Hi, I'm trying to setup a AVAudioEngine for USB Audio recording and monitoring playthrough. As soon as I try to setup playthough I get an error in the console: AVAEInternal.h:83 required condition is false: [AVAudioEngineGraph.mm:1361:Initialize: (IsFormatSampleRateAndChannelCountValid(outputHWFormat))] Any ideas how to fix it? // Input-Device setzen try? setupInputDevice(deviceID: inputDevice) let input = audioEngine.inputNode // Stereo-Format erzwingen let inputHWFormat = input.inputFormat(forBus: 0) let stereoFormat = AVAudioFormat(commonFormat: inputHWFormat.commonFormat, sampleRate: inputHWFormat.sampleRate, channels: 2, interleaved: inputHWFormat.isInterleaved) guard let format = stereoFormat else { throw AudioError.deviceSetupFailed(-1) } print("Input format: \(inputHWFormat)") print("Forced stereo format: \(format)") audioEngine.attach(monitorMixer) audioEngine.connect(input, to: monitorMixer, format: format) // MonitorMixer -> MainMixer (Output) // Problem here, format: format also breaks. audioEngine.connect(monitorMixer, to: audioEngine.mainMixerNode, format: nil)

Hello, I'm working on a Flutter app targeting both Android and iOS, where I implemented ShazamKit. In order to achieve that, I first tried with the flutter_shazam_kit package, but since it's not maintained anymore, I forked it here, and tried to update it to meet the Google Play Store requirements, as you can see here: https://github.com/mregnauld/flutter_shazam_kit/tree/fix-16k Unfortunately, after trying everything, my app still doesn't meet the (not so) new 16 KB native library alignment. Also, I'm 100% sure it comes from that because the error message disappears if I remove that package from my app. So after investigating, it seems that the problem comes from the ShazamKit for Android (that you can find here: https://developer.apple.com/download/all/?q=Android%20ShazamKit), and especially the .so files in the .aar file. Is there anything I can do to fix that, or should I wait before the ShazamKit team fix that? I'm totally stuck with that so any help is highly appreciated. Thanks.

Hi all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!

It appears iOS only comes with low quality voices installed. iOS requires the user to go into settings to download higher quality voices to be used with AVSpeechUtterance. There doesn't seem to be any api that can be used to make this process easier for the app user. Is there a way / api that would allow an app to download and use a higher quality voice? Will apple ever install on default higher quality voices? We really want to use the text to speech api in iOS however the very high amount of user friction to use high quality voices is stopping us. I would appreciate a response. Thanks

I'm getting hundreds of the message below in Xcode. I've narrowed it down to when I instantiate the following AVAudioUnitComponentManager.shared() Message send exceeds rate-limit threshold and will be dropped. { reporterID=231700600717315, rateLimit=32hz }

I’m developing a VoIP app that uses Linphone and CallKit. Everything works as expected until the user enables the speaker on the native CallKit screen. After that, all subsequent calls start with the speaker already on. Even if I call AVAudioSession.sharedInstance().overrideOutputAudioPort(.none), it gets overridden when the call starts (when Linphone begins playing the ringtone). I tested this behavior in WhatsApp, and it seems to work correctly there.

I’m using the shared instance of AVAudioSession. After activating it with .setActive(true), I observe the outputVolume, and it correctly reports the device’s volume. However, after deactivating the session using .setActive(false), changing the volume, and then reactivating it again, the outputVolume returns the previous volume (before deactivation), not the current device volume. The correct volume is only reported after the user manually changes it again using physical buttons or Control Center, which triggers the observer. What I need is a way to retrieve the actual current device volume immediately after reactivating the audio session, even on the second and subsequent activations. Disabling and re-enabling the audio session is essential to how my application functions. I’ve tested this behavior with my colleagues, and the issue is consistently reproducible on iOS 18.0.1, iOS 18.1, iOS 18.3, iOS 18.5 and iOS 18.6.2. On devices running iOS 17.6.1 and iOS 16.0.3, outputVolume correctly reflects the current volume immediately after calling .setActive(true) multiple times.

Hello, I want to know if there are any restrictions with MusicKit to be used in a mobile app to be able to manipulate audio with an EQ on tracks coming from Apple Music, without modifying the actual track structure/data of course, just the audio output.

I am trying to debug the AAX version of my plugin (MIDI effect) on Pro Tools, but I am getting the following error (Mac console) when attempting to load it: dlsym cannot find symbol g_dwILResult in CFBundle etc.. I used Xcode 16.4 to build the plugin. Has anybody come across the same or a similar message? Best, Achillefs Axart Labs

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!

I’m running HomePod OS 26 on two HomePod minis and OS 18.6 on main HomePod (original) I’ve enabled Crossfade in the Home app. I’m playing Apple Music directly in the HomePod mini. Crossfade just doesn’t work on any HomePod. I can understand it not working on the HomePod - but why isn’t it working on the minis running OS 26? I’ve tried disabling and enabling Crossfade, rebooting HomePods etc but nothing?!

AVAudioSessionCategoryOptionAllowBluetooth is marked as deprecated in iOS 8 in iOS 26 beta 5 when this option was not deprecated in iOS 18.6. I think this is a mistake and the deprecation is in iOS 26. Am I right? It seems that the substitute for this option is "AVAudioSessionCategoryOptionAllowBluetoothHFP". The documentation does not make clear if the behaviour is exactly the same or if any difference should be expected... Has anyone used this option in iOS 26? Should I expect any difference with the current behaviour of "AVAudioSessionCategoryOptionAllowBluetooth"? Thank you.

I want to use Apple ringtones in my app so that the app can play some of them. Could this lead to any legal issues or App Store rejection? What are the guidelines and potential concerns I should be aware of?

