AVAudioSessionCategoryOptionAllowBluetooth is marked as deprecated in iOS 8 in iOS 26 beta 5 when this option was not deprecated in iOS 18.6. I think this is a mistake and the deprecation is in iOS 26. Am I right? It seems that the substitute for this option is "AVAudioSessionCategoryOptionAllowBluetoothHFP". The documentation does not make clear if the behaviour is exactly the same or if any difference should be expected... Has anyone used this option in iOS 26? Should I expect any difference with the current behaviour of "AVAudioSessionCategoryOptionAllowBluetooth"? Thank you.

When I try to send a DRM-protected video via Airplay to an Apple TV, the license request is made twice instead of once as it normally does on iOS. We only allow one request per session for security reasons, this causes the second request to fail and the video won't play. We've tested DRM-protected videos without token usage limits and it works, but this creates a security hole in our system. Why does it request the license twice in function: func contentKeySession(_ session: AVContentKeySession, didProvide keyRequest: AVContentKeyRequest)? Is there a way to prevent this?

In my app I use AVAssetReaderTrackOutput to extract PCM audio from a user-provided video or audio file and display it as a waveform. Recently a user reported that the waveform is not in sync with his video, and after receiving the video I noticed that the waveform is in fact double as long as the video duration, i.e. it shows the audio in slow-motion, so to speak. Until now I was using CMFormatDescription.audioStreamBasicDescription.mSampleRate which for this particular user video returns 22'050. But in this case it seems that this value is wrong... because the audio file has two audio channels with different sample rates, as returned by CMFormatDescription.audioFormatList.map({ $0.mASBD.mSampleRate }) The first channel has a sample rate of 44'100, the second one 22'050. If I use the first sample rate, the waveform is perfectly in sync with the video. The problem is given by the fact that the ratio between the audio data length and the sample rate multiplied by the audio duration is 8, double the ratio for the first audio file (4). In the code below this ratio is given by Double(length) / (sampleRate * asset.duration.seconds) When commenting out the line with the sampleRate variable definition in the code below and uncommenting the following line, the ratios for both audio files are 4, which is the expected result. I would expect audioStreamBasicDescription to return the correct sample rate, i.e. the one used by AVAssetReaderTrackOutput, which (I think) somehow merges the stereo tracks. The documentation is sparse, and in particular it’s not documented whether the lower or higher sample rate is used; in this case, it seems like the higher one is used, but audioStreamBasicDescription for some reason returns the lower one. Does anybody know why this is the case or how I should extract the sample rate of the produced PCM audio data? Should I always take the higher one? I created FB19620455. let openPanel = NSOpenPanel() openPanel.allowedContentTypes = [.audiovisualContent] openPanel.runModal() let url = openPanel.urls[0] let asset = AVURLAsset(url: url) let assetTrack = asset.tracks(withMediaType: .audio)[0] let assetReader = try! AVAssetReader(asset: asset) let readerOutput = AVAssetReaderTrackOutput(track: assetTrack, outputSettings: [AVFormatIDKey: Int(kAudioFormatLinearPCM), AVLinearPCMBitDepthKey: 16, AVLinearPCMIsBigEndianKey: false, AVLinearPCMIsFloatKey: false, AVLinearPCMIsNonInterleaved: false]) readerOutput.alwaysCopiesSampleData = false assetReader.add(readerOutput) let formatDescriptions = assetTrack.formatDescriptions as! [CMFormatDescription] let sampleRate = formatDescriptions[0].audioStreamBasicDescription!.mSampleRate //let sampleRate = formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate }).max()! print(formatDescriptions[0].audioStreamBasicDescription!.mSampleRate) print(formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate })) if !assetReader.startReading() { preconditionFailure() } var length = 0 while assetReader.status == .reading { guard let sampleBuffer = readerOutput.copyNextSampleBuffer(), let blockBuffer = sampleBuffer.dataBuffer else { break } length += blockBuffer.dataLength } print(Double(length) / (sampleRate * asset.duration.seconds))

I'm working on a video player app that has the basic functionality of viewing a video and then be able to trim and crop that video and then save it. My flow of trimming a video and then saving it works well with any and every video. Cropping, however, doesn't work in the sense that I am unable to Save the video and export it. Whenever I crop a video, in the video player, I can see the cropped version of the video (it plays too!) but on saving said video, I get the error: Export failed with status: 4, error: Cannot Decode I've been debugging for 2 days now but I'm still unsure as to why this happens. I'm almost certain the bug is somewhere cause of cropping and then saving/exporting. If anyone has dealt with this before, please let me know what the best step to do is! If you could help me refine the flow for cropping and exporting, that'd be really helpful too. Thanks!

