I have an application that includes custom artwork for the album cover and text details setup with the MPRemoteCommandCenter.shared() reference. I need the user to have a full featured "now playing" display to see all of this. My experience is that cannot find a set of parameters for AVAudioSession.setCategory() that route audio successfully, and display the full featured now playing deck. If I use .playAndRecord, the audio I send out plays out on the radio. But, the now-playing deck is empty and nothing I do with the command center seems to change that. If I instead use .playback, I cannot use .defaultToSpeaker option which is the only way I've found to cause the "now-playing" navigation button to appear so that the full featured deck will display. But, of course setCategory() fails with an error about .defaultToSpeaker only available with .playAndRecord, so some default or intermediate state is entered and I see the full featured deck, but no audio goes out to the radio. What combination is supposed to be used here and is this more likely a problem with thread use (@MainActor) and/or some ordering of operations that I've overlooked?

Overview We are producing audio in real time from an editing application and are trying to put that on an HLS stream. We attempt to submit PCM samples through an audio writer but are getting a crash after a select number of samples have been appended. Depending on the number of audio frames in the PCM buffer, we might get more iterations before the crash but it always has the same traceback (see below). Code The setup is rather simple. We took inspiration from a few sources around the web. NSMutableDictionary *audio = [[NSMutableDictionary alloc] init]; [audio setObject:@(kAudioFormatMPEG4AAC) forKey:AVFormatIDKey]; [audio setObject:[NSNumber numberWithInt:config.audioSampleRate] // 48000 forKey:AVSampleRateKey]; [audio setObject:[NSNumber numberWithInt:config.audioChannels] // 2 forKey:AVNumberOfChannelsKey]; [audio setObject:@160000 forKey:AVEncoderBitRateKey]; m_audioConfig = [[NSDictionary alloc] initWithDictionary:audio]; m_audio = [[AVAssetWriterInput alloc] initWithMediaType:AVMediaTypeAudio outputSettings:m_audioConfig]; AVAudioFrameCount audioFrames = BUFFER_SAMPLES * bCount; AVAudioPCMBuffer *pcmBuffer = [[AVAudioPCMBuffer alloc] initWithPCMFormat:m_full.pcmFormat frameCapacity:audioFrames]; pcmBuffer.frameLength = pcmBuffer.frameCapacity; AudioChannelLayout layout; memset(&layout, 0, sizeof(layout)); layout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo; CMFormatDescriptionRef format; OSStatus stats = CMAudioFormatDescriptionCreate( kCFAllocatorDefault, pcmBuffer.format.streamDescription, sizeof(layout), &layout, 0, nil, nil, &format ); for (int i = 0; i < bCount; i++) { AudioPCM pcm; audioCallback->callback(pcm); memcpy(*(pcmBuffer.int16ChannelData) + (bufferSize * i), pcm.data, bufferSize); } size_t samplesConsumed = BUFFER_SAMPLES * bCount; CMSampleBufferRef sampleBuffer; CMSampleTimingInfo timing; timing.duration = CMTimeMake(1, config.audioSampleRate); timing.presentationTimeStamp = presentationTime; timing.decodeTimeStamp = kCMTimeInvalid; OSStatus ostatus = CMSampleBufferCreate( kCFAllocatorDefault, nil, false, nil, nil, format, (CMItemCount)pcmBuffer.frameLength, 1, &timing, 0, nil, &sampleBuffer ); //// ostatus = CMSampleBufferSetDataBufferFromAudioBufferList( sampleBuffer, kCFAllocatorDefault, kCFAllocatorDefault, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, pcmBuffer.audioBufferList ); if (ostatus != noErr) { NSLog(@"fill