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

Audio Documentation

Posts under Audio subtopic

Post

Replies

Boosts

Views

Activity

_MPRemoteCommandEventDispatch crashes on iOS 26.x devices.
I'm seeing crashes in _MPRemoteCommandEventDispatch on iOS 26.x devices in 3 apps. According to Bugsnag logs they are: NSInternalInconsistencyException: event dispatch <_MPRemoteCommandEventDispatch: <MPRemoteCommandEvent: 0x11c049500 commandID=THV0 command=<MPRemoteCommand: 0x109ad1ea0 type=Play (0) enabled=YES handlers=[0x109b6a310]> sourceID=(null) ([HostedRoutingSessionDataSource] handleControlSendingCommand<2W5E>)> state:201> deallocated without calling continuation I attached a log from Xcode organizer matching Bugsnag crash. mpr_remote_command_event.crash When I set the brakpoint on the -[_MPRemoteCommandEventDispatch dealloc] I can see it it's hit every time I tap play or pause on locked screen play button. Thread 0 Crashed: 0 libsystem_kernel.dylib 0x00000002370420cc __pthread_kill + 8 (:-1) 1 libsystem_pthread.dylib 0x00000001e975c810 pthread_kill + 268 (pthread.c:1721) 2 libsystem_c.dylib 0x0000000198f8ff64 abort + 124 (abort.c:122) 3 libc++abi.dylib 0x000000018a7cf808 __abort_message + 132 (abort_message.cpp:66) 4 libc++abi.dylib 0x000000018a7be484 demangling_terminate_handler() + 304 (cxa_default_handlers.cpp:76) 5 libobjc.A.dylib 0x000000018a6cff78 _objc_terminate() + 156 (objc-exception.mm:496) 6 xxxxxxxxxxxxxx 0x00000001003a7db8 CPPExceptionTerminate() + 416 (BSG_KSCrashSentry_CPPException.mm:156) 7 libc++abi.dylib 0x000000018a7cebdc std::__terminate(void (*)()) + 16 (cxa_handlers.cpp:59) 8 libc++abi.dylib 0x000000018a7ceb80 std::terminate() + 108 (cxa_handlers.cpp:88) 9 CoreFoundation 0x000000018d7341c4 __CFRunLoopPerCalloutARPEnd + 256 (CFRunLoop.c:769) 10 CoreFoundation 0x000000018d70bb5c __CFRunLoopRun + 1976 (CFRunLoop.c:3179) 11 CoreFoundation 0x000000018d70aa6c _CFRunLoopRunSpecificWithOptions + 532 (CFRunLoop.c:3462) 12 GraphicsServices 0x000000022e31c498 GSEventRunModal + 120 (GSEvent.c:2049) 13 UIKitCore 0x00000001930ceba4 -[UIApplication _run] + 792 (UIApplication.m:3902) 14 UIKitCore 0x0000000193077a78 UIApplicationMain + 336 (UIApplication.m:5577) 15 xxxxxxxxxxxxxx 0x00000001000c0134 main + 308 (main.swift:15) 16 dyld 0x000000018a722e28 start + 7116 (dyldMain.cpp:1477) Is the crash happening when the app is being terminated? Thank you!
4
2
941
2w
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
4
0
1.2k
Nov ’25
Delay in Microphone Input When Talking While Receiving Audio in PTT Framework (Full Duplex Mode)
Context: I am currently developing an app using the Push-to-Talk (PTT) framework. I have reviewed both the PTT framework documentation and the CallKit demo project to better understand how to properly manage audio session activation and AVAudioEngine setup. I am not activating the audio session manually. The audio session configuration is handled in the incomingPushResult or didBeginTransmitting callbacks from the PTChannelManagerDelegate. I am using a single AVAudioEngine instance for both input and playback. The engine is started in the didActivate callback from the PTChannelManagerDelegate. When I receive a push in full duplex mode, I set the active participant to the user who is speaking. Issue When I attempt to talk while the other participant is already speaking, my input tap on the input node takes a few seconds to return valid PCM audio data. Initially, it returns an empty PCM audio block. Details: The audio session is already active and configured with .playAndRecord. The input tap is already installed when the engine is started. When I talk from a neutral state (no one is speaking), the system plays the standard "microphone activation" tone, which covers this initial delay. However, this does not happen when I am already receiving audio. Assumptions / Current Setup Because the audio session is active in play and record, I assumed that microphone input would be available immediately, even while receiving audio. However, there seems to be a delay before valid input is delivered to the tap, only occurring when switching from a receive state to simultaneously talking. Questions Is this expected behavior when using the PTT framework in full duplex mode with a shared AVAudioEngine? Should I be restarting or reconfiguring the engine or audio session when beginning to talk while receiving audio? Is there a recommended pattern for managing microphone readiness in this scenario to avoid the initial empty PCM buffer? Would using separate engines for input and output improve responsiveness? I would like to confirm the correct approach to handling simultaneous talk and receive in full duplex mode using PTT framework and AVAudioEngine. Specifically, I need guidance on ensuring the microphone is ready to capture audio immediately without the delay seen in my current implementation. Relevant Code Snippets Engine Setup func setup() { let input = audioEngine.inputNode do { try input.setVoiceProcessingEnabled(true) } catch { print("Could not enable voice processing \(error)") return } input.isVoiceProcessingAGCEnabled = false let output = audioEngine.outputNode let mainMixer = audioEngine.mainMixerNode audioEngine.connect(pttPlayerNode, to: mainMixer, format: outputFormat) audioEngine.connect(beepNode, to: mainMixer, format: outputFormat) audioEngine.connect(mainMixer, to: output, format: outputFormat) // Initialize converters converter = AVAudioConverter(from: inputFormat, to: outputFormat)! f32ToInt16Converter = AVAudioConverter(from: outputFormat, to: inputFormat)! audioEngine.prepare() } Input Tap Installation func installTap() { guard AudioHandler.shared.checkMicrophonePermission() else { print("Microphone not granted for recording") return } guard !isInputTapped else { print("[AudioEngine] Input is already tapped!") return } let input = audioEngine.inputNode let microphoneFormat = input.inputFormat(forBus: 0) let microphoneDownsampler = AVAudioConverter(from: microphoneFormat, to: outputFormat)! let desiredFormat = outputFormat let inputFramesNeeded = AVAudioFrameCount((Double(OpusCodec.DECODED_PACKET_NUM_SAMPLES) * microphoneFormat.sampleRate) / desiredFormat.sampleRate) input.installTap(onBus: 0, bufferSize: inputFramesNeeded, format: input.inputFormat(forBus: 0)) { [weak self] buffer, when in guard let self = self else { return } // Output buffer: 1920 frames at 16kHz guard let outputBuffer = AVAudioPCMBuffer(pcmFormat: desiredFormat, frameCapacity: AVAudioFrameCount(OpusCodec.DECODED_PACKET_NUM_SAMPLES)) else { return } outputBuffer.frameLength = outputBuffer.frameCapacity let inputBlock: AVAudioConverterInputBlock = { inNumPackets, outStatus in outStatus.pointee = .haveData return buffer } var error: NSError? let converterResult = microphoneDownsampler.convert(to: outputBuffer, error: &error, withInputFrom: inputBlock) if converterResult != .haveData { DebugLogger.shared.print("Downsample error \(converterResult)") } else { self.handleDownsampledBuffer(outputBuffer) } } isInputTapped = true }
