Our capture application records system audio via HAL plugin, however, with the latest macOS 15 Sequoia, all audio buffer values are zero. I am attaching sample code that replicates the problem. Compile as a Command Line Tool application with Xcode. STEPS TO REPRODUCE Install BlackHole 2ch audio driver: https://existential.audio/blackhole/download/?code=1579271348 Start some system audio, e.g. YouTube. Compile and run the sample application. On macOS up to Sonoma, you will hear audio via loopback and see audio values in the debug/console window. On macOS Sequoia, you will not hear audio and the audio values are 0. #import <AVFoundation/AVFoundation.h> #import <CoreAudio/CoreAudio.h> #define BLACKHOLE_UID @"BlackHole2ch_UID" #define DEFAULT_OUTPUT_UID @"BuiltInSpeakerDevice" @interface AudioCaptureDelegate : NSObject <AVCaptureAudioDataOutputSampleBufferDelegate> @end void setDefaultAudioDevice(NSString *deviceUID); @implementation AudioCaptureDelegate // receive samples from CoreAudio/HAL driver and print amplitute values for testing // this is where samples would normally be copied and passed downstream for further processing which // is not needed in this simple sample application - (void)captureOutput:(AVCaptureOutput *)captureOutput didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer fromConnection:(AVCaptureConnection *)connection { // Access the audio data in the sample buffer CMBlockBufferRef blockBuffer = CMSampleBufferGetDataBuffer(sampleBuffer); if (!blockBuffer) { NSLog(@"No audio data in the sample buffer."); return; } size_t length; char *data; CMBlockBufferGetDataPointer(blockBuffer, 0, NULL, &length, &data); // Process the audio samples to calculate the average amplitude int16_t *samples = (int16_t *)data; size_t sampleCount = length / sizeof(int16_t); int64_t sum = 0; for (size_t i = 0; i < sampleCount; i++) { sum += abs(samples[i]); } // Calculate and log the average amplitude float averageAmplitude = (float)sum / sampleCount; NSLog(@"Average Amplitude: %f", averageAmplitude); } @end // set the default audio device to Blackhole while testing or speakers when done // called by main void setDefaultAudioDevice(NSString *deviceUID) { AudioObjectPropertyAddress address; AudioDeviceID deviceID = kAudioObjectUnknown; UInt32 size; CFStringRef uidString = (__bridge CFStringRef)deviceUID; // Gets the device corresponding to the given UID. AudioValueTranslation translation; translation.mInputData = &uidString; translation.mInputDataSize = sizeof(uidString); translation.mOutputData = &deviceID; translation.mOutputDataSize = sizeof(deviceID); size = sizeof(translation); address.mSelector = kAudioHardwarePropertyDeviceForUID; address.mScope = kAudioObjectPropertyScopeGlobal; //???? address.mElement = kAudioObjectPropertyElementMain; OSStatus status = AudioObjectGetPropertyData(kAudioObjectSystemObject, &address, 0, NULL, &size, &translation); if (status != noErr) { NSLog(@"Error: Could not retrieve audio device ID for UID %@. Status