Dive into the technical aspects of audio on your device, including codecs, format support, and customization options.

Audio Documentation

Posts under Audio subtopic

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MusicKit + AirPlay
Hello, I'm working on a MusicKit based SwiftUI app. I've integrated AirPlay using the AVRoutePickerView like so: struct UIKitAirPlayPickerView: UIViewRepresentable { func makeUIView(context: Context) -> AVRoutePickerView { let routePickerView = AVRoutePickerView() routePickerView.prioritizesVideoDevices = false return routePickerView } func updateUIView(_ uiView: AVRoutePickerView, context: Context) {} } The AirPlay menu appears as expected, and selecting an AirPlay device functions as expected. I'm currently sending audio from my app to a HomePod. However, the state of the AVRoutePickerView does not reflect the playback state. There is no cover art and it says "Not Playing". When my device is locked, my lock screen shows the album art, metadata and AirPlay routing as expected. My app uses the ApplicationMusicPlayer however I encounter the same behavior using the SystemMusicPlayer. Any guidance on how to troubleshoot this? Is there any other way to integrate the system AirPlay picker into my app, or is this my only option? Thank you for reading.
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386
Feb ’26
Add icon to DEXT based on AudioDriverKit
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
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1.2k
Jul ’25
Live Translations on VOIP on iOS26
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)
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170
Aug ’25
Unique identifier of a MIDI device
Hello, I need to know what is a unique identifier of a MIDI device (source/destination). Important note: I want to get the same ID when a device is reconnected (unplugged and then plugged again). The main candidate is kMIDIPropertyUniqueID property. But I don't know if it meets the requirement above or not. Additional question: is it always available for any endpoint? Also there is kMIDIPropertyDeviceID property. What about it? And one more option is just MIDIEndpointRef returned by MIDIGetSource or MIDIGetDestination. So what is the proper way to get ID which persists between device reconnections?
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86
Jan ’26
Essentials of macOS to read and write mp3 and mp4 audio files
Hi, On macOS I used to open MP3 and MP4 files with ExtAudioFile. For a few years it doesn't work anymore. So I decided to try different macOS API using the AudioFileID of AudioToolbox framework. I decided to write a test: https://gist.github.com/joelkraehemann/7f5b241b52ca38c3a765c138fb647588 It fails right here: AudioFileOpenWithCallbacks() By telling OSStatus error 1954115647, which means kAudioFileUnsupportedFileTypeError. The filename was set to an MP4 file: ~/Music/test.mp4 Howto fix this? regards, Joël
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446
Jun ’25
App Randomly Crashes During Continuous Sound Playback Using AVAudioPlayer
Environment→ ・Device: iPad 10th generation ・OS:**iOS18.3.2 We're using AVAudioPlayer to play a sound when a button is tapped. In our use case, this button can be tapped very frequently — roughly every 0.1 to 0.2 seconds. Each tap triggers the following function: var audioPlayer: AVAudioPlayer? func soundPlay(resource: String, type: String){ guard let path = Bundle.main.path(forResource: resource, ofType: type) else { return } do { audioPlayer = try AVAudioPlayer(contentsOf: URL(fileURLWithPath: path)) audioPlayer!.delegate = self try audioSession.setCategory(.playback) } catch { return } self.audioPlayer!.play() } The issue is that under high-frequency tapping (especially around 0.1–0.15s intervals), the app occasionally crashes. The crash does not occur every time, but it happens randomly — sometimes within 30 seconds, within 1 minute, or even 3 minutes of continuous tapping. Interestingly, adding a delay of 0.2 seconds between button taps seems to prevent the crash entirely. Delays shorter than 0.2 seconds (e.g.,0.15s,0.18s) still result in occasional crashes. My questions are: **Is this expected behavior from AVAudioPlayer or AVAudioSession? Could this be a known issue or a limitation in AVFoundation? Is there any documentation or guidance on handling frequent sound playback safely?** Any insights or recommendations on how to handle rapid, repeated audio playback more reliably would be appreciated.