audio sample from buffer list failed: %s", logAudioError(ostatus)); return; } ostatus = CMSampleBufferSetDataReady(sampleBuffer); if (ostatus != noErr) { NSLog(@"set sample buffer ready failed: %s", logAudioError(ostatus)); return; } // Finally we can attach it, then shove the presentation time forward [m_audio appendSampleBuffer:sampleBuffer]; The Crash The crash points towards some level of deallocation when the conversion tooling is done or has enough samples to process an output packet? It's had to say. 0 caulk 0x1a1e9532c caulk::alloc::tiered_allocator<caulk::alloc::size_range_tier<0ul, 1008ul, caulk::alloc::tree_allocator<caulk::alloc::chunk_allocator<caulk::alloc::page_allocator, caulk::alloc::bitmap_allocator, caulk::alloc::embed_block_memory, 16384ul, 16ul, 6ul>>>, caulk::alloc::size_range_tier<1009ul, 256000ul, caulk::alloc::guarded_edges_allocator<caulk::alloc::consolidating_free_map<caulk::alloc::page_allocator, 10485760ul>, 4ul>>, caulk::alloc::tracking_allocator<caulk::alloc::page_allocator>>::deallocate(caulk::alloc::block, unsigned long) + 636 1 AudioToolboxCore 0x1993fbfe4 ExtendedAudioBufferList_Destroy + 112 2 AudioToolboxCore 0x1993d5fe0 std::__1::__optional_destruct_base<ACCodecOutputBuffer, false>::~__optional_destruct_base[abi:ne180100]() + 68 3 AudioToolboxCore 0x1993d5f48 acv2::CodecConverter::~CodecConverter() + 196 4 AudioToolboxCore 0x1993d5e5c acv2::CodecConverter::~CodecConverter() + 16 5 AudioToolboxCore 0x1992574d8 std::__1::vector<std::__1::unique_ptr<acv2::AudioConverterBase, std::__1::default_delete<acv2::AudioConverterBase>>, std::__1::allocator<std::__1::unique_ptr<acv2::AudioConverterBase, std::__1::default_delete<acv2::AudioConverterBase>>>>::__clear[abi:ne180100]() + 84 6 AudioToolboxCore 0x199259acc acv2::AudioConverterChain::RebuildConverterChain(acv2::ChainBuildSettings const&) + 116 7 AudioToolboxCore 0x1992596ec acv2::AudioConverterChain::SetProperty(unsigned int, unsigned int, void const*) + 1808 8 AudioToolboxCore 0x199324acc acv2::AudioConverterV2::setProperty(unsigned int, unsigned int, void const*) + 84 9 AudioToolboxCore 0x199327f08 with_resolved(OpaqueAudioConverter*, caulk::function_ref<int (AudioConverterAPI*)>) + 60 10 AudioToolboxCore 0x1993281e4 AudioConverterSetProperty + 72 11 MediaToolbox 0x1a7566c2c FigSampleBufferProcessorCreateWithAudioCompression + 2296 12 MediaToolbox 0x1a754db08 0x1a70b5000 + 4819720 13 MediaToolbox 0x1a754dab4 FigMediaProcessorCreateForAudioCompressionWithFormatWriter + 100 14 MediaToolbox 0x1a77ebb98 0x1a70b5000 + 7564184 15 MediaToolbox 0x1a7804158 0x1a70b5000 + 7663960 16 MediaToolbox 0x1a7801da0 0x1a70b5000 + 7654816 17 AVFCore 0x1ada530c4 -[AVFigAssetWriterTrack addSampleBuffer:error:] + 192 18 AVFCore 0x1ada55164 -[AVFigAssetWriterAudioTrack _flushPendingSampleBuffersReturningError:] + 500 19 AVFCore 0x1ada55354 -[AVFigAssetWriterAudioTrack addSampleBuffer:error:] + 472 20 AVFCore 0x1ada4ebf0 -[AVAssetWriterInputWritingHelper appendSampleBuffer:error:] + 128 21 AVFCore 0x1ada4c354 -[AVAssetWriterInput appendSampleBuffer:] + 168 22 lib_devapple_hls.dylib 0x115d2c7cc detail::AppleHLSImplementation::audioRuntime() + 1052 23 lib_devapple_hls.dylib 0x115d2d094 void* std::__1::__thread_proxy[abi:ne180100]<std::__1::tuple<std::__1::unique_ptr<std::__1::__thread_struct, std::__1::default_delete<std::__1::__thread_struct>>, void (detail::AppleHLSImplementation::*)(), detail::AppleHLSImplementation*>>(void*) + 72 24 libsystem_pthread.dylib 0x196e5b2e4 _pthread_start + 136 Any insight would be welcome!

