I'm developing a TTS Audio Unit Extension that needs to write trace/log files to a shared App Group container. While the main app can successfully create and write files to the container, the extension gets sandbox denied errors despite having proper App Group entitlements configured. Setup: Main App (Flutter) and TTS Audio Unit Extension share the same App Group App Group is properly configured in developer portal and entitlements Main app successfully creates and uses files in the container Container structure shows existing directories (config/, dictionary/) with populated files Both targets have App Group capability enabled and entitlements set Current behavior: Extension can access/read the App Group container Extension can see existing directories and files All write attempts are blocked with "sandbox deny(1) file-write-create" errors Code example: const char* createSharedGroupPathWithComponent(const char* groupId, const char* component) { NSString* groupIdStr = [NSString stringWithUTF8String:groupId]; NSString* componentStr = [NSString stringWithUTF8String:component]; NSURL* url = [[NSFileManager defaultManager] containerURLForSecurityApplicationGroupIdentifier:groupIdStr]; NSURL* fullPath = [url URLByAppendingPathComponent:componentStr]; NSError *error = nil; if (![[NSFileManager defaultManager] createDirectoryAtPath:fullPath.path withIntermediateDirectories:YES attributes:nil error:&error]) { NSLog(@"Unable to create directory %@", error.localizedDescription); } return [[fullPath path] UTF8String]; } Error output: Sandbox: simaromur-extension(996) deny(1) file-write-create /private/var/mobile/Containers/Shared/AppGroup/36CAFE9C-BD82-43DD-A962-2B4424E60043/trace Key questions: Are there additional entitlements required for TTS Audio Unit Extensions to write to App Group containers? Is this a known limitation of TTS Audio Unit Extensions? What is the recommended way to handle logging/tracing in TTS Audio Unit Extensions? If writing to App Group containers is not supported, what alternatives are available? Current entitlements: <dict> <key>com.apple.security.application-groups</key> <array> <string>group.com.<company>.<appname></string> </array> </dict>

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?

I am developing an app that uses MusicKit to play music and then I need to have spoken words played to the user, while ducking the audio coming from MusicKit (application music player) the built in Siri voices are not off sufficient quality so I am using an external service to create an mp3 file and then play this back using AVAudioSession Sample code below the problem I am having is that .duckOthers is not ducking the Application Music Player output Is this a bug or am I doing this wrong? // Configure audio session for system-wide ducking try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, options: [.duckOthers, .mixWithOthers]) try AVAudioSession.sharedInstance().setActive(true) // Set the ducking level to maximum try AVAudioSession.sharedInstance().setPreferredIOBufferDuration(0.005) // Create and configure audio player self.audioPlayer = try AVAudioPlayer(data: audioData) self.audioPlayer?.delegate = self self.audioPlayer?.volume = 1.0 // Ensure full volume for speech self.audioPlayer?.prepareToPlay() // Set the audio player's settings for maximum clarity self.audioPlayer?.enableRate = false self.audioPlayer?.pan = 0.0 // Center the audio self.audioPlayer?.play()

FaceTime’s screen-share audio balance is insanely absurd right now. Whenever I share media, the system audio that gets sent through FaceTime is a tiny whisper even at full volume (or even when connected to my speaker or headphones). The moment anyone on the call makes any noise at all, the shared audio ducks so hard it disappears, while the voice (or rustling or air conditioning noise) spikes to painful levels. It’s impossible to watch or listen to anything together. Also, the feature where FaceTime would shrink to a square during screen-sharing has been completely removed. That was a good feature and I'm really confused why it's gone. Now, the FaceTime window stays as a long rectangle that covers part of the content I'm trying to share (unless I do full screen tile, but then I can't pull up any other windows during the call) and can't be made smaller than about a third of the screen. You can't resize the window or adjust its dimensions, so it ends up blocking the actual media you're trying to watch. Here are some feature requests/fixes that would greatly improve the FaceTime screen-share experience: Option to adjust the shared media volume independently of call audio. Disable/toggle the extreme automatic audio docking while screen-sharing Reintroduce the minimized “floating square” mode or allow full manual resizing and repositioning of the FaceTime window during screen-share sessions. Overall, this setup makes FaceTime screen-sharing basically unusable. The audio balance is so inconsistent that it’s easier to switch to Zoom or Google Meet, which both handle shared sound correctly and let you move the call window out of the way. Until these issues are fixed, there’s no practical reason to use FaceTime for shared viewing at all.

