Dive into the technical aspects of audio on your device, including codecs, format support, and customization options.

Audio Documentation

Posts under Audio subtopic

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Is Call Translation API available for VOIP?
I might have misunderstood the docs, but is Call Translation going to be available for VOIP applications? Eg in an already connected VOIP call, would it be possible for Call Translations to be enabled on an iOS 26 and Apple Intelligence supported device? I have personally tried it and it doesn’t look like it supported VOIP but would love to confirm this. reference: https://aninterestingwebsite.com/documentation/callkit/cxsettranslatingcallaction/
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79
Jun ’25
Audio driver based on AudioDriverKit sometimes hangs after sleep
Dear Sirs, I’ve written a virtual audio driver based on AudioDriverKit and running as dext in my MacOS app. Sometimes when waking up from a sleep state the recording side of my driver extension seems to hang and I don’t see any calls to my io_operation callback. Then the recording app like a DAW seems to hang when trying to start a recording. This doesn’t happen after short sleep states or after a complete new start of my MacBook. I already opened a case in Feedback-Assistant on 5th of May (FB17503622) which also includes a sysdiagnose and a ktrace but I didn't get any feedback so far. Meanwhile some of our customers are getting angry and I'd like to know if there's anything I could do to fix this problem on my side. We’re not sure whether this worked in previous MacOS versions, we think we didn’t observe this before 15.3.1 but at least since 15.3.1. we’ve seen this problem. Best regards, Johannes
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198
Aug ’25
Why does AVAudioRecorder show 8 kHz when iPhone hardware is 48 kHz?
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!
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274
Sep ’25
Application tones start when I get incoming call or message
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.
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116
May ’25
iOS 26.0 (23A5276f) - Bluetooth Call Audio Broken (AirPods + Car)
iOS 26.0 (23A5276f) – Bluetooth Call Audio Issue I’m experiencing a Bluetooth audio issue on iOS 26.0 (build 23A5276f). I cannot make or receive phone calls properly using Bluetooth devices — this affects both my car’s Bluetooth system and my AirPods Pro (2nd generation). Notably: Regular phone calls have no audio (either I can’t hear the other person, or they can’t hear me). WhatsApp and other VoIP apps work fine with the same Bluetooth devices. Media playback (music, video, etc.) works without issues over Bluetooth. It seems this bug is limited to the native Phone app or the system audio routing for regular cellular calls. Please advise if this is a known issue or if a fix is expected in upcoming beta releases.
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338
Jun ’25
How to get PID from AudioObjectID on macOS pre Sonoma
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?
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Sep ’25
Random EXC_BAD_ACCESS using AVFoundation
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.
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SpeechAnalyzer.start(inputSequence:) fails with _GenericObjCError nilError, while the same WAV succeeds with start(inputAudioFile:)
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
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2w
Incoming calls thrue Jisti Meet and locked screen
Problem: When the screen is locked, an incoming call does not initiate the launch of the Flutter application required for audio and video communication through Jitsi Meet. In the unlocked state, the application functions correctly. The current implementation does not have a mechanism for activating the Flutter engine when receiving a call via CallKit while the screen is locked. Although CallKit UI displays the call acceptance interface and the audio session is configured, the Flutter application remains in a suspended state, making it impossible to connect to the media server. Audio session activated using didActivateAudioSession method.
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3w
How to capture audio from the stream that's playing on the speakers?
