Hi, I am trying to remove the audio controls for my app on the lock screen. Since I use WKWebView, there are 3 audio tags in my html and I play and pause em via JS. However, if I do not play any sound since app launch, there are no audio controls on the lock screen. But if I play one of those 3 files (they are even less then 3 Sec sound effects e.g. for buttons) the audio controls appears on lock screen. Note even when the sounds on pause() or not playing they were listed on the lock screen. What I have tried so far without success MPNowPlayingInfoCenter.default().nowPlayingInfo = [:] and ``try audioSession.setCategory(.playback, mode: .default, options: []) try audioSession.setActive(false, options: .notifyOthersOnDeactivation)`` and UIApplication.shared.endReceivingRemoteControlEvents() Another problem is that the app scales with iOS system settings "display zoom". Is there a way to deny it? It is latest Xcode verion 16.3 and iOS 18. I have no background mode in my Capabilities. Nothing worked so far. Has anyone an idea? Greetings

Hi, our CourAudio server plugin utilizes the SystemConfiguration.framework to store and restore specific shared system wide settings. While our application can authenticate to utilize the SystemConfiguration.framework to gain write access to the shared configuration settings the CoreAudio server plugin obviously can't have any user interaction and therefor does not authenticate. Is it possible to authenticate the CoreAudio server plugin to gain write permissions? Are there any entitlements or other means that would allow this? Thanks!

Hi, when using ApplicationMusicPlayer from MusicKit my app automatically gets the media controls on the lock screen: Play/ Pause, Skip Buttons, Playback Position etc. I would like to customize these. Tried a bunch of things, e.g. using MPRemoteCommandCenter. So far I haven't had any success. Does anyone know how I can customize the media controls of ApplicationMusicPlayer. Thank you.

I'm using a 4 channel USB Audio interface, with 4 microphones, and want to process them through 4 independent effect chains. However the output from AVAudioInputNode is a single 4 channel bus. How can I split this into 4 mono busses? The following code splits the input into 4 copies, and routes them through the effects, but each bus contains all four channels. How can I remap the channels to remove the unwanted channels from the bus? I tried using channelMap on the mixer node but that had no effect. I'm currently using this code primarily on iOS but it should be portable between iOS and MacOS. It would be possible to do this through a Matrix Mixer Node, but that seems completely overkill, for such a basic operation. I'm already using a Matrix Mixer to combine the inputs, and it's not well supported in AVAudioEngine. AVAudioInputNode *inputNode=[engine inputNode]; [inputNode setVoiceProcessingEnabled:NO error:nil]; NSMutableArray *micDestinations=[NSMutableArray arrayWithCapacity:trackCount]; for(i=0;i<trackCount;i++) { fixMicFormat[i]=[AVAudioMixerNode new]; [engine attachNode:fixMicFormat[i]]; // And create reverb/compressor and eq the same way... [engine connect:reverb[i] to:matrixMixerNode fromBus:0 toBus:i format:nil]; [engine connect:eq[i] to:reverb[i] fromBus:0 toBus:0 format:nil]; [engine connect:compressor[i] to:eq[i] fromBus:0 toBus:0 format:nil]; [engine connect:fixMicFormat[i] to:compressor[i] fromBus:0 toBus:0 format:nil]; [micDestinations addObject:[[AVAudioConnectionPoint alloc] initWithNode:fixMicFormat[i] bus:0] ]; } AVAudioFormat *inputFormat = [inputNode outputFormatForBus: 1]; [engine connect:inputNode toConnectionPoints:micDestinations fromBus:1 format:inputFormat];

So, I've been wondering how fast a an offline STT -> ML Prompt -> TTS roundtrip would be. Interestingly, for many tests, the SpeechTranscriber (STT) takes the bulk of the time, compared to generating a FoundationModel response and creating the Audio using TTS. E.g. InteractionStatistics: - listeningStarted: 21:24:23 4480 2423 - timeTillFirstAboveNoiseFloor: 01.794 - timeTillLastNoiseAboveFloor: 02.383 - timeTillFirstSpeechDetected: 02.399 - timeTillTranscriptFinalized: 04.510 - timeTillFirstMLModelResponse: 04.938 - timeTillMLModelResponse: 05.379 - timeTillTTSStarted: 04.962 - timeTillTTSFinished: 11.016 - speechLength: 06.054 - timeToResponse: 02.578 - transcript: This is a test. - mlModelResponse: Sure! I'm ready to help with your test. What do you need help with? Here, between my audio input ending and the Text-2-Speech starting top play (using AVSpeechUtterance) the total response time was 2.5s. Of that time, it took the SpeechAnalyzer 2.1s to get the transcript finalized, FoundationModel only took 0.4s to respond (and TTS started playing nearly instantly). I'm already using reportingOptions: [.volatileResults, .fastResults] so it's probably as fast as possible right now? I'm just surprised the STT takes so much longer compared to the other parts (all being CoreML based, aren't they?)