code: %d", deviceUID, (int)status); return; } AudioObjectPropertyAddress propertyAddress; propertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice; propertyAddress.mScope = kAudioObjectPropertyScopeGlobal; status = AudioObjectSetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL, sizeof(AudioDeviceID), &deviceID); if (status == noErr) { NSLog(@"Default audio device set to %@", deviceUID); } else { NSLog(@"Failed to set default audio device: %d", status); } } // sets Blackhole device as default and configures it as AVCatureDeviceInput // sets the speakers as loopback so we can hear what is being captured // sets up queue to receive capture samples // runs session for 30 seconds, then restores speakers as default output int main(int argc, const char * argv[]) { @autoreleasepool { // Create the capture session AVCaptureSession *session = [[AVCaptureSession alloc] init]; // Select the audio device AVCaptureDevice *audioDevice = nil; NSString *audioDriverUID = nil; audioDriverUID = BLACKHOLE_UID; setDefaultAudioDevice(audioDriverUID); audioDevice = [AVCaptureDevice deviceWithUniqueID:audioDriverUID]; if (!audioDevice) { NSLog(@"Audio device %s not found!", [audioDriverUID UTF8String]); return -1; } else { NSLog(@"Using Audio device: %s", [audioDriverUID UTF8String]); } // Configure the audio input with the selected device (Blackhole) NSError *error = nil; AVCaptureDeviceInput *audioInput = [AVCaptureDeviceInput deviceInputWithDevice:audioDevice error:&error]; if (error || !audioInput) { NSLog(@"Failed to create audio input: %@", error); return -1; } [session addInput:audioInput]; // Configure the audio data output AVCaptureAudioDataOutput *audioOutput = [[AVCaptureAudioDataOutput alloc] init]; AudioCaptureDelegate *delegate = [[AudioCaptureDelegate alloc] init]; dispatch_queue_t queue = dispatch_queue_create("AudioCaptureQueue", NULL); [audioOutput setSampleBufferDelegate:delegate queue:queue]; [session addOutput:audioOutput]; // Set audio settings NSDictionary *audioSettings = @{ AVFormatIDKey: @(kAudioFormatLinearPCM), AVSampleRateKey: @48000, AVNumberOfChannelsKey: @2, AVLinearPCMBitDepthKey: @16, AVLinearPCMIsFloatKey: @NO, AVLinearPCMIsNonInterleaved: @NO }; [audioOutput setAudioSettings:audioSettings]; AVCaptureAudioPreviewOutput * loopback_output = nil; loopback_output = [[AVCaptureAudioPreviewOutput alloc] init]; loopback_output.volume = 1.0; loopback_output.outputDeviceUniqueID = DEFAULT_OUTPUT_UID; [session addOutput:loopback_output]; const char *deviceID = loopback_output.outputDeviceUniqueID ? [loopback_output.outputDeviceUniqueID UTF8String] : "nil"; NSLog(@"session addOutput for preview/loopback: %s", deviceID); // Start the session [session startRunning]; NSLog(@"Capturing audio data for 30 seconds..."); [[NSRunLoop currentRunLoop] runUntilDate:[NSDate dateWithTimeIntervalSinceNow:30.0]]; // Stop the session [session stopRunning]; NSLog(@"Capture session stopped."); setDefaultAudioDevice(DEFAULT_OUTPUT_UID); } return 0; }

