So I'm using AVAudioEngine. When playing audio I become the 'now playing' app using MPNowPlayingInfoCenter/MPRemoteCommandCenter APIs. When configuring MPRemoteCommandCenter I add a play/pause command target via -addTargetWithHandler on the togglePlayPauseCommand property. Now I also have a play/pause button in my app's UI. When I pause playback from my app's UI (which means I'm the active app, I'm in the foreground), what I do is this: -I pause the AVAudioPlayerNode I'm using with AVAudioEngine. I do not, stop, reset, etc. the AVAudioEngine. I only pause the player node. My thought process here is that the user just pressed pause and it is very likely that he will hit 'play' to resume playback in the near future because My app is in the foreground and the user just hit the pause button. Now if my app moves to the background and if I receive a memory warning I presume it'd make sense to tear down the engine or pause it. Perhaps I'm wrong about this? So when I initially hit the play button from my app's UI I also activate my AVAudioSession. I do this in high priority NSOperation since the documentation warns that "we recommend that applications not activate their session from a thread where a long blocking operation will be problematic." So now I'm playing and I hit pause from my app's UI. Then I quickly bring up the "Now Playing" center and I see I'm the "Now Playing" app but the play-pause button is showing the pause icon instead of the play icon but I'm in the pause state. I do set MPNowPlayingInfoCenter's playbackState to MPNowPlayingPlaybackStatePaused when I pause. Not surprisingly this doesn't work. The documentation states this is for macOS only. So the only way to get MPRemoteCommandCenter to show the "play" image for the play-pause button is to deactivate my AVAudioSession when I pause playback? Since I change the active state of my audio session in a NSOperation because documentation recommends "we recommend that applications not activate their session from a thread where a long blocking operation will be problematic." the play-pause toggle in the remote command center won't immediately update since I'm doing it on another thread. IMO it feels kind of inappropriate for a play-pause button to wait on a NSOperation activating the audio session before updating its UI when I already know my play/paused state, it should update right away like the button in my app does. Wouldn't it be nicer to just use MPNowPlayingInfoCenter's playbackState property on iOS too? If I'm no the longer the now playing app/active audio session it doesn't matter since I'm not in the now playing UI, just ignore it? Also is it recommended that I deactivate my audio session explicitly every time the user pauses audio in my app (when I'm in the foreground)? Also when I do deactivate the audio session I get an error: AVAudioSessionErrorCodeIsBusy (but the button in the now playing center updates to the proper image). I do this : -(void)pause { [self.playerNode pause]; [self runOperationToDeactivateAudioSession]; // This does nothing on iOS: MPNowPlayingInfoCenter *nowPlayingCenter = [MPNowPlayingInfoCenter defaultCenter]; nowPlayingCenter.playbackState = MPNowPlayingPlaybackStatePaused; } So in -runOperationToDeactivateAudioSession I get the AVAudioSessionErrorCodeIsBusy. According to the documentation Starting in iOS 8, if the session has running I/Os at the time that deactivation is requested, the session will be deactivated, but the method will return NO and populate the NSError with the code property set to AVAudioSessionErrorCodeIsBusy to indicate the misuse of the API. So pausing the player node when pausing isn't enough to meet the deactivation criteria. I guess I have to pause or stop the audio engine. I could probably wait until I receive a scene went to background notification or something before deactivating my audio session (which is async, so the button may not update to the correct image in time). This seems like a lot of code to have to write to get a play-pause toggle to update, especially in iPad-multi window scene environment. What's the recommended approach? Should I pause the AudioEngine instead of the player node always? Should I always explicitly deactivate my audio session when the user pauses playback from my app's UI even if I'm in the foreground? I personally like the idea of just being able to set [MPNowPlayingInfoCenter defaultCenter].playbackState = MPNowPlayingPlaybackStatePaused; But maybe that's because that would just make things easier on me. This does feels overcomplicated though. If anyone can share some tips on how I should handle this, I'd appreciate it.

I have spent a long time refactoring lots of older Swift code to compile without error in Swift 6. The app is a v3 audio unit host and audio unit. Having installed Sonoma and XCode 16 I compile the code using Swift 6 and it compiles and runs without any warnings or errors. My host will load my AU no problem. LOGIC PRO is still the ONLY audio unit host that will load native Mac V3 audio units and so I like to test my code using Logic. In Sonoma with XCode 16... My AU passes the most stringent AUVAL tests both in terminal and Logic pro. If I compile the AU source in Swift 5 Logic will see the AU, load it and run it without problems. But when I compile the AU in Swift 6 Logic sees the AU, will scan it and verify it passes the tests but will not load the AU. In XCode I see a log message that a "helper application failed to run" but the debugger never connects to the AU and I don't think Logic even gets as far as instantiating the AU. So... what is causing this? I'm stumped.. Developing AUv3 is a brain-aching maze of undocumented hurdles and I'm hoping someone might have found a solution for this one. Meanwhile I guess my only option is to continue using the Swift 5 compiler. (appending a little note just to mention that all the DSP code is written in C/C++, Swift is used mainly for the user interface and also does some offline thready work )