Hello everyone, I'm implementing the new AVInputPickerInteraction API on iOS 26 to allow users to select their microphone from a custom settings menu before recording. The implementation seems correct, but I'm encountering a strange issue where the input selection immediately reverts to the previous device. The Situation: The picker is presented correctly via a manual call to .present(). I can see all available inputs (e.g., "iPhone Microphone" and "AirPods"). The current input is "iPhone Microphone". I tap on "AirPods". The UI updates to show "AirPods" as selected for a fraction of a second, then immediately jumps back to "iPhone Microphone". The same thing happens in reverse. It seems like the system is automatically reverting the audio route change requested by the picker. My Implementation: My setup follows the standard pattern discussed in the WWDC sessions. Setup Code: This setup is performed once before the user can trigger the picker. @available(iOS 26.0, *) var inputPickerInteraction: AVInputPickerInteraction? // Note: The AVAudioSession is configured to .playAndRecord // and set to active elsewhere in the code before this setup is called. if #available(iOS 26.0, *) { // Setup the picker let picker = AVInputPickerInteraction() self.inputPickerInteraction = picker self.view.addInteraction(picker) // Added to establish context } Presentation Code: When a user selects "Change Input" from my custom settings menu, I call .present() on the main thread. // In a delegate method from a custom menu if #available(iOS 26.0, *) { DispatchQueue.main.async { self.inputPickerInteraction?.present(animated: true) } } What I've already checked: The AVAudioSession is active and its category is .playAndRecord. The inputPickerInteraction object is not nil. The .present() method is being called on the main thread. The picker is added to a view using view.addInteraction() in the setup phase. I've reviewed my code to ensure there is no other logic that could be manually resetting the AVAudioSession's preferred input. Has anyone else experienced this behavior? I suspect this might be a bug in the new API, but I want to make sure I'm not missing a crucial step in managing the AVAudioSession state. Any insights or potential workarounds would be greatly appreciated. Thank you.

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. 😅

After upgrading to iOS 18.4, I'm no longer able to establish an AirPlay v1 connection to an audio system. The symptom is that the AirPlay route picker just spins when trying to connect to an audio system. It eventually gives up. I tested this on an iPhone 14, connecting to a HomePod, AirPort express, AppleTV and a Wiim Pro. If I try connecting with AirPlay v2, ex: using Apple Music, the connection succeeds and audio can be played. I'm the developer of an app that plays audio over AirPlay while also recording. My app has to use AirPlay v1 because AvAudioSession doesn't allow the policy .longFormAudio when the category is .playAndRecord. This issue is a real pain as it means my app is suddenly broken for many thousands of users. Is anyone else seeing this issue? Any suggestions for a workaround?

The presentation "create audio drivers with DriverKit" from WWDC 2021 demonstrates how to use a dext to implement a virtual audio driver. It also says " If a virtual audio driver or device is all that is needed, the audio server plug-in driver model should continue to be used". Indeed, in AudioDriverKit/AudioDriverKitTypes.h, there is no IOUserAudioTransportType Virtual, although CoreAudio/AudioHardwareBase.h includes kAudioDeviceTransportTypeVirtual. For one of our products, we require virtual devices to implement a software loopback "cable". We've implemented this using the "traditional" HAL plugin, and as a proof-of-concept, also using a dext. In the dext, I tried setting the transport type to 'virt', which seems to only have the effect of changing the icon shown in Audio Midi Setup. HAL plugins require an installer, and the installer has to kill coreaudiod in a post-install script. You have to turn off SIP to debug them. Just like AudioDriverKit drivers, they are out-of-process and run in a process not owned by the hosting app. Our HAL plugin's interface is property based; we had to write a lot of boiler-plate code to implement required properties. Writing an AudioDriverKit driver is in most respects easier - a lot of the scaffolding is implemented in the base driver, which we only alter where required. Debugging and installation is much easier. The dext works just fine, as far as we can ascertain, just as well as a HAL plugin. So, my question is - is the advice to use a HAL plugin for a virtual device still correct in 2025? And if so, what's the objection? We'd really prefer to ship the AudioDriverKit virtual audio device.

