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() } } }

Hi, when a CoreMIDI driver controls physical HW it is probably quite commune to have to control the amount of MIDI data received from the system. What comes to mind is to just delay returning control of the MIDIDriverInterface::Send() callback to the calling process. While the application trying to send MIDI really stalls until the callback returns it seems only to be a side effect of a generally stalled CoreMIDI server. Between the callbacks the application can send as much MIDI data as it wants to CoreMIDI, it's buffering seems to be endless... However the HW might not be able to play out all the data. It seems there is no way to indicate an overflow/full buffer situation back the application/CoreMIDI. How is this supposed to work? Thanks, any hints or pointers are highly appreciated! Hagen.

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 need to read wether the transport is playing or stopped but my current method that works for vst does not work for au. is there a lpx resource available for developers anywhere? if (auto* playHead = processor->getPlayHead()) { juce::AudioPlayHead::CurrentPositionInfo posInfo; if (playHead->getCurrentPosition(posInfo)) { bool isCurrentlyPlaying = posInfo.isPlaying; if (isCurrentlyPlaying != wasTransportPlaying) { if (isCurrentlyPlaying) { wasTransportPlaying = isCurrentlyPlaying; startAllTimers(); } else { wasTransportPlaying = isCurrentlyPlaying; stopAllTimers(); } } } } thanks :)

My app Balletrax is a music player for people to use while they teach ballet. Used to be you could silence notifications during use, but now the customer seems to have to know how to use Focus mode, remember to turn it on and off, and have to check the notifications one does and doesn't want to use. Is there no way to silence all notifications when the app is in use?

Hi, I'm still stuck getting a basic record-with-playthrouh pipeline to work. Has anyone a sample of setting up a AVAudioEngine pipeline for recording with playthrough? Plkaythrough works with AVPlayerNode as input but not with any microphone input. The docs mention the "enabled state" of the outputNode of the engine without explaining the concept, i.e. how to enable an output. When the engine renders to and from an audio device, the AVAudioSession category and the availability of hardware determines whether an app performs output. Check the output node’s output format (specifically, the hardware format) for a nonzero sample rate and channel count to see if output is in an enabled state. Well, in my setup the output is NOT enabled, and any attempt to switch (e.g. audioEngine.outputNode.auAudioUnit.setDeviceID(deviceID) )/ attach a dedicated device / ... results in exceptions / errors

I prefer to use the album fetched from the library instead of the catalog since this is faster. If doing so, how can I check if all tracks of an album are added to the library. In this case I'd like to fetch the catalog version or throw an error (for example when offline). Using .with(.tracks) on the library album fetches the tracks added to the library. The trackCount property is referring to the tracks that can be fetched from the library. The isComplete property is always nil when fetching from the library. One possible way is checking the trackNumber and discCount properties. However this only detects that not all tracks of an album are added to the library if there is a song not added ahead of one that is. I'd like to be able to handle this edge case as well. Is there currently a way to do this? I'd prefer to not rely on the apple music catalog for this since this is supposed to work offline as well. Fetching and storing all trackIDs when connected and later comparing against these would work, but this would potentially mean storing tens of thousands of track ids. Thank you

A bit of a novice to app development here but I have a paid developer account, I have registered the identifier for MusicKit on the developer website (using the bundle identifier I've selected in Xcode) but the option to add MusicKit as a capability is not available in Xcode? I've manually updated the certificates, closed the app and reopened it, started a new project and tried with a different demo project? Apologies if I am missing something obvious but could someone help me get this capability added?

Hi, I’m an iOS developer building an app with an use case that needs advanced playback on Apple Music subscription streams, specifically: • Real-time tempo change (BPM) during playback — i.e., time-stretch with key-lock, not just crossfade. • Beat-matched transitions between tracks. From what I can tell, this capability seems to exist only for approved partners and isn’t available through public MusicKit. Question: What’s the official request path to be evaluated for that restricted partner entitlement (application form, questionnaire, NDA, or internal team/BD contact)? If the entitlement identifier is internal, how can I get my account routed to the right Apple Music team? For reference, publicly announced partners include Algoriddim djay, Serato DJ Pro, rekordbox (AlphaTheta), and Engine DJ—all of which appear to implement mixing features that imply advanced playback (tempo/beat-matching) on Apple Music content. I’d prefer not to share product details publicly for the moment and can provide specifics privately if needed. Thanks in advance!

