I have some tried-and-tested code that records and plays back audio via AUHAL which breaks on Tahoe on Intel. The same code works fine on Sequioa and also works on Tahoe on Apple Silicon. To start with something simple, the following code to request access to the Microphone doesn't work as it should: bool RequestMicrophoneAccess () { __block AVAuthorizationStatus status = [AVCaptureDevice authorizationStatusForMediaType: AVMediaTypeAudio]; if (status == AVAuthorizationStatusAuthorized) return true; __block bool done = false; [AVCaptureDevice requestAccessForMediaType: AVMediaTypeAudio completionHandler: ^ (BOOL granted) { status = (granted) ? AVAuthorizationStatusAuthorized : AVAuthorizationStatusDenied; done = true; }]; while (!done) CFRunLoopRunInMode (kCFRunLoopDefaultMode, 2.0, true); return status == AVAuthorizationStatusAuthorized; } On Tahoe on Intel, the code runs to completion but granted is always returned as NO. Tellingly, the popup to ask the user to grant microphone access is never displayed, even though the app is not present in the Privacy pane and never appears there. On Apple Silicon, everything works fine. There are some other problems, but I'm hoping they have a common underlying cause and that the Apple guys can figure out what's wrong from the information in this post. I'd be happy to test any potential fix. Thanks.

Currently, I have successfully used ChannelMap to map hardware input channels and obtained audio data from the hardware device's MIC and OTG inputs. Additionally, I have used ChannelMap to map output channels to freely feed data for playback to each output channel. However, I now have a problem. I have a hardware device that only has output channels (no input channels), and the system has set this hardware device as the default playback device. In this case, how can I obtain the audio data being played to the output channels for modification?

Is there a way to destroy MIDIUMPMutableEndpoint again? In my app, the user has a setting to enable and disable MIDI 2.0. If MIDI 2.0 should not be supported (or if iOS version < 18), it creates a virtual destination and a virtual source. And if MIDI 2.0 should be enabled, it instead creates a MIDIUMPMutableEndpoint, which itself creates the virtual destination and source automatically. So here is my problem: I didn't find any way to destroy the MIDIUMPMutableEndpoint again. There is a method to disable it (setEnabled:NO), but that doesn't destroy or hide the virtual destination and source. So when the user turns MIDI 2.0 support off, I will have two virtual destinations and sources, and cannot get rid of the 2.0 ones. What is the correct way to get rid of the MIDIUMPMutableEndpoint once it is created?

Hi everyone, I’m trying to use AVAssetResourceLoaderDelegate to handle a live radio stream (e.g. Icecast/HTTP stream). My goal is to have access to the last 30 seconds of audio data during playback, so I can analyze it for specific audio patterns in near-real-time. I’ve implemented a custom resource loader that works fine for podcasts and static files, where the file size and content length are known. However, for infinite live streams, my current implementation stops receiving new loading requests after the first one is served. As a result, the playback either stalls or fails to continue. Has anyone successfully used AVAssetResourceLoaderDelegate with a continuous radio stream? Or maybe you can suggest betterapproach for buffering and analyzing live audio? Any tips, examples, or advice would be appreciated. Thanks!

Among the millions of users of our online product, we have identified through data metrics that the silent audio data capture rate on iPadOS 18.4.1 or 18.5 has increased abnormally. However, we are unable to reproduce the issue. Has anyone encountered a similar issue? The parameters we used are as follows: AudioSession: category:AVAudioSessionCategoryPlayAndRecord mode:AVAudioSessionModeDefault option:77 preferredSampleRate:48000.000000 preferredIOBufferDuration:0.010000 AudioUnit format.mFormatID = kAudioFormatLinearPCM; format.mSampleRate = 48000.0; format.mChannelsPerFrame = 2; format.mBitsPerChannel = 16; format.mFramesPerPacket = 1; format.mBytesPerFrame = format.mChannelsPerFrame * 16 / 8; format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket; format.mFormatFlags = kAudioFormatFlagsNativeEndian | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger; component.componentType = kAudioUnitType_Output; component.componentSubType = kAudioUnitSubType_RemoteIO; component.componentManufacturer = kAudioUnitManufacturer_Apple; component.componentFlags = 0; component.componentFlagsMask = 0;

