Hi, I've had a new deck installed in my car for about 1.5 weeks. I'm having compatibility issues with my 15PM. It happens both wired and wirelessly, I get the error "Accessory not supported by this device". It used to happen all the time, now it's 50/50. Sometimes it works. I've removed and added Bluetooth multiple times on phone and deck, I bought a belkin usb-c to usb-a cable today and it seems to fix it but the problem comes back. I've changed the setting "FaceID and passcode-allow access when locked-accessories." The car stereo guy reckons it's definitely an issue with the phone not the deck, I'm inclined to believe him since the error states "by this device". Any advice appreciated.

Hello All, It seems that it's "very easy" (😬) to implement a little Swift code inside the prepared AU using Xcode 16.2 on Sequoia 15.1.1 and a Mac Studio M1 Ultra, but my issue is that I finally don't know... where. The documentation says that I've to find the AudioUnitViewController.swift file and then modify the render block : audioUnit.renderBlock = { (numFrames, ioData) in // Process audio here } in the Xcode project automatically generated, but I didn't find such a file... If somebody can help me in showing where is the file to be modified, I'll be very grateful ! Thank you very much. J

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 there! We have a suite of AudioUnit v2 plugins that have been shipped for some time as aufx plugins, and we are looking into MIDI-related platform upgrades, so we need a way to update these plugins to request MIDI from Logic (and other AU hosts) but avoid changing our AU type and subtype so we don't break existing sessions. Any ideas on how we can do this?

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)

Hello, I am wondering if it is possible to have audio from my AirPods be sent to my speech to text service and at the same time have the built in mic audio input be sent to recording a video? I ask because I want my users to be able to say "CAPTURE" and I start recording a video (with audio from the built in mic) and then when the user says "STOP" I stop the recording.

My app encountered problems when trying to open an x86 audioUnit v2 on a Silicon Mac (although Rosetta is installed). There seems to be a XPC connection issue with the AUHostingService that I don't know how to fix. I observed other host apps opening the same plugins without problem, so there is probably something wrong or incompatible in my codes. I noticed that: The issue occurs whether or not the app is sandboxed. The issue does no longer occur when the app itself runs under Rosetta. There is no error reported by CoreAudio during allocation and initialization of the audio unit. The first notified errors appears when the unit calls AudioUnitRender from the rendering callback. With most x86 plugins, the error is on first call: kAudioUnitErr_RenderTimeout and on any subsequent call: kAudioComponentErr_InstanceInvalidated On the UI side, when the Cocoa View is loaded, it appears shortly, then disappears immediately leaving its superview empty. With another x86 plugin, the Cocoa View is loaded normally, but CoreAudio still emits kAudioUnitErr_NoConnection from AudioUnitRender, whether the view has been loaded or not, and the plugin produces no sound. I also find these messages in the console (printed in that order): CLIENT ERROR: RemoteAUv2ViewController does not override - and thus cannot react to catastrophic errors beyond logging them AUAudioUnit_XPC.mm:641 Crashed AU possible component description: aumu/Helm/Tyte My app uses the AUv2 API and I suspect that working with the AUv3 API would spare me these problems. However, considering how my audio system is built (audio units are wrapped into C++ classes and most connections between units are managed on the fly from the rendering callback), it would be a lot of work to convert, and I’m even not sure that all I do with the AUv2 API would be possible with the AUv3 API. I could possibly find an intermediate solution, but in the immediate future I'm looking for the simplest and fastest possible fix. If I cannot find better, I see two fallback options: In this part of the doc: “Beginning with macOS 11, the system loads audio units into a separate process that depends on the architecture or host preference”, does “host preference” means that it would be possible to disable the “out of process” behavior, for example from the app entitlements or info.plist? Otherwise, as a last resort, I could completely disable the use of x86 audioUnits when my app runs under ARM64, for at least making things cleaner. But the Audio Component API doesn’t give any info about the plugin architecture, how could I found it? Any tip or idea about this issue will be much appreciated. Thanks in advance!

Since the last update to IOS 26.0 (23A5276f) the AirPods connect to my IPhone and the Audio is still running through the phone. They are shown in the Bluetooth Icon that they’re paired.