I'm experiencing a significant limitation with MusicKit's Dolby Atmos implementation on macOS and would appreciate clarification on whether this is intended behavior or if there are solutions available. When streaming Dolby Atmos content through MusicKit's ApplicationMusicPlayer, the output is limited to 2-channel stereo, even when: Audio MIDI Setup is configured for 7.1.4 (12-channel) output The same tracks play in full multichannel through the native Apple Music app Dolby Atmos is set to "Automatic" in Apple Music preferences Please let me know if there is anyway to enable this. If not, is this documented anywhere? Thanks!

Hello, I’m new here. I'm developing an iOS app and I’d like to know whether it is possible to detect if a phone call is being recorded by another app running in the background. I’ve already reviewed the documentation for CallKit and AVAudioSession, but I couldn’t find anything related. My expectation was that iOS might provide some callback or API to indicate if a call is being recorded (third-party apps), but so far I haven’t found a way. My questions are: Does iOS expose any API to detect if a call is being recorded? If not, is there any indirect, Apple's policy compliant method (e.g., microphone usage events) that can be relied upon? Or is this something that iOS explicitly prevents for privacyreasons? Expecting solutions that align with Apple’s policies and would be accepted under the App Store Review Guidelines. Thanks in advance for any guidance.

Hi, I am creating an app that can include videos or images in it's data. While @Attribute(.externalStorage) helps with images, with AVAssets I actually would like access to the URL behind that data. (as it would be stupid to load and then save the data again just to have a URL) One key component is to keep all of this clean enough so that I can use (private) CloudKit syncing with the resulting model. All the best Christoph

I have a SwiftUI app - (https://youtu.be/VbAfUk_eYl0?si=JxUBh0Bpb-vc1E1U) - which I thought was almost ready for release - a manager for airdropped audio files from Logic Pro or other music creation applications. It uses AVAudioEngine and AVAudioPlayerNode to play audio, and the MediaPlayer API to integrate with car audio and similar, all of which works well. It does not currently have an explicit CarPlay integration (and I'm slightly horrified at the amount of work that is going to require). I had the good or bad luck of getting a loaner car with carplay while mine is being repaired yesterday, and lo and behold, when connected to the vehicle via CarPlay, there is no audio output in the vehicle at all. The now playing panel correctly shows the information my app provides about the currently playing song; the player node believes it is playing, the AVAudioSession is configured as it should be. But there is no sound. Obviously I cannot ship it in this state. I've tried fiddling with the parameters the AVAudioSession is configured with, in case there was some parameter that was preventing audio output, to no avail - currently: var options = AVAudioSession.CategoryOptions() options.insert(.allowAirPlay) options.insert(.allowBluetooth) options.insert(.allowBluetoothA2DP) try session.setCategory(.playback, mode: .default, options: options) try? session.setPreferredIOBufferDuration(0.002) // ~96 samples at 44.1kHz try? session.setPrefersNoInterruptionsFromSystemAlerts(true) try? session.setPrefersInterruptionOnRouteDisconnect(false) try session.setActive(true, options: [.notifyOthersOnDeactivation]) All diagnostics within the app show the player operating correctly - files are played and flushed; AVAudioPlayerNodeCompletionCallbacks are called when they should be. But the output is not audible in the vehicle. I would much prefer to ship this app without full-blown CarPlay integration, but with working audio when connected via CarPlay, and work on full CarPlay integration for the next release. Is there some secret handshake I am just missing to make this work?