Good day, ladies and gents. I have an application that reads audio from the microphone. I'd like it to also be able to read from the Mac's audio output stream. (A bonus would be if it could detect when the Mac is playing music.) I'd eventually be able to figure it out reading docs, but if someone can give a hint, I'd be very grateful, and would owe you the libation of your choice. Here's the code used to set up the AudioUnit: -(NSString*) configureAU { AudioComponent component = NULL; AudioComponentDescription description; OSStatus err = noErr; UInt32 param; AURenderCallbackStruct callback; if( audioUnit ) { AudioComponentInstanceDispose( audioUnit ); audioUnit = NULL; } // was CloseComponent // Open the AudioOutputUnit description.componentType = kAudioUnitType_Output; description.componentSubType = kAudioUnitSubType_HALOutput; description.componentManufacturer = kAudioUnitManufacturer_Apple; description.componentFlags = 0; description.componentFlagsMask = 0; if( component = AudioComponentFindNext( NULL, &description ) ) { err = AudioComponentInstanceNew( component, &audioUnit ); if( err != noErr ) { audioUnit = NULL; return [ NSString stringWithFormat: @"Couldn't open AudioUnit component (ID=%d)", err] ; } } // Configure the AudioOutputUnit: // You must enable the Audio Unit (AUHAL) for input and output for the same device. // When using AudioUnitSetProperty the 4th parameter in the method refers to an AudioUnitElement. // When using an AudioOutputUnit for input the element will be '1' and the output element will be '0'. param = 1; // Enable input on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &param, sizeof(UInt32) ); chkerr("Couldn't set first EnableIO prop (enable inpjt) (ID=%d)"); param = 0; // Disable output on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(UInt32) ); chkerr("Couldn't set second EnableIO property on the audio unit (disable ootpjt) (ID=%d)"); param = sizeof(AudioDeviceID); // Select the default input device AudioObjectPropertyAddress OutputAddr = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; err = AudioObjectGetPropertyData( kAudioObjectSystemObject, &OutputAddr, 0, NULL, &param, &inputDeviceID ); chkerr("Couldn't get default input device (ID=%d)"); // Set the current device to the default input unit err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDeviceID, sizeof(AudioDeviceID) ); chkerr("Failed to hook up input device to our AudioUnit (ID=%d)"); callback.inputProc = AudioInputProc; // Setup render callback, to be called when the AUHAL has input data callback.inputProcRefCon = self; err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(AURenderCallbackStruct) ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); param = sizeof(AudioStreamBasicDescription); // get hardware device format err = AudioUnitGetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &deviceFormat, &param ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); audioChannels = MAX( deviceFormat.mChannelsPerFrame, 2 ); // Twiddle the format to our liking actualOutputFormat.mChannelsPerFrame = audioChannels; actualOutputFormat.mSampleRate = deviceFormat.mSampleRate; actualOutputFormat.mFormatID = kAudioFormatLinearPCM; actualOutputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved; if( actualOutputFormat.mFormatID == kAudioFormatLinearPCM && audioChannels == 1 ) actualOutputFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved; #if __BIG_ENDIAN__ actualOutputFormat.mFormatFlags |= kAudioFormatFlagIsBigEndian; #endif actualOutputFormat.mBitsPerChannel = sizeof(Float32) * 8; actualOutputFormat.mBytesPerFrame = actualOutputFormat.mBitsPerChannel / 8; actualOutputFormat.mFramesPerPacket = 1; actualOutputFormat.mBytesPerPacket = actualOutputFormat.mBytesPerFrame; // Set the AudioOutputUnit output data format err = AudioUnitSetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &actualOutputFormat, sizeof(AudioStreamBasicDescription)); chkerr("Could not change the stream format of the output device (ID=%d)"); param = sizeof(UInt32); // Get the number of frames in the IO buffer(s) err = AudioUnitGetProperty( audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &audioSamples, &param ); chkerr("Could not determine audio sample size (ID=%d)"); err = AudioUnitInitialize( audioUnit ); // Initialize the AU chkerr("Could not initialize the AudioUnit (ID=%d)"); // Allocate our audio buffers audioBuffer = [self allocateAudioBufferListWithNumChannels: actualOutputFormat.mChannelsPerFrame size: audioSamples * actualOutputFormat.mBytesPerFrame]; if( audioBuffer == NULL ) { [ self cleanUp ]; return [NSString stringWithFormat: @"Could not allocate buffers for recording (ID=%d)", err]; } return nil; } (...again, it would be nice to know if audio output is active and thereby choose the clean output stream over the noisy mic, but that would be a different chunk of code, and my main question may just be a quick edit to this chunk.) Thanks for your attention! ==Dave [p.s. if i get more than one useful answer, can i "Accept" more than one, to spread the credit around?] {pps: of course, the code lines up prettier in a monospaced font!}
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Jun ’25
Improving Speech Analyzer Transcription for technical terms
I am developing an app with transcription and I am exploring ways to improve the transcription from the SpeechAnalyzer/Transcriber for technical terms. SFSpeech... recognition had the capability of being augmented by contextualStrings. Does something similar exist for SpeechAnalyzer/Transcriber? If so please point me towards the documentation and any sample code that may exist for this. If there are other options, please let me know.
