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

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

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

After update,WeChat voice chatting no sounds, please help

3 I am working on an application to get when input audio device is being used. Basically I want to know the application using the microphone (built-in or external) This app runs on macOS. For Mac versions starting from Sonoma I can use this code: int getAudioProcessPID(AudioObjectID process) { pid_t pid; if (@available(macOS 14.0, *)) { constexpr AudioObjectPropertyAddress prop { kAudioProcessPropertyPID, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMain }; UInt32 dataSize = sizeof(pid); OSStatus error = AudioObjectGetPropertyData(process, &prop, 0, nullptr, &dataSize, &pid); if (error != noErr) { return -1; } } else { // Pre sonoma code goes here } return pid; } which works. However, kAudioProcessPropertyPID was added in macOS SDK 14.0. Does anyone know how to achieve the same functionality on previous versions?

I'm trying to use the new Speech framework for streaming transcription on macOS 26.3, and I can reproduce a failure with SpeechAnalyzer.start(inputSequence:). What is working: SpeechAnalyzer + SpeechTranscriber offline path using start(inputAudioFile:finishAfterFile:) same Spanish WAV file transcribes successfully and returns a coherent final result What is not working: SpeechAnalyzer + SpeechTranscriber stream path using start(inputSequence:) same WAV, replayed as AnalyzerInput(buffer:bufferStartTime:) fails once replay starts with: _GenericObjCError domain=Foundation._GenericObjCError code=0 detail=nilError I also tried: DictationTranscriber instead of SpeechTranscriber no realtime pacing during replay Both still fail in stream mode with the same error. So this does not currently look like a ScreenCaptureKit issue or a Python integration issue. I reduced it to a pure Swift CLI repro. Environment: macOS 26.3 (25D122) Xcode 26.3 Swift 6.2.4 Apple Silicon Mac Has anyone here gotten SpeechAnalyzer.start(inputSequence:) working reliably on macOS 26.x? If so, I'd be interested in any workaround or any detail that differs from the obvious setup: prepareToAnalyze(in:) bestAvailableAudioFormat(...) AnalyzerInput(buffer:bufferStartTime:) replaying a known-good WAV in chunks I already filed Feedback Assistant: FB22149971

I have both apple devices, AirPods Pro 3 is up to date and Ultra 3 is on watch os 26.1 latest public beta. Each morning when I would go on my mindfulness app and start a meditation or listen to Apple Music on my watch and AirPods Pro 3, it will play for a few seconds then disconnects. My bluetooth settings on my watch says my AirPods is connected to my watch. I also have removed the tick about connecting automatically to iPhone on the AirPods setting in my iPhone. To fix this I invariably turn off my Apple Watch Ultra 3 and turn it on again. Then the connection becomes stable. I am not sure why I have to do this each morning. It is frustrating. I am not sure why this fix does not last long? Is there something wrong with my AirPods? Has anyone encountered this before?

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!