4
0
510
Aug ’25
How to use the SpeechDetector Module
I am trying to use SpeechDetector Module in Speech framework along with SpeechTranscriber. and it is giving me an error Cannot convert value of type 'SpeechDetector' to expected element type 'Array.ArrayLiteralElement' (aka 'any SpeechModule') Below is how I am using it let speechDetector = Speech.SpeechDetector() let transcriber = SpeechTranscriber(locale: Locale.current, transcriptionOptions: [], reportingOptions: [.volatileResults], attributeOptions: [.audioTimeRange]) speechAnalyzer = try SpeechAnalyzer(modules: [transcriber,speechDetector])
4
2
487
Aug ’25
On iOS 18, Mandarin is read aloud as Cantonese
Please include the line below in follow-up emails for this request. Case-ID: 11089799 When using AVSpeechUtterance and setting it to play in Mandarin, if Siri is set to Cantonese on iOS 18, it will be played in Cantonese. There is no such issue on iOS 17 and 16. 1.let utterance = AVSpeechUtterance(string: textView.text) let voice = AVSpeechSynthesisVoice(language: "zh-CN") utterance.voice = voice 2.In the phone settings, Siri is set to Cantonese
4
1
816
Mar ’26
Audio Unit v3 host v2 third party plugins
Hi, I have just implemented an Audio Unit v3 host. AgsAudioUnitPlugin *audio_unit_plugin; AVAudioUnitComponentManager *audio_unit_component_manager; NSArray<AVAudioUnitComponent *> *av_component_arr; AudioComponentDescription description; guint i, i_stop; if(!AGS_AUDIO_UNIT_MANAGER(audio_unit_manager)){ return; } audio_unit_component_manager = [AVAudioUnitComponentManager sharedAudioUnitComponentManager]; /* effects */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_Effect; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } /* instruments */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_MusicDevice; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } But this doesn't show me Audio Unit v2 plugins, why? regards, Joël
3
0
755
Aug ’25
iOS 17 camera capture assertions and issues
Hello, Starting in iOS 17, our application started having some issue publishing to our video session. More specifically the video capture seems to be broken in some, but not all sessions. What's troubling is that we're seeing that it fails consistently every 4 sessions. It also fails silently, without reporting any problems to the app. We only notice that there are no frames being rendered or sent to the remote devices. Here's what shows-up in the console: <<<< FigCaptureSourceRemote >>>> Fig assert: "! storage->connectionDied" at bail (FigCaptureSourceRemote.m:235) - (err=0) <<<< FigCaptureSourceRemote >>>> Fig assert: "err == 0 " at bail (FigCaptureSourceRemote.m:253) - (err=-16453) Anyone seeing this? Any idea what could be the cause? Our sessions work perfectly on iOS16 and below. Thanks
3
1
1.4k
Oct ’25
AVAudioSession.outputVolume not reporting correctly in iOS 18+ devices
I’m using the shared instance of AVAudioSession. After activating it with .setActive(true), I observe the outputVolume, and it correctly reports the device’s volume. However, after deactivating the session using .setActive(false), changing the volume, and then reactivating it again, the outputVolume returns the previous volume (before deactivation), not the current device volume. The correct volume is only reported after the user manually changes it again using physical buttons or Control Center, which triggers the observer. What I need is a way to retrieve the actual current device volume immediately after reactivating the audio session, even on the second and subsequent activations. Disabling and re-enabling the audio session is essential to how my application functions. I’ve tested this behavior with my colleagues, and the issue is consistently reproducible on iOS 18.0.1, iOS 18.1, iOS 18.3, iOS 18.5 and iOS 18.6.2. On devices running iOS 17.6.1 and iOS 16.0.3, outputVolume correctly reflects the current volume immediately after calling .setActive(true) multiple times.
3
1
353
Feb ’26
AVAudioUnit host - PCM buffer output silent
Hi, I just started to develop audio unit hosting support in my application. Offline rendering seems to work except that I hear no output, but why? I suspect with the player goes something wrong. I connect to CoreAudio in a different location in the code. Here are some error messages I faced so far: 2025-08-14 19:42:04.132930+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:42:04.151171+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:43:08.344530+0200 com.gsequencer.GSequencer[34358:18614927] AUAudioUnit.mm:1417 Cannot set maximumFramesToRender while render resources allocated. 2025-08-14 19:43:08.346583+0200 com.gsequencer.GSequencer[34358:18614927] [avae] AVAEInternal.h:104 [AVAudioSequencer.mm:121:-[AVAudioSequencer(AVAudioSequencer_Player) startAndReturnError:]: (impl->Start()): error -10852 ** (<unknown>:34358): WARNING **: 19:43:08.346: error during audio sequencer start - -10852 I have implemented an AVAudioEngine based AudioUnit host. Here I instantiate player and effect: /* audio engine */ audio_engine = [[AVAudioEngine alloc] init]; fx_audio_unit_audio->audio_engine = (gpointer) audio_engine; av_format = (AVAudioFormat *) fx_audio_unit_audio->av_format; /* av audio player node */ av_audio_player_node = [[AVAudioPlayerNode alloc] init]; /* av audio unit */ av_audio_unit_effect = [[AVAudioUnitEffect alloc] initWithAudioComponentDescription:[((AVAudioUnitComponent *) AGS_AUDIO_UNIT_PLUGIN(base_plugin)->component) audioComponentDescription]]; av_audio_unit = (AVAudioUnit *) av_audio_unit_effect; fx_audio_unit_audio->av_audio_unit = av_audio_unit; /* audio sequencer */ av_audio_sequencer = [[AVAudioSequencer alloc] initWithAudioEngine:audio_engine]; fx_audio_unit_audio->av_audio_sequencer = (gpointer) av_audio_sequencer; /* output node */ [[AVAudioOutputNode alloc] init]; /* audio player and audio unit */ [audio_engine attachNode:av_audio_player_node]; [audio_engine attachNode:av_audio_unit]; [audio_engine connect:av_audio_player_node to:av_audio_unit format:av_format]; [audio_engine connect:av_audio_unit to:[audio_engine outputNode] format:av_format]; ns_error = NULL; [audio_engine enableManualRenderingMode:AVAudioEngineManualRenderingModeOffline format:av_format maximumFrameCount:buffer_size error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("enable manual rendering mode error - %d", [ns_error code]); } ns_error = NULL; [[av_audio_unit