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

Hello everyone, I've written an audio unit plugin that needs to be aware of any upstream latency caused by heavy plugins before it on the channel. Is there any way to query this? I know that Logic applies PDC at the channel's output (summing point), but I need to know what the accumulated latency is at the point the audio enters my plugin. Thanks!

We are currently working on a CarPlay navigation app and so far everything is working well except for speaking turn notifications. Our TTS implementation works fine on the phone and works fine on CarPlay if the voice is spoken over the speaker in the car. If users connect a BT headset to the car and listen through that headset, then the voice commands are chopped up / stutter. Why would users use BT headset? Well, we are working on a motorcycle app, and there are no speakers usually on a motorcycle. It sounds like the BT channel is opened and closed repeatedly for every character / word spoken. This happens on different CarPlay devices and different Bluetooth headsets, we have reports from multiple users that they find this behavior annoying and that other apps work fine. Is this a known issue? Are there possible workaround?

Many Apple users own both Bluetooth earphones (AirPods) and traditional wired earphones. While Bluetooth audio provides freedom of movement, some users still prefer wired earphones for comfort, sound profile, or personal preference. However, plugging wired earphones directly into an iPhone can feel restrictive and inconvenient during daily use. This proposal suggests a hybrid audio approach where wired earphones can be connected to a Bluetooth-enabled AirPods charging case (or a similar Apple-designed module), allowing users to enjoy wired earphones without a physical connection to the iPhone. #Problem Statement *Wired earphones offer consistent audio quality and zero latency *Bluetooth earphones provide freedom from cables *Users must currently choose one or the other *Plugging wired earphones into an iPhone limits movement and can feel intrusive in daily scenarios (walking, commuting, working) There is no native Apple solution that allows wired earphones to function wirelessly while maintaining Apple’s audio experience standards. #Proposed Solution Introduce a Wired-to-Wireless Audio Mode through the AirPods charging case or a dedicated Apple Bluetooth audio bridge. How it works: User plugs wired earphones into the AirPods case (or a future AirPods accessory port) The case acts as a Bluetooth audio transmitter Audio is streamed wirelessly from iPhone to the case The case outputs audio to the wired earphones #User experiences: No cable connected to the iPhone Familiar wired earphone sound Freedom of movement similar to Bluetooth earbuds User Experience (UX Flow) Plug wired earphones into the AirPods case iPhone automatically detects: “Wired Earphones via AirPods Case” Seamless pairing using existing AirPods framework Audio controls, volume, and switching handled through iOS No additional apps required #Key Benefits Combines wired sound reliability with wireless convenience Reduces physical cable disturbance during use Extends usefulness of existing wired earphones Minimal learning curve for users Fits naturally into Apple’s ecosystem and design philosophy #Privacy & Performance Considerations On-device audio processing only No cloud involvement Low-latency audio using Apple’s proprietary Bluetooth codecs Power-efficient usage leveraging AirPods case battery #Target Users Users who prefer wired earphones but want wireless freedom Commuters and walkers Developers and professionals who multitask Users sensitive to Bluetooth earbud fit or comfort #Ecosystem Fit Builds on existing AirPods pairing and audio stack Aligns with Apple’s focus on seamless UX Could be implemented via: New AirPods hardware Firmware update + accessory Dedicated Apple audio bridge

Is it possible to find IDR frame (CMSampleBuffer) in AVAsset h264 video file?

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?

I need to implement a solution through an API or custom driver to completely block out the built-in speakers and microphone of Mac, because I need other apps to use specified external devices as audio input and output. Is there a way to achieve this requirement? What I mean is that even in system preferences, it should not be possible to choose the built-in microphone and speakers; only my external device can be used.

I neet to take pcm data from aac data, but this api has fossy me deeply.

Hey everyone, I'm encountering an issue with audio sample rate conversion that I'm hoping someone can help with. Here's the breakdown: Issue Description: I've installed a tap on an input device to convert audio to an optimal sample rate. There's a converter node added on top of this setup. The problem arises when joining Zoom or FaceTime calls—the converter gets deallocated from memory, causing the program to crash. Symptoms: The converter node is being deallocated during video calls. The program crashes entirely when this happens. Traditional methods of monitoring sample rate changes (tracking nominal or actual sample rates) aren't working as expected. The Big Challenge: I can't figure out how to properly monitor sample rate changes. Listeners set up to track these changes don't trigger when the device joins a Zoom or FaceTime call. Please, if anyone has experience with this or knows a solution, I'd really appreciate your help. Thanks in advance! ⁠

Recently, after the update of 26.3 Mac OS (Tahoe), the ordering of my music app has been horrible at best - music disappearing, tracks not aligning with albums (even if the albums are different years). It's created quite a problem, because the disappearing tracks issue seems to be replicating to iOS devices as well (although track numbering and album association seem to be stable). Has anyone else heard of this issue?