I want to build an app for ios using react native. preferably expo. The app will be for recording user experiences with technology. the SLUDGE that they face while navigating through technology. I want to have basic login, signup. The main feature would be to have 2 recording modes. First is record the screen and the front camera simultaneously. Second is to record the back camera and the front camera simultaneously. I can then patch the two outputs later on that is the screen recording and the front camera clip in post processing. I want to know if this is possible as I was told that react native and expo does not have the support yet. if not is there any library or another approach to make this app come alive.

I am developing an app that plays HLS audio. When using AVPlayerItem with AVURLAsset, can AVAssetResourceLoaderDelegate correctly handle HLS segments? My goal is to use AVAssetResourceLoaderDelegate to add authentication HTTP headers when accessing HLS .m3u8 and .ts files. I can successfully download the files, but playback fails with errors. Specifically, I am observing the following cases: A. AVAssetResourceLoaderDelegate is canceled, and CoreMediaErrorDomain -12881 occurs In NSURLConnectionDataDelegate’s didReceiveResponse method, set contentInformationRequest In didReceiveData, call dataRequest respondWithData resourceLoader didCancelLoadingRequest is called CoreMediaErrorDomain -12881 occurs B. CoreMediaErrorDomain -12881 occurs In NSURLConnectionDataDelegate’s didReceiveResponse method, set contentInformationRequest In connection didReceiveData, buffer all received data until the end In connectionDidFinishLoading, pass the buffered data to respondWithData Call loadingRequest finishLoading CoreMediaErrorDomain -12881 occurs In both cases, dataRequest.requestsAllDataToEndOfResource is YES. For this use case, I am not using AVURLAssetHTTPHeaderFieldsKey because I need to apply the most up-to-date authentication data at the moment each file is accessed. I would appreciate any advice or suggestions you might have. Thank you in advance!

In iOS 26, AVSpeechSynthesizer read Mandarin into Cantonese pronunciation. No matter how you set the language, and change the settings of my phone system, it doesn't work. let utterance = AVSpeechUtterance(string: "你好啊") //let voice = AVSpeechSynthesisVoice(language: "zh-CN") // not work let voice = AVSpeechSynthesisVoice(language: "zh-Hans") // not work too utterance.voice = voice et synth = AVSpeechSynthesizer() synth.speak(utterance)

After updating to iOS 18.5, we’ve observed that outgoing audio from our app intermittently stops being transmitted during VoIP calls using AVAudioSession configured with .playAndRecord and .voiceChat. The session is set active without errors, and interruptions are handled correctly, yet audio capture suddenly ceases mid-call. This was not observed in earlier iOS versions (≤ 18.4). We’d like to confirm if there have been any recent changes in AVAudioSession, CallKit, or related media handling that could affect audio input behavior during long-running calls. func configureForVoIPCall() throws { try setCategory( .playAndRecord, mode: .voiceChat, options: [.allowBluetooth, .allowBluetoothA2DP, .defaultToSpeaker]) try setActive(true) }

Since iOS 18, the system setting “Allow Audio Playback” (enabled by default) allows third-party app audio to continue playing while the user is recording video with the Camera app. This has created a problem for the app I’m developing. ➡️ The problem: My app plays continuous audio in both foreground and background states. If the user starts recording video using the iOS Camera app, the app’s audio — still playing in the background — gets captured in the video — obviously an unintended behavior. Yes, the user could stop the app manually before starting the video recording, but that can’t be guaranteed. As a developer, I need a way to stop the app’s audio before the video recording begins. So far, I haven’t found a reliable way to detect when video recording starts if ‘Allow Audio Playback’ is ON. ➡️ What I’ve tried: — AVAudioSession.interruptionNotification → doesn’t fire — devicesChangedEventStream → not triggered I don’t want to request mic permission (app doesn’t use mic). also, disabling the app from playing audio in the background isn’t an option as it is a crucial part of the user experience ➡️ What I need: A reliable, supported way to detect when the Camera app begins video recording, without requiring mic access — so I can stop audio and avoid unintentional overlap with the user’s recordings. Any official guidance, workarounds, or AVFoundation techniques would be greatly appreciated. Thanks.