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232
May ’25
MusicKit - Skipping Forwards or Backwards does not update
Hello everyone, I am working on an app that allows you to review your own music using Apple Music. Currently I am running into an issue with the skipping forwards and backwards outside of the app. How it should work: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song on an album should play and the information should change to reflect that in the app. If you play a song in Apple Music, you can see a Now Playing view in the lock screen. When you skip forward or backwards, it will do either action and it would reflect that when you see a little frequency icon on artwork image of a song. What it's doing: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song is reflected outside of the app, but not in the app. When skipping a song outside of the app, it works correctly to head to the next song. But when I return to the app, it is not reflected NOTE: I am not using MusicKit variables such as Track, Album to display the songs. Since I want to grab the songs and review them I need a rating so I created my own that grabs the MusicItemID, name, artist(s), etc. NOTE: I am using ApplicationMusicPlayer.shared Is there a way to get the song to reflect in my app? (If its easier, a simple example of it would be nice. No need to create an entire xprod file)
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107
Apr ’25
AU MIDI Plugin UI not showing
Hello, I am having an issue with a very small percentage of our users not being able to view the UI of our MIDI Plugin Chord Prism. I have looked this up and seen to where it has been resolved within Logic for AU Instrument and Effect plugins by switching out of "Controls" view, but my situation is different and there is no option on what is displayed to switch out of "Controls" view. Is this something that can be fixed by adjusting settings within Logic?
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156
Mar ’26
Ducking MusicKit output when playing another sound
I am developing an app that uses MusicKit to play music and then I need to have spoken words played to the user, while ducking the audio coming from MusicKit (application music player) the built in Siri voices are not off sufficient quality so I am using an external service to create an mp3 file and then play this back using AVAudioSession Sample code below the problem I am having is that .duckOthers is not ducking the Application Music Player output Is this a bug or am I doing this wrong? // Configure audio session for system-wide ducking try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, options: [.duckOthers, .mixWithOthers]) try AVAudioSession.sharedInstance().setActive(true) // Set the ducking level to maximum try AVAudioSession.sharedInstance().setPreferredIOBufferDuration(0.005) // Create and configure audio player self.audioPlayer = try AVAudioPlayer(data: audioData) self.audioPlayer?.delegate = self self.audioPlayer?.volume = 1.0 // Ensure full volume for speech self.audioPlayer?.prepareToPlay() // Set the audio player's settings for maximum clarity self.audioPlayer?.enableRate = false self.audioPlayer?.pan = 0.0 // Center the audio self.audioPlayer?.play()
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66
Apr ’25
How to safely switch between mic configurations on iOS?
I have an iPadOS M-processor application with two different running configurations. In config1, the shared AVAudioSession is configured for .videoChat mode using the built-in microphone. The input/output nodes of the AVAudioEngine are configured with voice processing enabled. The built-in mic is formatted for 1 channel at 48KHz. In config2, the shared AVAudioSession is configured for .measurement mode using an external USB microphone. The input/output nodes of the AVAudioEngine are configured with voice processing disabled. The external mic is formatted for 2 channels at 44.1KHz I've written a configuration manager designed to safely switch between these two configurations. It works by stopping AVAudioEngine and detaching all but the input and output nodes, updating the shared audio session for the desired mic and sample-rates, and setting the appropriate state for voice processing to either true or false as required by the configuration. Finally the new audio graph is constructed by attaching appropriate nodes, connecting them, and re-starting AVAudioEngine I'm experiencing what I believe is a race-condition between switching voice processing on or off and then trying to re-build and start the new audio graph. Even though notifications, which are dumped to the console indicate that my requested input and sample-rate settings are in place, I crash when trying to start the audio engine because the sample-rate is wrong. Investigating further it looks like the switch from remote I/O to voice-processing I/O or vice-versa has not yet actually completed. I introduced a 100ms second delay and that seems to help but is obviously not a reliable way to build software that must work consistently. How can I make sure that what are apparently asynchronous configuration changes to the shared audio session and the input/output nodes have completed before I go on? I tried using route change notifications from the shared AVAudioSession but these lie. They say my preferred mic input and sample-rate setting is in place but when I dump the AVAudioEngine graph to the debugger console, I still see the wrong sample rate assigned to the input/output nodes. Also these are the wrong AU nodes. That is, VPIO is still in place when RIO should be, or vice-versa. How can I make the switch reliable without arbitrary time delays? Is my configuration manager approach appropriate (question for Apple engineers)?