My app utilizes background audio to play music files. I have the audio background mode enabled and I initialize the AVAudioSession in playback mode with the mixWithOthers option. And it usually works great while the app is backgrounded. I listen for audio interruptions as well as route changes and I am able to handle them appropriately and I can usually resume my background audio no problem. I discovered an issue while connected to CarPlay though. Roughly 50% of the time when I disconnect from a phone call while connected to CarPlay I get the following error after calling the play() method of my AVAudioPlayer instance: "ATAudioSessionClientImpl.mm:281 activation failed. status = 561015905" If I instead try to start a new audio session I get a similar error: Error Domain=NSOSStatusErrorDomain Code=561015905 "Session activation failed" UserInfo={NSLocalizedDescription=Session activation failed} Like I said, this isn't reproducible 100% of the time and is so far only seen while connected to CarPlay. I don't think Im forgetting so additional capability or plist setting, but if anyone has any clues it would be greatly appreciated. Otherwise this is likely just a bug that I need to report to Apple. One very important note, and reason I believe it's just a bug, is that while I was testing I found that other music apps like Spotify will also fail to resume their audio at the same time my app fails. Another important detail is that when it works successfully I receive the audio session interruption ended notification, and when it doesn't work I only receive a route configuration change or route override notification. From there I am able to still successfully granted background time to execute code, but my call to resume audio fails with the above mentioned error codes.

When multiple identical songs are added to a playlist, Playlist.Entry.id uses a suffix-based identifier (e.g. songID_0, songID_1, etc.). Removing one entry causes others to shift, changing their .id values. This leads to diffing errors and collection view crashes in SwiftUI or UIKit when entries are updated. Steps to Reproduce: Add the same song to a playlist multiple times. Observe .id.rawValue of entries (e.g. i.SONGID_0, i.SONGID_1). Remove one entry. Fetch playlist again — note the other IDs have shifted. FB18879062

Hi, I'm trying to setup a AVAudioEngine for USB Audio recording and monitoring playthrough. As soon as I try to setup playthough I get an error in the console: AVAEInternal.h:83 required condition is false: [AVAudioEngineGraph.mm:1361:Initialize: (IsFormatSampleRateAndChannelCountValid(outputHWFormat))] Any ideas how to fix it? // Input-Device setzen try? setupInputDevice(deviceID: inputDevice) let input = audioEngine.inputNode // Stereo-Format erzwingen let inputHWFormat = input.inputFormat(forBus: 0) let stereoFormat = AVAudioFormat(commonFormat: inputHWFormat.commonFormat, sampleRate: inputHWFormat.sampleRate, channels: 2, interleaved: inputHWFormat.isInterleaved) guard let format = stereoFormat else { throw AudioError.deviceSetupFailed(-1) } print("Input format: \(inputHWFormat)") print("Forced stereo format: \(format)") audioEngine.attach(monitorMixer) audioEngine.connect(input, to: monitorMixer, format: format) // MonitorMixer -> MainMixer (Output) // Problem here, format: format also breaks. audioEngine.connect(monitorMixer, to: audioEngine.mainMixerNode, format: nil)

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

I am developing a VOD playback app, but when I stream video to an external monitor connected via HDMI with Lightning on iOS 18 or later, the screen goes dark and I cannot confirm playback. The app I am developing does not detect the HDMI and display the Player separately, but simply mirrors the video. We have confirmed that the same phenomenon occurs with other services, but we were able to confirm playback with some services such as Apple TV. Please let us know if there are any other necessary settings such as video certificates required for video playback. We would also like to know if the problem occurs with iOS 18 or later.

Hello, I'm observing an intermittent memory leak being reported in the iOS Simulator when initializing and starting an AVAudioEngine. Even with minimal setup—just attaching a single AVAudioPlayerNode and connecting it to the mainMixerNode—Xcode's memory diagnostics and Instruments sometimes flag a leak. Here is a simplified version of the code I'm using: // This function is called when the user taps a button in the view controller: #import "ViewController.h" @interface ViewController () @end @implementation ViewController - (void)viewDidLoad { [super viewDidLoad]; } - (IBAction)myButtonAction:(id)sender { NSLog(@"Test"); soundCreate(); } @end // media.m static AVAudioEngine *audioEngine = nil; void soundCreate(void) { if (audioEngine != nil) return; [[AVAudioSession sharedInstance] setCategory:AVAudioSessionCategoryAmbient error:nil]; [[AVAudioSession sharedInstance] setActive:YES error:nil]; audioEngine = [[AVAudioEngine alloc] init]; AVAudioPlayerNode* playerNode = [[AVAudioPlayerNode alloc] init]; [audioEngine attachNode:playerNode]; [audioEngine connect:playerNode to:(AVAudioNode *)[audioEngine mainMixerNode] format:nil]; [audioEngine startAndReturnError:nil]; } In the memory leak report, the following call stack is repeated, seemingly in a loop: ListenerMap::InsertEvent(XAudioUnitEvent const&, ListenerBinding*) AudioToolboxCore ListenerMap::AddParameter(AUListener*, void*, XAudioUnitEvent const&) AudioToolboxCore AUListenerAddParameter AudioToolboxCore addOrRemoveParameterListeners(OpaqueAudioComponentInstance*, AUListenerBase*, AUParameterTree*, bool) AudioToolboxCore 0x180178ddf

