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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How to inform Logic Pro that AU view does not have a fixed aspect ratio?
I have an AUv3 that passes all validation and can be loaded into Logic Pro without issue. The UI for the plug in can be any aspect ratio but Logic insists on presenting it in a view with a fixed aspect ratio. That is when resizing, both the height and width are resized. I have never managed to work out what it is I need to do specify to Logic to allow the user to resize width or height independently of each other. Can anyone tell me what I need to specify in the AU code that will inform Logic that the view can be resized from any side of the window/panel?
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210
Apr ’25
CoreMIDI driver - flow control
Hi, when a CoreMIDI driver controls physical HW it is probably quite commune to have to control the amount of MIDI data received from the system. What comes to mind is to just delay returning control of the MIDIDriverInterface::Send() callback to the calling process. While the application trying to send MIDI really stalls until the callback returns it seems only to be a side effect of a generally stalled CoreMIDI server. Between the callbacks the application can send as much MIDI data as it wants to CoreMIDI, it's buffering seems to be endless... However the HW might not be able to play out all the data. It seems there is no way to indicate an overflow/full buffer situation back the application/CoreMIDI. How is this supposed to work? Thanks, any hints or pointers are highly appreciated! Hagen.
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248
Oct ’25
iOS 26 Beta Personal Voice bug affecting AVSpeechSynthesizer
I have sent in a feedback report (FB18222398) but I have no idea if anyone has looked at it. I know from past experiences that Apple devs do look at these forums. This applies to each of the betas, 1, 2 and 3. I have created a new Personal Voice with each beta. I create a personal voice in English. When it's done processing, I tap Preview and it says in English what is expected. But after some time, an hour or a day, the language of the voice file changes languages and no longer works properly. If I press Preview it is no longer intelligible. I have a text to speech app and initially the created voice works but then when the language of the file changes, it no longer works. I have run an app on my iphone through Xcode that prints to the console the voices installed on the device with the language. Currently this is the voice file: Voice Identifier: com.apple.speech.personalvoice.AAA9C6F2-9125-475F-BA2F-22C63274991D Language: es-MX and on a second device the same personal voice is in a different language: Voice Identifier: com.apple.speech.personalvoice.AAA9C6F2-9125-475F-BA2F-22C63274991D Language: zh-CN Although, a previous personal voice file that listed as Spanish-Mexican played in English with a Spanish accent or when playing Spanish text, it sounded almost perfect. This current personal voice doesn't do that, and is unintelligible. Previous attempts have converted to Chinese. I hope someone can look into this.
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545
Dec ’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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3w
How can I find the user's "Favorite Songs" playlist?
It sounds simple but searching for the name "Favorite Songs" is a non-starter because it's called different names in different countries, even if I specify "&l=en_us" on the query. So is there another property, relationship or combination thereof which I can use to tell me when I've found the right playlist? Properties I've looked at so far: canEdit: will always be false so narrows things down a little inFavorites: not helpful as it depends on whether the user has favourite the favourites playlist, so not relevant hasCatalog: seems always true so again may narrow things down a bit isPublic: doesn't help Adding the catalog relationship doesn't seem to show anything immediately useful either. Can anyone help? Ideally I'd like to see this as a "kind" or "type" as it has different properties to other playlists, but frankly I'll take anything at this point.