AUAudioUnit] allocateRenderResourcesAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("Audio Unit allocate render resources returned error - ErrorCode %d", [ns_error code]); } Then I render in a dedicated thread. ns_error = NULL; [audio_engine startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio engine start - %d", [ns_error code]); } [av_audio_sequencer prepareToPlay]; ns_error = NULL; [av_audio_sequencer startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio sequencer start - %d", [ns_error code]); } [av_audio_player_node play]; while(is_running){ /* pre sync */ /* IO buffers */ av_output_buffer = (AVAudioPCMBuffer *) scope_data->av_output_buffer; av_input_buffer = (AVAudioPCMBuffer *) scope_data->av_input_buffer; /* fill input buffer */ /* schedule av input buffer */ frame_position = 0; // (gint64) ((note_offset * absolute_delay) + delay_counter) * buffer_size; av_audio_player_node = (AVAudioPlayerNode *) fx_audio_unit_audio->av_audio_player_node; AVAudioTime *av_audio_time = [[AVAudioTime alloc] initWithHostTime:frame_position sampleTime:frame_position atRate:((double) samplerate)]; [av_audio_player_node scheduleBuffer:av_input_buffer atTime:av_audio_time options:0 completionHandler:nil]; /* render */ ns_error = NULL; status = [audio_engine renderOffline:AGS_FX_AUDIO_UNIT_AUDIO_FIXED_BUFFER_SIZE toBuffer:av_output_buffer error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("render offline error - %d", [ns_error code]); } } regards, Joël
3
0
511
Aug ’25
Handling AVAudioEngine Configuration Change
Hi all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!
3
0
957
Oct ’25
AppleAVBAudio assertion information
Hi, I'm currently developping an AVB hardware device, and I'm currently stuck because because the apple AVB stack is throwing me errors without much informations. Is there any way to have more information about these assertions and why they are happening ? Furtermore is there any documentation on theAppleAVBAudio module ? It would be very handy Here are the logs shown in the console: Filtering the log data using "process == "coreaudiod"" Timestamp Thread Type Activity PID TTL 2025-12-05 15:44:27.087043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.087545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.088043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.088546+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.089043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.089545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.090043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.090545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.091043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.091545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.092044+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.092544+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.093044+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.093552+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.094050+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.094543+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533
3
0
311
Jan ’26
AVAudioSessionCategoryPlayback is not allowed while CallKit call is active
We require assistance in resolving a critical audio design conflict within our Push-to-Talk (PTT) application. Our current volume amplification strategy—which relies on applying a GAIN factor to PCM samples in conjunction with setting the AVAudioSession category to Playback—is working successfully when PTT is used independently. However, upon integrating and reporting the same PTT call through the CallKit framework, this amplification effect is lost. The CallKit integration appears to be forcing a different, non-amplifying audio session category or configuration, negatively impacting the user's perceived call volume. We need guidance on how to maintain the AVAudioSessionCategoryPlayback setting, or an equivalent high-volume configuration, while operating under the control of CallKit.
3
0
418
Nov ’25
ScreenCaptureKit System Audio Capture Crashes with EXC_BAD_ACCESS
Bug Report: ScreenCaptureKit System Audio Capture Crashes with EXC_BAD_ACCESS Summary When using ScreenCaptureKit to capture system audio for extended periods, the application crashes with EXC_BAD_ACCESS in Swift's error handling runtime. The crash occurs in swift_getErrorValue when trying to process an error from the SCStream delegate method didStopWithError. This appears to be a framework-level issue in ScreenCaptureKit or its underlying ReplayKit implementation. Environment macOS Sonoma 14.6.1 Swift 5.8 ScreenCaptureKit framework Detailed Description Our application captures system audio using ScreenCaptureKit's audio capture capabilities. After successfully capturing for several minutes (typically after 3-4 segments of 60-second recordings), the application crashes with an EXC_BAD_ACCESS error. The crash happens when the Swift runtime attempts to process an error in the SCStreamDelegate.stream(_:didStopWithError:) method. The crash consistently occurs in swift_getErrorValue when attempting to access the class of what appears to be a null object. This suggests that the error being passed from the system framework to our delegate method is malformed or contains invalid memory. Steps to Reproduce Create an SCStream with audio capture enabled Add audio output to the stream Start capture and write audio data to disk Allow the capture to run for several minutes (3-5 minutes typically triggers the issue) The app will crash with EXC_BAD_ACCESS in swift_getErrorValue Code Sample func stream(_ stream: SCStream, didStopWithError error: Error) { print("Stream stopped with error: \(error)") // Crash occurs before this line executes } func stream(_ stream: SCStream, didOutputSampleBuffer sampleBuffer: CMSampleBuffer, of type: SCStreamOutputType) { guard type == .audio, sampleBuffer.isValid else { return } // Process audio data... } Expected Behavior The error should be properly propagated to the delegate method, allowing for graceful error handling and recovery. Actual Behavior The application crashes with EXC_BAD_ACCESS when the Swift runtime attempts to process the error in swift_getErrorValue. Crash Log Details Thread #35, queue = 'com.apple.NSXPCConnection.m-user.com.apple.replayd', stop reason = EXC_BAD_ACCESS (code=1, address=0x0) frame #0: 0x0000000194c3088c libswiftCore.dylib`swift::_swift_getClass(void const*) + 8 frame #1: 0x0000000194c30104 libswiftCore.dylib`swift_getErrorValue + 40 frame #2: 0x00000001057fba30 shadow`NewScreenCaptureService.stream(stream=0x0000600002de6700, error=Swift.Error @ 0x000000016b7b5e30) at NEW+ScreenCaptureService.swift:365:15 frame #3: 0x00000001057fc050 