Hi team, With regards to Call (Live) Translations on VOIP: Is it possible to invoke live translations within the app? (without going into the Call System UI) Is it possible to navigate users from app to Call System UI via an API? (and also invoking the new live translations directly) Will Apple support more languages apart from the current ones? (Currently I see 4 supported languages)

I've filed this as FB21446798 but figured I'd post here too. In the first build of macOS 26.3, playback via ApplicationMusicPlayer is completely broken. When starting playback of anything at all, the console shows the following error: applicationController: xpc service connection interrupted Failed to obtain remoteObject: Error Domain=NSCocoaErrorDomain Code=4099 "The connection to service created from an endpoint was invalidated from this process." UserInfo={NSDebugDescription=The connection to service created from an endpoint was invalidated from this process.} Failed to prepareToPlay with error: Error Domain=MPMusicPlayerControllerErrorDomain Code=10 "(null)" UserInfo={NSUnderlyingError=0xc92910ff0 {Error Domain=NSCocoaErrorDomain Code=4099 "The connection to service created from an endpoint was invalidated from this process." UserInfo={NSDebugDescription=The connection to service created from an endpoint was invalidated from this process.}}} In addition, several crash logs for RemotePlayerService are generated, showing my app as the parent process. This issue is 100% repeatable. No matter how I load the queue, whether it’s catalog or library content, any variation I can think of all fails like this. I really hope this can be fixed before 26.3 comes out, otherwise my app will be totally unusable. 😅

Since iOS 18, the system setting “Allow Audio Playback” (enabled by default) allows third-party app audio to continue playing while the user is recording video with the Camera app. This has created a problem for the app I’m developing. ➡️ The problem: My app plays continuous audio in both foreground and background states. If the user starts recording video using the iOS Camera app, the app’s audio — still playing in the background — gets captured in the video — obviously an unintended behavior. Yes, the user could stop the app manually before starting the video recording, but that can’t be guaranteed. As a developer, I need a way to stop the app’s audio before the video recording begins. So far, I haven’t found a reliable way to detect when video recording starts if ‘Allow Audio Playback’ is ON. ➡️ What I’ve tried: — AVAudioSession.interruptionNotification → doesn’t fire — devicesChangedEventStream → not triggered I don’t want to request mic permission (app doesn’t use mic). also, disabling the app from playing audio in the background isn’t an option as it is a crucial part of the user experience ➡️ What I need: A reliable, supported way to detect when the Camera app begins video recording, without requiring mic access — so I can stop audio and avoid unintentional overlap with the user’s recordings. Any official guidance, workarounds, or AVFoundation techniques would be greatly appreciated. Thanks.

I have an iPadOS M-processor application with two different running configurations. In config1, the shared AVAudioSession is configured for .videoChat mode using the built-in microphone. The input/output nodes of the AVAudioEngine are configured with voice processing enabled. The built-in mic is formatted for 1 channel at 48KHz. In config2, the shared AVAudioSession is configured for .measurement mode using an external USB microphone. The input/output nodes of the AVAudioEngine are configured with voice processing disabled. The external mic is formatted for 2 channels at 44.1KHz I've written a configuration manager designed to safely switch between these two configurations. It works by stopping AVAudioEngine and detaching all but the input and output nodes, updating the shared audio session for the desired mic and sample-rates, and setting the appropriate state for voice processing to either true or false as required by the configuration. Finally the new audio graph is constructed by attaching appropriate nodes, connecting them, and re-starting AVAudioEngine I'm experiencing what I believe is a race-condition between switching voice processing on or off and then trying to re-build and start the new audio graph. Even though notifications, which are dumped to the console indicate that my requested input and sample-rate settings are in place, I crash when trying to start the audio engine because the sample-rate is wrong. Investigating further it looks like the switch from remote I/O to voice-processing I/O or vice-versa has not yet actually completed. I introduced a 100ms second delay and that seems to help but is obviously not a reliable way to build software that must work consistently. How can I make sure that what are apparently asynchronous configuration changes to the shared audio session and the input/output nodes have completed before I go on? I tried using route change notifications from the shared AVAudioSession but these lie. They say my preferred mic input and sample-rate setting is in place but when I dump the AVAudioEngine graph to the debugger console, I still see the wrong sample rate assigned to the input/output nodes. Also these are the wrong AU nodes. That is, VPIO is still in place when RIO should be, or vice-versa. How can I make the switch reliable without arbitrary time delays? Is my configuration manager approach appropriate (question for Apple engineers)?

I have an AUv3 that passes all validation and can be loaded into Logic Pro without issue. The UI for the plug in can be any aspect ratio but Logic insists on presenting it in a view with a fixed aspect ratio. That is when resizing, both the height and width are resized. I have never managed to work out what it is I need to do specify to Logic to allow the user to resize width or height independently of each other. Can anyone tell me what I need to specify in the AU code that will inform Logic that the view can be resized from any side of the window/panel?