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289
Jul ’25
Error -50 writing to AVAudioFile
I'm trying to write 16-bit interleaved 2-channel data captured from a LiveSwitch audio source to a AVAudioFile. The buffer and file formats match but I get a bad parameter error from the API. Does this API not support the specified format or is there some other issue? Here is the debugger output. (lldb) po audioFile.url ▿ file:///private/var/mobile/Containers/Data/Application/1EB14379-0CF2-41B6-B742-4C9A80728DB3/tmp/Heart%20Sounds%201 - _url : file:///private/var/mobile/Containers/Data/Application/1EB14379-0CF2-41B6-B742-4C9A80728DB3/tmp/Heart%20Sounds%201 - _parseInfo : nil - _baseParseInfo : nil (lldb) po error Error Domain=com.apple.coreaudio.avfaudio Code=-50 "(null)" UserInfo={failed call=ExtAudioFileWrite(_impl->_extAudioFile, buffer.frameLength, buffer.audioBufferList)} (lldb) po buffer.format <AVAudioFormat 0x302a12b20: 2 ch, 44100 Hz, Int16, interleaved> (lldb) po audioFile.fileFormat <AVAudioFormat 0x302a515e0: 2 ch, 44100 Hz, Int16, interleaved> (lldb) po buffer.frameLength 882 (lldb) po buffer.audioBufferList ▿ 0x0000000300941e60 - pointerValue : 12894608992 This code handles the details of converting the Live Switch frame into an AVAudioPCMBuffer. extension FMLiveSwitchAudioFrame { func convertedToPCMBuffer() -> AVAudioPCMBuffer { Self.convertToAVAudioPCMBuffer(from: self)! } static func convertToAVAudioPCMBuffer(from frame: FMLiveSwitchAudioFrame) -> AVAudioPCMBuffer? { // Retrieve the audio buffer and format details from the FMLiveSwitchAudioFrame guard let buffer = frame.buffer(), let format = buffer.format() as? FMLiveSwitchAudioFormat else { return nil } // Extract PCM format details from FMLiveSwitchAudioFormat let sampleRate = Double(format.clockRate()) let channelCount = AVAudioChannelCount(format.channelCount()) // Determine bytes per sample based on bit depth let bitsPerSample = 16 let bytesPerSample = bitsPerSample / 8 let bytesPerFrame = bytesPerSample * Int(channelCount) let frameLength = AVAudioFrameCount(Int(buffer.dataBuffer().length()) / bytesPerFrame) // Create an AVAudioFormat from the FMLiveSwitchAudioFormat guard let avAudioFormat = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: sampleRate, channels: channelCount, interleaved: true) else { return nil } // Create an AudioBufferList to wrap the existing buffer let audioBufferList = UnsafeMutablePointer<AudioBufferList>.allocate(capacity: 1) audioBufferList.pointee.mNumberBuffers = 1 audioBufferList.pointee.mBuffers.mNumberChannels = channelCount audioBufferList.pointee.mBuffers.mDataByteSize = UInt32(buffer.dataBuffer().length()) audioBufferList.pointee.mBuffers.mData = buffer.dataBuffer().data().mutableBytes // Directly use LiveSwitch buffer // Transfer ownership of the buffer to AVAudioPCMBuffer let pcmBuffer = AVAudioPCMBuffer(pcmFormat: avAudioFormat, bufferListNoCopy: audioBufferList) /* { buffer in // Ensure the buffer is freed when AVAudioPCMBuffer is deallocated buffer.deallocate() // Only call this if LiveSwitch allows manual deallocation } */ pcmBuffer?.frameLength = frameLength return pcmBuffer } } This is the handler that is invoked with every frame in order to convert it for use with AVAudioFile and optionally update a scrolling signal display on the screen. private func onRaisedFrame(obj: Any!) -> Void { // Bail out early if no one is interested in the data. guard isMonitoring else { return } // Convert LS frame to AVAudioPCMBuffer (no-copy) let frame = obj as! FMLiveSwitchAudioFrame let buffer = frame.convertedToPCMBuffer() // Hand subscribers a reference to the buffer for rendering to display. bufferPublisher?.send(buffer) // If we have and output file, store the data there, as well. guard let audioFile = self.audioFile else { return } do { try audioFile.write(from: buffer) // FIXME: This call is throwing error -50 } catch { FMLiveSwitchLog.error(withMessage: "Failed to write buffer to audio file at \(audioFile.url): \(error)") self.audioFile = nil } } This is how the audio file is being setup. static var recordingFormat: AVAudioFormat = { AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44_100, channels: 2, interleaved: true)! }() let audioFile = try AVAudioFile(forWriting: outputURL, settings: Self.recordingFormat.settings)
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457
Mar ’25
Logic Pro - discover channel upstream latency
Hello everyone, I've written an audio unit plugin that needs to be aware of any upstream latency caused by heavy plugins before it on the channel. Is there any way to query this? I know that Logic applies PDC at the channel's output (summing point), but I need to know what the accumulated latency is at the point the audio enters my plugin. Thanks!