shadow`@objc NewScreenCaptureService.stream(_:didStopWithError:) at <compiler-generated>:0 frame #4: 0x0000000219ec5ca0 ScreenCaptureKit`-[SCStreamManager stream:didStopWithError:] + 456 frame #5: 0x00000001ca68a5cc ReplayKit`-[RPScreenRecorder stream:didStopWithError:] + 84 frame #6: 0x00000001ca696ff8 ReplayKit`-[RPDaemonProxy stream:didStopWithError:] + 224 Printing description of stream._streamQueue: error: ObjectiveC.id:4294967281:18: note: 'id' has been explicitly marked unavailable here public typealias id = AnyObject ^ error: /var/folders/v4/3xg1hmp93gjd8_xlzmryf_wm0000gn/T/expr23-dfa421..cpp:1:65: 'id' is unavailable in Swift: 'id' is not available in Swift; use 'Any' Swift._DebuggerSupport.stringForPrintObject(Swift.UnsafePointer<id>(bitPattern: 0x104ae08c0)!.pointee) ^~ ObjectiveC.id:2:18: note: 'id' has been explicitly marked unavailable here public typealias id = AnyObject ^ warning: /var/folders/v4/3xg1hmp93gjd8_xlzmryf_wm0000gn/T/expr23-dfa421..cpp:5:7: initialization of variable '$__lldb_error_result' was never used; consider replacing with assignment to '_' or removing it var $__lldb_error_result = __lldb_tmp_error ~~~~^~~~~~~~~~~~~~~~~~~~ _ Before the crash, we observed this error message in the console: [ERROR] *****SCStream*****RemoteAudioQueueOperationHandlerWithError:1015 Error received from the remote queue -16665 Additional Context The issue occurs consistently after approximately 3-4 successful audio segment recordings of 60 seconds each Commenting out custom segment rotation logic does not prevent the crash The crash involves XPC communication with Apple's ReplayKit daemon The error appears to be corrupted or malformed when crossing the XPC boundary Workarounds Attempted Added proper thread safety for all published properties using DispatchQueue.main.async Implemented more robust error handling in the delegate methods None of these approaches prevented the crash since it occurs at the Swift runtime level before our code executes. Impact This issue prevents reliable long-duration audio capture using ScreenCaptureKit. This bug significantly limits the usefulness of ScreenCaptureKit for any application requiring continuous system audio capture for more than a few minutes. Perhaps this issue might be related to a macOS bug where the system dialog indicates that the screen is being shared, even though nothing is actually being shared. Moreover, when attempting to stop sharing, nothing happens.
3
0
813
1w
SpeechTranscriber/SpeechAnalyzer being relatively slow compared to FoundationModel and TTS
So, I've been wondering how fast a an offline STT -> ML Prompt -> TTS roundtrip would be. Interestingly, for many tests, the SpeechTranscriber (STT) takes the bulk of the time, compared to generating a FoundationModel response and creating the Audio using TTS. E.g. InteractionStatistics: - listeningStarted: 21:24:23 4480 2423 - timeTillFirstAboveNoiseFloor: 01.794 - timeTillLastNoiseAboveFloor: 02.383 - timeTillFirstSpeechDetected: 02.399 - timeTillTranscriptFinalized: 04.510 - timeTillFirstMLModelResponse: 04.938 - timeTillMLModelResponse: 05.379 - timeTillTTSStarted: 04.962 - timeTillTTSFinished: 11.016 - speechLength: 06.054 - timeToResponse: 02.578 - transcript: This is a test. - mlModelResponse: Sure! I'm ready to help with your test. What do you need help with? Here, between my audio input ending and the Text-2-Speech starting top play (using AVSpeechUtterance) the total response time was 2.5s. Of that time, it took the SpeechAnalyzer 2.1s to get the transcript finalized, FoundationModel only took 0.4s to respond (and TTS started playing nearly instantly). I'm already using reportingOptions: [.volatileResults, .fastResults] so it's probably as fast as possible right now? I'm just surprised the STT takes so much longer compared to the other parts (all being CoreML based, aren't they?)
3
0
723
1w
SpeechTranscriber not providing audioTimeRange for most results
I started playing which transcription of audio files on macOS today, latest beta of Xcode and latest beta of Tahoe. Transcription itself works really well, but for some reason the majority of the results contain no audioTimeRange. I got 22 single-word results with time ranges, spread out all over total file of 53 minutes. Is there something I can do to improve this? To my understanding, I have followed sample code and instructions very closely, but the SwiftTranscriptionSampleApp and other examples I've seen lead me to believe I should be getting a lot more time ranges than I actually do.
3
0
206
Aug ’25
Can backgrounded apps record audio?
I'd like to find out: Can backgrounded apps record audio? In the past as I recall, I found that backgrounded apps were pretty restricted and couldn't do much of anything. However I'm not familiar with the current state of affairs. With iOS 15.8 and above, can backgrounded apps record audio if they've been given permission by the user to access the microphone? Thanks.
3
0
579
Jan ’26
Mac OS Tahoe 26.0 (25A354) Sound Glitches When opening the simulator app
Hey there, I just upgraded to Mac OS Tahoe ,son an apple MacBook Pro 2019 16inch. am using IntellijIDEA and Flutter to develop a mobile app which I test on the simulator app running iOS 18.4 . the issue: when I start the simulator app. ( while in the loading phase and in the operation phase as well ), the audio from an already open YouTube tab on safari (this happens on chrome browser as well). the sound glitches and becomes Noise. a fix I found online is to kill the audio deamon on Mac OS, This works using the command: "sudo killall coreaudiod" this kills the audio process, (while the emulator is operational), then the macOS restarts the audio deamon then the audio works fine alongside with the simulator being open. I just want to ask is there a permanent fix for this? is Apple working on a fix for this in the upcoming update?
3
5
1.3k
Oct ’25
Question about PT Framework channel tone behaviour
I've been wondering if there is a way to modify or even disable tones for indicating channel states. The behaviour regarding tones seems like a black box with little documentation. During migration to Apple's PT Framework we've noticed that there are few scenarios where a tone is played which doesn't match certain certifications. For example; moving from a channel to another produces a tone which would fail a test case. I understand the reasoning fully, as it marks that the channel is ready to transmit or receive, but this doesn't mirror the behaviour of TETRA which would be wanted in this case. I'm also wondering if there would be any way to directly communicate feedback regarding PT Framework?
3
0
418
Oct ’25
_MPRemoteCommandEventDispatch crashes on iOS 26.x devices.