Since upgrading to tvOS 18, the above function isn't working for me in converting a stream with these formats. It does work in decoding AAC, however. https://developer.apple.com/documentation/audiotoolbox/1503098-audioconverterfillcomplexbuffer?language=objc I pass a valid ioOutputDataPacketSize in, but it always comes out as zero. Has anyone else observed this too? I wonder if this is related to the issue being discussed widely about 5.1 sound being broken for many people after upgrading to tvOS 18? https://discussions.apple.com/thread/255769102?login=true&sortBy=rank EDIT: further information; the callback gets called once, asking for 1 packet (which is ok). I give it one packet and return noErr. However, after this, the callback is never invoked again. Must be a bug? EDIT2: the same code continues to work correctly on macOS in decoding the same audio stream.

Hi, I'm working on an audio mixing app, that comes with bundled audio units that provide some of the app's core functionality. For the next release of that app, we are planning to make two changes: make the app sandboxed package the bundled audio units as .appex bundles instead as .component bundles, so we don't need to take care of the installation at the correct spot in the file system When trying this new approach, we run into problems where [[AVAudioUnitEffect alloc] initWithAudioComponentDescription:] crashes when trying to load our audio unit with the exception: AVAEInternal.h:109 [AUInterface.mm:468:AUInterfaceBaseV3: (AudioComponentInstanceNew(comp, &_auv2)): error -10863 Our audio unit has the `sandboxSafe flag enabled, and loads fine when the host app is not sandboxed, so I'm guessing I got the bundle id/code signing requirements for the .appex correct. It seems, that my .appex isn't even loaded, and the system rejects it because of its metadata. Maybe there something wrong the Info.plist generated by Juice? "BuildMachineOSBuild" => "23H222" "CFBundleDisplayName" => "elgato_sample_recorder" "CFBundleExecutable" => "ElgatoSampleRecorder" "CFBundleIdentifier" => "com.iwascoding.EffectLoader.samplerecorderAUv3" "CFBundleName" => "elgato_sample_recorder" "CFBundlePackageType" => "XPC!" "CFBundleShortVersionString" => "1.0.0.0" "CFBundleSignature" => "????" "CFBundleSupportedPlatforms" => [ 0 => "MacOSX" ] "CFBundleVersion" => "1.0.0.0" "DTCompiler" => "com.apple.compilers.llvm.clang.1_0" "DTPlatformBuild" => "24C94" "DTPlatformName" => "macosx" "DTPlatformVersion" => "15.2" "DTSDKBuild" => "24C94" "DTSDKName" => "macosx15.2" "DTXcode" => "1620" "DTXcodeBuild" => "16C5032a" "LSMinimumSystemVersion" => "10.13" "NSExtension" => { "NSExtensionAttributes" => { "AudioComponents" => [ 0 => { "description" => "Elgato Sample Recorder" "factoryFunction" => "elgato_sample_recorderAUFactoryAUv3" "manufacturer" => "Manu" "name" => "Elgato: Elgato Sample Recorder" "sandboxSafe" => 1 "subtype" => "Znyk" "tags" => [ 0 => "Effects" ] "type" => "aufx" "version" => 65536 } ] } "NSExtensionPointIdentifier" => "com.apple.AudioUnit-UI" "NSExtensionPrincipalClass" => "elgato_sample_recorderAUFactoryAUv3" } "NSHighResolutionCapable" => 1 } Any ideas what I am missing?

So experimenting with the new SpeechTranscriber, if I do: let transcriber = SpeechTranscriber( locale: locale, transcriptionOptions: [], reportingOptions: [.volatileResults], attributeOptions: [.audioTimeRange] ) only the final result has audio time ranges, not the volatile results. Is this a performance consideration? If there is no performance problem, it would be nice to have the option to also get speech time ranges for volatile responses. I'm not presenting the volatile text at all in the UI, I was just trying to keep statistics about the non-speech and the speech noise level, this way I can determine when the noise level falls under the noisefloor for a while. The goal here was to finalize the recording automatically, when the noise level indicate that the user has finished speaking.

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?

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 tried adding watermarks to the recorded video. Appending sample buffers using AVAssetWriterInput's append method fails and when I inspect the AVAssetWriter's error property, I get the following: Error Domain=AVFoundation Error Domain Code=-11800 "This operation cannot be completed" UserInfo={NSLocalizedFailureReason=An unknown error occurred (-12780), NSLocalizedDDescription=This operation cannot be completed, NSUnderlyingError=0x302399a70 {Error Domain=NSOSStatusErrorDomain Code=-12780 "(null)"}} As far as I can tell -11800 indicates an AVErrorUknown, however I have not been able to find information about the -12780 error code, which as far as I can tell is undocumented. Thanks! Here is the code

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!

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, 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

So get a swift file and put this in it import Foundation import AVFoundation let synthesizer = AVSpeechSynthesizer() let utterance = AVSpeechUtterance(string: "Hello, testing speech synthesis on macOS.") if let voice = AVSpeechSynthesisVoice(identifier: "com.apple.voice.compact.en-GB.Daniel") { utterance.voice = voice print("Using voice: \(voice.name), \(voice.language)") } else { print("Daniel voice not found on macOS.") } synthesizer.speak(utterance) I get no speech output and this log output Error reading languages in for local resources. Error reading languages in for local resources. Using voice: Daniel, en-GB Program ended with exit code: 0 Why? and whats with "Error reading languages in for local resources." ?

Hi! I am creating a aumi AUv3 extension and I am trying to achieve simultaneous connections to multiple other avaudionodes. I would like to know it is possible to route the midi to different outputs inside the render process in the AUv3. I am using connectMIDI(_:to:format:eventListBlock:) to connect the output of the AUv3 to multiple AvAudioNodes. However, when I send midi out of the AUv3, it gets sent to all the AudioNodes connected to it. I can't seem to find any documentation on how to route the midi only to one of the connected nodes. Is this possible?