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?

When using AVSampleBufferDisplayLayer to play uncompressed H.264 and H.265 video with B-frames more than 7, frame drops occur. The more B-frames there are, the more noticeable the frame drops become, for example 15 bframes. Use FFmpeg to transcode a video file with visible timestamps and frame numbers (x264 or x265 ): ffmpeg -i test.mp4 -vf "drawtext=fontsize=45:text=%{pts} %{n}:y=400" -c:v libx264 -x264-params "bframes=15:b-adapt=0" -crf 30 -y x264_bf15.mp4 ffmpeg -i test.mp4 -vf "drawtext=fontsize=45:text=%{pts} %{n}:y=400" -c:v libx265 -x265-params "bframes=15:b-adapt=0" -crf 30 -y x265_bf15.mp4 Use the demo player from this repository to reproduce the issue: https://github.com/msfrms/CustomPlayer frame drops can be observed. And following log can be found in devices console. mediaserverd <<<< IQ-CA >>>> piqca_gmstats_dump: FIQCA(0x1266f4000) recent frames: enqueued: 184, displayed: 138, dropped: 42, flushed: 0, evicted: 3, >16ms late: 2 PS. I was using iphone11 iOS14.6, to replay this issue. May I ask why frame drops occur in this case? Is there any configuration or API usage change that could help fix the frame drop issue? Many thanks!

I am creating a barcode reader using the AVfoundation framework for iOS and IPadOS. The read result goes into payloadstringvalue, but I want to check the control characters contained in the symbol, so I am using the raw data of the description, which is a property of NSObjectProtocol inherited by VNBarcodeObservation. However, I noticed that if the length set in the raw data exceeds 26, some of the raw data in the description is omitted. So my question is, is it possible to set it so that all the raw data in the description is written out without omitting any raw data? If so, could you please tell me how to set this up? Also, if you know of any other way to extract the raw barcode data, I would appreciate it if you could let me know. Thank you.

Recurring crash on install of any app with the new sourceVideoTrackProvider.next() dyld[41966]: Symbol not found: _$sSo19AVAssetReaderOutputC12AVFoundationE8ProviderC4nextxSgyYaKFTjTu Referenced from: <79AA2BE0-A6B4-32F5-A804-E84BBE5D1AEA> /Users/<username>/Library/Developer/Xcode/DerivedData/TrackProviderCrash-bbbhjptcxnmfdcackxtpucnunxyc/Build/Products/Debug-maccatalyst/TrackProviderCrash.app/Contents/MacOS/TrackProviderCrash.debug.dylib Expected in: <1B847AF9-7973-3B28-95C2-09E73F6DD50B> /usr/lib/swift/libswiftAVFoundation.dylib Can be reproduced with the current Xcode Beta 4 by running on to MacCatalyst and macOS https://developer.apple.com/documentation/AVFoundation/converting-projected-video-to-apple-projected-media-profile Crash goes away of you comment out lines 154-158 and 164-170 which are while let sampleBuffer = try await sourceVideoTrackProvider.next(){/*other code*/} Can also be reproduced if you add the code below to a MacCatalyst project import AVKit let asset: AVURLAsset = .init(url: Bundle.main.url(forResource: "SomeVideo.mp4", withExtension: nil)!) let videoReader = try! AVAssetReader(asset: asset) let videoTracks = try! await asset.loadTracks(withMediaCharacteristic: .visual) // Get the side-by-side video track. let videoTrack = videoTracks.first! let videoInputTrack = AVAssetReaderTrackOutput(track: videoTrack, outputSettings: nil) let sourceVideoTrackProvider: AVAssetReaderOutput.Provider<CMReadySampleBuffer<CMSampleBuffer.DynamicContent>> = videoReader.outputProvider(for: videoInputTrack) //Comment out this while let sb = try! await sourceVideoTrackProvider.next() { }

