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?

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?

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.

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?

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

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

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.

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!

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.

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!

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.

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!

I'm developing the VisionOS app. I want to know how to play spatial audio in addition to RealityKit? If it's iOS or macOS, how to play spatial audio in addition to RealityKit?

Hello, I have an iOS app that is recording audio that is working fine on iPads/iPhones. It asks for microphone permission and after that recording works. I installed the same app on my M3 MacBook via TestFlight since iPad apps are supposed to work without a change that way. The app starts fine and everything, but it never asks for Microphone permission, so I can't record. Do I need to do something to make this happen (this is not macCatalyst, its running the arm64 iPhone binary on macOS) thanks

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

I'm trying to implement airplay into my app. I can successfully playback sound and trigger the airplay selector sheet. If the target device is a Bluetooth only device I can connect with no problem and stream the audio to the Bluetooth device, but if the audio device is a airplay specific device like a HomePod or an Apple TV when I select it, I get a spinning icon, indicating that it is trying to connect, and eventually it times out and stops without connecting. I don't believe it is an AirPlay audio issue because if I go to a different app, for example a podcast app and select my HomePods for output, and then switch back to my app. My audio will correctly stream to the HomePod. Not only that, I have it so that my icon will change color to indicate that it is connected via airplay and it is correctly indicating that it is connected via AirPlay. But I cannot then disconnect it using the Airplay selector. The issue appears to be in the AirPlay selection side, which I have spent several days attempting to troubleshoot mostly using ChatGPT to suggest code different than what I have to maybe work around the issue. Mostly it is focused on the audio player section, but it doesn't seem like that is really the route that is the problem.

AVAudioSessionCategoryOptionAllowBluetooth is marked as deprecated in iOS 8 in iOS 26 beta 5 when this option was not deprecated in iOS 18.6. I think this is a mistake and the deprecation is in iOS 26. Am I right? It seems that the substitute for this option is "AVAudioSessionCategoryOptionAllowBluetoothHFP". The documentation does not make clear if the behaviour is exactly the same or if any difference should be expected... Has anyone used this option in iOS 26? Should I expect any difference with the current behaviour of "AVAudioSessionCategoryOptionAllowBluetooth"? Thank you.

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

Hi all, with my app ScreenFloat, you can record your screen, along with system- and microphone audio. Those two audio feeds are recorded into separate audio tracks in order to individually remove or edit them later on. Now, these recordings you create with ScreenFloat can be drag-and-dropped to other apps instantly. So far, so good, but some apps, like Slack, or VLC, or even websites like YouTube, do not play back multiple audio tracks, just one. So what I'm trying to do is, on dragging the video recording file out of ScreenFloat, instantly baking together the two individual audio tracks into one, and offering that new file as the drag and drop file, so that all audio is played in the target app. But it's slow. I mean, it's actually quite fast, but for drag and drop, it's slow. My approach is this: "Bake together" the two audio tracks into a one-track m4a audio file using AVMutableAudioMix and AVAssetExportSession Take the video track, add the new audio file as an audio track to it, and render that out using AVAssetExportSession For a quick benchmark, a 3'40'' movie, step 1 takes ~1.7 seconds, and step two adds another ~1.5 seconds, so we're at ~3.2 seconds. That's an eternity for a drag and drop, where the user might cancel if there's no immediate feedback. I could also do it in one step, but then I couldn't use the AV*Passthrough preset, and that makes it take around 32 seconds then, because I assume it touches the video data (which is unnecessary in this case, so I think the two-step approach here is the fastest). So, my question is, is there a faster way? The best idea I can come up with right now is, when initially recording the screen with system- and microphone audio as separate tracks, to also record both of them into a third, muted, "hidden" track I could use later on, basically eliminating the need for step one and just ripping the two single audio tracks out of the movie and only have the video and the "hidden" track (then unmuted), but I'd still have a ~1.5 second delay there. Also, there's the processing and data overhead (basically doubling the movie's audio data). All this would be great for an export operation (where one expects it to take a little time), but for a drag-and-drop operation, it's not ideal. I've discarded the idea of doing a promise file drag, because many apps do not accept those, and I want to keep wide compatibility with all sorts of apps. I'd appreciate any ideas or pointers. Thank you kindly, Matthias

I have a new 2725QC (Dell) Monitor that uses USB-C connection to connect with the iMac (2019, 27 inch) through the back port but the problem is that the volume control can currently only be done from the hardware, not the software control using the Apple keyboard. What should I do in terms of writing code to do this (Swift or Obj-C)? Is there a third-party solution for Intel iMac and ARM Mac?