Periodically when testing I am running into a situation where the app hangs and beach balls forever when using AVAudioEngine. This seems to log out when this affect happens: Now when this happens if I pause the debugger it's hanging at a call to: [engine connect:playerNode to:engine.mainMixerNode format:buffer.format]; #0 0x000000019391ca9c in __psynch_mutexwait () #1 0x0000000104d49100 in _pthread_mutex_firstfit_lock_wait () #2 0x0000000104d49014 in _pthread_mutex_firstfit_lock_slow () #3 0x00000001938928ec in std::__1::recursive_mutex::lock () #4 0x00000001ef80e988 in CADeprecated::RealtimeMessenger::_PerformPendingMessages () #5 0x00000001ef818868 in AVAudioNodeTap::Uninitialize () #6 0x00000001ef7fdc68 in AUGraphNodeBase::Uninitialize () #7 0x00000001ef884f38 in AVAudioEngineGraph::PerformCommand () #8 0x00000001ef88e780 in AVAudioEngineGraph::_Connect () #9 0x00000001ef8b7e70 in AVAudioEngineImpl::Connect () #10 0x00000001ef8bc05c in -[AVAudioEngine connect:to:format:] () Current all my audio engine related calls are on the main queue (though I am curious about this https://forums.developer.apple.com/forums/thread/123540?answerId=816827022#816827022). In any case, anyone know where I'm going wrong here?

The following is my playground code. Any of the apple audio units show the plugin view, however anything else (i.e. kontakt, spitfire, etc.) does not. It does not error, just where the visual is expected is blank. import AppKit import PlaygroundSupport import AudioToolbox import AVFoundation import CoreAudioKit let manager = AVAudioUnitComponentManager.shared() let description = AudioComponentDescription(componentType: kAudioUnitType_MusicDevice, componentSubType: 0, componentManufacturer: 0, componentFlags: 0, componentFlagsMask: 0) var deviceComponents = manager.components(matching: description) var names = deviceComponents.map{$0.name} let pluginName: String = "AUSampler" // This works //let pluginName: String = "Kontakt" // This does not var plugin = deviceComponents.filter{$0.name.contains(pluginName)}.first! print("Plugin name: \(plugin.name)") var customViewController:NSViewController? AVAudioUnit.instantiate(with: plugin.audioComponentDescription, options: []){avAudioUnit, error in var ilip = avAudioUnit!.auAudioUnit.isLoadedInProcess print("Loaded in process: \(ilip)") guard error == nil else { print("Error: \(error!.localizedDescription)") return } print("AudioUnit successfully created.") let audioUnit = avAudioUnit!.auAudioUnit audioUnit.requestViewController{ vc in if let viewCtrl = vc { customViewController = vc var b = vc?.view.bounds PlaygroundPage.current.liveView = vc print("Successfully added view controller.") }else{ print("Failed to load controller.") } } }

It’s been established that generally speaking background apps cannot record audio while the foreground app is already reading audio data from the microphone, but are there exceptions? For instance, is there an exception for certain Apple apps? If so, and there’s a special exception that most programmers don’t know about but some Apple’s engineers do and perhaps some hackers do as well, wouldn’t the mechanism that allows that eventually be exploited?

I'd like to know: Let's say there's a backgrounded app which has microphone access, such as Signal or SoundHound or Shazam. It's established that these apps are allowed to record audio in the user's environment even after being backgrounded, seemingly for as long as they want and even upload that sound data. But can they ALSO continue recording even while another app that is in the foreground is using the microphone, such as the Phone app or Signal?