Thread 5 Crashed: 0 libobjc.A.dylib 0x19af7b038 objc_msgSend + 56 1 CoreFoundation 0x19dfdb618 cow_cleanup + 135 2 CoreFoundation 0x19dfdb6fc -[__NSDictionaryM dealloc] + 147 3 MediaToolbox 0x1b167636c FigRemotePropertyCacheTeardown + 31 4 MediaToolbox 0x1b1c5b648 remoteXPCAsset_Finalize + 107 5 CoreMedia 0x1b1e9166c FigBaseObjectFinalize + 275 6 CoreFoundation 0x19dfcc5ec _CFRelease + 295 7 AVFCore 0x1b1054d64 -[AVFigAssetTrackInspector dealloc] + 151 8 AVFCore 0x1b0f818d8 -[AVAssetTrack dealloc] + 63 9 CoreFoundation 0x19dfdba28 RELEASE_OBJECTS_IN_THE_ARRAY + 115 10 CoreFoundation 0x19dfdb7e0 -[__NSArrayM dealloc] + 147 11 AVFCore 0x1b0f52e04 -[AVURLAsset dealloc] + 167 12 libobjc.A.dylib 0x19af821f8 object_cxxDestructFromClass(objc_object*, objc_class*) + 115 13 libobjc.A.dylib 0x19af7df20 objc_destructInstance_nonnull_realized(objc_object*) + 75 14 libobjc.A.dylib 0x19af7d4a4 _objc_rootDealloc + 71 15 AVFCore 0x1b0fef988 -[AVAssetReaderOutput dealloc] + 415 16 AVFCore 0x1b0ff11ec -[AVAssetReaderTrackOutput dealloc] + 127 17 CoreFoundation 0x19dfe20a4 -[__NSSingleObjectArrayI dealloc] + 63 18 libobjc.A.dylib 0x19af7d3f8 AutoreleasePoolPage::releaseUntil(objc_object**) + 203

Hi all, I've developed an audio DSP application in C++ using AudioToolbox and CoreAudio on MacOS 14.4.1 with Xcode 15. I use an AudioQueue for input and another for output. This works great. I'm now adding real-time audio analysis eg spectral analysis. I want this to run independently of my audio processing so it can not interfere with audio playback. Taps on AudioQueues seem to be a good way of doing this... Since the analytics won't modify the audio data, I am using a Siphon Tap by setting the AudioQueueProcessingTapFlags to kAudioQueueProcessingTap_PreEffects | kAudioQueueProcessingTap_Siphon; This works fine on my output queue. However, on my input queue the Tap callback is called once and then a EXC_BAD_ACCESS occurs - screen shot below. NB: I believe that a callback should only call AudioQueueProcessingTapGetSourceAudio when not using a Siphon, so I don't call it. Relevant code: AudioQueueProcessingTapCallback tap_callback) { // Makes an audio tap for a queue void * tap_data_ptr = NULL; AudioQueueProcessingTapFlags tap_flags = kAudioQueueProcessingTap_PostEffects | kAudioQueueProcessingTap_Siphon; uint32_t max_frames = 0; AudioStreamBasicDescription asbd; AudioQueueProcessingTapRef tap_ref; OSStatus status = AudioQueueProcessingTapNew(queue_ref, tap_callback, tap_data_ptr, tap_flags, &max_frames, &asbd, &tap_ref); if (status != noErr) printf("Error while making Tap\n"); else printf("Successfully made tap\n"); } void tapper(void * tap_data, AudioQueueProcessingTapRef tap_ref, uint32_t number_of_frames_in, AudioTimeStamp * ts_ptr, AudioQueueProcessingTapFlags * tap_flags_ptr, uint32_t * number_of_frames_out_ptr, AudioBufferList * buf_list) { // Callback function for audio queue tap printf("Tap callback"); }``` Image of exception stack provided by Xcode: ![]("https://aninterestingwebsite.com/forums/content/attachment/27479e8d-a118-459b-aa2d-7e30528910e3" "title=Screenshot 2025-06-14 at 1.29.14 PM.png;width=932;height=562") What have I missed? Appreciate any help you learned folks may be able to provide. Best, Geoff.

