I use the Apple Music API to poll my listening history at regular intervals. Every morning between 5:30AM and 7:30AM, I observe a strange pattern in the API responses. During this window, one or more of the regular polling intervals returns a response that differs significantly from the prior history response, even though I had no listening activity at that time. I'm using this endpoint: https://api.music.apple.com/v1/me/recent/played/tracks?types=songs,library-songs&include[library-songs]=catalog&include[songs]=albums,artists Here’s a concrete example from this morning: Time: 5:45AM Fetch 1 Tracks (subset): 1799261990, 1739657416, 1786317143, 1784288789, 1743250261, 1738681804, 1789325498, 1743036755, ... Time: 5:50AM Fetch 2 Tracks (subset): 1799261990, 1739657416, 1786317143, 1623924746, 1635185172, 1574004238, 1198763630, 1621299055, ... Time: 5:55AM Fetch 3 Tracks (subset): 1799261990, 1739657416, 1786317143, 1784288789, 1743250261, 1738681804, 1789325498, 1743036755, ... At 5:50, a materially different history is returned, then it returns back to the prior history at the next poll. I've listened to all of the tracks in each set, but the 5:50 history drops some tracks and returns some from further back in history. I've connected other accounts and the behavior is consistent and repeatable every day across them. It appears the API is temporarily returning a different (possibly outdated or cached?) view of the user's history during that early morning window. Has anyone seen this behavior before? Is this a known issue with the Apple Music API or MusicKit backend? I'd love any insights into what might cause this, or recommendations on how to work around it.

Hi, On macOS I used to open MP3 and MP4 files with ExtAudioFile. For a few years it doesn't work anymore. So I decided to try different macOS API using the AudioFileID of AudioToolbox framework. I decided to write a test: https://gist.github.com/joelkraehemann/7f5b241b52ca38c3a765c138fb647588 It fails right here: AudioFileOpenWithCallbacks() By telling OSStatus error 1954115647, which means kAudioFileUnsupportedFileTypeError. The filename was set to an MP4 file: ~/Music/test.mp4 Howto fix this? regards, Joël

I am developing a VOD playback app, but when I stream video to an external monitor connected via HDMI with Lightning on iOS 18 or later, the screen goes dark and I cannot confirm playback. The app I am developing does not detect the HDMI and display the Player separately, but simply mirrors the video. We have confirmed that the same phenomenon occurs with other services, but we were able to confirm playback with some services such as Apple TV. Please let us know if there are any other necessary settings such as video certificates required for video playback. We would also like to know if the problem occurs with iOS 18 or later.

The device is connected to Bluetooth A and Bluetooth B, currently the audio is played through Bluetooth A, click the interface button, how to realize the code to switch to Bluetooth B？

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

Hi Team, We are using AVFoundation to read metadata from a stream and have noticed some delay between when the stream provides metadata and when the app receives it. Could someone from the team advise on ways to reduce this? Thanks

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)

Xcode Version 16.3 (16E140) App developed in Flutter Flutter 3.29.3 Test iPhone device: iPhone 16 Pro running iOS 18.5 I have an app that requires Camera access. This used to work before with iOS 18.4.x. I have dumbed down my app to just get Camera permission. Even then it fails flutter: Camera permission: PermissionStatus.denied flutter: Photos permission: PermissionStatus.denied flutter: Microphone permission: PermissionStatus.denied flutter: --- End Debug Info --- flutter: Loaded translations from asset for en_US container_create_or_lookup_app_group_path_by_app_group_identifier: client is not entitled container_create_or_lookup_app_group_path_by_app_group_identifier: client is not entitled container_create_or_lookup_app_group_path_by_app_group_identifier: client is not entitled container_create_or_lookup_app_group_path_by_app_group_identifier: client is not entitled container_create_or_lookup_app_group_path_by_app_group_identifier: client is not entitled container_create_or_lookup_app_group_path_by_app_group_identifier: client is not entitled flutter: CAMERA PERMISSION STATUS: PermissionStatus.permanentlyDenied Camera permissions don't show up in my App settings or under "Settings -&gt; Privacy and Security -&gt; Camera" and I am at loss to understand why this is happening.

Hello, I'm trying to create a webbrowser but currently when signed into apple music webplayer I get the following message when I attempt to play on any versions of my webbrowser: Not available on the web You can listen to this in the Apple Music app. Is there a way to setup DRM (assuming this is the issue) with apple to allow my webbrowser to play this content? I believe Apple TV is also affected. Thank you ahead of time.

