains: How can I check that?

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

private var audioEngine = AVAudioEngine() private var inputNode: AVAudioInputNode! func startAnalyzing() { inputNode = audioEngine.inputNode let recordingFormat = inputNode.outputFormat(forBus: 0) let hardwareSampleRate = recordingSession.sampleRate inputNode.removeTap(onBus: 0) if recordingFormat.sampleRate != hardwareSampleRate { print("。") let newFormat = AVAudioFormat(commonFormat: recordingFormat.commonFormat, sampleRate: hardwareSampleRate, channels: recordingFormat.channelCount, interleaved: recordingFormat.isInterleaved) inputNode.installTap(onBus: 0, bufferSize: 1024, format: newFormat) { buffer, time in self.processAudioBuffer(buffer, time: time) } } else { inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { buffer, time in self.processAudioBuffer(buffer, time: time) } } do { audioEngine.prepare() try audioEngine.start() } catch { print(": \(error)") } } I back the app to the background and then call startAnalyzing(), which reports an error and the background recording permissions are configured。 error： [10429:570139] [aurioc] AURemoteIO.cpp:1668 AUIOClient_StartIO failed (561145187) [10429:570139] [avae] AVAEInternal.h:109 [AVAudioEngineGraph.mm:1545:Start: (err = PerformCommand(*ioNode, kAUStartIO, NULL, 0)): error 561145187 Audio engine couldn't start. Is background boot not allowed？

The sysEx struct in the MIDIUniversalMessage struct has a channel member but the System Exclusive (7-Bit) Message doesn't have a channel field. The System Exclusive (7-Bit) Message has a # of bytes field but the sysEx struct doesn't have a nrOfBytes, byteCount or bytesUsed member. It looks like the channel member of the sysEx struct contains the number of used bytes. Is this a mistake in the header or did I misunderstand something?