When trying to load an ambisonics file using this project: https://github.com/robertncoomber/NativeiOSAmbisonicPlayback/ I get "Unexpected Ambisonics format". Interestingly, loading a 3rd order ambisonics file works fine: let ambisonicLayoutTag = kAudioChannelLayoutTag_HOA_ACN_SN3D | 16 let AmbisonicLayout = AVAudioChannelLayout(layoutTag: ambisonicLayoutTag) let StereoLayout = AVAudioChannelLayout(layoutTag: kAudioChannelLayoutTag_Stereo) So it's purely related to the kAudioChannelLayoutTag_Ambisonic_B_Format

My audio app shows a control bar at the bottom of the window. The controls show nicely, but there is a black "slab" appearing behind the inline controls, the same size as the playerView. Setting the player view background color does nothing: playerView.wantsLayer = true playerView.layer?.backgroundColor = NSColor.clear.cgColor How can I clear the background? If I use .floating controlsStyle, I don't get the background "slab".

We’ve encountered a reproducible issue where the iPhone fails to reconnect to a Wi-Fi access point under the following conditions: The device is connected to a 2.4GHz Wi-Fi network. A Bluetooth audio accessory is connected (e.g. headset). AVAudioSession is active (such as during a voice call or when using the Voice Memos app). The user moves away from the access point, causing a disconnect. Upon returning within range, the access point is no longer recognized or reconnected while AVAudioSession remains active. However, if the Bluetooth device is disconnected or the AVAudioSession is deactivated, the Wi-Fi access point is immediately recognized again. We confirmed this behavior not only in my app but also using Apple's built-in Voice Memos app, suggesting this is not specific to our implementation. It appears that the Wi-Fi system deprioritizes reconnection while AVAudioSession is engaged. Could this be by design? Or is this a known issue or limitation with Wi-Fi and AVAudioSession interaction? Test Environment: Device: iPhone 13 mini iOS: 17.5.1 Wi-Fi: 2.4GHz band Accessories: Bluetooth headset We’d appreciate clarification on whether this is expected behavior or a bug. Thank you!

I am developing an app that uses MusicKit to play music and then I need to have spoken words played to the user, while ducking the audio coming from MusicKit (application music player) the built in Siri voices are not off sufficient quality so I am using an external service to create an mp3 file and then play this back using AVAudioSession Sample code below the problem I am having is that .duckOthers is not ducking the Application Music Player output Is this a bug or am I doing this wrong? // Configure audio session for system-wide ducking try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, options: [.duckOthers, .mixWithOthers]) try AVAudioSession.sharedInstance().setActive(true) // Set the ducking level to maximum try AVAudioSession.sharedInstance().setPreferredIOBufferDuration(0.005) // Create and configure audio player self.audioPlayer = try AVAudioPlayer(data: audioData) self.audioPlayer?.delegate = self self.audioPlayer?.volume = 1.0 // Ensure full volume for speech self.audioPlayer?.prepareToPlay() // Set the audio player's settings for maximum clarity self.audioPlayer?.enableRate = false self.audioPlayer?.pan = 0.0 // Center the audio self.audioPlayer?.play()

I need to apply headphone-specific scenario only when headphones are the sole active playback device in my iOS audio app. Problem that there is no absolute way to definitively understand that headphones are the sole active playback device AVAudioSession.currentRoute.outputs portTypes don't guarantee headphones: let session = AVAudioSession.sharedInstance() let outputs = session.currentRoute.outputs let headphonesOnly = outputs.count == 1 && (outputs.first?.portType == .headphones || outputs.first?.portType == .bluetoothA2DP || outputs.first?.portType == .bluetoothHFP || outputs.first?.portType == .bluetoothLE) The issue in code above that listed bluetooth profiles (A2DP, HFP, LE) can be used by any audio device, not only headphones Is there any public API on iOS that can: Distinguish Bluetooth headphones vs Bluetooth speakers when both use A2DP/LE? Expose the user’s “Device Type” classification (headphones / speaker / car stereo, etc.) that is shown in Settings → Bluetooth → Device Type? Provide a more reliable way to know “this route is definitely headphones” for A2DP devices, beyond portType and portName string heuristics?

Hi everyone 👋 I’m building an iOS app in Swift where I want to do the following: Record the user’s voice Transcribe the spoken sentence (speech-to-text) Auto-detect the spoken language Translate it to another language selected by the user (e.g., English → Spanish or Hindi → English) Speak back (text-to-speech) the translated text on the same device Is this possible to record via phone mic and play the transcribe voice into headphone's audio?

