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];

Hello, Has anyone else experienced variations in the accuracy of the playbackTime value? After a few seconds of playback, the reported time adjusts by a fraction of a second, making it difficult to calculate the actual playbackTime of the audio. This can be recreated by playing a song in MusicKit, recording the start time of the audio, playing for at least 10-20 seconds, and then comparing the playbackTime value to one calculated using the start time of the audio. In my experience this jump occurs after about 10 seconds of playback. Any help would be appreciated. Thanks!

I'm developing an iOS app that requires continuous audio recording. Currently, when a phone call comes in, the AVAudioSession is interrupted and recording stops completely during the ringing phase. While I understand recording should stop if the call is answered, my app needs to continue recording while the phone is merely ringing. I've observed that Apple's Voice Memos app maintains recording during incoming call rings. This indicates the hardware and iOS are capable of supporting this functionality. Request Please advise on any available AVAudioSession configurations or APIs that would allow my app to: Continue recording during an incoming call ring Only stop recording if/when the call is actually answered Impact This interruption significantly impacts the user experience and core functionality of my app. Workarounds like asking users to enable airplane mode are impractical and create a poor user experience. Questions Is there an approved way to maintain microphone access during call rings? If not currently possible, could this capability be considered for addition to a future iOS SDK? Are there any interim solutions or best practices Apple recommends for this use case? Thank you for your help. SUPPORT INFORMATION Did someone from Apple ask you to submit a code-level support request? No Do you have a focused test project that demonstrates your issue? Yes, I have a focused test project to submit with my request What code level support issue are you having? Problems with an Apple framework API in my app

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.

Hi 👋! We have a SpriteKit-based app where we play AVAudio sounds in three different ways: Effects (incl. UI sounds) with AVAudioPlayer. Long looping tracks with AVAudioPlayer. Short animation effects on the timeline of SpriteKit's SKScene files (effectively SKAudioNode nodes). We've found that when you exit the app or otherwise interrupt audio plays, future audio plays often fail. For example, there's a WebKit-based video trailer inside the app, and if you play it, our looping background music track (2.) will stop playing, and won't resume as you close the trailer (return from WebKit). This is probably due to us not manually restarting the track (so may well be easily fixed). Periodically played AVAudioPlayer audio (1.) are not affected. However, the more concerning thing is that the audio tracks on SKScene file timelines (3.) will no longer play. My hypothesis is that AVAudioEngine gets interrupted, and needs to be restarted for those AVAudioNode elements to regain functionality. Thing is, we don't deal with AVAudioEngine at all currently in the app, meaning it is never initiated to begin with. Obviously things return to normal when you remove the app from short-term memory and restart it. However, it seems many of our users aren't doing this, and often report audio failing presumably due to some interruption in the past without the app ever being cleared from memory. Any idea why timeline-run SKAudioNodes would fail like this? Should the app react to app backgrounding/foregrounding regarding audio? Any help would be very much appreciated ✌️!

I have an app that records a health provider’s conversation with a patient. I am using Audio Queue Services for this. If a phone call comes in while recording, the doctor wants to be able to ignore the call and continue the conversation without touching the phone. If the doctor answers the call, that’s fine – I will stop the recording. I can detect when the call comes in and ends using CXCallObserver and AVAudioSession.interruptionNotification. Unfortunately, when a call comes in and before it is answered or dismissed, the audio is suppressed. After the call is dismissed, the audio continues to be suppressed. How can I continue to get audio from the mic as long as the user does not answer the phone call?