In iOS 18, CarPlay shows an error: “There was a problem loading this content” after playback starts. Audio works fine, but the Now Playing screen doesn’t load. I’m using MPPlayableContentManager. This worked fine in iOS 17. Anyone else seeing this error in iOS 18?

I’m developing a hybrid app (WebView / Turbo Native) that uses getUserMedia to access the back camera for a PPG/heart rate measurement feature (the user places their finger on the camera). Problem: Even when I specify constraints like: { video: { deviceId: '...', facingMode: { exact: 'environment' }, advanced: [{ zoom: 1.0 }] }, audio: false } On iPhone 15 (iOS 18), iOS unexpectedly switches between the wide, ultra-wide, and telephoto lenses during the measurement. This breaks the heart rate detection, and it forces the user to move their finger in the middle of the measurement. Question: Is there any way, via getUserMedia/WebRTC, to force iOS to use only the wide-angle lens and prevent automatic lens switching? I know that with AVFoundation (Swift) you can pick .builtInWideAngleCamera, but I’m hoping to avoid building a custom native layer and would prefer to stick with WebView/JavaScript if possible to save time and complexity. Any suggestions, workarounds, or updates from Apple would be greatly appreciated! Thanks a lot!

I’m currently working on a project where I capture both depth frames and RGB frames using AVCaptureDataOutputSynchronizer. Depth frames are stored as raw binary data and RGB frames are saved with AVAssetWriter. The issue I’m facing is that AVAssetWriter enforces a fixed framerate, meaning it adds or discards frames to maintain that rate (as I understand it). This causes a desynchronization between the depth and RGB frames, which is a problem because I need each depth frame to be exactly matched with the corresponding RGB frame as they were captured. How can I ensure that the RGB frames are saved without AVAssetWriter modifying the frame count?

I want to create a Live Photo. The project includes a .jpg image and a .mov video (2 seconds). Two permissions in xcode have been added: Privacy - Photo Library Usage Description Privacy - Photo Library Additions Usage Description Simulate: iphone 16, ios 18.3 The codes in ContentView.swift : private func saveLivePhoto(imageURL: URL, videoURL: URL, completion: @escaping (Bool, Error?) -> Void) { PHPhotoLibrary.shared().performChanges { let creationRequest = PHAssetCreationRequest.forAsset() let options = PHAssetResourceCreationOptions() options.shouldMoveFile = false creationRequest.addResource(with: .photo, fileURL: imageURL, options: options) creationRequest.addResource(with: .pairedVideo, fileURL: videoURL, options: options) } completionHandler: { success, error in DispatchQueue.main.async { print(error) completion(success, error) } } } guard let imageURL = Bundle.main.url(forResource: "livephoto", withExtension: "jpeg"), let videoURL = Bundle.main.url(forResource: "livephoto", withExtension: "mov") else { showAlertMessage(title: "error", message: "cant find Live Photo ") return } print("imageURL: \(imageURL)") print("videoURL: \(videoURL)") saveLivePhoto(imageURL: imageURL, videoURL: videoURL) { success, error in if success { xxxxx } else { xxxxx } } Really need help, thanks

The documentation of the Apple Music Feed API states that the Album object would include songs — a list of songs in the album. As per the doc, this would be an array of object Album.RelatedSong, which would include both an id and nameDefault. Unfortunately, the album parquet files only include the song id and not the name of the song itself. Please advise - is the documentation incorrect or are the recently posted parquet files in error?

t has been quite some time since I requested the Apple FPS package, yet I haven’t received it. I haven’t received any email either. Is there a developer support inquiry center where I can check the status of the process? Alternatively, could you share approximately how long it took for you to receive a response email?