RealityKit spatial audio crackles and pops on iOS 26.0 beta 5. It works correctly on iOS 18.6 and visionOS 26.0 beta 5. The APIs used are AudioPlaybackController, Entity.prepareAudio, Entity.play Videos of the expected and observed behavior are attached to the feedback FB19423059. The audio should be a consistent, repeating sound, but it seems oddly abbreviated and the volume varies unexpectedly. Thank you for investigating this issue.

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.

<Version / Build> macOS 26 beta 4 (25A5316i) usbaudiod.txt CMAudioFormatDescriptionCreate.txt

I'm streaming mp3 audio data using URLSession/AudioFileStream/AVAudioConverter and getting occasional silent buffers and glitches (little bleeps and whoops as opposed to clicks). The issues are present in an offline test, so this isn't an issue of underruns. Doing some buffering on the input coming from the URLSession (URLSessionDataTask) reduces the glitches/silent buffers to rather infrequent, but they do still happen occasionally. var bufferedData = Data() func parseBytes(data: Data) { bufferedData.append(data) // XXX: this buffering reduces glitching // to rather infrequent. But why? if bufferedData.count > 32768 { bufferedData.withUnsafeBytes { (bytes: UnsafeRawBufferPointer) in guard let baseAddress = bytes.baseAddress else { return } let result = AudioFileStreamParseBytes(audioStream!, UInt32(bufferedData.count), baseAddress, []) if result != noErr { print("❌ error parsing stream: \(result)") } } bufferedData = Data() } } No errors are returned by AudioFileStream or AVAudioConverter. func handlePackets(data: Data, packetDescriptions: [AudioStreamPacketDescription]) { guard let audioConverter else { return } var maxPacketSize: UInt32 = 0 for packetDescription in packetDescriptions { maxPacketSize = max(maxPacketSize, packetDescription.mDataByteSize) if packetDescription.mDataByteSize == 0 { print("EMPTY PACKET") } if Int(packetDescription.mStartOffset) + Int(packetDescription.mDataByteSize) > data.count { print("❌ Invalid packet: offset \(packetDescription.mStartOffset) + size \(packetDescription.mDataByteSize) > data.count \(data.count)") } } let bufferIn = AVAudioCompressedBuffer(format: inFormat!, packetCapacity: AVAudioPacketCount(packetDescriptions.count), maximumPacketSize: Int(maxPacketSize)) bufferIn.byteLength = UInt32(data.count) for i in 0 ..< Int(packetDescriptions.count) { bufferIn.packetDescriptions![i] = packetDescriptions[i] } bufferIn.packetCount = AVAudioPacketCount(packetDescriptions.count) _ = data.withUnsafeBytes { ptr in memcpy(bufferIn.data, ptr.baseAddress, data.count) } if verbose { print("handlePackets: \(data.count) bytes") } // Setup input provider closure var inputProvided = false let inputBlock: AVAudioConverterInputBlock = { packetCount, statusPtr in if !inputProvided { inputProvided = true statusPtr.pointee = .haveData return bufferIn } else { statusPtr.pointee = .noDataNow return nil } } // Loop until converter runs dry or is done while true { let bufferOut = AVAudioPCMBuffer(pcmFormat: outFormat, frameCapacity: 4096)! bufferOut.frameLength = 0 var error: NSError? let status = audioConverter.convert(to: bufferOut, error: &error, withInputFrom: inputBlock) switch status { case .haveData: if verbose { print("✅ convert returned haveData: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(haveData) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } case .inputRanDry: if verbose { print("🔁 convert returned inputRanDry: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(inputRanDry) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } return // wait for next handlePackets case .endOfStream: if verbose { print("✅ convert returned endOfStream") } return case .error: if verbose { print("❌ convert returned error") } if let error = error { print("error converting: \(error.localizedDescription)") } return @unknown default: fatalError() } } }

Hi everyone, I'm looking for a way to programmatically set the left/right audio balance to perfect center (50/50) using either a Terminal command or AppleScript. Background: The audio balance slider in System Settings &gt; Sound &gt; Output &amp; Input works functionally, but I have difficulty determining when it's positioned at the exact center point. The visual nature of the slider makes it challenging for me to achieve the precision I need, and I end up adjusting it repeatedly trying to get it perfectly centered. What I'm looking for: A Terminal command that can set the audio balance to exact center An AppleScript that accomplishes the same thing Any other programmatic method to ensure perfect 50/50 balance I've tried searching through the defaults command documentation and Core Audio frameworks but haven't found the right approach yet. Has anyone successfully automated this setting before? Any help would be greatly appreciated! Thanks in advance, Dylan

Hello, I have an AppIntent that uses the AudioPlaybackIntent to trigger my app to open and initiate an AVPlayer that plays back a media stream I control. When the phone is unlocked, everything works as I expect. The app opens and plays the audio. However, when the phone is locked, any attempt to invoke the intent causes a "Request Code" dialog to be displayed. This seems counter to what I would expect with the AudioPlaybackIntent usage. Am I able to accomplish what I'm after here with AppIntents? Does the fact that I'm using openAppWhenRun require me to have the phone unlocked somehow? import AppIntents import Foundation struct PlayStationAppIntent: AudioPlaybackIntent { static var title: LocalizedStringResource = "Play radio station" static var description: IntentDescription = .init("Play radio station") static var notification: Notification.Name = .init("playStation") static var openAppWhenRun: Bool = true init() {} func perform() async throws -> some IntentResult { AudioPlayerService.shared.play() return .result() } }

It's been an ask for a few years and I'm wondering if there are any plans, or whether the '26 SDKs/Tools allow Apple Music to work in the simulator? I develop for the Vision Pro so the usual 'fix' of running on the device is a bit of a hard ask. At the very least a small sample library that works in the simulator would be welcome (similar to how photos works) Cheers