I'm working in Swift/SwiftUI, running XCode 16.3 on macOS 15.4 and I've seen this when running in the iOS simulator and in a macOS app run from XCode. I've also seen this behaviour with 3 different audio files. Nothing in the documentation says that the speechRecognitionMetadata property on an SFSpeechRecognitionResult will be nil until isFinal, but that's the behaviour I'm seeing. I've stripped my class down to the following: private var isAuthed = false // I call this in a .task {} in my SwiftUI View public func requestSpeechRecognizerPermission() { SFSpeechRecognizer.requestAuthorization { authStatus in Task { self.isAuthed = authStatus == .authorized } } } public func transcribe(from url: URL) { guard isAuthed else { return } let locale = Locale(identifier: "en-US") let recognizer = SFSpeechRecognizer(locale: locale) let recognitionRequest = SFSpeechURLRecognitionRequest(url: url) // the behaviour occurs whether I set this to true or not, I recently set // it to true to see if it made a difference recognizer?.supportsOnDeviceRecognition = true recognitionRequest.shouldReportPartialResults = true recognitionRequest.addsPunctuation = true recognizer?.recognitionTask(with: recognitionRequest) { (result, error) in guard result != nil else { return } if result!.isFinal { //speechRecognitionMetadata is not nil } else { //speechRecognitionMetadata is nil } } } } Further, and this isn't documented either, the SFTranscriptionSegment values don't have correct timestamp and duration values until isFinal. The values aren't all zero, but they don't align with the timing in the audio and they change to accurate values when isFinal is true. The transcription otherwise "works", in that I get transcription text before isFinal and if I wait for isFinal the segments are correct and speechRecognitionMetadata is filled with values. The context here is I'm trying to generate a transcription that I can then highlight the spoken sections of as audio plays and I'm thinking I must be just trying to use the Speech framework in a way it does not work. I got my concept working if I pre-process the audio (i.e. run it through until isFinal and save the results I need to json), but being able to do even a rougher version of it 'on the fly' - which requires segments to have the right timestamp/duration before isFinal - is perhaps impossible?

Hello, I have been running into issues with setting nowPlayingInfo information, specifically updating information for CarPlay and the CPNowPlayingTemplate. When I start playback for an item, I see lock screen information update as expected, along with the CarPlay now playing information. However, the playing items are books with collections of tracks. When I select a new track(chapter) within the book, I set the MPMediaItemPropertyTitle to the new chapter name. This change is reflected correctly on the lock screen, but almost never appears correctly on the CarPlay CPNowPlayingTemplate. The previous chapter title remains set and never updates. I see "Application exceeded audio metadata throttle limit." in the debug console fairly frequently. From that a I figured that I need to minimize updates to the nowPlayingInfo dictionary. What I did: I store the metadata dictionary in a local dictionary and only set values in the main nowPlayingInfo dictionary when they are different from the current value. I kick off the nowPlayingInfo update via a task that initially sleeps for around 2 seconds (not a final value, just for my current testing). If a previous Task is active, it gets cancelled, so that only one update can happen within that time window. Neither of these things have been sufficient. I can switch between different titles entirely and the information updates (including cover art). But when I switch chapters within a title, the MPMediaItemPropertyTitle continues to get dropped. I know the value is getting set, because it updates on the lock screen correctly. In total, I have 12 keys I update for info, though with the above changes, usually 2-4 of them actually get updated with high frequency. I am running out of ideas to satisfy the throttling thresholds to accurately display metadata. I could use some advice. Thanks.

I am work an app development on an app which request an audio function in background as an alert sound. during debug testing , the function work fine, but once I testing standalone without debugging , The function not work , it will play out the sound when I back to app. does any way to trace the issues ?

ApplicationMusicPlayer is not available on watchOS but all other platforms. Is there a technical reason for that like battery life? Same goes for SystemMusicPlayer and MPMusicPlayerController. I already filed feedbacks for that.

I created a virtual audio device to capture system audio with a sample rate of 44.1 kHz. After capturing the audio, I forward it to the hardware sound card using AVAudioEngine, also with a sample rate of 44.1 kHz. However, due to the clock sources being unsynchronized, problems occur after a period of playback. How can I retrieve the clock source of the hardware device and set it for the virtual device?