Hi all! I have been experiencing some issues when using the AVAudioEngine to play audio and record input while doing a voice chat (through the PTT Interface). I noticed if I connect any players to the AudioGraph OR call start that the audio session becomes active (this is on iOS). I don't see anything in the docs or the header files in the AVFoundation, but is it possible that calling the stop method on an engine deactivates the audio session too? In a normal app this behavior seems logical, but when using PTT all activation and deactivation of the audio session must go through the framework and its delegate methods. The issue I am debugging is that when the engine with the input node tapped gets stopped, and there is a gap between the input and when the server replies with inbound audio to be played and something seems to be getting the hardware/audio session into a jammed state. Thanks for any feedback and/or confirmation on this behavior!

Hi everyone, I wanted to bring up a question about Core Audio and its potential for future updates or improvements, specifically regarding latency optimization. As someone who relies on Core Audio for real-time audio processing, any enhancements in this area would be incredibly beneficial for professionals in the industry. Does anyone know if Apple has shared any plans or updates regarding Core Audio’s performance, particularly for low-latency applications? I’d appreciate any insights or advice from the community! Thanks so much! Best, Michael

Overview We are producing audio in real time from an editing application and are trying to put that on an HLS stream. We attempt to submit PCM samples through an audio writer but are getting a crash after a select number of samples have been appended. Depending on the number of audio frames in the PCM buffer, we might get more iterations before the crash but it always has the same traceback (see below). Code The setup is rather simple. We took inspiration from a few sources around the web. NSMutableDictionary *audio = [[NSMutableDictionary alloc] init]; [audio setObject:@(kAudioFormatMPEG4AAC) forKey:AVFormatIDKey]; [audio setObject:[NSNumber numberWithInt:config.audioSampleRate] // 48000 forKey:AVSampleRateKey]; [audio setObject:[NSNumber numberWithInt:config.audioChannels] // 2 forKey:AVNumberOfChannelsKey]; [audio setObject:@160000 forKey:AVEncoderBitRateKey]; m_audioConfig = [[NSDictionary alloc] initWithDictionary:audio]; m_audio = [[AVAssetWriterInput alloc] initWithMediaType:AVMediaTypeAudio outputSettings:m_audioConfig]; AVAudioFrameCount audioFrames = BUFFER_SAMPLES * bCount; AVAudioPCMBuffer *pcmBuffer = [[AVAudioPCMBuffer alloc] initWithPCMFormat:m_full.pcmFormat frameCapacity:audioFrames]; pcmBuffer.frameLength = pcmBuffer.frameCapacity; AudioChannelLayout layout; memset(&layout, 0, sizeof(layout)); layout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo; CMFormatDescriptionRef format; OSStatus stats = CMAudioFormatDescriptionCreate( kCFAllocatorDefault, pcmBuffer.format.streamDescription, sizeof(layout), &layout, 0, nil, nil, &format ); for (int i = 0; i < bCount; i++) { AudioPCM pcm; audioCallback->callback(pcm); memcpy(*(pcmBuffer.int16ChannelData) + (bufferSize * i), pcm.data, bufferSize); } size_t samplesConsumed = BUFFER_SAMPLES * bCount; CMSampleBufferRef sampleBuffer; CMSampleTimingInfo timing; timing.duration = CMTimeMake(1, config.audioSampleRate); timing.presentationTimeStamp = presentationTime; timing.decodeTimeStamp = kCMTimeInvalid; OSStatus ostatus = CMSampleBufferCreate( kCFAllocatorDefault, nil, false, nil, nil, format, (CMItemCount)pcmBuffer.frameLength, 1, &timing, 0, nil, &sampleBuffer ); //// ostatus = CMSampleBufferSetDataBufferFromAudioBufferList( sampleBuffer, kCFAllocatorDefault, kCFAllocatorDefault, kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment, pcmBuffer.audioBufferList ); if (ostatus != noErr) { NSLog(@"fill audio sample from buffer list failed: %s", logAudioError(ostatus)); return; } ostatus = CMSampleBufferSetDataReady(sampleBuffer); if (ostatus != noErr) { NSLog(@"set sample buffer ready failed: %s", logAudioError(ostatus)); return; } // Finally we can attach it, then shove the presentation time forward [m_audio appendSampleBuffer:sampleBuffer]; The Crash The crash points towards some level of deallocation when the conversion tooling is done or has enough samples to process an output packet? It's had to say. 0 caulk 0x1a1e9532c caulk::alloc::tiered_allocator<caulk::alloc::size_range_tier<0ul, 1008ul, caulk::alloc::tree_allocator<caulk::alloc::chunk_allocator<caulk::alloc::page_allocator, caulk::alloc::bitmap_allocator, caulk::alloc::embed_block_memory, 16384ul, 16ul, 6ul>>>, caulk::alloc::size_range_tier<1009ul, 256000ul, caulk::alloc::guarded_edges_allocator<caulk::alloc::consolidating_free_map<caulk::alloc::page_allocator, 10485760ul>, 4ul>>, caulk::alloc::tracking_allocator<caulk::alloc::page_allocator>>::deallocate(caulk::alloc::block, unsigned long) + 636 1 AudioToolboxCore 0x1993fbfe4 ExtendedAudioBufferList_Destroy + 112 2 AudioToolboxCore 0x1993d5fe0 std::__1::__optional_destruct_base<ACCodecOutputBuffer, false>::~__optional_destruct_base[abi:ne180100]() + 68 3 AudioToolboxCore 0x1993d5f48 acv2::CodecConverter::~CodecConverter() + 196 4 AudioToolboxCore 0x1993d5e5c acv2::CodecConverter::~CodecConverter() + 16 5 AudioToolboxCore 0x1992574d8 std::__1::vector<std::__1::unique_ptr<acv2::AudioConverterBase, std::__1::default_delete<acv2::AudioConverterBase>>, std::__1::allocator<std::__1::unique_ptr<acv2::AudioConverterBase, std::__1::default_delete<acv2::AudioConverterBase>>>>::__clear[abi:ne180100]() + 84 6 AudioToolboxCore 0x199259acc acv2::AudioConverterChain::RebuildConverterChain(acv2::ChainBuildSettings const&) + 116 7 AudioToolboxCore 0x1992596ec acv2::AudioConverterChain::SetProperty(unsigned int, unsigned int, void const*) + 1808 8 AudioToolboxCore 0x199324acc acv2::AudioConverterV2::setProperty(unsigned int, unsigned int, void const*) + 84 9 AudioToolboxCore 0x199327f08 with_resolved(OpaqueAudioConverter*, caulk::function_ref<int (AudioConverterAPI*)>) + 60 10 AudioToolboxCore 0x1993281e4 AudioConverterSetProperty + 72 11 MediaToolbox 0x1a7566c2c FigSampleBufferProcessorCreateWithAudioCompression + 2296 12 MediaToolbox 0x1a754db08 0x1a70b5000 + 4819720 13 MediaToolbox 0x1a754dab4 FigMediaProcessorCreateForAudioCompressionWithFormatWriter + 100 14 MediaToolbox 0x1a77ebb98 0x1a70b5000 + 7564184 15 MediaToolbox 0x1a7804158 0x1a70b5000 + 7663960 16 MediaToolbox 0x1a7801da0 0x1a70b5000 + 7654816 17 AVFCore 0x1ada530c4 -[AVFigAssetWriterTrack addSampleBuffer:error:] + 192 18 AVFCore 0x1ada55164 -[AVFigAssetWriterAudioTrack _flushPendingSampleBuffersReturningError:] + 500 19 AVFCore 0x1ada55354 -[AVFigAssetWriterAudioTrack addSampleBuffer:error:] + 472 20 AVFCore 0x1ada4ebf0 -[AVAssetWriterInputWritingHelper appendSampleBuffer:error:] + 128 21 AVFCore 0x1ada4c354 -[AVAssetWriterInput appendSampleBuffer:] + 168 22 lib_devapple_hls.dylib 0x115d2c7cc detail::AppleHLSImplementation::audioRuntime() + 1052 23 lib_devapple_hls.dylib 0x115d2d094 void* std::__1::__thread_proxy[abi:ne180100]<std::__1::tuple<std::__1::unique_ptr<std::__1::__thread_struct, std::__1::default_delete<std::__1::__thread_struct>>, void (detail::AppleHLSImplementation::*)(), detail::AppleHLSImplementation*>>(void*) + 72 24 libsystem_pthread.dylib 0x196e5b2e4 _pthread_start + 136 Any insight would be welcome!