I'm seeing crashes in _MPRemoteCommandEventDispatch on iOS 26.x devices in 3 apps. According to Bugsnag logs they are: NSInternalInconsistencyException: event dispatch <_MPRemoteCommandEventDispatch: <MPRemoteCommandEvent: 0x11c049500 commandID=THV0 command=<MPRemoteCommand: 0x109ad1ea0 type=Play (0) enabled=YES handlers=[0x109b6a310]> sourceID=(null) ([HostedRoutingSessionDataSource] handleControlSendingCommand<2W5E>)> state:201> deallocated without calling continuation I attached a log from Xcode organizer matching Bugsnag crash. mpr_remote_command_event.crash When I set the brakpoint on the -[_MPRemoteCommandEventDispatch dealloc] I can see it it's hit every time I tap play or pause on locked screen play button. Thread 0 Crashed: 0 libsystem_kernel.dylib 0x00000002370420cc __pthread_kill + 8 (:-1) 1 libsystem_pthread.dylib 0x00000001e975c810 pthread_kill + 268 (pthread.c:1721) 2 libsystem_c.dylib 0x0000000198f8ff64 abort + 124 (abort.c:122) 3 libc++abi.dylib 0x000000018a7cf808 __abort_message + 132 (abort_message.cpp:66) 4 libc++abi.dylib 0x000000018a7be484 demangling_terminate_handler() + 304 (cxa_default_handlers.cpp:76) 5 libobjc.A.dylib 0x000000018a6cff78 _objc_terminate() + 156 (objc-exception.mm:496) 6 xxxxxxxxxxxxxx 0x00000001003a7db8 CPPExceptionTerminate() + 416 (BSG_KSCrashSentry_CPPException.mm:156) 7 libc++abi.dylib 0x000000018a7cebdc std::__terminate(void (*)()) + 16 (cxa_handlers.cpp:59) 8 libc++abi.dylib 0x000000018a7ceb80 std::terminate() + 108 (cxa_handlers.cpp:88) 9 CoreFoundation 0x000000018d7341c4 __CFRunLoopPerCalloutARPEnd + 256 (CFRunLoop.c:769) 10 CoreFoundation 0x000000018d70bb5c __CFRunLoopRun + 1976 (CFRunLoop.c:3179) 11 CoreFoundation 0x000000018d70aa6c _CFRunLoopRunSpecificWithOptions + 532 (CFRunLoop.c:3462) 12 GraphicsServices 0x000000022e31c498 GSEventRunModal + 120 (GSEvent.c:2049) 13 UIKitCore 0x00000001930ceba4 -[UIApplication _run] + 792 (UIApplication.m:3902) 14 UIKitCore 0x0000000193077a78 UIApplicationMain + 336 (UIApplication.m:5577) 15 xxxxxxxxxxxxxx 0x00000001000c0134 main + 308 (main.swift:15) 16 dyld 0x000000018a722e28 start + 7116 (dyldMain.cpp:1477) Is the crash happening when the app is being terminated? Thank you!
Replies
4
Boosts
2
Views
941
Activity
2w
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
Replies
4
Boosts
0
Views
1.2k
Activity
Nov ’25
Delay in Microphone Input When Talking While Receiving Audio in PTT Framework (Full Duplex Mode)
Context: I am currently developing an app using the Push-to-Talk (PTT) framework. I have reviewed both the PTT framework documentation and the CallKit demo project to better understand how to properly manage audio session activation and AVAudioEngine setup. I am not activating the audio session manually. The audio session configuration is handled in the incomingPushResult or didBeginTransmitting callbacks from the PTChannelManagerDelegate. I am using a single AVAudioEngine instance for both input and playback. The engine is started in the didActivate callback from the PTChannelManagerDelegate. When I receive a push in full duplex mode, I set the active participant to the user who is speaking. Issue When I attempt to talk while the other participant is already speaking, my input tap on the input node takes a few seconds to return valid PCM audio data. Initially, it returns an empty PCM audio block. Details: The audio session is already active and configured with .playAndRecord. The input tap is already installed when the engine is started. When I talk from a neutral state (no one is speaking), the system plays the standard "microphone activation" tone, which covers this initial delay. However, this does not happen when I am already receiving audio. Assumptions / Current Setup Because the audio session is active in play and record, I assumed that microphone input would be available immediately, even while receiving audio. However, there seems to be a delay before valid input is delivered to the tap, only occurring when switching from a receive state to simultaneously talking. Questions Is this expected behavior when using the PTT framework in full duplex mode with a shared AVAudioEngine? Should I be restarting or reconfiguring the engine or audio session when beginning to talk while receiving audio? Is there a recommended pattern for managing microphone readiness in this scenario to avoid the initial empty PCM buffer? Would using separate engines for input and output improve responsiveness? I would like to confirm the correct approach to handling simultaneous talk and receive in full duplex mode using PTT framework and AVAudioEngine. Specifically, I need guidance on ensuring the microphone is ready to capture audio immediately without the delay seen in my current implementation. Relevant Code Snippets Engine Setup func setup() { let input = audioEngine.inputNode do { try input.setVoiceProcessingEnabled(true) } catch { print("Could not enable voice processing \(error)") return } input.isVoiceProcessingAGCEnabled = false let output = audioEngine.outputNode let mainMixer = audioEngine.mainMixerNode audioEngine.connect(pttPlayerNode, to: mainMixer, format: outputFormat) audioEngine.connect(beepNode, to: mainMixer, format: outputFormat) audioEngine.connect(mainMixer, to: output, format: outputFormat) // Initialize converters converter = AVAudioConverter(from: inputFormat, to: outputFormat)! f32ToInt16Converter = AVAudioConverter(from: outputFormat, to: inputFormat)! audioEngine.prepare() } Input Tap Installation func installTap() { guard AudioHandler.shared.checkMicrophonePermission() else { print("Microphone not granted for recording") return } guard !isInputTapped else { print("[AudioEngine] Input is already tapped!") return } let input = audioEngine.inputNode let microphoneFormat = input.inputFormat(forBus: 0) let microphoneDownsampler = AVAudioConverter(from: microphoneFormat, to: outputFormat)! let desiredFormat = outputFormat let inputFramesNeeded = AVAudioFrameCount((Double(OpusCodec.DECODED_PACKET_NUM_SAMPLES) * microphoneFormat.sampleRate) / desiredFormat.sampleRate) input.installTap(onBus: 0, bufferSize: inputFramesNeeded, format: input.inputFormat(forBus: 0)) { [weak self] buffer, when in guard let self = self else { return } // Output buffer: 1920 frames at 16kHz guard let outputBuffer = AVAudioPCMBuffer(pcmFormat: desiredFormat, frameCapacity: AVAudioFrameCount(OpusCodec.DECODED_PACKET_NUM_SAMPLES)) else { return } outputBuffer.frameLength = outputBuffer.frameCapacity let inputBlock: AVAudioConverterInputBlock = { inNumPackets, outStatus in outStatus.pointee = .haveData return buffer } var error: NSError? let converterResult = microphoneDownsampler.convert(to: outputBuffer, error: &error, withInputFrom: inputBlock) if converterResult != .haveData { DebugLogger.shared.print("Downsample error \(converterResult)") } else { self.handleDownsampledBuffer(outputBuffer) } } isInputTapped = true }
Replies
4
Boosts
0
Views
510
Activity
Aug ’25
How to use the SpeechDetector Module
I am trying to use SpeechDetector Module in Speech framework along with SpeechTranscriber. and it is giving me an error Cannot convert value of type 'SpeechDetector' to expected element type 'Array.ArrayLiteralElement' (aka 'any SpeechModule') Below is how I am using it let speechDetector = Speech.SpeechDetector() let transcriber = SpeechTranscriber(locale: Locale.current, transcriptionOptions: [], reportingOptions: [.volatileResults], attributeOptions: [.audioTimeRange]) speechAnalyzer = try SpeechAnalyzer(modules: [transcriber,speechDetector])
Replies
4
Boosts
2
Views
