Hi all, I’ve implemented the new Core Audio Tap API (AudioHardwareCreateProcessTap with CATapDescription) and I’m seeing consistent level attenuation that scales with the number of stereo output pairs exposed by the target device. What I observe Device with 4 stereo pairs (8 outs) → tap shows −12.04 dB relative to source. True 2-ch devices (built-in speakers, AirPods) → ~0 dB attenuation. The attenuation appears regardless of whether I: Create a global (default-output) tap via initStereoGlobalTapButExcludeProcesses: Or create a per-process/per-device tap via initWithProcesses:andDeviceUID:withStream: Additionally, the routing choice inside the sending app matters: App output to “System/Default Output” → I often see no attenuation. App output directly to a multi-out interface (e.g., RME Fireface) → I see the pair-count-scaled attenuation. I can query Core Audio for the number of output channels/pairs and gain-compensate (+20·log10(N_pairs) dB) and that matches my measurements for many cases. However, this compensation is not universally correct because it seems to depend on where each process routes its audio (Default Output vs. direct device), even when those processes are included in the same tap aggregate. Question Is there a supported way to obtain the raw, unattenuated streams for all processes through the Tap API—i.e., to bypass this automatic headroom/attenuation behavior entirely? If this attenuation is expected by design: Is there a documented rule for when it applies (global vs. device taps, per-process taps, stream selection, etc.)? Is there a property/flag to disable it, or a reliable, official method to compute the exact compensation (beyond counting stereo pairs)? Any guidance on ensuring consistent levels when multiple processes route differently (Default Output vs. direct device) but are captured by the same tap? Environment API: AudioHardwareCreateProcessTap + CATapDescription Devices: built-in output (2-ch), RME Fireface (8+ outs / 4+ stereo pairs) Behavior reproducible with both global and per-process/per-device tap descriptions. Attenuation example: 4 stereo pairs → −12.04 dB observed. Happy to provide a minimal sample, measurements, and device logs. Thanks! — David

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!

hi all, as soon an audio is played in a whatever app, coreaudiod inserts a sleep prevent assertion for both, the system AND the display. can i somehow stop the insertion of the display sleep assertion? pid 223(coreaudiod): [0x00004e9e00058dc2] 00:03:18 PreventUserIdleDisplaySleep named: "com.apple.audio.AppleGFXHDAEngineOutputDP:10001:0:{B31A-08C6-00000000}.context.preventuseridledisplaysleep" Created for PID: 4145. where PID 4145 is spotify. but it doesn't matter which app is playing the audio. any help would be appreciated thanks

I'm using AVFoundation to make a multi-track editor app, which can insert multiple track and clip, including scale some clip to change the speed of the clip, (also I'm not sure whether AVFoundation the best choice for me) but after making the scale with scaleTimeRange API, there is some short noise sound in play back. Also, sometimes it's fine when play AVMutableCompostion using AVPlayer with AVPlayerItem, but after exporting with AVAssetReader, will catch some short noise sounds in result file.... Not sure why. Here is the example project, which can build and run directly. https://github.com/luckysmg/daily_images/raw/refs/heads/main/TestDemo.zip

Hello, I am having an issue with a very small percentage of our users not being able to view the UI of our MIDI Plugin Chord Prism. I have looked this up and seen to where it has been resolved within Logic for AU Instrument and Effect plugins by switching out of "Controls" view, but my situation is different and there is no option on what is displayed to switch out of "Controls" view. Is this something that can be fixed by adjusting settings within Logic?

I'm working on a project to support spatial audio editing, using this sample project as a reference: https://aninterestingwebsite.com/documentation/Cinematic/editing-spatial-audio-with-an-audio-mix This sample works well on an unedited capture, but does not work for a capture that has already been edited. The failure is occurring at "let audioInfo = try await CNAssetSpatialAudioInfo(asset: myAsset)", which is throwing "no eligible audio tracks in asset". I also find that for already edited captures, if i use CNAssetSpatialAudioInfo.assetContainsSpatialAudio, it returns false. What i mean by "already edited" is that if I take a spatial capture with my iPhone 16, and then edit that capture in the Photos app using the Cinematic effect, and then save the edited output (e.g. edited_capture.mov), I can't import that edited_capture.mov into my project as a spatial audio asset. Is this intentional behavior or a bug? If it's intentional, can you describe why?