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) {

Hello, I'm trying to receive parquet files using the example that provided in documentation. I've done all required steps but receive constantly error 500 with "Upstream Service Error". By looking into the issues list, seems this error exists for months. Is it possible to get it working?

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.

Hello, i can successfully match music using shazamkit on Apple using SwiftUI, a simple app that let user to load an audio file and exctracts the relative match, while i am unable to match music using shamzamkit on Android. I am trying to make the same simple app but i cannot match music as i get MATCH_ATTEMPT_FAILED every time i try to. I don't know what i am doing wrong but the shazam part in the kotlin Android code is in this method : suspend fun processAudioFileInBackground( filePath: String, developerTokenProvider: DeveloperTokenProvider ) = withContext(Dispatchers.IO) { val bufferSize = 1024 * 1024 val audioFile = FileInputStream(filePath) val byteBuffer = ByteBuffer.allocate(bufferSize) byteBuffer.order(ByteOrder.LITTLE_ENDIAN) var bytesRead: Int while (audioFile.read(byteBuffer.array()).also { bytesRead = it } != -1) { val signatureGenerator = (ShazamKit.createSignatureGenerator(AudioSampleRateInHz.SAMPLE_RATE_44100) as ShazamKitResult.Success).data signatureGenerator.append(byteBuffer.array(), bytesRead, System.currentTimeMillis()) val signature = signatureGenerator.generateSignature() println("Signature: ${signature.durationInMs}") val catalog = ShazamKit.createShazamCatalog(developerTokenProvider, Locale.ENGLISH) val session = (ShazamKit.createSession(catalog) as ShazamKitResult.Success).data val matchResult = session.match(signature) println("MatchResult : $matchResult") setMatchResult(matchResult) byteBuffer.clear() } audioFile.close() } I noticed that changing Locale in catalog creation results in different result as i get NoMatch without exception. Can you please help me with this? Do i need to create a custom catalog?

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! I'm experiencing an issue with iOS's audio routing system when trying to use Bluetooth headphones for audio output while also recording environmental audio from the built-in microphone. Desired behavior: Play audio through Bluetooth headset (AirPods) Record unprocessed environmental audio from the iPhone's built-in microphone Actual behavior: When explicitly selecting the built-in microphone, iOS reports it's using it (in currentRoute.inputs) However, the actual audio data received is clearly still coming from the AirPods microphone The audio is heavily processed with voice isolation/noise cancellation, removing environmental sounds Environment Details Device: iPhone 12 Pro Max iOS Version: 18.4.1 Hardware: AirPods Audio Framework: AVAudioEngine (also tried AudioQueue) Code Attempted I've tried multiple approaches to force the correct routing: func configureAudioSession() { let session = AVAudioSession.sharedInstance() // Configure to allow Bluetooth output but use built-in mic try? session.setCategory(.playAndRecord, options: [.allowBluetoothA2DP, .defaultToSpeaker]) try? session.setActive(true) // Explicitly select built-in microphone if let inputs = session.availableInputs, let builtInMic = inputs.first(where: { $0.portType == .builtInMic }) { try? session.setPreferredInput(builtInMic) print("Selected input: \(builtInMic.portName)") } // Log the current route let route = session.currentRoute print("Current input: \(route.inputs.first?.portName ?? "None")") // Configure audio engine with native format let inputNode = audioEngine.inputNode let nativeFormat = inputNode.inputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: nativeFormat) { buffer, time in // Process audio buffer // Despite showing "Built-in Microphone" in route, audio appears to be // coming from AirPods with voice isolation applied - welp! } try? audioEngine.start() } I've also tried various combinations of: Different audio session modes (.default, .measurement, .voiceChat) Different option combinations (with/without .allowBluetooth, .allowBluetoothA2DP) Setting session.setPreferredInput() both before and after activation Diagnostic Observations When AirPods are connected: AVAudioSession.currentRoute.inputs correctly shows "Built-in Microphone" after setPreferredInput() The actual audio data received shows clear signs of AirPods' voice isolation processing Background/environmental sounds are actively filtered out... When recording a test audio played near the phone (not through the app), the recording is nearly silent. Only headset voice goes through. Questions Is there a workaround to force iOS to actually use the built-in microphone while maintaining Bluetooth output? Are there any lower-level configurations that might resolve this issue? Any insights, workarounds, or suggestions would be greatly appreciated. This is blocking a critical feature in my application that requires environmental audio recording while providing audio feedback through headphones 😅