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 all, I've developed an audio DSP application in C++ using AudioToolbox and CoreAudio on MacOS 14.4.1 with Xcode 15. I use an AudioQueue for input and another for output. This works great. I'm now adding real-time audio analysis eg spectral analysis. I want this to run independently of my audio processing so it can not interfere with audio playback. Taps on AudioQueues seem to be a good way of doing this... Since the analytics won't modify the audio data, I am using a Siphon Tap by setting the AudioQueueProcessingTapFlags to kAudioQueueProcessingTap_PreEffects | kAudioQueueProcessingTap_Siphon; This works fine on my output queue. However, on my input queue the Tap callback is called once and then a EXC_BAD_ACCESS occurs - screen shot below. NB: I believe that a callback should only call AudioQueueProcessingTapGetSourceAudio when not using a Siphon, so I don't call it. Relevant code: AudioQueueProcessingTapCallback tap_callback) { // Makes an audio tap for a queue void * tap_data_ptr = NULL; AudioQueueProcessingTapFlags tap_flags = kAudioQueueProcessingTap_PostEffects | kAudioQueueProcessingTap_Siphon; uint32_t max_frames = 0; AudioStreamBasicDescription asbd; AudioQueueProcessingTapRef tap_ref; OSStatus status = AudioQueueProcessingTapNew(queue_ref, tap_callback, tap_data_ptr, tap_flags, &max_frames, &asbd, &tap_ref); if (status != noErr) printf("Error while making Tap\n"); else printf("Successfully made tap\n"); } void tapper(void * tap_data, AudioQueueProcessingTapRef tap_ref, uint32_t number_of_frames_in, AudioTimeStamp * ts_ptr, AudioQueueProcessingTapFlags * tap_flags_ptr, uint32_t * number_of_frames_out_ptr, AudioBufferList * buf_list) { // Callback function for audio queue tap printf("Tap callback"); }``` Image of exception stack provided by Xcode: ![]("https://developer.apple.com/forums/content/attachment/27479e8d-a118-459b-aa2d-7e30528910e3" "title=Screenshot 2025-06-14 at 1.29.14 PM.png;width=932;height=562") What have I missed? Appreciate any help you learned folks may be able to provide. Best, Geoff.

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

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

I've got a problem with my app where I'm testing it on my own phone. I'm using audio kit to generate tones as part of the app. Everything seems to work fine. Sounds start, Stop, etc. They play when the app is closed and when the phone is locked, so background is working. However, I'm seeing an issue where, even when STOP is pressed and the application exited, if I get a notification such as a text message, the base tone for the app starts to play. If I then open the app, check the Start/Stop button - it says start so that. hasnt' been activated. If I click Start, then a 2nd tone starts. This one stops with the Stop button. However the original tone that was set off by an incoming message carries on playing. Until I go to the Open Apps View on the phone and slide the application upwards. For the life of me, I can't figure out whats happening here.

Using an iPhone Pro 12 running iOS 26.0.1, with AirPods Pro 3. Camera app does capture video with what seems to be "Studio Quality Recording". Am trying to replicate that SQR with my own Camera like app, and while I can pull audio in from the APP3 mic, and my video capture app is recording a 48,000Hz high-bitrate video, the audio still sounds non-SQR. I'm seeing bluetoothA2DP , bluetoothLE , bluetoothHFP as portType, and not sure if SQR depends on one of those? Is there sample code demonstrating a SQR capture? Nevermind video and camera, just audio even? Also, I don't understand what SQR is doing between the APP3 and the iPhone. What codec is that? What bitrate is that? If I capture video using Capture and inspect the audio stream I see mono 74.14 kbit/s MPEG-4 AAC, 48000 Hz. But I assume that's been recompressed and not really giving me any insight into the APP3 H2 transmission?

I have an iPadOS M-processor application with two different running configurations. In config1, the shared AVAudioSession is configured for .videoChat mode using the built-in microphone. The input/output nodes of the AVAudioEngine are configured with voice processing enabled. The built-in mic is formatted for 1 channel at 48KHz. In config2, the shared AVAudioSession is configured for .measurement mode using an external USB microphone. The input/output nodes of the AVAudioEngine are configured with voice processing disabled. The external mic is formatted for 2 channels at 44.1KHz I've written a configuration manager designed to safely switch between these two configurations. It works by stopping AVAudioEngine and detaching all but the input and output nodes, updating the shared audio session for the desired mic and sample-rates, and setting the appropriate state for voice processing to either true or false as required by the configuration. Finally the new audio graph is constructed by attaching appropriate nodes, connecting them, and re-starting AVAudioEngine I'm experiencing what I believe is a race-condition between switching voice processing on or off and then trying to re-build and start the new audio graph. Even though notifications, which are dumped to the console indicate that my requested input and sample-rate settings are in place, I crash when trying to start the audio engine because the sample-rate is wrong. Investigating further it looks like the switch from remote I/O to voice-processing I/O or vice-versa has not yet actually completed. I introduced a 100ms second delay and that seems to help but is obviously not a reliable way to build software that must work consistently. How can I make sure that what are apparently asynchronous configuration changes to the shared audio session and the input/output nodes have completed before I go on? I tried using route change notifications from the shared AVAudioSession but these lie. They say my preferred mic input and sample-rate setting is in place but when I dump the AVAudioEngine graph to the debugger console, I still see the wrong sample rate assigned to the input/output nodes. Also these are the wrong AU nodes. That is, VPIO is still in place when RIO should be, or vice-versa. How can I make the switch reliable without arbitrary time delays? Is my configuration manager approach appropriate (question for Apple engineers)?