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350
Jan ’26
occasional glitches and empty buffers when using AudioFileStream + AVAudioConverter
I'm streaming mp3 audio data using URLSession/AudioFileStream/AVAudioConverter and getting occasional silent buffers and glitches (little bleeps and whoops as opposed to clicks). The issues are present in an offline test, so this isn't an issue of underruns. Doing some buffering on the input coming from the URLSession (URLSessionDataTask) reduces the glitches/silent buffers to rather infrequent, but they do still happen occasionally. var bufferedData = Data() func parseBytes(data: Data) { bufferedData.append(data) // XXX: this buffering reduces glitching // to rather infrequent. But why? if bufferedData.count > 32768 { bufferedData.withUnsafeBytes { (bytes: UnsafeRawBufferPointer) in guard let baseAddress = bytes.baseAddress else { return } let result = AudioFileStreamParseBytes(audioStream!, UInt32(bufferedData.count), baseAddress, []) if result != noErr { print("❌ error parsing stream: \(result)") } } bufferedData = Data() } } No errors are returned by AudioFileStream or AVAudioConverter. func handlePackets(data: Data, packetDescriptions: [AudioStreamPacketDescription]) { guard let audioConverter else { return } var maxPacketSize: UInt32 = 0 for packetDescription in packetDescriptions { maxPacketSize = max(maxPacketSize, packetDescription.mDataByteSize) if packetDescription.mDataByteSize == 0 { print("EMPTY PACKET") } if Int(packetDescription.mStartOffset) + Int(packetDescription.mDataByteSize) > data.count { print("❌ Invalid packet: offset \(packetDescription.mStartOffset) + size \(packetDescription.mDataByteSize) > data.count \(data.count)") } } let bufferIn = AVAudioCompressedBuffer(format: inFormat!, packetCapacity: AVAudioPacketCount(packetDescriptions.count), maximumPacketSize: Int(maxPacketSize)) bufferIn.byteLength = UInt32(data.count) for i in 0 ..< Int(packetDescriptions.count) { bufferIn.packetDescriptions![i] = packetDescriptions[i] } bufferIn.packetCount = AVAudioPacketCount(packetDescriptions.count) _ = data.withUnsafeBytes { ptr in memcpy(bufferIn.data, ptr.baseAddress, data.count) } if verbose { print("handlePackets: \(data.count) bytes") } // Setup input provider closure var inputProvided = false let inputBlock: AVAudioConverterInputBlock = { packetCount, statusPtr in if !inputProvided { inputProvided = true statusPtr.pointee = .haveData return bufferIn } else { statusPtr.pointee = .noDataNow return nil } } // Loop until converter runs dry or is done while true { let bufferOut = AVAudioPCMBuffer(pcmFormat: outFormat, frameCapacity: 4096)! bufferOut.frameLength = 0 var error: NSError? let status = audioConverter.convert(to: bufferOut, error: &error, withInputFrom: inputBlock) switch status { case .haveData: if verbose { print("✅ convert returned haveData: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(haveData) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } case .inputRanDry: if verbose { print("🔁 convert returned inputRanDry: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(inputRanDry) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } return // wait for next handlePackets case .endOfStream: if verbose { print("✅ convert returned endOfStream") } return case .error: if verbose { print("❌ convert returned error") } if let error = error { print("error converting: \(error.localizedDescription)") } return @unknown default: fatalError() } } }
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561
Jul ’25
tvOS AVQueuePlayer Now Playing Info in Control Center?
I have a music app I'm developing and having a weird issue where I can see now playing info for every other platform than tvOS. As far as I can tell I have correctly configured the MPNowPlayingInfoCenter MPNowPlayingInfoCenter.default().nowPlayingInfo = nowPlayingInfo MPNowPlayingInfoCenter.default().playbackState = .playing Are there any extra requirements to get my app's now-playing info showing in control center on tvOS? Another strange issue that might be related is I can use the apple TV remote to pause audio but not resume playback, so I feel like there's something I'm missing about registering audio playback on tvOS specifically.
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100
Jun ’25
What is the best approach to multi-channel, per-channel volume control.