private var audioEngine = AVAudioEngine() private var inputNode: AVAudioInputNode! func startAnalyzing() { inputNode = audioEngine.inputNode let recordingFormat = inputNode.outputFormat(forBus: 0) let hardwareSampleRate = recordingSession.sampleRate inputNode.removeTap(onBus: 0) if recordingFormat.sampleRate != hardwareSampleRate { print("。") let newFormat = AVAudioFormat(commonFormat: recordingFormat.commonFormat, sampleRate: hardwareSampleRate, channels: recordingFormat.channelCount, interleaved: recordingFormat.isInterleaved) inputNode.installTap(onBus: 0, bufferSize: 1024, format: newFormat) { buffer, time in self.processAudioBuffer(buffer, time: time) } } else { inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { buffer, time in self.processAudioBuffer(buffer, time: time) } } do { audioEngine.prepare() try audioEngine.start() } catch { print(": \(error)") } } I back the app to the background and then call startAnalyzing(), which reports an error and the background recording permissions are configured。 error： [10429:570139] [aurioc] AURemoteIO.cpp:1668 AUIOClient_StartIO failed (561145187) [10429:570139] [avae] AVAEInternal.h:109 [AVAudioEngineGraph.mm:1545:Start: (err = PerformCommand(*ioNode, kAUStartIO, NULL, 0)): error 561145187 Audio engine couldn't start. Is background boot not allowed？

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];

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

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

It's only occurs on iOS 18+. Backtrace attached below. Exception Codes: 0x0000000000000000, 0x0000000000000000 Termination Reason: SIGNAL 6 Abort trap: 6 Terminating Process: NoteKeys [24384] Triggered by Thread: 0 Last Exception Backtrace: 0 CoreFoundation 0x1a2d4c7cc __exceptionPreprocess + 164 (NSException.m:249) 1 libobjc.A.dylib 0x1a001f2e4 objc_exception_throw + 88 (objc-exception.mm:356) 2 CoreFoundation 0x1a2e47748 +[NSException raise:format:] + 128 (NSException.m:0) 3 AVFAudio 0x1bd41f4c8 -[AVMIDIPlayer play:] + 300 (AVMIDIPlayer.mm:145) 4 NoteKeys 0x1023c0670 SoundGenerator.playData() + 20 (SoundGenerator.swift:170) 5 NoteKeys 0x1023c0670 EditViewController.playBtnTapped(startIndex:) + 940 (EditViewController.swift:2034) 6 NoteKeys 0x1024497fc specialized Keyboard.playBtnTapped(sender:) + 1904 (Keyboard.swift:1249) 7 NoteKeys 0x10244631c Keyboard.playBtnTapped(sender:) + 4 (<compiler-generated>:0) 8 NoteKeys 0x10244631c @objc Keyboard.playBtnTapped(sender:) + 48 9 UIKitCore 0x1a58739cc -[UIApplication sendAction:to:from:forEvent:] + 100 (UIApplication.m:5816) 10 UIKitCore 0x1a58738a4 -[UIControl sendAction:to:forEvent:] + 112 (UIControl.m:942) 11 UIKitCore 0x1a58736f4 -[UIControl _sendActionsForEvents:withEvent:] + 324 (UIControl.m:1013) 12 UIKitCore 0x1a5fe8d8c -[UIButton _sendActionsForEvents:withEvent:] + 124 (UIButton.m:4198) 13 UIKitCore 0x1a5fea5a0 -[UIControl touchesEnded:withEvent:] + 400 (UIControl.m:692) 14 UIKitCore 0x1a57bb9ac -[UIWindow _sendTouchesForEvent:] + 852 (UIWindow.m:3318) 15 UIKitCore 0x1a57bb3d8 -[UIWindow sendEvent:] + 2964 (UIWindow.m:3641) 16 UIKitCore 0x1a564fb70 -[UIApplication sendEvent:] + 376 (UIApplication.m:12972) 17 UIKitCore 0x1a565009c __dispatchPreprocessedEventFromEventQueue + 1048 (UIEventDispatcher.m:2686) 18 UIKitCore 0x1a5659f3c __processEventQueue + 5696 (UIEventDispatcher.m:3044) 19 UIKitCore 0x1a5552c60 updateCycleEntry + 160 (UIEventDispatcher.m:133) 20 UIKitCore 0x1a55509d8 _UIUpdateSequenceRun + 84 (_UIUpdateSequence.mm:136) 21 UIKitCore 0x1a5550628 schedulerStepScheduledMainSection + 172 (_UIUpdateScheduler.m:1171) 22 UIKitCore 0x1a555159c runloopSourceCallback + 92 (_UIUpdateScheduler.m:1334) 23 CoreFoundation 0x1a2d20328 __CFRUNLOOP_IS_CALLING_OUT_TO_A_SOURCE0_PERFORM_FUNCTION__ + 28 (CFRunLoop.c:1970) 24 CoreFoundation 0x1a2d202bc __CFRunLoopDoSource0 + 176 (CFRunLoop.c:2014) 25 CoreFoundation 0x1a2d1ddc0 __CFRunLoopDoSources0 + 244 (CFRunLoop.c:2051) 26 CoreFoundation 0x1a2d1cfbc __CFRunLoopRun + 840 (CFRunLoop.c:2969) 27 CoreFoundation 0x1a2d1c830 CFRunLoopRunSpecific + 588 (CFRunLoop.c:3434) 28 GraphicsServices 0x1eecfc1c4 GSEventRunModal + 164 (GSEvent.c:2196) 29 UIKitCore 0x1a5882eb0 -[UIApplication _run] + 816 (UIApplication.m:3844) 30 UIKitCore 0x1a59315b4 UIApplicationMain + 340 (UIApplication.m:5496) 31 NoteKeys 0x10254bc10 main + 68 (AppDelegate.swift:15) 32 dyld 0x1c870aec8 start + 2724 (dyldMain.cpp:1334) Thanks very much for any help: )

