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?

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!

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

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.

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?

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

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

I'm encountering errors while using AVAudioEngine with voice processing enabled (setVoiceProcessingEnabled(true)) in scenarios where the input and output audio devices are not the same. This issue arises specifically with mismatched devices, preventing the application from functioning as expected. Works: Paired devices (e.g., MacBook Pro mic → MacBook Pro speakers) Fails: Mismatched devices (e.g., AirPods mic → MacBook Pro speakers) When using paired input and output devices: The setup works as expected. Example: MacBook Pro microphone → MacBook Pro speakers. When using mismatched devices: AVAudioEngine setup fails during aggregate device construction. Example: AirPods microphone → MacBook Pro speakers. Error logs indicate a channel count mismatch. Here are the partial logs. Due to the content limit, I cannot post the entire logs. AUVPAggregate.cpp:1000 client-side input and output formats do not match (err=-10875) AUVPAggregate.cpp:1036 err=-10875 AVAEInternal.h:109 [AVAudioEngineGraph.mm:1344:Initialize: (err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)): error -10875 AggregateDevice.mm:329 Failed expectation of constructed aggregate (312): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (312): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (336): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (336): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AudioHardware-mac-imp.cpp:3484 AudioDeviceSetProperty: no device with given ID AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (348): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (348): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) Is it possible to use voice processing with different input/output devices? If yes, are there any specific configurations required to handle mismatched devices? How can we resolve channel count mismatch errors during aggregate device construction? Are there settings or API adjustments to enforce compatibility between input/output devices? Are there any workarounds or alternative approaches to achieve voice processing functionality with mismatched devices? For instance, can we force an intermediate channel configuration or downmix input/output formats?

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.

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.

Please Update Andorid MusicKit，the version 1.1.2 will complied fail。the error msg：•SDKUriHandlerActivity>. Apps targeting Android 12 and higher are required to specify an explicit value for android:exported when the corres

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!

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

We have application using PTT Framework to record audio messages when app is backgrounded. Right now we are using AVAudioRecorder for that purpose. And problem is one specific user has frequent issue - recorded audio contains only silence. I've checked almost everything I can imagine but didn't find any possible reason of issue. Conditions: AVAudioRecorder uses following configuration: [ AVEncoderAudioQualityKey: AVAudioQuality.low.rawValue, AVFormatIDKey : kAudioFormatMPEG4AAC, AVNumberOfChannelsKey: 1, AVSampleRateKey: 16000.0 ] App waits both didBeginTransmitting and didActivate audioSession from PTChannelManager (audio session has playback category at that moment) App does AVAudioSession category change to playAndRecord App gets routeChangeNotification with categoryChange and category = playAndRecord There is no any interruption notifications from AVAudioSession during recording There is no any error notification from AVAudioRecorder Any idea what exactly I do wrong? Is there anything else I should check? Thanks in advance. P.S. it looks like recording audio with AudioUnit has the same issue, but let's exclude it from question atm for simplicity.

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

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.