i'm trying to work on a simple screen recording app on macOS that always records the last 'x' seconds of your screen and saves it whenever you want, as a way to get comfortable with swift programming and apple APIs. i was able to get it running for the past '30 seconds' and record and store it. however i realised that there was a core issue with my solution: i was defining the SCStreamConfiguration.queueDepth = 900 (to account for 30fps for 30 seconds) which goes completely against apple's instructions: https://developer.apple.com/documentation/screencapturekit/scstreamconfiguration/queuedepth?language=objc now when i changed queueDepth back to 8, i am only able to record 8 frames and it saves only those first 8 frames. i am unsure what the flow of the apis should be while dealing with screenCaptureKit. for context, here's my recording manager code that handles this logic (queueDepth = 900) import Foundation import ScreenCaptureKit import AVFoundation class RecordingManager: NSObject, ObservableObject, SCStreamDelegate { static let shared = RecordingManager() @Published var isRecording = false private var isStreamActive = false // Custom state flag private var stream: SCStream? private var streamOutputQueue = DispatchQueue(label: "com.clipback.StreamOutput", qos: .userInteractive) private var screenStreamOutput: ScreenStreamOutput? // Strong reference to output private var lastDisplayID: CGDirectDisplayID? private let displayCheckQueue = DispatchQueue(label: "com.clipback.DisplayCheck", qos: .background) // In-memory rolling buffer for last 30 seconds private var rollingFrameBuffer: [(CMSampleBuffer, CMTime)] = [] private let rollingFrameBufferQueue = DispatchQueue(label: "com.clipback.RollingBuffer", qos: .userInteractive) private let rollingBufferDuration: TimeInterval = 30.0 // seconds // Track frame statistics private var frameCount: Int = 0 private var lastReportTime: Date = Date() // Monitor for display availability private var displayCheckTimer: Timer? private var isWaitingForDisplay = false func startRecording() { print("[DEBUG] startRecording called.") guard !isRecording && !isWaitingForDisplay else { print("[DEBUG] Already recording or waiting, ignoring startRecording call") return } isWaitingForDisplay = true isStreamActive = true // Set active state checkForDisplay() } private func setupAndStartRecording(for display: SCDisplay, excluding appToExclude: SCRunningApplication?) { print("[DEBUG] setupAndStartRecording called for display: \(display.displayID)") let excludedApps = [appToExclude].compactMap { $0 } let filter = SCContentFilter(display: display, excludingApplications: excludedApps, exceptingWindows: []) let config = SCStreamConfiguration() config.width = display.width config.height = display.height config.minimumFrameInterval = CMTime(value: 1, timescale: 30) // 30 FPS config.queueDepth = 900 config.showsCursor = true print("[DEBUG] SCStreamConfiguration created: width=\(config.width), height=\(config.height), FPS=\(config.minimumFrameInterval.timescale)") stream = SCStream(filter: filter, configuration: config, delegate: self) print("[DEBUG] SCStream initialized.") self.screenStreamOutput = ScreenStreamOutput { [weak self] sampleBuffer, outputType in guard let self = self else { return } guard outputType == .screen else { return } guard sampleBuffer.isValid else { return } guard let attachments = CMSampleBufferGetSampleAttachmentsArray(sampleBuffer, createIfNecessary: false) as? [[SCStreamFrameInfo: Any]], let statusRawValue = attachments.first?[.status] as? Int, let status = SCFrameStatus(rawValue: statusRawValue), status == .complete else { return } self.trackFrameRate() self.handleFrame(sampleBuffer) } do { try stream?.addStreamOutput(screenStreamOutput!, type: .screen, sampleHandlerQueue: streamOutputQueue) stream?.startCapture { [weak self] error in print("[DEBUG] SCStream.startCapture completion handler.") guard error == nil else { print("[DEBUG] Failed to start capture: \(error!.localizedDescription)") self?.handleStreamError(error!) return } DispatchQueue.main.async { self?.isRecording = true self?.isStreamActive = true // Update state on successful start print("[DEBUG] Recording started. isRecording = true.") } } } catch { print("[DEBUG] Error adding stream output: \(error.localizedDescription)") handleStreamError(error) } } private func handleFrame(_ sampleBuffer: CMSampleBuffer) { rollingFrameBufferQueue.async { [weak self] in guard let self = self else { return } let pts = CMSampleBufferGetPresentationTimeStamp(sampleBuffer) var retainedBuffer: CMSampleBuffer? CMSampleBufferCreateCopy(allocator: kCFAllocatorDefault, sampleBuffer: sampleBuffer, sampleBufferOut: &retainedBuffer) guard let buffer = retainedBuffer else { print("[DEBUG] Failed to copy sample buffer") return } self.rollingFrameBuffer.append((buffer, pts)) if let lastPTS = self.rollingFrameBuffer.last?.1 { while let firstPTS = self.rollingFrameBuffer.first?.1, CMTimeGetSeconds(CMTimeSubtract(lastPTS, firstPTS)) > self.rollingBufferDuration { self.rollingFrameBuffer.removeFirst() } } } } func stream(_ stream: SCStream, didStopWithError error: Error) { print("[DEBUG] Stream stopped with error: \(error.localizedDescription)") displayCheckQueue.async { [weak self] in // Move to displayCheckQueue for synchronization self?.handleStreamError(error) } } what could be the reason for this and what would be the possible fix logically? i dont understand why it's dependant on queueDepth, and if it is, how can I empty and append new recorded frames to it so that it continues working? any help or resource is greatly appreciated!