I’m currently developing an iOS metronome app using DispatchSourceTimer as the timer. The interval is set very small, around 50 milliseconds, and I’m using CFAbsoluteTimeGetCurrent to calculate the elapsed time to ensure the beat is played within a ±0.003-second margin. The problem is that once the app goes to the background, the timing becomes unstable—it slows down noticeably, then recovers after 1–2 seconds. When coming back to the foreground, it suddenly speeds up, and again, it takes 1–2 seconds to return to normal. It feels like the app is randomly “powering off” and then “overclocking.” It’s super frustrating. I’ve noticed that some metronome apps in the App Store have similar issues, but there’s one called “Professional Metronome” that’s rock solid with no such problems. What kind of magic are they using? Any experts out there who can help? Thanks in advance! P.S. I’ve already enabled background audio permissions. The professional metronome that has no issues: https://link.zhihu.com/?target=https%3A//apps.apple.com/cn/app/pro-metronome-%25E4%25B8%2593%25E4%25B8%259A%25E8%258A%2582%25E6%258B%258D%25E5%2599%25A8/id477960671

I'm working with modern Core Audio API introduced in macOS Sequoia. I have an AudioHadwareDevice which has several controls of type AudioHardwareControl. I figured out to filter only volume controls I can use classID == kAudioVolumeControlClassID condition. Some devices have volume controls for both input and output. How I can determine the direction of the control? Streams, i.e. AudioHardwareStream object have direction, but I didn't found a way to map controls to streams. There are kAudioObjectPropertyScopeInput and kAudioObjectPropertyScopeOutput property scopes, but no matter what I tried controls always return false to any control.hasProperty(address: whatever). Any other ideas?

Hello! I used the Apple CA Playthrough example code that pipes audio between devices. It uses AudioUnit callbacks to pipe the input to an output device, and I created a system equalizer with it - however users reported it stopped working in macOS 15. I am getting the error HALPlugIn.cpp:552 HALPlugIn::DeviceGetCurrentTime: got an error from the plug-in routine, Error: 1937010544 (stop) for the output device and no sound coming out of the speakers. The error only occurs when using a virtual device as an input, not using the microphone. First I thought the problem was in the loopback driver, but it also does not work with other loopback drivers like Blackhole. STEPS TO REPRODUCE Install a virtual device, for example "brew install blackhole-2ch" and run the CAPlayThrough example code (you need to add Mic Permission in the info.plist). Then set your system audio output to the virtual device, select the device as input in CAPlayThrough and hit start. You should see the error in console. My question: What did change in macOS 15 that could cause this? Is it something with the new permission handling maybe?

If I call AudioDeviceStart on an AudioDevice in my application then "Hey Siri!" will not wake Siri up. Our users have complained that Siri does not get activated with my application is running. We found that calling AudioDeviceStart is causing the issue. How should we handle this?

Hi everyone, I wanted to bring up a question about Core Audio and its potential for future updates or improvements, specifically regarding latency optimization. As someone who relies on Core Audio for real-time audio processing, any enhancements in this area would be incredibly beneficial for professionals in the industry. Does anyone know if Apple has shared any plans or updates regarding Core Audio’s performance, particularly for low-latency applications? I’d appreciate any insights or advice from the community! Thanks so much! Best, Michael

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

Hello, I'm developing a Command Line Tool in XCode, in order to capture system audio and save it to a file, which will then be used by a separate process. Everything works perfectly when running it from either XCode or the native terminal application (see image below), but as soon as I try to run it from any 3rd party application, it doesn't ask for permissions to record sound, and the resultant file ends up soundless. When archiving it and then running it from other 3rd party applications, e.g Warp (terminal) or spawning it as a child process from a bundled Electron application, it doesn't ask for permissions. Things of note: I've codesigned the application with "Developer ID Application" I've added NSAudioCaptureUsageDescriptionto Info.plist I've included Info.plist in the binary (see image below) I've added the com.apple.security.device.audio-input entitlement I've used the following resources as inspiration: https://github.com/insidegui/AudioCap https://developer.apple.com/documentation/coreaudio/capturing-system-audio-with-core-audio-taps As my use-case involves spawning the executable from Electron as a child process, I've tried to include the appropriate permissions to the parent application too, without success. I'm really at a loss here, it feels like I've tried everything. Any pointers are much appreciated! Thanks