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327
Nov ’25
Incoming calls thrue Jisti Meet and locked screen
Problem: When the screen is locked, an incoming call does not initiate the launch of the Flutter application required for audio and video communication through Jitsi Meet. In the unlocked state, the application functions correctly. The current implementation does not have a mechanism for activating the Flutter engine when receiving a call via CallKit while the screen is locked. Although CallKit UI displays the call acceptance interface and the audio session is configured, the Flutter application remains in a suspended state, making it impossible to connect to the media server. Audio session activated using didActivateAudioSession method.
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3w
AirPods Pro 3 Disconnecting from Apple Ultra 3 consistently
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?
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790
Oct ’25
AVAssetWriterInput Crash on appendSampleBuffer Converting PCM
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!
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342
Jun ’25
Spatial Audio on iOS 18 don't work as inteneded
I’m facing a problem while trying to achieve spatial audio effects in my iOS 18 app. I have tried several approaches to get good 3D audio, but the effect never felt good enough or it didn’t work at all. Also what mostly troubles me is I noticed that AirPods I have doesn’t recognize my app as one having spatial audio (in audio settings it shows "Spatial Audio Not Playing"). So i guess my app doesn't use spatial audio potential. First approach uses AVAudioEnviromentNode with AVAudioEngine. Chaining position of player as well as changing listener’s doesn’t seem to change anything in how audio plays. Here's simple how i initialize AVAudioEngine import Foundation import AVFoundation class AudioManager: ObservableObject { // important class variables var audioEngine: AVAudioEngine! var environmentNode: AVAudioEnvironmentNode! var playerNode: AVAudioPlayerNode! var audioFile: AVAudioFile? ... //Sound set up func setupAudio() { do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } audioEngine = AVAudioEngine() environmentNode = AVAudioEnvironmentNode() playerNode = AVAudioPlayerNode() audioEngine.attach(environmentNode) audioEngine.attach(playerNode) audioEngine.connect(playerNode, to: environmentNode, format: nil) audioEngine.connect(environmentNode, to: audioEngine.mainMixerNode, format: nil) environmentNode.listenerPosition = AVAudio3DPoint(x: 0, y: 0, z: 0) environmentNode.listenerAngularOrientation = AVAudio3DAngularOrientation(yaw: 0, pitch: 0, roll: 0) environmentNode.distanceAttenuationParameters.referenceDistance = 1.0 environmentNode.distanceAttenuationParameters.maximumDistance = 100.0 environmentNode.distanceAttenuationParameters.rolloffFactor = 2.0 // example.mp3 is mono sound guard let audioURL = Bundle.main.url(forResource: "example", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: audioURL) } catch { print("Failed to load audio file: \(error)") } } ... //Playing sound func playSpatialAudio(pan: Float ) { guard let audioFile = audioFile else { return } // left side playerNode.position = AVAudio3DPoint(x: pan, y: 0, z: 0) playerNode.scheduleFile(audioFile, at: nil, completionHandler: nil) do { try audioEngine.start() playerNode.play() } catch { print("Failed to start audio engine: \(error)") } ... } Second more complex approach using PHASE did better. I’ve made an exemplary app that allows players to move audio player in 3D space. I have added reverb, and sliders changing audio position up to 10 meters each direction from listener but audio seems to only really change left to right (x axis) - again I think it might be trouble with the app not being recognized as spatial. //Crucial class Variables: class PHASEAudioController: ObservableObject{ private var soundSourcePosition: simd_float4x4 = matrix_identity_float4x4 private var audioAsset: PHASESoundAsset! private let phaseEngine: PHASEEngine private let params = PHASEMixerParameters() private var soundSource: PHASESource private var phaseListener: PHASEListener! private var soundEventAsset: PHASESoundEventNodeAsset? // Initialization of PHASE init{ do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } // Init