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

Environment→ ・Device: iPad 10th generation ・OS：**iOS18.3.2 We're using AVAudioPlayer to play a sound when a button is tapped. In our use case, this button can be tapped very frequently — roughly every 0.1 to 0.2 seconds. Each tap triggers the following function: var audioPlayer: AVAudioPlayer? func soundPlay(resource: String, type: String){ guard let path = Bundle.main.path(forResource: resource, ofType: type) else { return } do { audioPlayer = try AVAudioPlayer(contentsOf: URL(fileURLWithPath: path)) audioPlayer!.delegate = self try audioSession.setCategory(.playback) } catch { return } self.audioPlayer!.play() } The issue is that under high-frequency tapping (especially around 0.1–0.15s intervals), the app occasionally crashes. The crash does not occur every time, but it happens randomly — sometimes within 30 seconds, within 1 minute, or even 3 minutes of continuous tapping. Interestingly, adding a delay of 0.2 seconds between button taps seems to prevent the crash entirely. Delays shorter than 0.2 seconds (e.g.,0.15s,0.18s) still result in occasional crashes. My questions are: **Is this expected behavior from AVAudioPlayer or AVAudioSession? Could this be a known issue or a limitation in AVFoundation? Is there any documentation or guidance on handling frequent sound playback safely?** Any insights or recommendations on how to handle rapid, repeated audio playback more reliably would be appreciated.

In iOS 18, CarPlay shows an error: “There was a problem loading this content” after playback starts. Audio works fine, but the Now Playing screen doesn’t load. I’m using MPPlayableContentManager. This worked fine in iOS 17. Anyone else seeing this error in iOS 18?

I have a music app I'm developing and having a weird issue where I can see now playing info for every other platform than tvOS. As far as I can tell I have correctly configured the MPNowPlayingInfoCenter MPNowPlayingInfoCenter.default().nowPlayingInfo = nowPlayingInfo MPNowPlayingInfoCenter.default().playbackState = .playing Are there any extra requirements to get my app's now-playing info showing in control center on tvOS? Another strange issue that might be related is I can use the apple TV remote to pause audio but not resume playback, so I feel like there's something I'm missing about registering audio playback on tvOS specifically.

Your draft looks great! Here's a refined version with the iOS 17 comparison emphasized and slightly better flow: Hi Apple Engineers and fellow developers, I'm experiencing a critical regression with ShazamKit's background operation on iOS 18. ShazamKit's SHManagedSession stops identifying songs in the background after approximately 20 seconds on iOS 18, while the exact same code works perfectly on iOS 17. The behavior is consistent: the app works perfectly in the foreground, but when backgrounded or device is locked, it initially works for about 20 seconds then stops identifying new songs. The microphone indicator remains active suggesting audio access is maintained, but ShazamKit doesn't send identified songs in the background until you open the app again. Detection immediately resumes when bringing the app to foreground. My technical setup uses SHManagedSession for continuous matching with background modes properly configured in Info.plist including audio mode, and Background App Refresh enabled. I've tested this on physical devices running iOS 18.0 through 18.5 with the same results across all versions. The exact same code running on iOS 17 devices works flawlessly in the background. To reproduce: initialize SHManagedSession and start matching, begin song identification in foreground, background the app or lock device, play different songs which are initially detected for about 20 seconds, then after the timeout period new songs are no longer identified until you bring the app to foreground. This regression has impacted my production app as users who rely on continuous background music identification are experiencing a broken feature. I submitted this as Feedback ID FB15255903 last September with no solution so far. I've created a minimal demo project that reproduces this issue: https://github.com/tfmart/ShazamKitBackground Has anyone else experienced this ShazamKit background regression on iOS 18? Are there any known workarounds or alternative approaches? Given the time this issue has persisted, could we please get acknowledgment of this regression, expected timeline for a fix, or any recommended workarounds? Testing environment is Xcode 16.0+ on iOS 18.0-18.5 across multiple physical device models. Any guidance would be greatly appreciated.

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

I found that the aggregated device correctly obtains input channels in the standard microphone mode. However, in voice isolation mode, it only retrieves channels from the first sub-device in the aggregated device's list. If I want to properly obtain channel information in voice isolation mode, how should I do it?

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

When using the Apple Devices to sync Apple Music to iPhone where is the Apple Devices backup being written to? Apple Devices->music->sync. Not trying to backup the iPhone via Apple Devices app.