When multiple identical songs are added to a playlist, Playlist.Entry.id uses a suffix-based identifier (e.g. songID_0, songID_1, etc.). Removing one entry causes others to shift, changing their .id values. This leads to diffing errors and collection view crashes in SwiftUI or UIKit when entries are updated. Steps to Reproduce: Add the same song to a playlist multiple times. Observe .id.rawValue of entries (e.g. i.SONGID_0, i.SONGID_1). Remove one entry. Fetch playlist again — note the other IDs have shifted. FB18879062

I'm using MusicKit for DRM track playback in my iOS app and a third party library to play local user-owned music on the file system and from the music library. This app is also supporting accessory devices that offer Bluetooth remote media control. The wish is to achieve parity between how the remote interacts with user owned music and the DRM / cloud / Apple Music tracks in my application music player. Track navigation, app volume (rather than system volume), and scrubbing need to work consistently on a mix of tracks which could alternate DRM and cloud status within one album or playlist. Apple Music queue and track pickers are not useful tools in my app. How can I support playing DRM and Apple Music tracks while not surrendering the remote control features to the system?

I bought two "Apple USB-C to Headphone Jack Adapters". Upon closer inspection, they seems to be of different generations: The one with product ID 0x110a on top is working fine. The one with product ID 0x110b has two issues: There is a short but loud click noise on the headphone when I connect it to the iPad. When I play audio using AVAudioPlayer the first half of a second or so is cut off. Here's how I'm playing the audio: audioPlayer = try AVAudioPlayer(contentsOf: url) audioPlayer?.delegate = self audioPlayer?.prepareToPlay() audioPlayer?.play() Is this a known issue? Am I doing something wrong?

Hello everyone, I am working on an app that allows you to review your own music using Apple Music. Currently I am running into an issue with the skipping forwards and backwards outside of the app. How it should work: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song on an album should play and the information should change to reflect that in the app. If you play a song in Apple Music, you can see a Now Playing view in the lock screen. When you skip forward or backwards, it will do either action and it would reflect that when you see a little frequency icon on artwork image of a song. What it's doing: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song is reflected outside of the app, but not in the app. When skipping a song outside of the app, it works correctly to head to the next song. But when I return to the app, it is not reflected NOTE: I am not using MusicKit variables such as Track, Album to display the songs. Since I want to grab the songs and review them I need a rating so I created my own that grabs the MusicItemID, name, artist(s), etc. NOTE: I am using ApplicationMusicPlayer.shared Is there a way to get the song to reflect in my app? (If its easier, a simple example of it would be nice. No need to create an entire xprod file)

I’m seeing what appears to be an iOS audio-session issue that occurs only when a phone call happens while the app is in the background. API: AVAudioSession, AVAudioRecorder Background Modes: Audio enabled (UIBackgroundModes = audio) Category: .playAndRecord Microphone permission: granted Expected Behavior If the app is recording audio in the background and a phone call interrupts it: AVAudioSession.interruptionNotification(.began) fires Call ends AVAudioSession.interruptionNotification(.ended) fires App should be able to re-activate its audio session and resume or restart recording Apple documentation suggests this should be supported for background audio apps. Actual Behavior When the app is in the background and phone call is ended: AVAudioSession.interruptionNotification(.ended) does fire Attempting to reactivate the audio session always fails: Error Domain=NSOSStatusErrorDomain Code=560557684 ("!int") "Session activation failed" The session appears to remain permanently “interrupted” Retrying activation (with delays) does not help Recreating AVAudioRecorder does not help Reactivation works only after the app is opened again

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

Issue Description I'm implementing a system audio capture feature using AudioHardwareCreateProcessTap and AudioHardwareCreateAggregateDevice. The app successfully creates the tap and aggregate device, but when starting the IO procedure with AudioDeviceStart, it sometimes fails with OSStatus error 1852797029. (The operation couldn’t be completed. (OSStatus error 1852797029.)) The error occurs inconsistently, which makes it particularly difficult to debug and reproduce. Questions Has anyone encountered this intermittent "nope" error code (0x6e6f7065) when working with system audio capture? Are there specific conditions or system states that might trigger this error sporadically? Are there any known workarounds for handling this intermittent failure case? Any insights or guidance would be greatly appreciated. I'm wondering if anyone else has encountered this specific "nope" error code (0x6e6f7065) when working with system audio capture.