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?

Hi all, I'm working on an audio visualizer app that plays files from the user's music library utilizing MediaPlayer and AVAudioEngine. I'm working on getting the music library functionality working before the visualizer aspect. After setting up the engine for file playback, my app inexplicably crashes with an EXC_BREAKPOINT with code = 1. Usually this means I'm unwrapping a nil value, but I think I'm handling the optionals correctly with guard statements. I'm not able to pinpoint where it's crashing. I think it's either in the play function or the setupAudioEngine function. I removed the processAudioBuffer function and my code still crashes the same way, so it's not that. The device that I'm testing this on is running iOS 26 beta 3, although my app is designed for iOS 18 and above. After commenting out code, it seems that the app crashes at the scheduleFile call in the play function, but I'm not fully sure. Here is the setupAudioEngine function: private func setupAudioEngine() { do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .default) try AVAudioSession.sharedInstance().setActive(true) } catch { print("Audio session error: \(error)") } engine.attach(playerNode) engine.attach(analyzer) engine.connect(playerNode, to: analyzer, format: nil) engine.connect(analyzer, to: engine.mainMixerNode, format: nil) analyzer.installTap(onBus: 0, bufferSize: 1024, format: nil) { [weak self] buffer, _ in self?.processAudioBuffer(buffer) } } Here is the play function: func play(_ mediaItem: MPMediaItem) { guard let assetURL = mediaItem.assetURL else { print("No asset URL for media item") return } stop() do { audioFile = try AVAudioFile(forReading: assetURL) guard let audioFile else { print("Failed to create audio file") return } duration = Double(audioFile.length) / audioFile.fileFormat.sampleRate if !engine.isRunning { try engine.start() } playerNode.scheduleFile(audioFile, at: nil) playerNode.play() DispatchQueue.main.async { [weak self] in self?.isPlaying = true self?.startDisplayLink() } } catch { print("Error playing audio: \(error)") DispatchQueue.main.async { [weak self] in self?.isPlaying = false self?.stopDisplayLink() } } } Here is a link to my test project if you want to try it out for yourself: https://github.com/aabagdi/VisualMan-example Thanks!

We found that on some iPad devices (such as iPad8,9/iPad OS/18.5), when the media is playing, the recorded human voice is very small and is basically suppressed by the sound of the playing media. When the media is not playing, the human voice can be recorded. Likely when use screen recording with microphone, the ambient sound is also basically suppressed by the media sound.

Hi all, I'm capturing a photo using AVCapturePhotoOutput, and I've set: let photoSettings = AVCapturePhotoSettings() photoSettings.isDepthDataDeliveryEnabled = true Then I create the handler like this: let data = photo.fileDataRepresentation() let handler = try ImageRequestHandler(data: data, orientation: .right) Now I’m wondering: If depth data delivery is enabled, is it actually included and used when I pass the Data to ImageRequestHandler? Or do I need to explicitly pass the depth data using the other initializer? let handler = try ImageRequestHandler( cvPixelBuffer: photo.pixelBuffer!, depthData: photo.depthData, orientation: .right ) In short: Does ImageRequestHandler(data:) make use of embedded depth info from AVCapturePhoto.fileDataRepresentation() — or is the pixel buffer + explicit depth data required? Thanks for any clarification!

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'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() } } }