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Sep ’25
coreaudio-api mailing list search broken
Hello, The search functionality of the coreaudio-api mailing list archive has been broken for a very long time. Several of the lower-level audio APIs have only been discussed on this mailing list, making it critical for those of us maintaining old audio code. Steps to reproduce: Open https://lists.apple.com/archives/list/coreaudio-api@lists.apple.com/ in your web browser. Enter a search term in the "Search this list" field in the top-right corner of the page. The search will eventually time out with "502 Bad Gateway" Can somebody please forward this information to the current maintainer? I've tried to contact developer support but they weren't sure what to do. Thanks!
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Feb ’26
AirPods with H2 and studio-quality recording - how to replicate Camera video capture
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?
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Oct ’25
Mixing ScreenCaptureKit audio with microphone audio
Hi, I'm new to AVAudioEngine(and macOS programming in general). I'm trying to mix microphone audio with ScreenCaptureKit audio using AVAudioEngine without playing it back. I've created a AVAudioPlayerNode and scheduling buffers in my SCStream handler: playerNode.scheduleBuffer(samples) and have connected the playerNode to the mainMixerNode. audioEngine.connect(audioEngine.inputNode, to: audioEngine.mainMixerNode, format: micFormat) audioEngine.connect(playerNode, to: audioEngine.mainMixerNode, format: format) The problem is that mainMixerNode plays the audio to the speaker creating a feedback loop. How can I prevent the mixer output from being played back. Also: Is this the best way of mixing microphone input with some other input? I ran into AVAudioEngine's manual rendering mode, which seems like the way to go for mixing audio without playing it back. However, I couldn't figure out how to connect microphone input to the AVAudioEngine in manual rendering mode?
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MusicKit + AirPlay
Hello, I'm working on a MusicKit based SwiftUI app. I've integrated AirPlay using the AVRoutePickerView like so: struct UIKitAirPlayPickerView: UIViewRepresentable { func makeUIView(context: Context) -> AVRoutePickerView { let routePickerView = AVRoutePickerView() routePickerView.prioritizesVideoDevices = false return routePickerView } func updateUIView(_ uiView: AVRoutePickerView, context: Context) {} } The AirPlay menu appears as expected, and selecting an AirPlay device functions as expected. I'm currently sending audio from my app to a HomePod. However, the state of the AVRoutePickerView does not reflect the playback state. There is no cover art and it says "Not Playing". When my device is locked, my lock screen shows the album art, metadata and AirPlay routing as expected. My app uses the ApplicationMusicPlayer however I encounter the same behavior using the SystemMusicPlayer. Any guidance on how to troubleshoot this? Is there any other way to integrate the system AirPlay picker into my app, or is this my only option? Thank you for reading.
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Feb ’26
When to set AVAudioSession's preferredInput?
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.
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883
Oct ’25
Is AVAudioPCMFormatFloat32 required for playing a buffer with AVAudioEngine / AVAudioPlayerNode
I have a PCM audio buffer (AVAudioPCMFormatInt16). When I try to play it using AVPlayerNode / AVAudioEngine an exception is thrown: "[[busArray objectAtIndexedSubscript:(NSUInteger)element] setFormat:format error:&nsErr]: returned false, error Error Domain=NSOSStatusErrorDomain Code=-10868 (related thread https://forums.aninterestingwebsite.com/forums/thread/700497?answerId=780530022#780530022) If I convert the buffer to AVAudioPCMFormatFloat32 playback works. My questions are: Does AVAudioEngine / AVPlayerNode require AVAudioPCMBuffer to be in the Float32 format? Is there a way I can configure it to accept another format instead for my application? If 1 is YES is this documented anywhere? If 1 is YES is this required format subject to change at any point? Thanks! I was looking to watch the "AVAudioEngine in Practice" session video from WWDC 2014 but I can't find it anywhere (https://forums.aninterestingwebsite.com/forums/thread/747008).
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Oct ’25
Is Call Translation API available for VOIP?