Recently, after the update of 26.3 Mac OS (Tahoe), the ordering of my music app has been horrible at best - music disappearing, tracks not aligning with albums (even if the albums are different years). It's created quite a problem, because the disappearing tracks issue seems to be replicating to iOS devices as well (although track numbering and album association seem to be stable). Has anyone else heard of this issue?

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.

Hi all, I’ve implemented the new Core Audio Tap API (AudioHardwareCreateProcessTap with CATapDescription) and I’m seeing consistent level attenuation that scales with the number of stereo output pairs exposed by the target device. What I observe Device with 4 stereo pairs (8 outs) → tap shows −12.04 dB relative to source. True 2-ch devices (built-in speakers, AirPods) → ~0 dB attenuation. The attenuation appears regardless of whether I: Create a global (default-output) tap via initStereoGlobalTapButExcludeProcesses: Or create a per-process/per-device tap via initWithProcesses:andDeviceUID:withStream: Additionally, the routing choice inside the sending app matters: App output to “System/Default Output” → I often see no attenuation. App output directly to a multi-out interface (e.g., RME Fireface) → I see the pair-count-scaled attenuation. I can query Core Audio for the number of output channels/pairs and gain-compensate (+20·log10(N_pairs) dB) and that matches my measurements for many cases. However, this compensation is not universally correct because it seems to depend on where each process routes its audio (Default Output vs. direct device), even when those processes are included in the same tap aggregate. Question Is there a supported way to obtain the raw, unattenuated streams for all processes through the Tap API—i.e., to bypass this automatic headroom/attenuation behavior entirely? If this attenuation is expected by design: Is there a documented rule for when it applies (global vs. device taps, per-process taps, stream selection, etc.)? Is there a property/flag to disable it, or a reliable, official method to compute the exact compensation (beyond counting stereo pairs)? Any guidance on ensuring consistent levels when multiple processes route differently (Default Output vs. direct device) but are captured by the same tap? Environment API: AudioHardwareCreateProcessTap + CATapDescription Devices: built-in output (2-ch), RME Fireface (8+ outs / 4+ stereo pairs) Behavior reproducible with both global and per-process/per-device tap descriptions. Attenuation example: 4 stereo pairs → −12.04 dB observed. Happy to provide a minimal sample, measurements, and device logs. Thanks! — David

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

Dear Sirs, I'd like to add an icon to my audio driver based on AudioDriverKit. This icon should show up left of my audio device in the audio devices dialog. For an Audio Server Plugin I managed to do this using the property kAudioDevicePropertyIcon and CFBundleCopyResourceURL(...) but how would you do this with AudioDriverKit? Should I use IOUserAudioCustomProperty or IOUserAudioControl and how would I refer to the Bundle? Is there an example available somewhere? Thanks and best regards, Johannes

Hello, I have an existing AUv3 instrument plugin. In the plug in, users can access files (audio files, song projects) via a UIDocumentPickerViewController In Logic Pro, (and some other hosts, but not all), the document picker is unable to receive touches, while a keyboard case is attached to the iPad. Removing the case (this is an Apple brand iPad case) allows the interactions to resume and allows me to pick files in the usual way. One of my users reports this non-responsive behavior occurs even after disconnecting their keyboard. I have fiddled with entitlements all day, and have determined that is not the issue, since the keyboard disconnection appears to fix it every time for me. Here is my, very boilerplate, presentation code : guard let type = UTType("com.my.type") else { return } let fileBrowser = UIDocumentPickerViewController(forOpeningContentTypes: [type]) fileBrowser.overrideUserInterfaceStyle = .dark fileBrowser.delegate = self fileBrowser.directoryURL = myFileFolderURL() self.present(fileBrowser, animated: true) {

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?

I bought two "Apple USB-C to Headphone Jack Adapters". Upon closer inspection, they seems to be of different generations: The one with product ID 0x110a on top is working fine. The one with product ID 0x110b has two issues: There is a short but loud click noise on the headphone when I connect it to the iPad. When I play audio using AVAudioPlayer the first half of a second or so is cut off. Here's how I'm playing the audio: audioPlayer = try AVAudioPlayer(contentsOf: url) audioPlayer?.delegate = self audioPlayer?.prepareToPlay() audioPlayer?.play() Is this a known issue? Am I doing something wrong?

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, macOS (latest macOS, latest HW, but doesn't matter) seems to prevent CoreMIDI driver logging with standard logging procedures (syslog, unified logging). The only chance to log something is writing to a file at one of the rare write-accessible locations for CoreMIDI. How is this supposed to work? Any hint is highly appreciated. Thanks!

We have an VoiP application, our application can be configured to amplify the PCM samples before feeding it to the Player to achieve volume gain at the receiver. In order to support this, We follow as below. If User has configured this Gain Settings within application, Application applies the amplification for the samples to introduce the gain. Application will also set the AVAudioSessionCategory to AVAudioSessionCategoryPlayback Provided the User has chosen the output to Speaker. This settings was working for us but we see there is a difference in behaviour w.r.t Volume Level System Settings between OS 26.3.1 and OS 26.4 When user has chosen earpiece as Output, then we will set the AVAudioSessionCategory to AVAudioSessionCategoryPlayAndRecord. User would have set the volume level to minimum. When user will change the output to Speaker, then we will set the AVAudioSessionCategory to AVAudioSessionCategoryPlayback. The expectation is, the volume level should be of AVAudioSessionCategoryPlayback what was set earlier instead we are seeing the volume level stays as minimum which was set to AVAudioSessionCategoryPlayAndRecord Could you please explain about this inconsistency w.r.t Volume level.