I've got a setup using AVAudioEngine with several tone generator nodes, each with a chain of processing nodes, the chains then mixed into the main output. Generator ➡️ Effect ➡️... ➡️ .mainMixerNode ➡️ .outputNode). Generator ➡️ Effect ➡️... ⤴️ ... Generator ➡️ Effect ➡️... ⤴️ The user should be able to mute any chain individually. I've found several potential approaches to muting, but not terribly happy with any of them. Adjust the amplitudes directly in my tone generators. Issue: Consumes CPU even when completely muted. 4 generators adds ~15% cpu, even when all chains are muted. Detach/attach chains that are muted/unmuted. Issue: Causes loud clicking/popping sounds whenever muted/unmuted. Fade mixer output volume while detaching/attaching a chain (just cutting the volume immediately to 0 doesn't get rid of the clicking/popping). Issue: Causes all channels to fade during the transition, so not ideal. The rest of these ideas are variations on making volume control+detatch/attach work for individual chains, since approach #3 worked well. Add an AVAudioMixer to the end of each chain (just for volume control). Issue: Only the mixer on the final chain functions -- the others block all output. Not sure what's going on there. Use matrix mixer (for multi-input volume control). Plus detach/attach to reduce CPU if necessary. Not yet attempted, due to perceived complexity and reports of fragility in order of wiring in. A bunch of effort before I even know if it's going to work. Develop my own fader node to put on the end of each channel. Unlike the tone generator (simple AVSourceNode), developing an effect node seems complex and time consuming. Might not even fix CPU use. I'm not completely averse to the learning curve of either 5 or 6, but would rather get some guidance on best approach before diving in. They both seem likely to take more effort than I'd like for the simple behavior I'm trying to achieve.
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356
Jul ’25
access transport in Logic Pro
hi, i need to read wether the transport is playing or stopped but my current method that works for vst does not work for au. is there a lpx resource available for developers anywhere? if (auto* playHead = processor->getPlayHead()) { juce::AudioPlayHead::CurrentPositionInfo posInfo; if (playHead->getCurrentPosition(posInfo)) { bool isCurrentlyPlaying = posInfo.isPlaying; if (isCurrentlyPlaying != wasTransportPlaying) { if (isCurrentlyPlaying) { wasTransportPlaying = isCurrentlyPlaying; startAllTimers(); } else { wasTransportPlaying = isCurrentlyPlaying; stopAllTimers(); } } } } thanks :)
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306
Mar ’25
Audio / Video sync issue on iOS using AVSampleBufferRenderSynchronizer
My current app implements a custom video player, based on a AVSampleBufferRenderSynchronizer synchronising two renderers: an AVSampleBufferDisplayLayer receiving decoded CVPixelBuffer-based video CMSampleBuffers, and an AVSampleBufferAudioRenderer receiving decoded lpcm-based audio CMSampleBuffers. The AVSampleBufferRenderSynchronizer is started when the first image (in presentation order) is decoded and enqueued, using avSynchronizer.setRate(_ rate: Float, time: CMTime), with rate = 1 and time the presentation timestamp of the first decoded image. Presentation timestamps of video and audio sample buffers are consistent, and on most streams, the audio and video are correctly synchronized. However on some network streams, on iOS, the audio and video aren't synchronized, with a time difference that seems to increase with time. On the other hand, with the same player code and network streams on macOS, the synchronization always works fine. This reminds me of something I've read, about cases where an AVSampleBufferRenderSynchronizer could not synchronize audio and video, causing them to run with independent and potentially drifting clocks, but I cannot find it again. So, any help / hints on this sync problem will be greatly appreciated! :)
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1.3k
Apr ’25
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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873
Oct ’25
Why Does WebView Audio Get Quiet During RTC Calls? (AVAudioSession Analysis)
I developed an educational app that implements audio-video communication through RTC, while using WebView to display course materials during classes. However, some users are experiencing an issue where the audio playback from WebView is very quiet. I've checked that the AVAudioSessionCategory is set by RTC to AVAudioSessionCategoryPlayAndRecord, and the AVAudioSessionCategoryOption also includes AVAudioSessionCategoryOptionMixWithOthers. What could be causing the WebView audio to be suppressed, and how can this be resolved?