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 problem I have at the moment is that if a phone call comes in during my recording, even if I don't answer, my recording will be interrupted The phenomenon of recording interruption is that the picture is stuck, and the recording can be resumed automatically after the call is over. But it will cause the recorded video sound and painting out of sync Through the AVCaptureSessionWasInterrupted listening, I can get to record the types of alerts and interrupt As far as I can tell, a ringing or vibrating phone can block the audio channel. I found the same scenario in other apps, you can turn off the ring tone or vibration, but I don't know how to do it, I tried a lot of ways, but it doesn't work BlackmagicCam or ProMovie App, when a call comes in during recording, there will only be a notification menu, and there will be no ringtone or vibration, which solves the problem of recording interruption I don't know if this requires some configuration or application, please let me know if it does

I am unable to access the Int32 error from the errors that CoreAudio throws in Swift type AudioHardwareError. This is critical. There is no way to access the errors or even create an AudioHardwareError to test for errors. do { _ = try AudioHardwareDevice(id: 0).streams // will throw } catch { if let error = error as? AudioHardwareError { // cast to AudioHardwareError print(error) // prints error code but not the errorDescription } } How can get reliably get the error.Int32? Or create a AudioHardwareError with an error constant? There is no way for me to handle these error with code or run tests without knowing what the error is. On top of that, by default the error localizedDescription does not contain the errorDescription unless I extend AudioHardwareError with CustomStringConvertible. extension AudioHardwareError: @retroactive CustomStringConvertible { public var description: String { return self.localizedDescription } }

Description As of iOS 18, AVAudioSession.setPreferredIOBufferDuration ignores the requested buffer size when Sound Recognition or Vocal Shortcuts is enabled. This results in 1) much larger buffer sizes and 2) mismatched buffer sizes between input and output buffers, which causes ‘glitchy’ audio and increased latency. Additionally, when this issue occurs AVAudioSession.setPreferredIOBufferDuration continues to return ‘true’ and no error is produced. Steps to Reproduce: Enable Vocal Shortcuts on a device running iOS 18. Enable at least one shortcut (e.g. Control Center). Open or clone the example project (https://github.com/cwalo/SoundRecognitionBug) Build and install the example project Attach a headset and launch the application Observe console logs showing a requested buffer size of 0.005805 (256 samples @ 48k) an actual buffer size of 0.023220 (1104 samples @48k - this is regularly the resulting buffer size in all of our tests) Quit the app and detach the headset. Enable mutesOutput in AudioSystem.mm (to avoid feedback) Launch the application Observe Same result from step 4 Mismatched hardware buffer size of 1104 and recorded frame count of 1024 Mismatched playbackCount and recordCount Quit the app and disable vocal shortcuts Launch the app Observe IOBufferDuration matching the requested duration and matched buffer sizes (expected behavior) Expected results: Requested IOBufferDuration is respected or AVAudioSession returns false or error is produced Input and output buffer sizes match Device(s): iPhone 11 Pro, iPad Pro OS: iOS 18.0.1 Environment: Xcode 16.1 FB: FB15715421 Related to: https://forums.developer.apple.com/forums/thread/765477

I’m experiencing an unusual audio issue with AirPods on macOS Sequoia while developing VoIP applications like Zoom and FaceTime. When AirPods are connected, the other party’s voice sometimes sounds unnaturally stretched (approximately twice as long). This problem can be temporarily fixed by switching the sound output settings from AirPods to speakers and then back to AirPods. From our analysis, the issue appears to be related to the sample rate provided by AudioObjectGetPropertyData. Here’s what we’ve observed: When the issue occurs, the AudioStreamBasicDescription.sampleRate for AirPods is reported as 48000. Under normal conditions, it’s reported as 24000. It seems like the system is mistakenly returning a sample rate that doesn’t match the AirPods’ actual settings, perhaps defaulting to a system speaker value. Once the output setting is toggled, the correct sampleRate (24000) is retrieved. This discrepancy causes our application to transmit the audio stream at 48000, leading to the distorted playback. Has anyone encountered a similar issue or knows how to resolve it?

I'm trying to setup a listener for kAudioProcessPropertyIsRunningOutput but it's never triggered. I get calls for kAudioProcessPropertyIsRunning and kAudioProcessPropertyDevices but not for kAudioProcessPropertyIsRunningInput or kAudioProcessPropertyIsRunningOutput. class MyDelegate: PropertyListenerDelegate { func propertiesChanged(properties: [AudioObjectPropertyAddress]) { print(properties) } } var myDelegate = MyDelegate() var processes = try AudioHardwareSystem.shared.processes for process in processes { process.delegates += [myDelegate] try process.addListener(forProperties: [AudioObjectPropertyAddress(mSelector: kAudioPropertyWildcardPropertyID, mScope: kAudioObjectPropertyScopeWildcard, mElement: kAudioObjectPropertyElementWildcard)]) } Xcode 16.1 macOS 15.0.1

Hi, I have configured the stream as interleaved, but I am unsure if the function produces interleaved samples. So here my question: Does AudioDeviceCreateIOProcID produce interleaved samples with microphone input?