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 😅

When trying to load an ambisonics file using this project: https://github.com/robertncoomber/NativeiOSAmbisonicPlayback/ I get "Unexpected Ambisonics format". Interestingly, loading a 3rd order ambisonics file works fine: let ambisonicLayoutTag = kAudioChannelLayoutTag_HOA_ACN_SN3D | 16 let AmbisonicLayout = AVAudioChannelLayout(layoutTag: ambisonicLayoutTag) let StereoLayout = AVAudioChannelLayout(layoutTag: kAudioChannelLayoutTag_Stereo) So it's purely related to the kAudioChannelLayoutTag_Ambisonic_B_Format

I develop a application with an uvc camera, this camera is a webcam, I use the AVFoundation library ,but when I run the code "[self.mCaptureSession startRunning]" ,I can not get the buffer, I already set the delegate, any answer will help.

This is the code I use: @MainActor func picker()->UIViewController{ let pickerController = MPMediaPickerController(mediaTypes: .music) print(pickerController) pickerController.showsCloudItems=true pickerController.prompt = NSLocalizedString("Add pieces to queue", comment:""); pickerController.allowsPickingMultipleItems=true; pickerController.delegate=MPMusicPlayerControllerSingleton.sharedController(); return pickerController } MainActor @IBAction func handleBrowserTapped(_ sender: AnyObject){ if let pickerController=contentProvider?.picker(){ self.present(pickerController, animated:true, completion:nil) } } And this his is the crash log: *** Assertion failure in -[MPMediaPickerController_Appex requestRemoteViewController], MPMediaPickerController.m:523 *** Terminating app due to uncaught exception 'NSInternalInconsistencyException', reason: 'remoteViewController cannot be nil -- process will crash inserting in hierarchy. We likely got a nil remoteViewController because Music is crashing.' *** First throw call stack: (0x1869cac70 0x183499224 0x1844f9f50 0x1a6c6a060 0x18d45518c 0x18d4cd410 0x103354544 0x10336dccc 0x10338f748 0x103364204 0x103364144 0x186957a64 0x1868e5288 0x1868e41d0 0x22bde7498 0x18c5a5ca0 0x18c510254 0x18c71ce90 0x103854340 0x1038542b0 0x103854430 0x1834f1c1c) libc++abi: terminating due to uncaught exception of type NSException *** Terminating app due to uncaught exception 'NSInternalInconsistencyException', reason: 'remoteViewController cannot be nil -- process will crash inserting in hierarchy. We likely got a nil remoteViewController because Music is crashing.'

I'm using a 4 channel USB Audio interface, with 4 microphones, and want to process them through 4 independent effect chains. However the output from AVAudioInputNode is a single 4 channel bus. How can I split this into 4 mono busses? The following code splits the input into 4 copies, and routes them through the effects, but each bus contains all four channels. How can I remap the channels to remove the unwanted channels from the bus? I tried using channelMap on the mixer node but that had no effect. I'm currently using this code primarily on iOS but it should be portable between iOS and MacOS. It would be possible to do this through a Matrix Mixer Node, but that seems completely overkill, for such a basic operation. I'm already using a Matrix Mixer to combine the inputs, and it's not well supported in AVAudioEngine. AVAudioInputNode *inputNode=[engine inputNode]; [inputNode setVoiceProcessingEnabled:NO error:nil]; NSMutableArray *micDestinations=[NSMutableArray arrayWithCapacity:trackCount]; for(i=0;i<trackCount;i++) { fixMicFormat[i]=[AVAudioMixerNode new]; [engine attachNode:fixMicFormat[i]]; // And create reverb/compressor and eq the same way... [engine connect:reverb[i] to:matrixMixerNode fromBus:0 toBus:i format:nil]; [engine connect:eq[i] to:reverb[i] fromBus:0 toBus:0 format:nil]; [engine connect:compressor[i] to:eq[i] fromBus:0 toBus:0 format:nil]; [engine connect:fixMicFormat[i] to:compressor[i] fromBus:0 toBus:0 format:nil]; [micDestinations addObject:[[AVAudioConnectionPoint alloc] initWithNode:fixMicFormat[i] bus:0] ]; } AVAudioFormat *inputFormat = [inputNode outputFormatForBus: 1]; [engine connect:inputNode toConnectionPoints:micDestinations fromBus:1 format:inputFormat];

hi all, as soon an audio is played in a whatever app, coreaudiod inserts a sleep prevent assertion for both, the system AND the display. can i somehow stop the insertion of the display sleep assertion? pid 223(coreaudiod): [0x00004e9e00058dc2] 00:03:18 PreventUserIdleDisplaySleep named: "com.apple.audio.AppleGFXHDAEngineOutputDP:10001:0:{B31A-08C6-00000000}.context.preventuseridledisplaysleep" Created for PID: 4145. where PID 4145 is spotify. but it doesn't matter which app is playing the audio. any help would be appreciated thanks

Hi team, With regards to Call (Live) Translations on VOIP: Is it possible to invoke live translations within the app? (without going into the Call System UI) Is it possible to navigate users from app to Call System UI via an API? (and also invoking the new live translations directly) Will Apple support more languages apart from the current ones? (Currently I see 4 supported languages)