I might have misunderstood the docs, but is Call Translation going to be available for VOIP applications? Eg in an already connected VOIP call, would it be possible for Call Translations to be enabled on an iOS 26 and Apple Intelligence supported device? I have personally tried it and it doesn’t look like it supported VOIP but would love to confirm this. reference: https://developer.apple.com/documentation/callkit/cxsettranslatingcallaction/

I work on an iOS app that records video and audio. We've been getting reports for a while from users who are experiencing their video recordings being cut off. After investigating, I found that many users are receiving the AVAudioSessionMediaServicesWereResetNotification (.mediaServicesWereResetNotification) notification while recording. It's associated with the AVFoundationErrorDomain[-11819] error, which seems to indicate that the system audio daemon crashed. We have a handler registered to end the recording, show the user a prompt, and restart our AV sessions. However, from our logs this looks to be happening to hundreds of users every day and it's not an ideal user experience, so I would like to figure out why this is happening and if it's due to something that we're doing wrong. The debug menu option to trigger the audio session reset is not of much use, because it can't be triggered unless you leave the app and go to system settings. So our app can't be recording video when the debug reset is triggered. So far I haven't found a way to reproduced the issue locally, but I can see that it's happening to users from logs. I've found some posts online from developers experiencing similar issues, but none of them seem to directly address our issue. The system error doesn't include a userInfo dictionary, and as far as I can tell it's a system daemon crash so any logs would need to be captured from the OS. Is there any way that I could get more information about what may be causing this error that I may have missed?

Dear Sirs, I’ve written a virtual audio driver based on AudioDriverKit and running as dext in my MacOS app. Sometimes when waking up from a sleep state the recording side of my driver extension seems to hang and I don’t see any calls to my io_operation callback. Then the recording app like a DAW seems to hang when trying to start a recording. This doesn’t happen after short sleep states or after a complete new start of my MacBook. I already opened a case in Feedback-Assistant on 5th of May (FB17503622) which also includes a sysdiagnose and a ktrace but I didn't get any feedback so far. Meanwhile some of our customers are getting angry and I'd like to know if there's anything I could do to fix this problem on my side. We’re not sure whether this worked in previous MacOS versions, we think we didn’t observe this before 15.3.1 but at least since 15.3.1. we’ve seen this problem. Best regards, Johannes

I am trying to stream audio from local filesystem. For that, I am trying to use an AVAssetResourceLoaderDelegate for an AVURLAsset. However, Content-Length is not known at the start. To overcome this, I tried several methods: Set content length as nil, in the AVAssetResourceLoadingContentInformationRequest Set content length to -1, in the ContentInformationRequest Both of these cause the AVPlayerItem to fail with an error. I also tried setting Content-Length as INT_MAX, and setting a renewalDate = Date(timeIntervalSinceNow: 5). However, that seems to be buggy. Even after updating the Content-Length to the correct value (e.g. X bytes) and finishing that loading request, the resource loader keeps getting requests with requestedOffset = X with dataRequest.requestsAllDataToEndOfResource = true. These requests keep coming indefinitely, and as a result it seems that the next item in the queue does not get played. Also, .AVPlayerItemDidPlayToEndTime notification does not get called. I wanted to check if this is an expected behavior or is there a bug in this implementation. Also, what is the recommended way to stream audio of unknown initial length from local file system? Thanks!

Hi, On macOS I used to open MP3 and MP4 files with ExtAudioFile. For a few years it doesn't work anymore. So I decided to try different macOS API using the AudioFileID of AudioToolbox framework. I decided to write a test: https://gist.github.com/joelkraehemann/7f5b241b52ca38c3a765c138fb647588 It fails right here: AudioFileOpenWithCallbacks() By telling OSStatus error 1954115647, which means kAudioFileUnsupportedFileTypeError. The filename was set to an MP4 file: ~/Music/test.mp4 Howto fix this? regards, Joël

Hi everyone, I’m testing audio recording on an iPhone 15 Plus using AVFoundation. Here’s a simplified version of my setup: let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatLinearPCM), AVSampleRateKey: 8000, AVNumberOfChannelsKey: 1, AVLinearPCMBitDepthKey: 16, AVLinearPCMIsFloatKey: false ] audioRecorder = try AVAudioRecorder(url: fileURL, settings: settings) audioRecorder?.record() When I check the recorded file’s sample rate, it logs: Actual sample rate: 8000.0 However, when I inspect the hardware sample rate: try session.setCategory(.playAndRecord, mode: .default) try session.setActive(true) print("Hardware sample rate:", session.sampleRate) I consistently get: `Hardware sample rate: 48000.0 My questions are: Is the iPhone mic actually capturing at 8 kHz, or is it recording at 48 kHz and then downsampling to 8 kHz internally? Is there any way to force the hardware to record natively at 8 kHz? If not, what’s the recommended approach for telephony-quality audio (true 8 kHz) on iOS devices? Thanks in advance for your guidance!

There appears to be no method of going forward or backwards in Get Info in the Music application,