Not able to record audio in AAC format with 96 kHz sample rate using AVAudioRecorder or Extended Audio File services with 96 kHz input audio from input device. The audio recording settings used are let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatMPEG4AAC), AVSampleRateKey: sampleRate AVNumberOfChannelsKey: 1 AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue ] When tried using AVAudioEngine using AVAudioFile, AVAudioFile(forWriting: fileURL, // file extension .m4a settings: fileSettings, commonFormat: AVAudioCommonFormat.pcmFormatFloat32, interleaved: interleaved) else { return } got error CodecConverterFactory.cpp:977 unable to select compatible encoder sample rate AudioConverter.cpp:1017 Failed to create a new in process converter -> from 1 ch, 96000 Hz, Float32 to 1 ch, 96000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame, with status 1718449215

Hello. My app uses AVAudioRecorder to generate recording files, which are consistently only 4kb in size. Most users generate audio files normally, with only a few users experiencing this phenomenon occasionally. After uninstalling and installing the app, it will work normally, but it will reappear after a period of time. I have compared that the problematic audio files generated each time are fixed and cannot be played. Added the audioRecorderDidFinishRecording proxy method, which shows that the recording was completed normally. The user also reported that the recording is normal, but there is a problem with the generated file. How should I handle this issue? Look forward to your reply. - (void)startRecordWithOrderID:(NSString *)orderID { AVAudioSession *audioSession = [AVAudioSession sharedInstance]; [audioSession setCategory:AVAudioSessionCategoryRecord error:nil]; [audioSession setActive:YES error:nil]; NSMutableDictionary *settings = [[NSMutableDictionary alloc] init]; [settings setObject:[NSNumber numberWithFloat: 8000.0] forKey:AVSampleRateKey]; [settings setObject:[NSNumber numberWithInt: kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [settings setObject:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [settings setObject:[NSNumber numberWithInt: 1] forKey:AVNumberOfChannelsKey]; [settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; NSString *path = [WDUtility createDirInDocument:@"audios" withOrderID:orderID withPathExtension:@"wav"]; NSURL *tmpFile = [NSURL fileURLWithPath:path]; recorder = [[AVAudioRecorder alloc] initWithURL:tmpFile settings:settings error:nil]; [recorder setDelegate:self]; [recorder prepareToRecord]; [recorder record]; }

Hi all, with my app ScreenFloat, you can record your screen, along with system- and microphone audio. Those two audio feeds are recorded into separate audio tracks in order to individually remove or edit them later on. Now, these recordings you create with ScreenFloat can be drag-and-dropped to other apps instantly. So far, so good, but some apps, like Slack, or VLC, or even websites like YouTube, do not play back multiple audio tracks, just one. So what I'm trying to do is, on dragging the video recording file out of ScreenFloat, instantly baking together the two individual audio tracks into one, and offering that new file as the drag and drop file, so that all audio is played in the target app. But it's slow. I mean, it's actually quite fast, but for drag and drop, it's slow. My approach is this: "Bake together" the two audio tracks into a one-track m4a audio file using AVMutableAudioMix and AVAssetExportSession Take the video track, add the new audio file as an audio track to it, and render that out using AVAssetExportSession For a quick benchmark, a 3'40'' movie, step 1 takes ~1.7 seconds, and step two adds another ~1.5 seconds, so we're at ~3.2 seconds. That's an eternity for a drag and drop, where the user might cancel if there's no immediate feedback. I could also do it in one step, but then I couldn't use the AV*Passthrough preset, and that makes it take around 32 seconds then, because I assume it touches the video data (which is unnecessary in this case, so I think the two-step approach here is the fastest). So, my question is, is there a faster way? The best idea I can come up with right now is, when initially recording the screen with system- and microphone audio as separate tracks, to also record both of them into a third, muted, "hidden" track I could use later on, basically eliminating the need for step one and just ripping the two single audio tracks out of the movie and only have the video and the "hidden" track (then unmuted), but I'd still have a ~1.5 second delay there. Also, there's the processing and data overhead (basically doubling the movie's audio data). All this would be great for an export operation (where one expects it to take a little time), but for a drag-and-drop operation, it's not ideal. I've discarded the idea of doing a promise file drag, because many apps do not accept those, and I want to keep wide compatibility with all sorts of apps. I'd appreciate any ideas or pointers. Thank you kindly, Matthias

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

I’m working with the Push-to-Talk (PTT) framework and observing a consistent delay when starting audio capture. Scenario: A PTT call is already active The AVAudioSession is fully configured I request beginTransmission on the PTT channel I start my Audio Unit for recording (AudioOutputUnitStart) Observed behavior: AudioOutputUnitStart takes ~500 ms This happens whether I start the Audio Unit: after didBeginTransmission, or after AVAudioSession didActivate Comparison: Using the same Audio Unit, same format, and same configuration Without the PTT framework, AudioOutputUnitStart takes ~200 ms Additional notes: I am not modifying or reconfiguring AVAudioSession when requesting beginTransmission The audio session is already set up when the PTT call starts There are no interruptions or route changes at the time of starting the Audio Unit Impact: This extra latency is significant for Push-to-Talk use cases where fast transmit start is critical.