487
Activity
Aug ’25
On iOS 18, Mandarin is read aloud as Cantonese
Please include the line below in follow-up emails for this request. Case-ID: 11089799 When using AVSpeechUtterance and setting it to play in Mandarin, if Siri is set to Cantonese on iOS 18, it will be played in Cantonese. There is no such issue on iOS 17 and 16. 1.let utterance = AVSpeechUtterance(string: textView.text) let voice = AVSpeechSynthesisVoice(language: "zh-CN") utterance.voice = voice 2.In the phone settings, Siri is set to Cantonese
Replies
4
Boosts
1
Views
816
Activity
Mar ’26
Audio Unit v3 host v2 third party plugins
Hi, I have just implemented an Audio Unit v3 host. AgsAudioUnitPlugin *audio_unit_plugin; AVAudioUnitComponentManager *audio_unit_component_manager; NSArray<AVAudioUnitComponent *> *av_component_arr; AudioComponentDescription description; guint i, i_stop; if(!AGS_AUDIO_UNIT_MANAGER(audio_unit_manager)){ return; } audio_unit_component_manager = [AVAudioUnitComponentManager sharedAudioUnitComponentManager]; /* effects */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_Effect; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } /* instruments */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_MusicDevice; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } But this doesn't show me Audio Unit v2 plugins, why? regards, Joël
Replies
3
Boosts
0
Views
755
Activity
Aug ’25
iOS 17 camera capture assertions and issues
Hello, Starting in iOS 17, our application started having some issue publishing to our video session. More specifically the video capture seems to be broken in some, but not all sessions. What's troubling is that we're seeing that it fails consistently every 4 sessions. It also fails silently, without reporting any problems to the app. We only notice that there are no frames being rendered or sent to the remote devices. Here's what shows-up in the console: <<<< FigCaptureSourceRemote >>>> Fig assert: "! storage->connectionDied" at bail (FigCaptureSourceRemote.m:235) - (err=0) <<<< FigCaptureSourceRemote >>>> Fig assert: "err == 0 " at bail (FigCaptureSourceRemote.m:253) - (err=-16453) Anyone seeing this? Any idea what could be the cause? Our sessions work perfectly on iOS16 and below. Thanks
Replies
3
Boosts
1
Views
1.4k
Activity
Oct ’25
AVAudioSession.outputVolume not reporting correctly in iOS 18+ devices
I’m using the shared instance of AVAudioSession. After activating it with .setActive(true), I observe the outputVolume, and it correctly reports the device’s volume. However, after deactivating the session using .setActive(false), changing the volume, and then reactivating it again, the outputVolume returns the previous volume (before deactivation), not the current device volume. The correct volume is only reported after the user manually changes it again using physical buttons or Control Center, which triggers the observer. What I need is a way to retrieve the actual current device volume immediately after reactivating the audio session, even on the second and subsequent activations. Disabling and re-enabling the audio session is essential to how my application functions. I’ve tested this behavior with my colleagues, and the issue is consistently reproducible on iOS 18.0.1, iOS 18.1, iOS 18.3, iOS 18.5 and iOS 18.6.2. On devices running iOS 17.6.1 and iOS 16.0.3, outputVolume correctly reflects the current volume immediately after calling .setActive(true) multiple times.
Replies
3
Boosts
1
Views
353
Activity
Feb ’26
AVAudioUnit host - PCM buffer output silent
Hi, I just started to develop audio unit hosting support in my application. Offline rendering seems to work except that I hear no output, but why? I suspect with the player goes something wrong. I connect to CoreAudio in a different location in the code. Here are some error messages I faced so far: 2025-08-14 19:42:04.132930+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:42:04.151171+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:43:08.344530+0200 com.gsequencer.GSequencer[34358:18614927] AUAudioUnit.mm:1417 Cannot set maximumFramesToRender while render resources allocated. 2025-08-14 19:43:08.346583+0200 com.gsequencer.GSequencer[34358:18614927] [avae] AVAEInternal.h:104 [AVAudioSequencer.mm:121:-[AVAudioSequencer(AVAudioSequencer_Player) startAndReturnError:]: (impl->Start()): error -10852 ** (<unknown>:34358): WARNING **: 19:43:08.346: error during audio sequencer start - -10852 I have implemented an AVAudioEngine based AudioUnit host. Here I instantiate player and effect: /* audio engine */ audio_engine = [[AVAudioEngine alloc] init]; fx_audio_unit_audio->audio_engine = (gpointer) audio_engine; av_format = (AVAudioFormat *) fx_audio_unit_audio->av_format; /* av audio player node */ av_audio_player_node = [[AVAudioPlayerNode alloc] init]; /* av audio unit */ av_audio_unit_effect = [[AVAudioUnitEffect alloc] initWithAudioComponentDescription:[((AVAudioUnitComponent *) AGS_AUDIO_UNIT_PLUGIN(base_plugin)->component) audioComponentDescription]]; av_audio_unit = (AVAudioUnit *) av_audio_unit_effect; fx_audio_unit_audio->av_audio_unit = av_audio_unit; /* audio sequencer */ av_audio_sequencer = [[AVAudioSequencer alloc] initWithAudioEngine:audio_engine]; fx_audio_unit_audio->av_audio_sequencer = (gpointer) av_audio_sequencer; /* output node */ [[AVAudioOutputNode alloc] init]; /* audio player and audio unit */ [audio_engine attachNode:av_audio_player_node]; [audio_engine attachNode:av_audio_unit]; [audio_engine connect:av_audio_player_node to:av_audio_unit format:av_format]; [audio_engine connect:av_audio_unit to:[audio_engine outputNode] format:av_format]; ns_error = NULL; [audio_engine enableManualRenderingMode:AVAudioEngineManualRenderingModeOffline format:av_format maximumFrameCount:buffer_size error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("enable manual rendering mode error - %d", [ns_error code]); } ns_error = NULL; [[av_audio_unit AUAudioUnit] allocateRenderResourcesAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("Audio Unit allocate render resources returned error - ErrorCode %d", [ns_error code]); } Then I render in a dedicated thread. ns_error = NULL; [audio_engine startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio engine start - %d", [ns_error code]); } [av_audio_sequencer prepareToPlay]; ns_error = NULL; [av_audio_sequencer startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio sequencer start - %d", [ns_error code]); } [av_audio_player_node play]; while(is_running){ /* pre sync */ /* IO buffers */ av_output_buffer = (AVAudioPCMBuffer *) scope_data->av_output_buffer; av_input_buffer = (AVAudioPCMBuffer *) scope_data->av_input_buffer; /* fill input buffer */ /* schedule av input buffer */ frame_position = 0; // (gint64) ((note_offset * absolute_delay) + delay_counter) * buffer_size; av_audio_player_node = (AVAudioPlayerNode *) fx_audio_unit_audio->av_audio_player_node; AVAudioTime *av_audio_time = [[AVAudioTime alloc] initWithHostTime:frame_position sampleTime:frame_position atRate:((double) samplerate)]; [av_audio_player_node scheduleBuffer:av_input_buffer atTime:av_audio_time options:0 completionHandler:nil]; /* render */ ns_error = NULL; status = [audio_engine renderOffline:AGS_FX_AUDIO_UNIT_AUDIO_FIXED_BUFFER_SIZE toBuffer:av_output_buffer error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("render offline error - %d", [ns_error code]); } } regards, Joël
Replies
3
Boosts
0
Views
511
Activity
Aug ’25
Handling AVAudioEngine Configuration Change
Hi all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!