PHASE Engine phaseEngine = PHASEEngine(updateMode: .automatic) phaseEngine.defaultReverbPreset = .mediumHall phaseEngine.outputSpatializationMode = .automatic //nothing helps // Set listener position to (0,0,0) in World space let origin: simd_float4x4 = matrix_identity_float4x4 phaseListener = PHASEListener(engine: phaseEngine) phaseListener.transform = origin phaseListener.automaticHeadTrackingFlags = .orientation try! self.phaseEngine.rootObject.addChild(self.phaseListener) do{ try self.phaseEngine.start(); } catch { print("Could not start PHASE engine") } audioAsset = loadAudioAsset() // Create sound Source // Sphere soundSourcePosition.translate(z:3.0) let sphere = MDLMesh.newEllipsoid(withRadii: vector_float3(0.1,0.1,0.1), radialSegments: 14, verticalSegments: 14, geometryType: MDLGeometryType.triangles, inwardNormals: false, hemisphere: false, allocator: nil) let shape = PHASEShape(engine: phaseEngine, mesh: sphere) soundSource = PHASESource(engine: phaseEngine, shapes: [shape]) soundSource.transform = soundSourcePosition print(soundSourcePosition) do { try phaseEngine.rootObject.addChild(soundSource) } catch { print ("Failed to add a child object to the scene.") } let simpleModel = PHASEGeometricSpreadingDistanceModelParameters() simpleModel.rolloffFactor = rolloffFactor soundPipeline.distanceModelParameters = simpleModel let samplerNode = PHASESamplerNodeDefinition( soundAssetIdentifier: audioAsset.identifier, mixerDefinition: soundPipeline, identifier: audioAsset.identifier + "_SamplerNode") samplerNode.playbackMode = .looping do {soundEventAsset = try phaseEngine.assetRegistry.registerSoundEventAsset( rootNode: samplerNode, identifier: audioAsset.identifier + "_SoundEventAsset") } catch { print("Failed to register a sound event asset.") soundEventAsset = nil } } //Playing sound func playSound(){ // Fire new sound event with currently set properties guard let soundEventAsset else { return } params.addSpatialMixerParameters( identifier: soundPipeline.identifier, source: soundSource, listener: phaseListener) let soundEvent = try! PHASESoundEvent(engine: phaseEngine, assetIdentifier: soundEventAsset.identifier, mixerParameters: params) soundEvent.start(completion: nil) } ... } Also worth mentioning might be that I only own personal team account
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1.2k
Nov ’25
FaceTime Screen-Share Audio and Video Experience
FaceTime’s screen-share audio balance is insanely absurd right now. Whenever I share media, the system audio that gets sent through FaceTime is a tiny whisper even at full volume (or even when connected to my speaker or headphones). The moment anyone on the call makes any noise at all, the shared audio ducks so hard it disappears, while the voice (or rustling or air conditioning noise) spikes to painful levels. It’s impossible to watch or listen to anything together. Also, the feature where FaceTime would shrink to a square during screen-sharing has been completely removed. That was a good feature and I'm really confused why it's gone. Now, the FaceTime window stays as a long rectangle that covers part of the content I'm trying to share (unless I do full screen tile, but then I can't pull up any other windows during the call) and can't be made smaller than about a third of the screen. You can't resize the window or adjust its dimensions, so it ends up blocking the actual media you're trying to watch. Here are some feature requests/fixes that would greatly improve the FaceTime screen-share experience: Option to adjust the shared media volume independently of call audio. Disable/toggle the extreme automatic audio docking while screen-sharing Reintroduce the minimized “floating square” mode or allow full manual resizing and repositioning of the FaceTime window during screen-share sessions. Overall, this setup makes FaceTime screen-sharing basically unusable. The audio balance is so inconsistent that it’s easier to switch to Zoom or Google Meet, which both handle shared sound correctly and let you move the call window out of the way. Until these issues are fixed, there’s no practical reason to use FaceTime for shared viewing at all.
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441
Nov ’25
Unable to match music with shazamkit for Android
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?