I did watch WWDC 2019 Session 716 and understand that an active audio session is key to unlocking low‑level networking on watchOS. I’m configuring my audio session and engine as follows: private func configureAudioSession(completion: @escaping (Bool) -> Void) { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: []) try audioSession.setActive(true, options: .notifyOthersOnDeactivation) // Retrieve sample rate and configure the audio format. let sampleRate = audioSession.sampleRate print("Active hardware sample rate: \(sampleRate)") audioFormat = AVAudioFormat(standardFormatWithSampleRate: sampleRate, channels: 1) // Configure the audio engine. audioInputNode = audioEngine.inputNode audioEngine.attach(audioPlayerNode) audioEngine.connect(audioPlayerNode, to: audioEngine.mainMixerNode, format: audioFormat) try audioEngine.start() completion(true) } catch { print("Error configuring audio session: \(error.localizedDescription)") completion(false) } } private func setupUDPConnection() { let parameters = NWParameters.udp parameters.includePeerToPeer = true connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupTCPConnection() { let parameters = NWParameters.tcp connection = NWConnection(host: "***.***.xxxxx.***", port: 0000, using: parameters) setupNWConnectionHandlers() } private func setupWebSocketConnection() { guard let url = URL(string: "ws://***.***.xxxxx.***:0000") else { print("Invalid WebSocket URL") return } let session = URLSession(configuration: .default) webSocketTask = session.webSocketTask(with: url) webSocketTask?.resume() print("WebSocket connection initiated") sendAudioToServer() receiveDataFromServer() sendWebSocketPing(after: 0.6) } private func setupNWConnectionHandlers() { connection?.stateUpdateHandler = { [weak self] state in DispatchQueue.main.async { switch state { case .ready: print("Connected (NWConnection)") self?.isConnected = true self?.failToConnect = false self?.receiveDataFromServer() self?.sendAudioToServer() case .waiting(let error), .failed(let error): print("Connection error: \(error.localizedDescription)") DispatchQueue.main.asyncAfter(deadline: .now() + 2) { self?.setupNetwork() } case .cancelled: print("NWConnection cancelled") self?.isConnected = false default: break } } } connection?.start(queue: .main) } Duplex in this context refers to two-way audio transmission simultaneously recording and sending audio while also receiving and playing back incoming audio, similar to a VoIP/SIP call. The setup works fine on the simulator, which suggests that the core logic is correct. However, since the simulator doesn’t fully replicate WatchOS hardware behavior especially for audio sessions and networking issues might arise when running on a real device. The problem likely lies in either the Watch’s actual hardware limitations, permission constraints, or specific audio session configurations. I am reaching out to seek further assistance regarding the challenges I've been experiencing with establishing a UDP, TCP & web socket connection on watchOS using NWConnection for duplex audio streaming. Despite implementing the recommendations provided earlier, I am still encountering difficulties From what I can see, your implementation is focused on streaming audio playback with the server. In my case, I'm looking for a slightly different approach: I want to capture audio and send buffers of a specific size to the server while playing audio simultaneously, essentially achieving full duplex streaming similar to a VOIP call. Additionally, I’d like to ensure that if no external audio route is connected, the Apple Watch speaker is used by default. Any thoughts or insights on adapting this setup for those requirements would be very welcome.

We are seeing logs were on iOS devices we see some keyframes request. but on safari browser don’t see any request like this. could you please explain what is it. /d8ceb9244ff889b42b82eb807327531-c27dbcb10e0bbf3cde6c-1/d8ceb9244ff88e9b42b82eb807327531-c27dbcb10e0bbf3cde6c-1/keyframes/hls/.

In our logging tools (Firebase) I see a lot of errors reported when users are playing content and the app transitions to the background. A AVPlayerItemFailedToPlayToEndTime notification is fired with an error containing error codes like -1102 and 1852797029 which seem to correspond to NSURLErrorNoPermissionsToReadFile and kCMIOHardwareIllegalOperationError respectively. To me, it looks like these might have something to do with caching logic. The items being played are HLS streams and we make use of AVAssetDownloadTask to make any streamed content offline available. Our setup is similar to the sample provided here: https://developer.apple.com/documentation/avfoundation/using-avfoundation-to-play-and-persist-http-live-streams. Whenever an item is selected for playback the app will check if a cached version is available and if so gets the url to the stored file like the "localAssetForStream()" method in the example, or get the asset from a currently running AVAssetDownloadTask for the item, or else, starts a new AVAssetDownloadTask and returns a AVAsset from that task to play. This seems to work fine, and I can't reproduce the issues our users and our logging tools are reporting. Is there some case I am missing where AVAssetDownloadTask and associated AVAssets might become unreadable when the app transitions to the background? Or do these errors indicate a different problem entirely?

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!}

I am playing FairPlay + Multi-Key content (fMP4) in Safari browser. I want to implement the implementation to distinguish between SD and HD video quality, and play it in HD if HDCP is supported, and in SD if HDCP is not supported. I have already confirmed that HDCP support is the default, and that a black screen is output in non-HDCP environments. What I want is to improve the user experience by appropriately switching to SD/HD depending on HDCP support when playing DRM content. Question: Is there an API or function that can detect HDCP support in Safari through JavaScript or other methods? Or is there a way to indirectly guess it?