Replies
3
Boosts
0
Views
957
Activity
Oct ’25
SpeechTranscriber on Simulator
I am trying to use SpeechTranscriber from Speech framework. Is it possible to use it on Simulator of iOS 26 (Mac OS Tahoe)? Function "supportedLocales" returns an empty array.
Replies
3
Boosts
2
Views
995
Activity
Nov ’25
CoreMIDI: neither syslog nor unified logging works.
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!
Replies
3
Boosts
0
Views
345
Activity
Oct ’25
AppleAVBAudio assertion information
Hi, I'm currently developping an AVB hardware device, and I'm currently stuck because because the apple AVB stack is throwing me errors without much informations. Is there any way to have more information about these assertions and why they are happening ? Furtermore is there any documentation on theAppleAVBAudio module ? It would be very handy Here are the logs shown in the console: Filtering the log data using "process == "coreaudiod"" Timestamp Thread Type Activity PID TTL 2025-12-05 15:44:27.087043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.087545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.088043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.088546+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.089043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.089545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.090043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.090545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.091043+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.091545+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.092044+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.092544+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.093044+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.093552+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.094050+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533 2025-12-05 15:44:27.094543+0100 0x15ae74 Default 0x0 12965 0 coreaudiod: (AppleAVBAudio) Assert: <private> (value 0x0 0), <private> file: <private>, line: 1533
Replies
3
Boosts
0
Views
311
Activity
Jan ’26
AVAudioSessionCategoryPlayback is not allowed while CallKit call is active
We require assistance in resolving a critical audio design conflict within our Push-to-Talk (PTT) application. Our current volume amplification strategy—which relies on applying a GAIN factor to PCM samples in conjunction with setting the AVAudioSession category to Playback—is working successfully when PTT is used independently. However, upon integrating and reporting the same PTT call through the CallKit framework, this amplification effect is lost. The CallKit integration appears to be forcing a different, non-amplifying audio session category or configuration, negatively impacting the user's perceived call volume. We need guidance on how to maintain the AVAudioSessionCategoryPlayback setting, or an equivalent high-volume configuration, while operating under the control of CallKit.
Replies
3
Boosts
0
Views
418
Activity
Nov ’25
ScreenCaptureKit System Audio Capture Crashes with EXC_BAD_ACCESS
Bug Report: ScreenCaptureKit System Audio Capture Crashes with EXC_BAD_ACCESS Summary When using ScreenCaptureKit to capture system audio for extended periods, the application crashes with EXC_BAD_ACCESS in Swift's error handling runtime. The crash occurs in swift_getErrorValue when trying to process an error from the SCStream delegate method didStopWithError. This appears to be a framework-level issue in ScreenCaptureKit or its underlying ReplayKit implementation. Environment macOS Sonoma 14.6.1 Swift 5.8 ScreenCaptureKit framework Detailed Description Our application captures system audio using ScreenCaptureKit's audio capture capabilities. After successfully capturing for several minutes (typically after 3-4 segments of 60-second recordings), the application crashes with an EXC_BAD_ACCESS error. The crash happens when the Swift runtime attempts to process an error in the SCStreamDelegate.stream(_:didStopWithError:) method. The crash consistently occurs in swift_getErrorValue when attempting to access the class of what appears to be a null object. This suggests that the error being passed from the system framework to our delegate method is malformed or contains invalid memory. Steps to Reproduce Create an SCStream with audio capture enabled Add audio output to the stream Start capture and write audio data to disk Allow the capture to run for several minutes (3-5 minutes typically triggers the issue) The app will crash with EXC_BAD_ACCESS in swift_getErrorValue Code Sample func stream(_ stream: SCStream, didStopWithError error: Error) { print("Stream stopped with error: \(error)") // Crash occurs before this line executes } func stream(_ stream: SCStream, didOutputSampleBuffer sampleBuffer: CMSampleBuffer, of type: SCStreamOutputType) { guard type == .audio, sampleBuffer.isValid else { return } // Process audio data... } Expected Behavior The error should be properly propagated to the delegate method, allowing for graceful error handling and recovery. Actual Behavior The application crashes with EXC_BAD_ACCESS when the Swift runtime attempts to process the error in swift_getErrorValue. Crash Log Details Thread #35, queue = 'com.apple.NSXPCConnection.m-user.com.apple.replayd', stop reason = EXC_BAD_ACCESS (code=1, address=0x0) frame #0: 0x0000000194c3088c libswiftCore.dylib`swift::_swift_getClass(void const*) + 8 frame #1: 0x0000000194c30104 libswiftCore.dylib`swift_getErrorValue + 40 frame #2: 0x00000001057fba30 shadow`NewScreenCaptureService.stream(stream=0x0000600002de6700, error=Swift.Error @ 0x000000016b7b5e30) at NEW+ScreenCaptureService.swift:365:15 frame #3: 0x00000001057fc050 shadow`@objc NewScreenCaptureService.stream(_:didStopWithError:) at <compiler-generated>:0 frame #4: 0x0000000219ec5ca0 ScreenCaptureKit`-[SCStreamManager stream:didStopWithError:] + 456 frame #5: 0x00000001ca68a5cc ReplayKit`-[RPScreenRecorder stream:didStopWithError:] + 84 frame #6: 0x00000001ca696ff8 ReplayKit`-[RPDaemonProxy stream:didStopWithError:] + 224 Printing description of stream._streamQueue: error: ObjectiveC.id:4294967281:18: note: 'id' has been explicitly marked unavailable here public typealias id = AnyObject ^ error: /var/folders/v4/3xg1hmp93gjd8_xlzmryf_wm0000gn/T/expr23-dfa421..cpp:1:65: 'id' is unavailable in Swift: 'id' is not available in Swift; use 'Any' Swift._DebuggerSupport.stringForPrintObject(Swift.UnsafePointer<id>(bitPattern: 0x104ae08c0)!.pointee) ^~ ObjectiveC.id:2:18: note: 'id' has been explicitly marked unavailable here public typealias id = AnyObject ^ warning: /var/folders/v4/3xg1hmp93gjd8_xlzmryf_wm0000gn/T/expr23-dfa421..cpp:5:7: initialization of variable '$__lldb_error_result' was never used; consider replacing with assignment to '_' or removing it var $__lldb_error_result = __lldb_tmp_error ~~~~^~~~~~~~~~~~~~~~~~~~ _ Before the crash, we observed this error message in the console: [ERROR] *****SCStream*****RemoteAudioQueueOperationHandlerWithError:1015 Error received from the remote queue -16665 Additional Context The issue occurs consistently after approximately 3-4 successful audio segment recordings of 60 seconds each Commenting out custom segment rotation logic does not prevent the crash The crash involves XPC communication with Apple's ReplayKit daemon The error appears to be corrupted or malformed when crossing the XPC boundary Workarounds Attempted Added proper thread safety for all published properties using DispatchQueue.main.async Implemented more robust error handling in the delegate methods None of these approaches prevented the crash since it occurs at the Swift runtime level before our code executes. Impact This issue prevents reliable long-duration audio capture using ScreenCaptureKit. This bug significantly limits the usefulness of ScreenCaptureKit for any application requiring continuous system audio capture for more than a few minutes. Perhaps this issue might be related to a macOS bug where the system dialog indicates that the screen is being shared, even though nothing is actually being shared. Moreover, when attempting to stop sharing, nothing happens.
Replies
3
Boosts
0
Views
813
Activity
1w
SpeechTranscriber/SpeechAnalyzer being relatively slow compared to FoundationModel and TTS
So, I've been wondering how fast a an offline STT -> ML Prompt -> TTS roundtrip would be. Interestingly, for many tests, the SpeechTranscriber (STT) takes the bulk of the time, compared to generating a FoundationModel response and creating the Audio using TTS. E.g. InteractionStatistics: - listeningStarted: 21:24:23 4480 2423 - timeTillFirstAboveNoiseFloor: 01.794 - timeTillLastNoiseAboveFloor: 02.383 - timeTillFirstSpeechDetected: 02.399 - timeTillTranscriptFinalized: 04.510 - timeTillFirstMLModelResponse: 04.938 - timeTillMLModelResponse: 05.379 - timeTillTTSStarted: 04.962 - timeTillTTSFinished: 11.016 - speechLength: 06.054 - timeToResponse: 02.578 - transcript: This is a test. - mlModelResponse: Sure! I'm ready to help with your test. What do you need help with? Here, between my audio input ending and the Text-2-Speech starting top play (using AVSpeechUtterance) the total response time was 2.5s. Of that time, it took the SpeechAnalyzer 2.1s to get the transcript finalized, FoundationModel only took 0.4s to respond (and TTS started playing nearly instantly). I'm already using reportingOptions: [.volatileResults, .fastResults] so it's probably as fast as possible right now? I'm just surprised the STT takes so much longer compared to the other parts (all being CoreML based, aren't they?)
Replies
3
Boosts
0
Views
723
Activity
1w
SpeechTranscriber not providing audioTimeRange for most results
I started playing which transcription of audio files on macOS today, latest beta of Xcode and latest beta of Tahoe. Transcription itself works really well, but for some reason the majority of the results contain no audioTimeRange. I got 22 single-word results with time ranges, spread out all over total file of 53 minutes. Is there something I can do to improve this? To my understanding, I have followed sample code and instructions very closely, but the SwiftTranscriptionSampleApp and other examples I've seen lead me to believe I should be getting a lot more time ranges than I actually do.
Replies
3
Boosts
0
Views
206
Activity
Aug ’25
Can backgrounded apps record audio?
I'd like to find out: Can backgrounded apps record audio? In the past as I recall, I found that backgrounded apps were pretty restricted and couldn't do much of anything. However I'm not familiar with the current state of affairs. With iOS 15.8 and above, can backgrounded apps record audio if they've been given permission by the user to access the microphone? Thanks.
Replies
3
Boosts
0
Views
579
Activity
Jan ’26
Mac OS Tahoe 26.0 (25A354) Sound Glitches When opening the simulator app
Hey there, I just upgraded to Mac OS Tahoe ,son an apple MacBook Pro 2019 16inch. am using IntellijIDEA and Flutter to develop a mobile app which I test on the simulator app running iOS 18.4 . the issue: when I start the simulator app. ( while in the loading phase and in the operation phase as well ), the audio from an already open YouTube tab on safari (this happens on chrome browser as well). the sound glitches and becomes Noise. a fix I found online is to kill the audio deamon on Mac OS, This works using the command: "sudo killall coreaudiod" this kills the audio process, (while the emulator is operational), then the macOS restarts the audio deamon then the audio works fine alongside with the simulator being open. I just want to ask is there a permanent fix for this? is Apple working on a fix for this in the upcoming update?
Replies
3
Boosts
5
Views
1.3k
Activity
Oct ’25
Question about PT Framework channel tone behaviour
I've been wondering if there is a way to modify or even disable tones for indicating channel states. The behaviour regarding tones seems like a black box with little documentation. During migration to Apple's PT Framework we've noticed that there are few scenarios where a tone is played which doesn't match certain certifications. For example; moving from a channel to another produces a tone which would fail a test case. I understand the reasoning fully, as it marks that the channel is ready to transmit or receive, but this doesn't mirror the behaviour of TETRA which would be wanted in this case. I'm also wondering if there would be any way to directly communicate feedback regarding PT Framework?
Replies
3
Boosts
0
Views
418
Activity
Oct ’25