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154
May ’25
MusicKit + AirPlay
Hello, I'm working on a MusicKit based SwiftUI app. I've integrated AirPlay using the AVRoutePickerView like so: struct UIKitAirPlayPickerView: UIViewRepresentable { func makeUIView(context: Context) -> AVRoutePickerView { let routePickerView = AVRoutePickerView() routePickerView.prioritizesVideoDevices = false return routePickerView } func updateUIView(_ uiView: AVRoutePickerView, context: Context) {} } The AirPlay menu appears as expected, and selecting an AirPlay device functions as expected. I'm currently sending audio from my app to a HomePod. However, the state of the AVRoutePickerView does not reflect the playback state. There is no cover art and it says "Not Playing". When my device is locked, my lock screen shows the album art, metadata and AirPlay routing as expected. My app uses the ApplicationMusicPlayer however I encounter the same behavior using the SystemMusicPlayer. Any guidance on how to troubleshoot this? Is there any other way to integrate the system AirPlay picker into my app, or is this my only option? Thank you for reading.
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1
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0
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386
Activity
Feb ’26
Add icon to DEXT based on AudioDriverKit
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
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7
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0
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1.2k
Activity
Jul ’25
Audio Unit MIDI Plugin documentation
Hi folks - I'm having trouble finding specific documentation about Audio Unit MIDI plugins - as in MIDI -only. Any suggestions welcome as searches aren't returning much. (too niche? user error?)
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0
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0
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145
Activity
Dec ’25
Live Translations on VOIP on iOS26
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)
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1
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0
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170
Activity
Aug ’25
Unique identifier of a MIDI device
Hello, I need to know what is a unique identifier of a MIDI device (source/destination). Important note: I want to get the same ID when a device is reconnected (unplugged and then plugged again). The main candidate is kMIDIPropertyUniqueID property. But I don't know if it meets the requirement above or not. Additional question: is it always available for any endpoint? Also there is kMIDIPropertyDeviceID property. What about it? And one more option is just MIDIEndpointRef returned by MIDIGetSource or MIDIGetDestination. So what is the proper way to get ID which persists between device reconnections?
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0
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0
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86
Activity
Jan ’26
Essentials of macOS to read and write mp3 and mp4 audio files
Hi, On macOS I used to open MP3 and MP4 files with ExtAudioFile. For a few years it doesn't work anymore. So I decided to try different macOS API using the AudioFileID of AudioToolbox framework. I decided to write a test: https://gist.github.com/joelkraehemann/7f5b241b52ca38c3a765c138fb647588 It fails right here: AudioFileOpenWithCallbacks() By telling OSStatus error 1954115647, which means kAudioFileUnsupportedFileTypeError. The filename was set to an MP4 file: ~/Music/test.mp4 Howto fix this? regards, Joël
Replies
1
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0
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446
Activity
Jun ’25
App Randomly Crashes During Continuous Sound Playback Using AVAudioPlayer
Environment→ ・Device: iPad 10th generation ・OS:**iOS18.3.2 We're using AVAudioPlayer to play a sound when a button is tapped. In our use case, this button can be tapped very frequently — roughly every 0.1 to 0.2 seconds. Each tap triggers the following function: var audioPlayer: AVAudioPlayer? func soundPlay(resource: String, type: String){ guard let path = Bundle.main.path(forResource: resource, ofType: type) else { return } do { audioPlayer = try AVAudioPlayer(contentsOf: URL(fileURLWithPath: path)) audioPlayer!.delegate = self try audioSession.setCategory(.playback) } catch { return } self.audioPlayer!.play() } The issue is that under high-frequency tapping (especially around 0.1–0.15s intervals), the app occasionally crashes. The crash does not occur every time, but it happens randomly — sometimes within 30 seconds, within 1 minute, or even 3 minutes of continuous tapping. Interestingly, adding a delay of 0.2 seconds between button taps seems to prevent the crash entirely. Delays shorter than 0.2 seconds (e.g.,0.15s,0.18s) still result in occasional crashes. My questions are: **Is this expected behavior from AVAudioPlayer or AVAudioSession? Could this be a known issue or a limitation in AVFoundation? Is there any documentation or guidance on handling frequent sound playback safely?** Any insights or recommendations on how to handle rapid, repeated audio playback more reliably would be appreciated.