Hello, I have a CarPlay Navigation app and utilize the AVSpeechSynthesizer to speak directions to a user. Everything works great on my CarPlay simulator as well as when plugged into my GMC truck. However, I found out yesterday that one of my users with a Ford truck the audio would cut in an out. After much troubleshooting, I was able to replicate this on my own truck when using Bluetooth to connect to CarPlay. My user was also utilizing Bluetooth. Has anyone else experienced this? Is there a fix to the problem? import SwiftUI import AVFoundation class TextToSpeechService: NSObject, ObservableObject, AVSpeechSynthesizerDelegate { private var speechSynthesizer = AVSpeechSynthesizer() static let shared = TextToSpeechService() override init() { super.init() speechSynthesizer.delegate = self } func configureAudioSession() { speechSynthesizer.delegate = self do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .voicePrompt, options: [.mixWithOthers, .allowBluetooth]) } catch { print("Failed to set audio session category: \(error.localizedDescription)") } } func speak(_ text: String) { Task(priority: .high) { let speechUtterance = AVSpeechUtterance(string: text) speechUtterance.voice = AVSpeechSynthesisVoice(language: AVSpeechSynthesisVoice.currentLanguageCode()) try AVAudioSession.sharedInstance().setActive(true, options: .notifyOthersOnDeactivation) speechSynthesizer.speak(speechUtterance) } } func speechSynthesizer(_ synthesizer: AVSpeechSynthesizer, didFinish utterance: AVSpeechUtterance) { Task { stopSpeech() try AVAudioSession.sharedInstance().setActive(false) } } func stopSpeech() { speechSynthesizer.stopSpeaking(at: .immediate) } }

Hello, I'm trying to determine the best/recommended AVAudioSession configuration (i.e category, mode, and options) for the following use-case. Essentially, I'd like to switch between periods of playing an audio file and then recognizing speech. The audio file is typically speech and I don't intend for playback and speech recognition to occur simultaneously. I'd like for the user to sill be able to interact with Siri and I'd like for it to work with CarPlay where navigation prompts can occur. I would assume the category to use is 'playAndRecord', but I'm not sure if it's better to just set that once for the entire lifecycle, or set to 'playback' for audio file playback and then switch to 'playAndRecord' for speech recognition . I'm also not sure on the best 'mode' and 'options' to set. Any suggestions would be appreciated. Thanks.