Hi everyone! I’ve developed a location-based Audio AR app in Unity with FMOD & Resonance Audio and AirPods Pro Head-Tracking to create a ubiquitous augmented soundscape experience. Think of it as an audio version of Pokémon Go, but with a more precise location requirement to ensure spatial audio is placed correctly. I want this experience to run in the background on iOS, but from what I’ve gathered, it seems Unity doesn’t support this well. So, I’m considering developing a Swift version instead. Since this is primarily for research purposes, privacy concerns are not a major issue in my case. However, I’ve come across some potential challenges: Real-time precise location updates – Can iOS provide fully instantaneous, high-accuracy location updates in the background? Continuous real-time data processing – Can an app continuously process spatial audio, head-tracking, and location data while running in the background? I’m not sure if newer iOS versions have improved in these areas or if there are workarounds to achieve this. Would this kind of experience be feasible to run in the background on iOS? Any insights or pointers would be greatly appreciated! I’m very new to iOS development, so apologies if this is a basic question. Thanks in advance!

I want the audio session to always use the built-in microphone. However, when using the setPreferredInput() method like in this example private func enableBuiltInMic() { // Get the shared audio session. let session = AVAudioSession.sharedInstance() // Find the built-in microphone input. guard let availableInputs = session.availableInputs, let builtInMicInput = availableInputs.first(where: { $0.portType == .builtInMic }) else { print("The device must have a built-in microphone.") return } // Make the built-in microphone input the preferred input. do { try session.setPreferredInput(builtInMicInput) } catch { print("Unable to set the built-in mic as the preferred input.") } } and calling that function once in the initializer, the audio session still switches to the external microphone once one is plugged in. The session's preferredInput is nil again at that point, even if the built-in microphone is still listed in availableInputs. So, why is the preferredInput suddenly reset? when would be the appropriate time to set the preferredInput again? Observing the session’s availableInputs did not work and setting the preferredInput again in the routeChangeNotification handler seems a bad choice as it’s already a bit too late then.

Hello, I have an existing AUv3 instrument plugin. In the plug in, users can access files (audio files, song projects) via a UIDocumentPickerViewController In Logic Pro, (and some other hosts, but not all), the document picker is unable to receive touches, while a keyboard case is attached to the iPad. Removing the case (this is an Apple brand iPad case) allows the interactions to resume and allows me to pick files in the usual way. One of my users reports this non-responsive behavior occurs even after disconnecting their keyboard. I have fiddled with entitlements all day, and have determined that is not the issue, since the keyboard disconnection appears to fix it every time for me. Here is my, very boilerplate, presentation code : guard let type = UTType("com.my.type") else { return } let fileBrowser = UIDocumentPickerViewController(forOpeningContentTypes: [type]) fileBrowser.overrideUserInterfaceStyle = .dark fileBrowser.delegate = self fileBrowser.directoryURL = myFileFolderURL() self.present(fileBrowser, animated: true) {

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

Hi! I am writing a browser extension that allows you to control the playback of media content on a music service website. Unfortunately Safari does not support tracking changes to the audible property in an event tabs.onUpdated. Is there an alternative to this event? I'm looking for a way to track when the automatic inference engine interrupts playback on a music service website. That you.

I’m facing a strange audio routing issue that seems specific to iPhone 14 Pro / Pro Max. I’m using LiveKit (WebRTC) in a React Native app, which uses AVAudioSession internally for audio capture (VoIP / call-style usage). 🔍 What’s happening: I’m using an external USB microphone. On these devices: iPhone 11 → ✅ USB mic works iPhone 13 → ✅ USB mic works iPhone 17 Pro → ✅ USB mic works iPhone 14 Pro Max → ❌ USB mic does NOT work On iPhone 14 Pro Max: The same USB mic: ✅ Works in Voice Memos ✅ Works in Instagram Live ❌ Does NOT appear as an input option in my app ❌ Does NOT work in WhatsApp / Instagram calls Also: In my app on iPhone 14 Pro Max, iOS does not show the audio input selector UI On iPhone 17 Pro, the same app and same build does show the selector and the USB mic works ⚙️ My audio session config ( LiveKit ): await AudioSession.setAppleAudioConfiguration({ audioCategory: 'playAndRecord', audioMode: 'default', audioCategoryOptions: ['allowBluetooth', 'defaultToSpeaker'], }); await AudioSession.startAudioSession(); ❓ My questions: Is this a known limitation or behavior specific to iPhone 14 Pro / Pro Max? Does iPhone 14 Pro have different audio routing rules for call / VoIP mode compared to other devices? Why does the same USB mic work in recording apps (Voice Memos, Instagram Live) but not in call-style apps (LiveKit, WhatsApp, Instagram call)? Is there any documented difference in AVAudioSession behavior on iPhone 14 Pro regarding external USB audio inputs?

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?