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232
Activity
May ’25
Forward/Reverse Arrows missing in Music/Get Info
There appears to be no method of going forward or backwards in Get Info in the Music application,
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51
Activity
Jun ’25
MusicKit - Skipping Forwards or Backwards does not update
Hello everyone, I am working on an app that allows you to review your own music using Apple Music. Currently I am running into an issue with the skipping forwards and backwards outside of the app. How it should work: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song on an album should play and the information should change to reflect that in the app. If you play a song in Apple Music, you can see a Now Playing view in the lock screen. When you skip forward or backwards, it will do either action and it would reflect that when you see a little frequency icon on artwork image of a song. What it's doing: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song is reflected outside of the app, but not in the app. When skipping a song outside of the app, it works correctly to head to the next song. But when I return to the app, it is not reflected NOTE: I am not using MusicKit variables such as Track, Album to display the songs. Since I want to grab the songs and review them I need a rating so I created my own that grabs the MusicItemID, name, artist(s), etc. NOTE: I am using ApplicationMusicPlayer.shared Is there a way to get the song to reflect in my app? (If its easier, a simple example of it would be nice. No need to create an entire xprod file)
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107
Activity
Apr ’25
CoreAudio: Specification of Private Aggregate or Tap
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?
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235
Activity
3w
AU MIDI Plugin UI not showing
Hello, I am having an issue with a very small percentage of our users not being able to view the UI of our MIDI Plugin Chord Prism. I have looked this up and seen to where it has been resolved within Logic for AU Instrument and Effect plugins by switching out of "Controls" view, but my situation is different and there is no option on what is displayed to switch out of "Controls" view. Is this something that can be fixed by adjusting settings within Logic?
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156
Activity
Mar ’26
Ducking MusicKit output when playing another sound
I am developing an app that uses MusicKit to play music and then I need to have spoken words played to the user, while ducking the audio coming from MusicKit (application music player) the built in Siri voices are not off sufficient quality so I am using an external service to create an mp3 file and then play this back using AVAudioSession Sample code below the problem I am having is that .duckOthers is not ducking the Application Music Player output Is this a bug or am I doing this wrong? // Configure audio session for system-wide ducking try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, options: [.duckOthers, .mixWithOthers]) try AVAudioSession.sharedInstance().setActive(true) // Set the ducking level to maximum try AVAudioSession.sharedInstance().setPreferredIOBufferDuration(0.005) // Create and configure audio player self.audioPlayer = try AVAudioPlayer(data: audioData) self.audioPlayer?.delegate = self self.audioPlayer?.volume = 1.0 // Ensure full volume for speech self.audioPlayer?.prepareToPlay() // Set the audio player's settings for maximum clarity self.audioPlayer?.enableRate = false self.audioPlayer?.pan = 0.0 // Center the audio self.audioPlayer?.play()
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66
Activity
Apr ’25
How to safely switch between mic configurations on iOS?
I have an iPadOS M-processor application with two different running configurations. In config1, the shared AVAudioSession is configured for .videoChat mode using the built-in microphone. The input/output nodes of the AVAudioEngine are configured with voice processing enabled. The built-in mic is formatted for 1 channel at 48KHz. In config2, the shared AVAudioSession is configured for .measurement mode using an external USB microphone. The input/output nodes of the AVAudioEngine are configured with voice processing disabled. The external mic is formatted for 2 channels at 44.1KHz I've written a configuration manager designed to safely switch between these two configurations. It works by stopping AVAudioEngine and detaching all but the input and output nodes, updating the shared audio session for the desired mic and sample-rates, and setting the appropriate state for voice processing to either true or false as required by the configuration. Finally the new audio graph is constructed by attaching appropriate nodes, connecting them, and re-starting AVAudioEngine I'm experiencing what I believe is a race-condition between switching voice processing on or off and then trying to re-build and start the new audio graph. Even though notifications, which are dumped to the console indicate that my requested input and sample-rate settings are in place, I crash when trying to start the audio engine because the sample-rate is wrong. Investigating further it looks like the switch from remote I/O to voice-processing I/O or vice-versa has not yet actually completed. I introduced a 100ms second delay and that seems to help but is obviously not a reliable way to build software that must work consistently. How can I make sure that what are apparently asynchronous configuration changes to the shared audio session and the input/output nodes have completed before I go on? I tried using route change notifications from the shared AVAudioSession but these lie. They say my preferred mic input and sample-rate setting is in place but when I dump the AVAudioEngine graph to the debugger console, I still see the wrong sample rate assigned to the input/output nodes. Also these are the wrong AU nodes. That is, VPIO is still in place when RIO should be, or vice-versa. How can I make the switch reliable without arbitrary time delays? Is my configuration manager approach appropriate (question for Apple engineers)?