On macOS Sequoia, I'm having the hardest time getting this basic audio output to work correctly. I'm compiling in XCode using C99, and when I run this, I get audio for a split second, and then nothing, indefinitely. Any ideas what could be going wrong? Here's a minimum code example to demonstrate: #include &lt;AudioToolbox/AudioToolbox.h&gt; #include &lt;stdint.h&gt; #define RENDER_BUFFER_COUNT 2 #define RENDER_FRAMES_PER_BUFFER 128 // mono linear PCM audio data at 48kHz #define RENDER_SAMPLE_RATE 48000 #define RENDER_CHANNEL_COUNT 1 #define RENDER_BUFFER_BYTE_COUNT (RENDER_FRAMES_PER_BUFFER * RENDER_CHANNEL_COUNT * sizeof(f32)) void RenderAudioSaw(float* outBuffer, uint32_t frameCount, uint32_t channelCount) { static bool isInverted = false; float scalar = isInverted ? -1.f : 1.f; for (uint32_t frame = 0; frame &lt; frameCount; ++frame) { for (uint32_t channel = 0; channel &lt; channelCount; ++channel) { // series of ramps, alternating up and down. outBuffer[frame * channelCount + channel] = 0.1f * scalar * ((float)frame / frameCount); } } isInverted = !isInverted; } AudioStreamBasicDescription coreAudioDesc = { 0 }; AudioQueueRef coreAudioQueue = NULL; AudioQueueBufferRef coreAudioBuffers[RENDER_BUFFER_COUNT] = { NULL }; void coreAudioCallback(void* unused, AudioQueueRef queue, AudioQueueBufferRef buffer) { // 0's here indicate no fancy packet magic AudioQueueEnqueueBuffer(queue, buffer, 0, 0); } int main(void) { const UInt32 BytesPerSample = sizeof(float); coreAudioDesc.mSampleRate = RENDER_SAMPLE_RATE; coreAudioDesc.mFormatID = kAudioFormatLinearPCM; coreAudioDesc.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked; coreAudioDesc.mBytesPerPacket = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mFramesPerPacket = 1; coreAudioDesc.mBytesPerFrame = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mChannelsPerFrame = RENDER_CHANNEL_COUNT; coreAudioDesc.mBitsPerChannel = BytesPerSample * 8; coreAudioQueue = NULL; OSStatus result; // most of the 0 and NULL params here are for compressed sound formats etc. result = AudioQueueNewOutput(&amp;coreAudioDesc, &amp;coreAudioCallback, NULL, 0, 0, 0, &amp;coreAudioQueue); if (result != noErr) { assert(false == "AudioQueueNewOutput failed!"); abort(); } for (int i = 0; i &lt; RENDER_BUFFER_COUNT; ++i) { uint32_t bufferSize = coreAudioDesc.mBytesPerFrame * RENDER_FRAMES_PER_BUFFER; result = AudioQueueAllocateBuffer(coreAudioQueue, bufferSize, &amp;(coreAudioBuffers[i])); if (result != noErr) { assert(false == "AudioQueueAllocateBuffer failed!"); abort(); } } for (int i = 0; i &lt; RENDER_BUFFER_COUNT; ++i) { RenderAudioSaw(coreAudioBuffers[i]-&gt;mAudioData, RENDER_FRAMES_PER_BUFFER, RENDER_CHANNEL_COUNT); coreAudioBuffers[i]-&gt;mAudioDataByteSize = coreAudioBuffers[i]-&gt;mAudioDataBytesCapacity; AudioQueueEnqueueBuffer(coreAudioQueue, coreAudioBuffers[i], 0, 0); } AudioQueueStart(coreAudioQueue, NULL); sleep(10); // some time to hear the audio AudioQueueStop(coreAudioQueue, true); AudioQueueDispose(coreAudioQueue, true); return 0; }

Hi all! I have been experiencing some issues when using the AVAudioEngine to play audio and record input while doing a voice chat (through the PTT Interface). I noticed if I connect any players to the AudioGraph OR call start that the audio session becomes active (this is on iOS). I don't see anything in the docs or the header files in the AVFoundation, but is it possible that calling the stop method on an engine deactivates the audio session too? In a normal app this behavior seems logical, but when using PTT all activation and deactivation of the audio session must go through the framework and its delegate methods. The issue I am debugging is that when the engine with the input node tapped gets stopped, and there is a gap between the input and when the server replies with inbound audio to be played and something seems to be getting the hardware/audio session into a jammed state. Thanks for any feedback and/or confirmation on this behavior!

Hi, when using ApplicationMusicPlayer from MusicKit my app automatically gets the media controls on the lock screen: Play/ Pause, Skip Buttons, Playback Position etc. I would like to customize these. Tried a bunch of things, e.g. using MPRemoteCommandCenter. So far I haven't had any success. Does anyone know how I can customize the media controls of ApplicationMusicPlayer. Thank you.

Is there a recommended way on macOS 26 Tahoe to take a CoreAudio AudioObjectID and use it to lookup the underlying USB LocationID? I previously used AudioObjectID to query the corresponding DeviceUID with kAudioDevicePropertyDeviceUID. Then I queried for the IOService matching kIOAudioEngineClassName with property kIOAudioEngineGlobalUniqueIDKey matching DeviceUID, and I loaded kUSBDevicePropertyLocationID from the result. This fails on macOS 26, because the IO Registry for the device has an entry for usbaudiod rather than AppleUSBAudioEngine, and usbaudiod does not include a kIOAudioEngineGlobalUniqueIDKey property (or any other property to map it to a CoreAudio DeviceUID). My use-case here is a piece of audio recording software that allows configuring a set of supported audio devices via USB HID prior to recording. I present the user with a list of CoreAudio devices to use, but without a way to lookup the underlying USB LocationID, I cannot guarantee that the configured device matches the selected device (e.g. if the user plugged in two identical microphones).