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

Two issues: No matter what I set in try audioSession.setPreferredSampleRate(x) the sample rate on both iOS and macOS is always 48000 when the output goes through the speaker, and 24000 when my Airpods connect to an iPhone/iPad. Now, I'm checking the current output loudness to animate a 3D character, using mixerNode.installTap(onBus: 0, bufferSize: y, format: nil) { [weak self] buffer, time in Task { @MainActor in // calculate rms and animate character accordingly but any buffer size under 4800 is just ignored and the buffers I get are 4800 sized. This is ok, when the sampleRate is currently 48000, as 10 samples per second lead to decent visual results. But when AirPods connect, the samplerate is 24000, which means only 5 samples per second, so the character animation looks lame. My AVAudioEngine setup is the following: audioEngine.connect(playerNode, to: pitchShiftEffect, format: format) audioEngine.connect(pitchShiftEffect, to: mixerNode, format: format) audioEngine.connect(mixerNode, to: audioEngine.outputNode, format: nil) Now, I'd be fine if the outputNode runs at whatever if it needs, as long as my tap would get at least 10 samples per second. PS: Specifying my favorite format in the let format = AVAudioFormat(standardFormatWithSampleRate: 48_000, channels: 2)! mixerNode.installTap(onBus: 0, bufferSize: y, format: format) doesn't change anything either

My app utilizes background audio to play music files. I have the audio background mode enabled and I initialize the AVAudioSession in playback mode with the mixWithOthers option. And it usually works great while the app is backgrounded. I listen for audio interruptions as well as route changes and I am able to handle them appropriately and I can usually resume my background audio no problem. I discovered an issue while connected to CarPlay though. Roughly 50% of the time when I disconnect from a phone call while connected to CarPlay I get the following error after calling the play() method of my AVAudioPlayer instance: "ATAudioSessionClientImpl.mm:281 activation failed. status = 561015905" If I instead try to start a new audio session I get a similar error: Error Domain=NSOSStatusErrorDomain Code=561015905 "Session activation failed" UserInfo={NSLocalizedDescription=Session activation failed} Like I said, this isn't reproducible 100% of the time and is so far only seen while connected to CarPlay. I don't think Im forgetting so additional capability or plist setting, but if anyone has any clues it would be greatly appreciated. Otherwise this is likely just a bug that I need to report to Apple. One very important note, and reason I believe it's just a bug, is that while I was testing I found that other music apps like Spotify will also fail to resume their audio at the same time my app fails. Another important detail is that when it works successfully I receive the audio session interruption ended notification, and when it doesn't work I only receive a route configuration change or route override notification. From there I am able to still successfully granted background time to execute code, but my call to resume audio fails with the above mentioned error codes.

Hello, I'm observing an intermittent memory leak being reported in the iOS Simulator when initializing and starting an AVAudioEngine. Even with minimal setup—just attaching a single AVAudioPlayerNode and connecting it to the mainMixerNode—Xcode's memory diagnostics and Instruments sometimes flag a leak. Here is a simplified version of the code I'm using: // This function is called when the user taps a button in the view controller: #import "ViewController.h" @interface ViewController () @end @implementation ViewController - (void)viewDidLoad { [super viewDidLoad]; } - (IBAction)myButtonAction:(id)sender { NSLog(@"Test"); soundCreate(); } @end // media.m static AVAudioEngine *audioEngine = nil; void soundCreate(void) { if (audioEngine != nil) return; [[AVAudioSession sharedInstance] setCategory:AVAudioSessionCategoryAmbient error:nil]; [[AVAudioSession sharedInstance] setActive:YES error:nil]; audioEngine = [[AVAudioEngine alloc] init]; AVAudioPlayerNode* playerNode = [[AVAudioPlayerNode alloc] init]; [audioEngine attachNode:playerNode]; [audioEngine connect:playerNode to:(AVAudioNode *)[audioEngine mainMixerNode] format:nil]; [audioEngine startAndReturnError:nil]; } In the memory leak report, the following call stack is repeated, seemingly in a loop: ListenerMap::InsertEvent(XAudioUnitEvent const&, ListenerBinding*) AudioToolboxCore ListenerMap::AddParameter(AUListener*, void*, XAudioUnitEvent const&) AudioToolboxCore AUListenerAddParameter AudioToolboxCore addOrRemoveParameterListeners(OpaqueAudioComponentInstance*, AUListenerBase*, AUParameterTree*, bool) AudioToolboxCore 0x180178ddf

The sysEx struct in the MIDIUniversalMessage struct has a channel member but the System Exclusive (7-Bit) Message doesn't have a channel field. The System Exclusive (7-Bit) Message has a # of bytes field but the sysEx struct doesn't have a nrOfBytes, byteCount or bytesUsed member. It looks like the channel member of the sysEx struct contains the number of used bytes. Is this a mistake in the header or did I misunderstand something?