In Instruments, I'm seeing "Zero Time Stamp" events in the "Audio Server" lane. What does that mean?

I have a simple AVAudioEngine graph as follows: AVAudioPlayerNode -> AVAudioUnitEQ -> AVAudioUnitTimePitch -> AVAudioUnitReverb -> Main mixer node of AVAudioEngine. I noticed that whenever I have AVAudioUnitTimePitch or AVAudioUnitVarispeed in the graph, I noticed a very distinct crackling/popping sound in my Airpods Pro 2 when starting up the engine and playing the AVAudioPlayerNode and unable to find the reason why this is happening. When I remove the node, the crackling completely goes away. How do I fix this problem since i need the user to be able to control the pitch and rate of the audio during playback. import AVKit @Observable @MainActor class AudioEngineManager { nonisolated private let engine = AVAudioEngine() private let playerNode = AVAudioPlayerNode() private let reverb = AVAudioUnitReverb() private let pitch = AVAudioUnitTimePitch() private let eq = AVAudioUnitEQ(numberOfBands: 10) private var audioFile: AVAudioFile? private var fadePlayPauseTask: Task<Void, Error>? private var playPauseCurrentFadeTime: Double = 0 init() { setupAudioEngine() } private func setupAudioEngine() { guard let url = Bundle.main.url(forResource: "Song name goes here", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: url) } catch { print("Failed to load audio file: \(error)") return } reverb.loadFactoryPreset(.mediumHall) reverb.wetDryMix = 50 pitch.pitch = 0 // Increase pitch by 500 cents (5 semitones) engine.attach(playerNode) engine.attach(pitch) engine.attach(reverb) engine.attach(eq) // Connect: player -> pitch -> reverb -> output engine.connect(playerNode, to: eq, format: audioFile?.processingFormat) engine.connect(eq, to: pitch, format: audioFile?.processingFormat) engine.connect(pitch, to: reverb, format: audioFile?.processingFormat) engine.connect(reverb, to: engine.mainMixerNode, format: audioFile?.processingFormat) } func prepare() { guard let audioFile else { return } playerNode.scheduleFile(audioFile, at: nil) } func play() { DispatchQueue.global().async { [weak self] in guard let self else { return } engine.prepare() try? engine.start() DispatchQueue.main.async { [weak self] in guard let self else { return } playerNode.play() fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: true) // Ramp up volume until 1 is reached if volume >= 1 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 1 } } } } func pause() { fadePlayPauseTask?.cancel() playPauseCurrentFadeTime = 0 fadePlayPauseTask = Task { [weak self] in guard let self else { return } while true { let volume = updateVolume(for: playPauseCurrentFadeTime / 0.1, rising: false) // Ramp down volume until 0 is reached if volume <= 0 { break } engine.mainMixerNode.outputVolume = volume try await Task.sleep(for: .milliseconds(10)) playPauseCurrentFadeTime += 0.01 } engine.mainMixerNode.outputVolume = 0 playerNode.pause() // Shut down engine once ramp down completes DispatchQueue.global().async { [weak self] in guard let self else { return } engine.pause() } } } private func updateVolume(for x: Double, rising: Bool) -> Float { if rising { // Fade in return Float(pow(x, 2) * (3.0 - 2.0 * (x))) } else { // Fade out return Float(1 - (pow(x, 2) * (3.0 - 2.0 * (x)))) } } func setPitch(_ value: Float) { pitch.pitch = value } func setReverbMix(_ value: Float) { reverb.wetDryMix = value } } struct ContentView: View { @State private var audioManager = AudioEngineManager() @State private var pitch: Float = 0 @State private var reverb: Float = 0 var body: some View { VStack(spacing: 20) { Text("🎵 Audio Player with Reverb & Pitch") .font(.title2) HStack { Button("Prepare") { audioManager.prepare() } Button("Play") { audioManager.play() } .padding() .background(Color.green) .foregroundColor(.white) .cornerRadius(10) Button("Pause") { audioManager.pause() } .padding() .background(Color.red) .foregroundColor(.white) .cornerRadius(10) } VStack { Text("Pitch: \(Int(pitch)) cents") Slider(value: $pitch, in: -2400...2400, step: 100) { _ in audioManager.setPitch(pitch) } } VStack { Text("Reverb Mix: \(Int(reverb))%") Slider(value: $reverb, in: 0...100, step: 1) { _ in audioManager.setReverbMix(reverb) } } } .padding() } }