I’m looking to add DAW-like capabilities to my macOS music app, and AVAudioEngine seems like the right tool for the job. However, I haven’t been able to find any documentation on how to save the user’s AVAudioEngine configuration—specifically the connections between nodes and the internal states of each node—to a file. Does AVAudioEngine provide any API for saving and restoring this state, or does it need to be handled manually? If it’s manual, are there any sample "DAW" apps or resources that demonstrate how this can be implemented? Any guidance would be greatly appreciated. Thanks, BD

Hello We have an application that play some sound via the system sound APIs from the AudioToolbox framework. AudioServicesCreateSystemSoundID(url as CFURL, &soundID) AudioServicesPlaySystemSoundWithCompletion(soundID) Our make sure that an active audio session is available before playing the system sound. But when the device is connected to a BluetoothA2DP device. The sound are played on through the device speaker and not through the bluetooth A2DP device. Our AudioSesison is configured with the following categories [.allowBluetooth, .defaultToSpeaker, .allowBluetoothA2DP] Sound played from the AVAudioPlayer are played on the allowBluetoothA2DP device with similar code. Is this a bug in the AudioToolbox framework?

I'm trying to make an app that is able to quietly run in the background. It needs to detect other apps' or the system's incoming video and/or audio, using only on-device resources to determine if it might be a scam caller. It will tap into an escalating cascade of resources to do so. For video/image scam detection, it uses OpenCV to detect faces, then refers to a known database of reported scam imagery. For audio scam calls, we defer to known techniques of voice modulation in frequency and/or amplitude. Each video and/or audio result will be relayed via notification banner as well as recorded in-app. Crucially, if the results are uncertain, users have the option to submit it to a global collaborative cloud database for investigative teams; 60 second audio snippets or series of images where faces were detected (60 second equivalent). In the end, we expect to deploy this app across most parts of Asia and Africa, thereby protecting generations of iPhone and iPad users. However, we have not been able to find a method that does this, and there is no known correspondance able to provide such technical guidance. Please assist.

Hi, I use AudioQueueNewInput() with my very own run loop and dedicated thread. But now it doesn't show the mic alert window. Howto fix this? AudioQueueNewInput(&(core_audio_port->record_format), ags_core_audio_port_handle_input_buffer, core_audio_port, ags_core_audio_port_input_run_loop, kCFRunLoopDefaultMode, 0, &(core_audio_port->record_aq_ref));

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.

Hello, Using ShazamKit, based on a shazam catalog result, would it be possible to detect the audio-recorded FPS (speed)? I'm thinking that the shazam catalog which was created from an audio file can be used to compare the speed of a live recorded audio. Thank you!

I'm running into an issue where in some cases, when the AUHostingServiceXPC_arrow process is shut down by Logic, the process is terminated abruptly without calling AP_Close on all of the plugins hosted in the process. In our case, we have filesystem resources we need to clean up, and having stale files around from the last run can cause issues in new sessions, so this leak is having some pretty gnarly effects. I can reproduce the issue using only Apple sample plugins, and it seems to be triggered by a timeout. If I have two different AU plugins in the session, and I add a 1 second sleep to the destructor of one of the sample plugins, Logic will force terminate the process and the remaining destructors are not called (even for the plugins without the 1 second sleep). Is there a way to avoid this behavior? Or to safely clean up our plugin even if other plugins in the session take a second to tear down?

Hello, I have a question regarding the voice and sound recognition features on the iPhone 15 Pro. The iPhone 15 Pro is equipped with four microphones, and I understand that for features like Apple’s sound recognition and when invoking Siri, the microphone(s) must always be active. My question is whether the device uses a single microphone (mono channel) for these functions or if multiple microphones are activated simultaneously. I would appreciate clarification on how the microphones are utilized in sound and voice recognition features. Thank you for your assistance. Best regards.

Hello, I have a question regarding the voice and sound recognition features on the iPhone 15 Pro. The iPhone 15 Pro is equipped with four microphones, and I understand that for features like Apple’s sound recognition and when invoking Siri, the microphone(s) must always be active. My question is whether the device uses a single microphone (mono channel) for these functions or if multiple microphones are activated simultaneously. I would appreciate clarification on how the microphones are utilized in sound and voice recognition features. Thank you for your assistance. Best regards.