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327
Activity
Nov ’25
Incoming calls thrue Jisti Meet and locked screen
Problem: When the screen is locked, an incoming call does not initiate the launch of the Flutter application required for audio and video communication through Jitsi Meet. In the unlocked state, the application functions correctly. The current implementation does not have a mechanism for activating the Flutter engine when receiving a call via CallKit while the screen is locked. Although CallKit UI displays the call acceptance interface and the audio session is configured, the Flutter application remains in a suspended state, making it impossible to connect to the media server. Audio session activated using didActivateAudioSession method.
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116
Activity
3w
AirPods Pro 3 Disconnecting from Apple Ultra 3 consistently
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?
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790
Activity
Oct ’25
AVAssetWriterInput Crash on appendSampleBuffer Converting PCM
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!
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342
Activity
Jun ’25
Audio of AirPods won’t work
Since the last update to IOS 26.0 (23A5276f) the AirPods connect to my IPhone and the Audio is still running through the phone. They are shown in the Bluetooth Icon that they’re paired.
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94
Activity
Jun ’25
Spatial Audio on iOS 18 don't work as inteneded
I’m facing a problem while trying to achieve spatial audio effects in my iOS 18 app. I have tried several approaches to get good 3D audio, but the effect never felt good enough or it didn’t work at all. Also what mostly troubles me is I noticed that AirPods I have doesn’t recognize my app as one having spatial audio (in audio settings it shows "Spatial Audio Not Playing"). So i guess my app doesn't use spatial audio potential. First approach uses AVAudioEnviromentNode with AVAudioEngine. Chaining position of player as well as changing listener’s doesn’t seem to change anything in how audio plays. Here's simple how i initialize AVAudioEngine import Foundation import AVFoundation class AudioManager: ObservableObject { // important class variables var audioEngine: AVAudioEngine! var environmentNode: AVAudioEnvironmentNode! var playerNode: AVAudioPlayerNode! var audioFile: AVAudioFile? ... //Sound set up func setupAudio() { do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } audioEngine = AVAudioEngine() environmentNode = AVAudioEnvironmentNode() playerNode = AVAudioPlayerNode() audioEngine.attach(environmentNode) audioEngine.attach(playerNode) audioEngine.connect(playerNode, to: environmentNode, format: nil) audioEngine.connect(environmentNode, to: audioEngine.mainMixerNode, format: nil) environmentNode.listenerPosition = AVAudio3DPoint(x: 0, y: 0, z: 0) environmentNode.listenerAngularOrientation = AVAudio3DAngularOrientation(yaw: 0, pitch: 0, roll: 0) environmentNode.distanceAttenuationParameters.referenceDistance = 1.0 environmentNode.distanceAttenuationParameters.maximumDistance = 100.0 environmentNode.distanceAttenuationParameters.rolloffFactor = 2.0 // example.mp3 is mono sound guard let audioURL = Bundle.main.url(forResource: "example", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: audioURL) } catch { print("Failed to load audio file: \(error)") } } ... //Playing sound func playSpatialAudio(pan: Float ) { guard let audioFile = audioFile else { return } // left side playerNode.position = AVAudio3DPoint(x: pan, y: 0, z: 0) playerNode.scheduleFile(audioFile, at: nil, completionHandler: nil) do { try audioEngine.start() playerNode.play() } catch { print("Failed to start audio engine: \(error)") } ... } Second more complex approach using PHASE did better. I’ve made an exemplary app that allows players to move audio player in 3D space. I have added reverb, and sliders changing audio position up to 10 meters each direction from listener but audio seems to only really change left to right (x axis) - again I think it might be trouble with the app not being recognized as spatial. //Crucial class Variables: class PHASEAudioController: ObservableObject{ private var soundSourcePosition: simd_float4x4 = matrix_identity_float4x4 private var audioAsset: PHASESoundAsset! private let phaseEngine: PHASEEngine private let params = PHASEMixerParameters() private var soundSource: PHASESource private var phaseListener: PHASEListener! private var soundEventAsset: PHASESoundEventNodeAsset? // Initialization of PHASE init{ do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } // Init PHASE Engine phaseEngine = PHASEEngine(updateMode: .automatic) phaseEngine.defaultReverbPreset = .mediumHall phaseEngine.outputSpatializationMode = .automatic //nothing helps // Set listener position to (0,0,0) in World space let origin: simd_float4x4 = matrix_identity_float4x4 phaseListener = PHASEListener(engine: phaseEngine) phaseListener.transform = origin phaseListener.automaticHeadTrackingFlags = .orientation try! self.phaseEngine.rootObject.addChild(self.phaseListener) do{ try self.phaseEngine.start(); } catch { print("Could not start PHASE engine") } audioAsset = loadAudioAsset() // Create sound Source // Sphere soundSourcePosition.translate(z:3.0) let sphere = MDLMesh.newEllipsoid(withRadii: vector_float3(0.1,0.1,0.1), radialSegments: 14, verticalSegments: 14, geometryType: MDLGeometryType.triangles, inwardNormals: false, hemisphere: false, allocator: nil) let shape = PHASEShape(engine: phaseEngine, mesh: sphere) soundSource = PHASESource(engine: phaseEngine, shapes: [shape]) soundSource.transform = soundSourcePosition print(soundSourcePosition) do { try phaseEngine.rootObject.addChild(soundSource) } catch { print ("Failed to add a child object to the scene.") } let simpleModel = PHASEGeometricSpreadingDistanceModelParameters() simpleModel.rolloffFactor = rolloffFactor soundPipeline.distanceModelParameters = simpleModel let samplerNode = PHASESamplerNodeDefinition( soundAssetIdentifier: audioAsset.identifier, mixerDefinition: soundPipeline, identifier: audioAsset.identifier + "_SamplerNode") samplerNode.playbackMode = .looping do {soundEventAsset = try phaseEngine.assetRegistry.registerSoundEventAsset( rootNode: samplerNode, identifier: audioAsset.identifier + "_SoundEventAsset") } catch { print("Failed to register a sound event asset.") soundEventAsset = nil } } //Playing sound func playSound(){ // Fire new sound event with currently set properties guard let soundEventAsset else { return } params.addSpatialMixerParameters( identifier: soundPipeline.identifier, source: soundSource, listener: phaseListener) let soundEvent = try! PHASESoundEvent(engine: phaseEngine, assetIdentifier: soundEventAsset.identifier, mixerParameters: params) soundEvent.start(completion: nil) } ... } Also worth mentioning might be that I only own personal team account
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Activity
Nov ’25
FaceTime Screen-Share Audio and Video Experience
FaceTime’s screen-share audio balance is insanely absurd right now. Whenever I share media, the system audio that gets sent through FaceTime is a tiny whisper even at full volume (or even when connected to my speaker or headphones). The moment anyone on the call makes any noise at all, the shared audio ducks so hard it disappears, while the voice (or rustling or air conditioning noise) spikes to painful levels. It’s impossible to watch or listen to anything together. Also, the feature where FaceTime would shrink to a square during screen-sharing has been completely removed. That was a good feature and I'm really confused why it's gone. Now, the FaceTime window stays as a long rectangle that covers part of the content I'm trying to share (unless I do full screen tile, but then I can't pull up any other windows during the call) and can't be made smaller than about a third of the screen. You can't resize the window or adjust its dimensions, so it ends up blocking the actual media you're trying to watch. Here are some feature requests/fixes that would greatly improve the FaceTime screen-share experience: Option to adjust the shared media volume independently of call audio. Disable/toggle the extreme automatic audio docking while screen-sharing Reintroduce the minimized “floating square” mode or allow full manual resizing and repositioning of the FaceTime window during screen-share sessions. Overall, this setup makes FaceTime screen-sharing basically unusable. The audio balance is so inconsistent that it’s easier to switch to Zoom or Google Meet, which both handle shared sound correctly and let you move the call window out of the way. Until these issues are fixed, there’s no practical reason to use FaceTime for shared viewing at all.
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441
Activity
Nov ’25
Unable to match music with shazamkit for Android
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?
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154
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May ’25