My app uses the AVFoundation to pronounce some words. Running the app from Xcode, either to a simulator or device, I frequently get this crash at start-up: AXSpeech (13): EXC_BAD_ACCESS (code=EXC_I386_GPFLT). It seems to occur randomly, maybe 20%-30% of the time I launch the app. When it does not crash, using audio works as expected. When launched from the device, it never crashes (so far, at least). Here's the code that outputs speech: Declared at the top level of the View struct: @State var synth = AVSpeechSynthesizer() In the View, as part of a Button's action closure: let utterance = AVSpeechUtterance(string: answer) utterance.voice = AVSpeechSynthesisVoice(language: "en_US") synth.speak(utterance) Any idea on how to stop this? It's annoying having to launch the app multiple times to test on a simulator or device.

Hi, I try to record audio on the iPhone with the AVAudioRecorder and Xcode 26.0.1. Maybe the problem is that I can not record audio with the simulator. But there's a menu for audio. In the plist I added 'Privacy - Microphone Usage Description' and I ask for permission before recording. if await AVAudioApplication.requestRecordPermission() { print("permission granted") recordPermission = true } else { print("permission denied") } Permission is granted. let settings: [String : Any] = [ AVFormatIDKey: kAudioFormatMPEG4AAC, AVSampleRateKey: 12000, AVNumberOfChannelsKey: 1, AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue ] recorder = try AVAudioRecorder(url: filename, settings: settings) let prepared = recorder.prepareToRecord() print("prepared started: \(prepared)") let started = recorder.record() print("recording started: \(started)") started is always false and I tried many settings. Error messages AddInstanceForFactory: No factory registered for id <CFUUID 0x600000211480> F8BB1C28-BAE8-11D6-9C31-00039315CD46 AudioConverter.cpp:1052 Failed to create a new in process converter -> from 0 ch, 12000 Hz, .... (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame to 1 ch, 12000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame, with status -50 AudioQueueObject.cpp:1892 BuildConverter: AudioConverterNew returned -50 from: 0 ch, 12000 Hz, .... (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame to: 1 ch, 12000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame prepared started: true AudioQueueObject.cpp:7581 ConvertInput: aq@0x10381be00: AudioConverterFillComplexBuffer returned -50, packetCount 5 recording started: false All examples I find are the same, but apparently there must be something different.

{ "aps": { "content-available": 1 }, "audio_file_name": "ding.caf", "audio_url": "https://example.com/audio.mp3" } When the app is in the background or killed, it receives a remote APNs push. The data format is roughly as shown above. How can I play the MP3 audio file at the specified "audio_url"? The user does not need to interact with the device when receiving the APNs. How can I play the audio file immediately after receiving it?