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557
Jul ’25
failed to set category, reason: 未能完成操作。(OSStatus错误4097。)
When using the [AVAudioSession setCategory:withOptions:error:] API, the call hangs for a long time and eventually returns an error.This issue occurs on iOS 16, and did not appear in earlier versions. Thread 135: 0 libsystem_kernel.dylib 0x00000002478e3cd4 _mach_msg2_trap :8 (in libsystem_kernel.dylib) 1 libsystem_kernel.dylib 0x00000002478e7214 _mach_msg_overwrite :428 (in libsystem_kernel.dylib) 2 libsystem_kernel.dylib 0x00000002478e705c _mach_msg :24 (in libsystem_kernel.dylib) 3 libdispatch.dylib 0x00000001d63ffe84 __dispatch_mach_send_and_wait_for_reply :548 (in libdispatch.dylib) 4 libdispatch.dylib 0x00000001d6400224 _dispatch_mach_send_with_result_and_wait_for_reply :60 (in libdispatch.dylib) 5 libxpc.dylib 0x00000001b2114e04 _xpc_connection_send_message_with_reply_sync :256 (in libxpc.dylib) 6 Foundation 0x000000019b6249f0 ___NSXPCCONNECTION_IS_WAITING_FOR_A_SYNCHRONOUS_REPLY__ :16 (in Foundation) 7 Foundation 0x000000019c06d1b4 -[NSXPCConnection _sendInvocation:orArguments:count:methodSignature:selector:withProxy:] :2100 (in Foundation) 8 CoreFoundation 0x000000019dfcb1cc ____forwarding___ :1072 (in CoreFoundation) 9 CoreFoundation 0x000000019dfd3200 ___forwarding_prep_0___ :96 (in CoreFoundation) 10 AudioSession 0x00000001c77498b0 __ZN4avas6client11SessionCore10HandlePingEv :192 (in AudioSession) 11 AudioSession 0x00000001c77497b0 ____ZN4avas6client11SessionCore12DispatchPingEv_block_invoke :52 (in AudioSession) 12 libdispatch.dylib 0x00000001d63e4adc __dispatch_call_block_and_release :32 (in libdispatch.dylib) 13 libdispatch.dylib 0x00000001d63fe7ec __dispatch_client_callout :16 (in libdispatch.dylib) 14 libdispatch.dylib 0x00000001d63ed468 __dispatch_lane_serial_drain :740 (in libdispatch.dylib) 15 libdispatch.dylib 0x00000001d63edf78 __dispatch_lane_invoke :440 (in libdispatch.dylib) 16 libdispatch.dylib 0x00000001d63f6f48 __dispatch_root_queue_drain :364 (in libdispatch.dylib) 17 libdispatch.dylib 0x00000001d63f6d08 __dispatch_worker_thread :268 (in libdispatch.dylib) 18 libsystem_pthread.dylib 0x00000001f9ff144c __pthread_start :136 (in libsystem_pthread.dylib) 19 libsystem_pthread.dylib 0x00000001f9fed8cc _thread_start :8 (in libsystem_pthread.dylib) Thread 132: 0 libsystem_kernel.dylib 0x00000002478e3cd4 _mach_msg2_trap :8 (in libsystem_kernel.dylib) 1 libsystem_kernel.dylib 0x00000002478e7214 _mach_msg_overwrite :428 (in libsystem_kernel.dylib) 2 libsystem_kernel.dylib 0x00000002478e705c _mach_msg :24 (in libsystem_kernel.dylib) 3 libdispatch.dylib 0x00000001d63ffe84 __dispatch_mach_send_and_wait_for_reply :548 (in libdispatch.dylib) 4 libdispatch.dylib 0x00000001d6400224 _dispatch_mach_send_with_result_and_wait_for_reply :60 (in libdispatch.dylib) 5 libxpc.dylib 0x00000001b2114e04 _xpc_connection_send_message_with_reply_sync :256 (in libxpc.dylib) 6 Foundation 0x000000019b6249f0 ___NSXPCCONNECTION_IS_WAITING_FOR_A_SYNCHRONOUS_REPLY__ :16 (in Foundation) 7 Foundation 0x000000019c06d1b4 -[NSXPCConnection _sendInvocation:orArguments:count:methodSignature:selector:withProxy:] :2100 (in Foundation) 8 CoreFoundation 0x000000019dfcb1cc ____forwarding___ :1072 (in CoreFoundation) 9 CoreFoundation 0x000000019dfd3200 ___forwarding_prep_0___ :96 (in CoreFoundation) 10 AudioSession 0x00000001c7754198 __ZNK4avas6client11SessionCore18SetBatchPropertiesEP12NSDictionaryIP8NSStringPU25objcproto14NSSecureCoding11objc_objectEPU15__autoreleasingP7NSArrayIPS2_IS4_P8NSNumberEENS_30AVAudioSessionBatchSetStrategyEbb :548 (in AudioSession) 11 AudioSession 