Good day, ladies and gents. I have an application that reads audio from the microphone. I'd like it to also be able to read from the Mac's audio output stream. (A bonus would be if it could detect when the Mac is playing music.) I'd eventually be able to figure it out reading docs, but if someone can give a hint, I'd be very grateful, and would owe you the libation of your choice. Here's the code used to set up the AudioUnit: -(NSString*) configureAU { AudioComponent component = NULL; AudioComponentDescription description; OSStatus err = noErr; UInt32 param; AURenderCallbackStruct callback; if( audioUnit ) { AudioComponentInstanceDispose( audioUnit ); audioUnit = NULL; } // was CloseComponent // Open the AudioOutputUnit description.componentType = kAudioUnitType_Output; description.componentSubType = kAudioUnitSubType_HALOutput; description.componentManufacturer = kAudioUnitManufacturer_Apple; description.componentFlags = 0; description.componentFlagsMask = 0; if( component = AudioComponentFindNext( NULL, &description ) ) { err = AudioComponentInstanceNew( component, &audioUnit ); if( err != noErr ) { audioUnit = NULL; return [ NSString stringWithFormat: @"Couldn't open AudioUnit component (ID=%d)", err] ; } } // Configure the AudioOutputUnit: // You must enable the Audio Unit (AUHAL) for input and output for the same device. // When using AudioUnitSetProperty the 4th parameter in the method refers to an AudioUnitElement. // When using an AudioOutputUnit for input the element will be '1' and the output element will be '0'. param = 1; // Enable input on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &param, sizeof(UInt32) ); chkerr("Couldn't set first EnableIO prop (enable inpjt) (ID=%d)"); param = 0; // Disable output on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(UInt32) ); chkerr("Couldn't set second EnableIO property on the audio unit (disable ootpjt) (ID=%d)"); param = sizeof(AudioDeviceID); // Select the default input device AudioObjectPropertyAddress OutputAddr = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; err = AudioObjectGetPropertyData( kAudioObjectSystemObject, &OutputAddr, 0, NULL, &param, &inputDeviceID ); chkerr("Couldn't get default input device (ID=%d)"); // Set the current device to the default input unit err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDeviceID, sizeof(AudioDeviceID) ); chkerr("Failed to hook up input device to our AudioUnit (ID=%d)"); callback.inputProc = AudioInputProc; // Setup render callback, to be called when the AUHAL has input data callback.inputProcRefCon = self; err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(AURenderCallbackStruct) ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); param = sizeof(AudioStreamBasicDescription); // get hardware device format err = AudioUnitGetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &deviceFormat, &param ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); audioChannels = MAX( deviceFormat.mChannelsPerFrame, 2 ); // Twiddle the format to our liking actualOutputFormat.mChannelsPerFrame = audioChannels; actualOutputFormat.mSampleRate = deviceFormat.mSampleRate; actualOutputFormat.mFormatID = kAudioFormatLinearPCM; actualOutputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved; if( actualOutputFormat.mFormatID == kAudioFormatLinearPCM && audioChannels == 1 ) actualOutputFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved; #if __BIG_ENDIAN__ actualOutputFormat.mFormatFlags |= kAudioFormatFlagIsBigEndian; #endif actualOutputFormat.mBitsPerChannel = sizeof(Float32) * 8; actualOutputFormat.mBytesPerFrame = actualOutputFormat.mBitsPerChannel / 8; actualOutputFormat.mFramesPerPacket = 1; actualOutputFormat.mBytesPerPacket = actualOutputFormat.mBytesPerFrame; // Set the AudioOutputUnit output data format err = AudioUnitSetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &actualOutputFormat, sizeof(AudioStreamBasicDescription)); chkerr("Could not change the stream format of the output device (ID=%d)"); param = sizeof(UInt32); // Get the number of frames in the IO buffer(s) err = AudioUnitGetProperty( audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &audioSamples, &param ); chkerr("Could not determine audio sample size (ID=%d)"); err = AudioUnitInitialize( audioUnit ); // Initialize the AU chkerr("Could not initialize the AudioUnit (ID=%d)"); // Allocate our audio buffers audioBuffer = [self allocateAudioBufferListWithNumChannels: actualOutputFormat.mChannelsPerFrame size: audioSamples * actualOutputFormat.mBytesPerFrame]; if( audioBuffer == NULL ) { [ self cleanUp ]; return [NSString stringWithFormat: @"Could not allocate buffers for recording (ID=%d)", err]; } return nil; } (...again, it would be nice to know if audio output is active and thereby choose the clean output stream over the noisy mic, but that would be a different chunk of code, and my main question may just be a quick edit to this chunk.) Thanks for your attention! ==Dave [p.s. if i get more than one useful answer, can i "Accept" more than one, to spread the credit around?] {pps: of course, the code lines up prettier in a monospaced font!}