I might have misunderstood the docs, but is Call Translation going to be available for VOIP applications? Eg in an already connected VOIP call, would it be possible for Call Translations to be enabled on an iOS 26 and Apple Intelligence supported device? I have personally tried it and it doesn’t look like it supported VOIP but would love to confirm this. reference: https://aninterestingwebsite.com/documentation/callkit/cxsettranslatingcallaction/
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1
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79
Activity
Jun ’25
Audio driver based on AudioDriverKit sometimes hangs after sleep
Dear Sirs, I’ve written a virtual audio driver based on AudioDriverKit and running as dext in my MacOS app. Sometimes when waking up from a sleep state the recording side of my driver extension seems to hang and I don’t see any calls to my io_operation callback. Then the recording app like a DAW seems to hang when trying to start a recording. This doesn’t happen after short sleep states or after a complete new start of my MacBook. I already opened a case in Feedback-Assistant on 5th of May (FB17503622) which also includes a sysdiagnose and a ktrace but I didn't get any feedback so far. Meanwhile some of our customers are getting angry and I'd like to know if there's anything I could do to fix this problem on my side. We’re not sure whether this worked in previous MacOS versions, we think we didn’t observe this before 15.3.1 but at least since 15.3.1. we’ve seen this problem. Best regards, Johannes
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1
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198
Activity
Aug ’25
Should AVAudioFormat be Sendable?
AVAudioFormat has no Swift concurrency annotations but the documentation states "Instances of this class are immutable." This made me always assume it was safe to pass AVAudioFormat instances around. Is this the case? If so can it be marked as Sendable? Am I missing something?
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1
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1
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712
Activity
Aug ’25
Why does AVAudioRecorder show 8 kHz when iPhone hardware is 48 kHz?
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!
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1
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274
Activity
Sep ’25
Application tones start when I get incoming call or message
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.
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1
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116
Activity
May ’25
iOS 26.0 (23A5276f) - Bluetooth Call Audio Broken (AirPods + Car)
iOS 26.0 (23A5276f) – Bluetooth Call Audio Issue I’m experiencing a Bluetooth audio issue on iOS 26.0 (build 23A5276f). I cannot make or receive phone calls properly using Bluetooth devices — this affects both my car’s Bluetooth system and my AirPods Pro (2nd generation). Notably: Regular phone calls have no audio (either I can’t hear the other person, or they can’t hear me). WhatsApp and other VoIP apps work fine with the same Bluetooth devices. Media playback (music, video, etc.) works without issues over Bluetooth. It seems this bug is limited to the native Phone app or the system audio routing for regular cellular calls. Please advise if this is a known issue or if a fix is expected in upcoming beta releases.
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1
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1
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338
Activity
Jun ’25
How to get PID from AudioObjectID on macOS pre Sonoma
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?
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377
Activity
Sep ’25
Random EXC_BAD_ACCESS using AVFoundation
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.
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550
Activity
3w
SpeechAnalyzer.start(inputSequence:) fails with _GenericObjCError nilError, while the same WAV succeeds with start(inputAudioFile:)
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
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1
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358
Activity
2w
Incoming calls thrue Jisti Meet and locked screen
Problem: When the screen is locked, an incoming call does not initiate the launch of the Flutter application required for audio and video communication through Jitsi Meet. In the unlocked state, the application functions correctly. The current implementation does not have a mechanism for activating the Flutter engine when receiving a call via CallKit while the screen is locked. Although CallKit UI displays the call acceptance interface and the audio session is configured, the Flutter application remains in a suspended state, making it impossible to connect to the media server. Audio session activated using didActivateAudioSession method.
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1
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104
Activity
3w
How to capture audio from the stream that's playing on the speakers?