Hello! I'm experiencing an issue with iOS's audio routing system when trying to use Bluetooth headphones for audio output while also recording environmental audio from the built-in microphone. Desired behavior: Play audio through Bluetooth headset (AirPods) Record unprocessed environmental audio from the iPhone's built-in microphone Actual behavior: When explicitly selecting the built-in microphone, iOS reports it's using it (in currentRoute.inputs) However, the actual audio data received is clearly still coming from the AirPods microphone The audio is heavily processed with voice isolation/noise cancellation, removing environmental sounds Environment Details Device: iPhone 12 Pro Max iOS Version: 18.4.1 Hardware: AirPods Audio Framework: AVAudioEngine (also tried AudioQueue) Code Attempted I've tried multiple approaches to force the correct routing: func configureAudioSession() { let session = AVAudioSession.sharedInstance() // Configure to allow Bluetooth output but use built-in mic try? session.setCategory(.playAndRecord, options: [.allowBluetoothA2DP, .defaultToSpeaker]) try? session.setActive(true) // Explicitly select built-in microphone if let inputs = session.availableInputs, let builtInMic = inputs.first(where: { $0.portType == .builtInMic }) { try? session.setPreferredInput(builtInMic) print("Selected input: \(builtInMic.portName)") } // Log the current route let route = session.currentRoute print("Current input: \(route.inputs.first?.portName ?? "None")") // Configure audio engine with native format let inputNode = audioEngine.inputNode let nativeFormat = inputNode.inputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: nativeFormat) { buffer, time in // Process audio buffer // Despite showing "Built-in Microphone" in route, audio appears to be // coming from AirPods with voice isolation applied - welp! } try? audioEngine.start() } I've also tried various combinations of: Different audio session modes (.default, .measurement, .voiceChat) Different option combinations (with/without .allowBluetooth, .allowBluetoothA2DP) Setting session.setPreferredInput() both before and after activation Diagnostic Observations When AirPods are connected: AVAudioSession.currentRoute.inputs correctly shows "Built-in Microphone" after setPreferredInput() The actual audio data received shows clear signs of AirPods' voice isolation processing Background/environmental sounds are actively filtered out... When recording a test audio played near the phone (not through the app), the recording is nearly silent. Only headset voice goes through. Questions Is there a workaround to force iOS to actually use the built-in microphone while maintaining Bluetooth output? Are there any lower-level configurations that might resolve this issue? Any insights, workarounds, or suggestions would be greatly appreciated. This is blocking a critical feature in my application that requires environmental audio recording while providing audio feedback through headphones 😅

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 can successfully match music using shazamkit on Apple using SwiftUI, a simple app that let user to load an audio file and exctracts the relative match, while i am unable to match music using shamzamkit on Android. I am trying to make the same simple app but i cannot match music as i get MATCH_ATTEMPT_FAILED every time i try to. I don't know what i am doing wrong but the shazam part in the kotlin Android code is in this method : suspend fun processAudioFileInBackground( filePath: String, developerTokenProvider: DeveloperTokenProvider ) = withContext(Dispatchers.IO) { val bufferSize = 1024 * 1024 val audioFile = FileInputStream(filePath) val byteBuffer = ByteBuffer.allocate(bufferSize) byteBuffer.order(ByteOrder.LITTLE_ENDIAN) var bytesRead: Int while (audioFile.read(byteBuffer.array()).also { bytesRead = it } != -1) { val signatureGenerator = (ShazamKit.createSignatureGenerator(AudioSampleRateInHz.SAMPLE_RATE_44100) as ShazamKitResult.Success).data signatureGenerator.append(byteBuffer.array(), bytesRead, System.currentTimeMillis()) val signature = signatureGenerator.generateSignature() println("Signature: ${signature.durationInMs}") val catalog = ShazamKit.createShazamCatalog(developerTokenProvider, Locale.ENGLISH) val session = (ShazamKit.createSession(catalog) as ShazamKitResult.Success).data val matchResult = session.match(signature) println("MatchResult : $matchResult") setMatchResult(matchResult) byteBuffer.clear() } audioFile.close() } I noticed that changing Locale in catalog creation results in different result as i get NoMatch without exception. Can you please help me with this? Do i need to create a custom catalog?

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?

I have a music app I'm developing and having a weird issue where I can see now playing info for every other platform than tvOS. As far as I can tell I have correctly configured the MPNowPlayingInfoCenter MPNowPlayingInfoCenter.default().nowPlayingInfo = nowPlayingInfo MPNowPlayingInfoCenter.default().playbackState = .playing Are there any extra requirements to get my app's now-playing info showing in control center on tvOS? Another strange issue that might be related is I can use the apple TV remote to pause audio but not resume playback, so I feel like there's something I'm missing about registering audio playback on tvOS specifically.