0x00000001c7753e58 __ZNK4avas6client11SessionCore20SetBatchPropertiesMXEP12NSDictionaryIP8NSStringPU25objcproto14NSSecureCoding11objc_objectE :92 (in AudioSession) 12 AudioSession 0x00000001c775179c __ZN4avas6client11SessionCore11setCategoryEP8NSStringS3_32AVAudioSessionRouteSharingPolicym :472 (in AudioSession) 13 AudioSession 0x00000001c7768f88 -[AVAudioSession setCategory:withOptions:error:] :68 (in AudioSession) 14 AlipayWallet 0x000000010140580c -[AVAudioSession(APMHook) apmhook_setCategory:withOptions:error:] APMHookAudioSession.m:35 (in AlipayWallet) 15 AlipayWallet 0x00000001014001a4 -[APMAudioSessionManager resume] APMAudioSessionManager.m:718 (in AlipayWallet) 16 libdispatch.dylib 0x00000001d63e4adc __dispatch_call_block_and_release :32 (in libdispatch.dylib) 17 libdispatch.dylib 0x00000001d63fe7ec __dispatch_client_callout :16 (in libdispatch.dylib) 18 libdispatch.dylib 0x00000001d63ed468 __dispatch_lane_serial_drain :740 (in libdispatch.dylib) 19 libdispatch.dylib 0x00000001d63edf44 __dispatch_lane_invoke :388 (in libdispatch.dylib) 20 libdispatch.dylib 0x00000001d63f83ec __dispatch_root_queue_drain_deferred_wlh :292 (in libdispatch.dylib) 21 libdispatch.dylib 0x00000001d63f7ce4 __dispatch_workloop_worker_thread :692 (in libdispatch.dylib) 22 libsystem_pthread.dylib 0x00000001f9fee3b8 __pthread_wqthread :292 (in libsystem_pthread.dylib) 23 libsystem_pthread.dylib 0x00000001f9fed8c0 _start_wqthread :8 (in libsystem_pthread.dylib)
2
0
751
Feb ’26
AVAudioEngine failing with -10877 on macOS 26 beta, no devices detected via AVFoundation but HAL works
I’m developing a macOS audio monitoring app using AVAudioEngine, and I’ve run into a critical issue on macOS 26 beta where AVFoundation fails to detect any input devices, and AVAudioEngine.start() throws the familiar error 10877. FB#: FB19024508 Strange Behavior: AVAudioEngine.inputNode shows no channels or input format on bus 0. AVAudioEngine.start() fails with -10877 (AudioUnit connection error). AVCaptureDevice.DiscoverySession returns zero audio devices. Microphone permission is granted (authorized), and the app is properly signed and sandboxed with com.apple.security.device.audio-input. However, CoreAudio HAL does detect all input/output devices: Using AudioObjectGetPropertyDataSize and AudioObjectGetPropertyData with kAudioHardwarePropertyDevices, I can enumerate 14+ devices, including AirPods, USB DACs, and BlackHole. This suggests the lower-level audio stack is functional. I have tried: Resetting CoreAudio with sudo killall coreaudiod Rebuilding and re-signing the app Clearing TCC with tccutil reset Microphone Running on Apple Silicon and testing Rosetta/native detection via sysctl.proc_translated Using a fallback mechanism that logs device info from HAL and rotates logs for submission via Feedback Assistant I have submitted logs and a reproducible test case via Feedback Assitant : FB#: FB19024508]
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433
Jul ’25
How to record voice, auto-transcribe, translate (auto-detect input language), and play back translated audio on same device in iOS Swift?
Hi everyone 👋 I’m building an iOS app in Swift where I want to do the following: Record the user’s voice Transcribe the spoken sentence (speech-to-text) Auto-detect the spoken language Translate it to another language selected by the user (e.g., English → Spanish or Hindi → English) Speak back (text-to-speech) the translated text on the same device Is this possible to record via phone mic and play the transcribe voice into headphone's audio?
0
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275
Oct ’25