Good day, ladies and gents. I have an application that reads audio from the microphone. I'd like it to also be able to read from the Mac's audio output stream. (A bonus would be if it could detect when the Mac is playing music.) I'd eventually be able to figure it out reading docs, but if someone can give a hint, I'd be very grateful, and would owe you the libation of your choice. Here's the code used to set up the AudioUnit: -(NSString*) configureAU { AudioComponent component = NULL; AudioComponentDescription description; OSStatus err = noErr; UInt32 param; AURenderCallbackStruct callback; if( audioUnit ) { AudioComponentInstanceDispose( audioUnit ); audioUnit = NULL; } // was CloseComponent // Open the AudioOutputUnit description.componentType = kAudioUnitType_Output; description.componentSubType = kAudioUnitSubType_HALOutput; description.componentManufacturer = kAudioUnitManufacturer_Apple; description.componentFlags = 0; description.componentFlagsMask = 0; if( component = AudioComponentFindNext( NULL, &description ) ) { err = AudioComponentInstanceNew( component, &audioUnit ); if( err != noErr ) { audioUnit = NULL; return [ NSString stringWithFormat: @"Couldn't open AudioUnit component (ID=%d)", err] ; } } // Configure the AudioOutputUnit: // You must enable the Audio Unit (AUHAL) for input and output for the same device. // When using AudioUnitSetProperty the 4th parameter in the method refers to an AudioUnitElement. // When using an AudioOutputUnit for input the element will be '1' and the output element will be '0'. param = 1; // Enable input on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &param, sizeof(UInt32) ); chkerr("Couldn't set first EnableIO prop (enable inpjt) (ID=%d)"); param = 0; // Disable output on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(UInt32) ); chkerr("Couldn't set second EnableIO property on the audio unit (disable ootpjt) (ID=%d)"); param = sizeof(AudioDeviceID); // Select the default input device AudioObjectPropertyAddress OutputAddr = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; err = AudioObjectGetPropertyData( kAudioObjectSystemObject, &OutputAddr, 0, NULL, &param, &inputDeviceID ); chkerr("Couldn't get default input device (ID=%d)"); // Set the current device to the default input unit err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDeviceID, sizeof(AudioDeviceID) ); chkerr("Failed to hook up input device to our AudioUnit (ID=%d)"); callback.inputProc = AudioInputProc; // Setup render callback, to be called when the AUHAL has input data callback.inputProcRefCon = self; err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(AURenderCallbackStruct) ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); param = sizeof(AudioStreamBasicDescription); // get hardware device format err = AudioUnitGetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &deviceFormat, &param ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); audioChannels = MAX( deviceFormat.mChannelsPerFrame, 2 ); // Twiddle the format to our liking actualOutputFormat.mChannelsPerFrame = audioChannels; actualOutputFormat.mSampleRate = deviceFormat.mSampleRate; actualOutputFormat.mFormatID = kAudioFormatLinearPCM; actualOutputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved; if( actualOutputFormat.mFormatID == kAudioFormatLinearPCM && audioChannels == 1 ) actualOutputFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved; #if __BIG_ENDIAN__ actualOutputFormat.mFormatFlags |= kAudioFormatFlagIsBigEndian; #endif actualOutputFormat.mBitsPerChannel = sizeof(Float32) * 8; actualOutputFormat.mBytesPerFrame = actualOutputFormat.mBitsPerChannel / 8; actualOutputFormat.mFramesPerPacket = 1; actualOutputFormat.mBytesPerPacket = actualOutputFormat.mBytesPerFrame; // Set the AudioOutputUnit output data format err = AudioUnitSetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &actualOutputFormat, sizeof(AudioStreamBasicDescription)); chkerr("Could not change the stream format of the output device (ID=%d)"); param = sizeof(UInt32); // Get the number of frames in the IO buffer(s) err = AudioUnitGetProperty( audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &audioSamples, &param ); chkerr("Could not determine audio sample size (ID=%d)"); err = AudioUnitInitialize( audioUnit ); // Initialize the AU chkerr("Could not initialize the AudioUnit (ID=%d)"); // Allocate our audio buffers audioBuffer = [self allocateAudioBufferListWithNumChannels: actualOutputFormat.mChannelsPerFrame size: audioSamples * actualOutputFormat.mBytesPerFrame]; if( audioBuffer == NULL ) { [ self cleanUp ]; return [NSString stringWithFormat: @"Could not allocate buffers for recording (ID=%d)", err]; } return nil; } (...again, it would be nice to know if audio output is active and thereby choose the clean output stream over the noisy mic, but that would be a different chunk of code, and my main question may just be a quick edit to this chunk.) Thanks for your attention! ==Dave [p.s. if i get more than one useful answer, can i "Accept" more than one, to spread the credit around?] {pps: of course, the code lines up prettier in a monospaced font!}
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