I’m an amateur developer working on a free utility for composers/producers, for which the macOS release needs to create and name RTP-MIDI sessions in Audio MIDI Setup from the command line (so I can ship a small C helper instead of telling users to click through the UI). Here’s what I’ve tried so far, without luck: • Plist hacks: Injecting entries into ~/Library/Audio/MIDI Configurations/*.mcfg works when AMS is closed, but AMS immediately locks and reverts my changes when it’s open. • CoreMIDI C API: I can create virtual ports with MIDISourceCreate, but attempting MIDIObjectGetDataProperty on the apple.midirtp.session plugin always returns err –10836. • Obj-C & Swift: Loading MIDINetworkSession and calling defaultSession, init, setNetworkName: and setting enabled = YES doesn’t produce a new session object in the Network panel. • dlopen/dlsym: I extracted the real CoreMIDI binary out of the dyld shared cache and tried binding _MIDINetworkSessionCreate, _SetName, _SetEnabled, etc., but all the symbols come back null or my tool segfaults. • Plugin registration: I’ve pulled the factory UUID (70C9C5EA-7C65-11D8-B317-000393A34B5A) from /System/Library/Extensions/AppleMIDIRTPDriver.plugin/Contents/Info.plist and called CFPlugInRegisterFactories, but it still never exposes the session-creation calls. At this point I’m convinced I’m either loading the wrong binary or missing one critical step in registering the RTP-MIDI plugin’s private API. Can anyone point me to: The exact path of the dylib or bundle that actually exports the MIDINetworkSessionCreate/MIDINetworkSessionSetName/MIDINetworkSessionSetEnabled symbols? A minimal working snippet (C or Obj-C) that reliably creates and names a Network-MIDI session? Any pointers, sample code, or even ideas about where Apple hides this functionality on macOS 15 would be hugely appreciated. Thanks!

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.

I have an application that includes custom artwork for the album cover and text details setup with the MPRemoteCommandCenter.shared() reference. I need the user to have a full featured "now playing" display to see all of this. My experience is that cannot find a set of parameters for AVAudioSession.setCategory() that route audio successfully, and display the full featured now playing deck. If I use .playAndRecord, the audio I send out plays out on the radio. But, the now-playing deck is empty and nothing I do with the command center seems to change that. If I instead use .playback, I cannot use .defaultToSpeaker option which is the only way I've found to cause the "now-playing" navigation button to appear so that the full featured deck will display. But, of course setCategory() fails with an error about .defaultToSpeaker only available with .playAndRecord, so some default or intermediate state is entered and I see the full featured deck, but no audio goes out to the radio. What combination is supposed to be used here and is this more likely a problem with thread use (@MainActor) and/or some ordering of operations that I've overlooked?

With the Voice Conversational Entitlement, can a CarPlay app establish a turn-based audio interface that operates in two modes: Speaking mode: Audio Session configured for playback Buffered audio Listening mode: Switch Audio Session to .record or .playAndRecord Activate SFSpeechRecognizer And continue toggling back and forth. The app should listen for responses to questions or other audio cues, and assuming those answers are correct (based on analysis of results from SFSpeechRecognizer), continue this pattern of mode 1 and 2 alternating. This appears to be a valid use of this entitlement. Does this also require the Audio App Entitlement, or is the Voice Conversational Entitlement sufficient? Are there other obstacles to this type of app that I'm not seeing? Or perhaps this is technically possible, but unlikely to pass app store review?

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! ⁠

Is there any way for me to use an AutoMix api in my IOS apps, I would play tracks using the Apple Music api and use AutoMix to attempt to merge tracks. Is this feature/api available to developers.

I developed an educational app that implements audio-video communication through RTC, while using WebView to display course materials during classes. However, some users are experiencing an issue where the audio playback from WebView is very quiet. I've checked that the AVAudioSessionCategory is set by RTC to AVAudioSessionCategoryPlayAndRecord, and the AVAudioSessionCategoryOption also includes AVAudioSessionCategoryOptionMixWithOthers. What could be causing the WebView audio to be suppressed, and how can this be resolved?