We are using a VoiceProcessingIO audio unit in our VoIP application on Mac. In certain scenarios, the AudioComponentInstanceNew call blocks for up to five seconds (at least two). We are using the following code to initialize the audio unit: OSStatus status; AudioComponentDescription desc; AudioComponent inputComponent; desc.componentType = kAudioUnitType_Output; desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO; desc.componentFlags = 0; desc.componentFlagsMask = 0; desc.componentManufacturer = kAudioUnitManufacturer_Apple; inputComponent = AudioComponentFindNext(NULL, &desc); status = AudioComponentInstanceNew(inputComponent, &unit); We are having the issue with current MacOS versions on a host of different Macs (x86 and x64 alike). It takes two to three seconds until AudioComponentInstanceNew returns. We also see the following errors in the log multiple times: AUVPAggregate.cpp:2560 AggInpStreamsChanged wait failed and those right after (which I don't know if they matter to this issue): KeystrokeSuppressorCore.cpp:44 ERROR: KeystrokeSuppressor initialization was unsuccessful. Invalid or no plist was provided. AU will be bypassed. vpStrategyManager.mm:486 Error code 2003332927 reported at GetPropertyInfo

i have a CarPlay implementation eand I want to show previous/next track button on player UI MPRemoteCommandCenter.shared().seekForwardCommand.isEnabled = false MPRemoteCommandCenter.shared().seekBackwardCommand.isEnabled = false MPRemoteCommandCenter.shared().previousTrackCommand.isEnabled = true MPRemoteCommandCenter.shared().nextTrackCommand.isEnabled = true It works correctly on CarPlay simulator , but on some car only SEEK button are shown . I have to suppose that it is that a problem on the car side , but I would ask about your opinion , maybe there is some pieces I'm missing

I'm using AVAudioEngine to play AVAudioPCMBuffers. I'd like to synchronize some events with the playback. For example if the audio's frame position is >= some point && less than some point trigger some code. So I'm looking at - (void)installTapOnBus:(AVAudioNodeBus)bus bufferSize:(AVAudioFrameCount)bufferSize format:(AVAudioFormat * __nullable)format block:(AVAudioNodeTapBlock)tapBlock; Now I have frame positions calculated (predetermined before audio is scheduled I already made all necessary computations) . So I just need to fire code at certain points during playback: [playerNode installTapOnBus:bus bufferSize:bufferSize format:format block:^(AVAudioPCMBuffer * _Nonnull buffer, AVAudioTime * _Nonnull when) { //Inspect current audio here and fire... }]; [playerNode scheduleBuffer:fullbuffer atTime:startTime options:0 completionCallbackType:AVAudioPlayerNodeCompletionDataPlayedBack completionHandler:^(AVAudioPlayerNodeCompletionCallbackType callbackType) { // some code is here, not important to this question. }]; The problem I'm having is figuring out at what point in full buffer I'm at within the tap block. The tap block passes chunks (not the full audio buffer). I tried using the when parameter of the block to calculate the frame position relative to the entire audio but have be unsuccessful so far. I'm assuming the when parameter is relative to the buffer passed in the tap block (not my entire audio buffer I scheduled). Not installing a tap and just using a timer before scheduling my fullBuffer has given me good results but I'd rather avoid using a timer if possible and use sample time.

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

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.

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

Hi, Apple's engineer. Hoping that you can reply to this one. We're developing a Text-to-Speak app. Everything went well until the IOS got upgraded to 18. AVSpeechSynthesisVoice(language: "zh-CN") is running well under IOS 16 AND IOS 17. It speaks Mandarin correctly. In IOS 18, we noticed that Siri's Language setting interrupted the performance of AVSpeechSynthesisVoice. It plays Cantonese instead of Mandarin. Buggy language setting in Siri that affects the AVSpeechSynthesisVoice : Chinese (Cantonese - China mainland) Chinese (Cantonese -Hong Kong)

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.