Currently we tested iOS AAC LC encoder using AudioToolbox framework, no matter we set mManufacturer to kAppleHardwareAudioCodecManufacturer or kAppleSoftwareAudioCodecManufacturer, it always run on CPU.

Hello, As explained in this link, the AVAssetReaderTrackOutput.copyNextSampleBuffer() returns a CMSampleBuffer in linear PCM audio format. I want to place this audio buffer into an AVAssetWriterInput of type kAudioFormatMPEG4AAC, but I can't manage the conversion. Could you help me by providing an extension that returns a CMSampleBuffer converted from linear PCM audio format to kAudioFormatMPEG4AAC? Example: extension CMSampleBuffer { func fromPCMToAAC() -> CMSampleBuffer? { // Here, get a new AudioStreamBasicDescription, create a CMSampleBuffer and a CMBlockBuffer } } I've tried multiple times but without success. Software: iOS 18.1 XCode: 16.0 Thank you!

AudioQueueObject.cpp:1580 BuildConverter: AudioConverterNew returned -50 from: 0 ch, 16000 Hz, .... (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame to: 2 ch, 16000 Hz, Int16, interleaved AQMEIO_HAL.cpp:2773 iOSSimulatorAudioDevice-15111-0: Abandoning I/O cycle because reconfig pending (1). HALC_ProxySystem.cpp:163 HALC_ProxySystem::GetObjectInfo: got an error from the server, Error: 560947818 (!obj) HALC_ShellObject.mm:213 HALC_ShellObject::HasProperty: there is no proxy object AudioHardware-mac-imp.cpp:1224 AudioObjectRemovePropertyListener: no object with given ID 160 HALSystem.cpp:2216 AudioObjectPropertiesChanged: no such object why? Can't record on ios17. Normal recording before iOS 16.

Calls to ExtAudioFileRead are throwing OSStatus 561145203 (AVAudioSessionErrorCodeResourceNotAvailable) on iOS and iPadOS 18 -- earlier versions of iOS have not exhibited this behavior. This is a longstanding code path that has seen a spike of these error codes since iOS 18's release. The following is also printed to the Xcode 16 console:

Hello everyone, I'm new to Core Audio and still haven't found my footing. I'm learning how to capture audio from the default device, using Audio Units. On my MacBook, the default audio input is mono. But when I write a piece of code to capture audio using AUHAL, I'm discovering that I need to provide an AudioBufferList with two channels, not one. Also, when I try to capture audio from an audio interface with 20 audio inputs, I must provide an AudioBufferList with two channels, and not with 20 channels. To investigate the issue, I wrote a small diagnostic program, which opens the default audio device and probes it for the number of channels. Depending on which way I'm probing, I'm getting different results. When I probe the stream format, I'm getting information that there is 1 channels. But when I probe the input audio unit, I'm getting information that there are 2 input channels. Here's my program to demonstrate the issue: // InputDeviceChannels.m // Compile with: // clang -framework CoreAudio -framework AudioToolbox -framework CoreFoundation -framework AudioUnit -o InputDeviceChannels InputDeviceChannels.m // // On my system, this prints: // Device Name: MacBook Pro Microphone // Number of Channels (Stream Format): 1 // Number of Elements (Element Count): 2 #import <AudioToolbox/AudioToolbox.h> #import <AudioUnit/AudioUnit.h> #import <CoreAudio/CoreAudio.h> #import <Foundation/Foundation.h> void printDeviceInfo(AudioUnit audioUnit) { UInt32 size; OSStatus err; AudioStreamBasicDescription streamFormat; size = sizeof(streamFormat); err = AudioUnitGetProperty(audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &streamFormat, &size); if (err != noErr) { printf("Error getting stream format\n"); exit(1); } int numChannels = streamFormat.mChannelsPerFrame; UInt32 elementCount; size = sizeof(elementCount); err = AudioUnitGetProperty(audioUnit, kAudioUnitProperty_ElementCount, kAudioUnitScope_Input, 0, &elementCount, &size); if (err != noErr) { printf("Error getting element count\n"); exit(1); } printf("Number of Channels (Stream Format): %d\n", numChannels); printf("Number of Elements (Element Count): %d\n", elementCount); } void printDeviceName(AudioDeviceID deviceID) { UInt32 size; OSStatus err; CFStringRef deviceName = NULL; size = sizeof(deviceName); err = AudioObjectGetPropertyData( deviceID, &(AudioObjectPropertyAddress){kAudioDevicePropertyDeviceNameCFString, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMain}, 0, NULL, &size, &deviceName); if (err != noErr) { printf("Error getting device name\n"); exit(1); } char deviceNameStr[256]; if (!CFStringGetCString(deviceName, deviceNameStr, sizeof(deviceNameStr), kCFStringEncodingUTF8)) { printf("Error converting device name to C string\n"); exit(1); } CFRelease(deviceName); printf("Device Name: %s\n", deviceNameStr); } int main(int argc, const char *argv[]) { @autoreleasepool { OSStatus err; // Get the default input device ID AudioDeviceID input_device_id = kAudioObjectUnknown; { UInt32 property_size = sizeof(input_device_id); AudioObjectPropertyAddress input_device_property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMain, }; err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &input_device_property, 0, NULL, &property_size, &input_device_id); if (err != noErr || input_device_id == kAudioObjectUnknown) { printf("Error getting default input device ID\n"); exit(1); } } // Print the device name using the input device ID printDeviceName(input_device_id); // Open audio unit for the input device AudioComponentDescription desc = {kAudioUnitType_Output, kAudioUnitSubType_HALOutput, kAudioUnitManufacturer_Apple, 0, 0}; AudioComponent component = AudioComponentFindNext(NULL, &desc); AudioUnit audioUnit; err = AudioComponentInstanceNew(component, &audioUnit); if (err != noErr) { printf("Error creating AudioUnit\n"); exit(1); } // Enable IO for input on the AudioUnit and disable output UInt32 enableInput = 1; UInt32 disableOutput = 0; err = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableInput, sizeof(enableInput)); if (err != noErr) { printf("Error enabling input on AudioUnit\n"); exit(1); } err = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &disableOutput, sizeof(disableOutput)); if (err != noErr) { printf("Error disabling output on AudioUnit\n"); exit(1); } // Set the current device to the input device err = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &input_device_id, sizeof(input_device_id)); if (err != noErr) { printf("Error setting device for AudioUnit\n"); exit(1); } // Initialize AudioUnit err = AudioUnitInitialize(audioUnit); if (err != noErr) { printf("Error initializing AudioUnit\n"); exit(1); } // Print device info printDeviceInfo(audioUnit); // Clean up AudioUnitUninitialize(audioUnit); AudioComponentInstanceDispose(audioUnit); } return 0; } It prints: Device Name: MacBook Pro Microphone Number of Channels (Stream Format): 1 Number of Elements (Element Count): 2 I tried to set the number of channels to 1 on the input unit, but it didn’t change anything. After calling setNumberOfChannels(1, audioUnit), I’m still getting the same output. Note 1: I know that I can ignore one channel, etc, etc. My purpose here is not to "somehow get it to work", I already did that. My purpose is to understand the API, so that I'll be able to write code that handles any number of audio inputs. Note 2: I already read a bunch of documentation, especially this here: https://developer.apple.com/library/archive/technotes/tn2091/ - perhaps the channel map could help here, but I can’t make sense of it - I tried to use it based on my understanding but I only got the -50 OSStatus. How should I understand this? Is it that that audio unit is an abstraction layer and automatically converts mono input into stereo input? Can I ask AUHAL to provide me the same number of input channels that the audio device has?