Hi, I try to record audio on the iPhone with the AVAudioRecorder and Xcode 26.0.1. Maybe the problem is that I can not record audio with the simulator. But there's a menu for audio. In the plist I added 'Privacy - Microphone Usage Description' and I ask for permission before recording. if await AVAudioApplication.requestRecordPermission() { print("permission granted") recordPermission = true } else { print("permission denied") } Permission is granted. let settings: [String : Any] = [ AVFormatIDKey: kAudioFormatMPEG4AAC, AVSampleRateKey: 12000, AVNumberOfChannelsKey: 1, AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue ] recorder = try AVAudioRecorder(url: filename, settings: settings) let prepared = recorder.prepareToRecord() print("prepared started: \(prepared)") let started = recorder.record() print("recording started: \(started)") started is always false and I tried many settings. Error messages AddInstanceForFactory: No factory registered for id <CFUUID 0x600000211480> F8BB1C28-BAE8-11D6-9C31-00039315CD46 AudioConverter.cpp:1052 Failed to create a new in process converter -> from 0 ch, 12000 Hz, .... (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame to 1 ch, 12000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame, with status -50 AudioQueueObject.cpp:1892 BuildConverter: AudioConverterNew returned -50 from: 0 ch, 12000 Hz, .... (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame to: 1 ch, 12000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame prepared started: true AudioQueueObject.cpp:7581 ConvertInput: aq@0x10381be00: AudioConverterFillComplexBuffer returned -50, packetCount 5 recording started: false All examples I find are the same, but apparently there must be something different.

I am building an app where I am using AudioRecordingIntent to start audio recording from shortcuts / Action button etc. Whenever I set that up, I notice that I get an error - Unknown NSError Live Activity start failed: The operation couldn’t be completed. Target is not foreground I explicitly try to start the live activity and then start the audio recording and that's when I see this error. How can I make this work? I am unable to find any examples.

I’m building a teleprompter-style app that relies on Picture in Picture. PiP starts correctly on device. Everything works — until another app (e.g. TikTok / Instagram) starts active video recording. When camera capture begins in the foreground app, iOS terminates my PiP session. Some teleprompter apps appear to keep PiP active while recording in other apps, so I’m trying to understand the recommended architectural pattern for this scenario. Is there a documented approach or best practice to keep PiP stable during third-party camera capture? Looking specifically for guidance on the correct AVKit / AVAudioSession configuration for this use case.

Hello. To determine wether "AVB/EAV Mode" of a AV-capable network interfaces is turned on or off I query the IO registry and evaluate the property "AVBControllerState". I was wondering if this is the "correct" approach and if there is anything known about the values for this property? Network interfaces without AV capability may also carry this property (e.g.: for my WiFi adapter the value of 1) whereas the value for interfaces with AV capability can be 0 and 3. At least as far as I could observe with my limited amount of test devices at hand. Is it safe to assume that a value of 3 means this feature is turned on, 0 that it is turned off and ignore values of 1? Is there another approach to get to know the status of the "AVB/EAV Mode"? Thanks for any insight. Best regards, Ingo

Hello! We stumbled upon a problem with our karaoke app where user on iPhone 16e/iOS 18.5 has problem with mic capture, other users cannot hear him. The mic capture is working fine on 17.5, 16.8. Maybe there is something else we need when configuring AVAudioSession for iOS 18.5? Currently it's set up like this: override func viewDidLoad() { super.viewDidLoad() UIApplication.shared.isIdleTimerDisabled = true mRoomId = appDelegate.getRoomId() let audioSession = AVAudioSession.sharedInstance() try! audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: [.defaultToSpeaker]) try! audioSession.setPreferredSampleRate(48000) try! audioSession.setActive(true, options: []) }

The Android MusicKit documentation documents two functions that are not actually exposed/added to the SDK. https://aninterestingwebsite.com/musickit/android/com/apple/android/music/playback/controller/MediaPlayerController.html#canSetRadioLikeState-- https://aninterestingwebsite.com/musickit/android/com/apple/android/music/playback/controller/MediaPlayerController.html#setRadioLikeState-int- Is the documentation stale or is the SDK out of date?