Hi, I have just implemented an Audio Unit v3 host. AgsAudioUnitPlugin *audio_unit_plugin; AVAudioUnitComponentManager *audio_unit_component_manager; NSArray<AVAudioUnitComponent *> *av_component_arr; AudioComponentDescription description; guint i, i_stop; if(!AGS_AUDIO_UNIT_MANAGER(audio_unit_manager)){ return; } audio_unit_component_manager = [AVAudioUnitComponentManager sharedAudioUnitComponentManager]; /* effects */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_Effect; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } /* instruments */ description = (AudioComponentDescription) {0,}; description.componentType = kAudioUnitType_MusicDevice; av_component_arr = [audio_unit_component_manager componentsMatchingDescription:description]; i_stop = [av_component_arr count]; for(i = 0; i < i_stop; i++){ ags_audio_unit_manager_load_component(audio_unit_manager, (gpointer) av_component_arr[i]); } But this doesn't show me Audio Unit v2 plugins, why? regards, Joël

Hi, I just started to develop audio unit hosting support in my application. Offline rendering seems to work except that I hear no output, but why? I suspect with the player goes something wrong. I connect to CoreAudio in a different location in the code. Here are some error messages I faced so far: 2025-08-14 19:42:04.132930+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:42:04.151171+0200 com.gsequencer.GSequencer[34358:18611871] [avae] AVAudioEngineGraph.mm:4668 Can't retrieve source node to play sequence because there is no output node! 2025-08-14 19:43:08.344530+0200 com.gsequencer.GSequencer[34358:18614927] AUAudioUnit.mm:1417 Cannot set maximumFramesToRender while render resources allocated. 2025-08-14 19:43:08.346583+0200 com.gsequencer.GSequencer[34358:18614927] [avae] AVAEInternal.h:104 [AVAudioSequencer.mm:121:-[AVAudioSequencer(AVAudioSequencer_Player) startAndReturnError:]: (impl->Start()): error -10852 ** (<unknown>:34358): WARNING **: 19:43:08.346: error during audio sequencer start - -10852 I have implemented an AVAudioEngine based AudioUnit host. Here I instantiate player and effect: /* audio engine */ audio_engine = [[AVAudioEngine alloc] init]; fx_audio_unit_audio->audio_engine = (gpointer) audio_engine; av_format = (AVAudioFormat *) fx_audio_unit_audio->av_format; /* av audio player node */ av_audio_player_node = [[AVAudioPlayerNode alloc] init]; /* av audio unit */ av_audio_unit_effect = [[AVAudioUnitEffect alloc] initWithAudioComponentDescription:[((AVAudioUnitComponent *) AGS_AUDIO_UNIT_PLUGIN(base_plugin)->component) audioComponentDescription]]; av_audio_unit = (AVAudioUnit *) av_audio_unit_effect; fx_audio_unit_audio->av_audio_unit = av_audio_unit; /* audio sequencer */ av_audio_sequencer = [[AVAudioSequencer alloc] initWithAudioEngine:audio_engine]; fx_audio_unit_audio->av_audio_sequencer = (gpointer) av_audio_sequencer; /* output node */ [[AVAudioOutputNode alloc] init]; /* audio player and audio unit */ [audio_engine attachNode:av_audio_player_node]; [audio_engine attachNode:av_audio_unit]; [audio_engine connect:av_audio_player_node to:av_audio_unit format:av_format]; [audio_engine connect:av_audio_unit to:[audio_engine outputNode] format:av_format]; ns_error = NULL; [audio_engine enableManualRenderingMode:AVAudioEngineManualRenderingModeOffline format:av_format maximumFrameCount:buffer_size error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("enable manual rendering mode error - %d", [ns_error code]); } ns_error = NULL; [[av_audio_unit AUAudioUnit] allocateRenderResourcesAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("Audio Unit allocate render resources returned error - ErrorCode %d", [ns_error code]); } Then I render in a dedicated thread. ns_error = NULL; [audio_engine startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio engine start - %d", [ns_error code]); } [av_audio_sequencer prepareToPlay]; ns_error = NULL; [av_audio_sequencer startAndReturnError:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("error during audio sequencer start - %d", [ns_error code]); } [av_audio_player_node play]; while(is_running){ /* pre sync */ /* IO buffers */ av_output_buffer = (AVAudioPCMBuffer *) scope_data->av_output_buffer; av_input_buffer = (AVAudioPCMBuffer *) scope_data->av_input_buffer; /* fill input buffer */ /* schedule av input buffer */ frame_position = 0; // (gint64) ((note_offset * absolute_delay) + delay_counter) * buffer_size; av_audio_player_node = (AVAudioPlayerNode *) fx_audio_unit_audio->av_audio_player_node; AVAudioTime *av_audio_time = [[AVAudioTime alloc] initWithHostTime:frame_position sampleTime:frame_position atRate:((double) samplerate)]; [av_audio_player_node scheduleBuffer:av_input_buffer atTime:av_audio_time options:0 completionHandler:nil]; /* render */ ns_error = NULL; status = [audio_engine renderOffline:AGS_FX_AUDIO_UNIT_AUDIO_FIXED_BUFFER_SIZE toBuffer:av_output_buffer error:&ns_error]; if(ns_error != NULL && [ns_error code] != noErr){ g_warning("render offline error - %d", [ns_error code]); } } regards, Joël

I have an SCStreamDelegate for capturing frames from applications. On recent point releases of macOS Sonoma, I've noticed that the stream is being cancelled with no user action being taken. I started trying to debug it and when my on error method is called, the error parameter being passed is null: func stream(_ stream: SCStream, didStopWithError error: Error) { /*debugger shows this and segfaults if I try to print "\(error)" error (Error) > error = (Builtin.RawPointer) 0x0 */ From what I can tell, error should be a valid NSError so I can check the error code, based on similar code I've seen in, for example OBS (https://github.com/obsproject/obs-studio/blob/265239d4174f8d291b0de437088c5b78f8e27687/plugins/mac-capture/mac-sck-common.m#L29) Usually when this happens, the menubar icon for screen sharing (where I would click to change sharing window, etc) stays there even after my app has closed an no apps are doing sharing stuff. Has anyone come across this before? Am I misinterpreting what the debugger is saying about the error parameter? I'm running macos 14.7.3, but I just updated from 14.7.2 earlier and had basically the same issue on both macos versions

One thing I've noticed on tvOS 26 is that if you try to set the AVPlayerViewController customInfoViewControllers property while the Content Tabs are on screen, your app will crash. *** Terminating app due to uncaught exception 'UIViewControllerHierarchyInconsistency', reason: 'trying to add child view controller that is already presented: <AVInfoPanelViewController: 0x1030cdc00>' *** First throw call stack: (0x18a7167bc 0x189a77510 0x18a7166a8 0x1ab425658 0x1b2ee9d54 0x1b2efcd60 0x1b2eaf3f0 0x1080f744c 0x107e021a8 0x107e01b3c 0x18de41c14 0x18de41ba8 0x18de48d28 0x18ad9e358 0x101fac5f0 0x101fc6228 0x101fe7278 0x101fbc6fc 0x101fbc63c 0x18a67a2e0 0x18a679418 0x18a673b34 0x1937e4d5c 0x1abb36588 0x1abb3ae80 0x1aae9dec4 0x108610174 0x1086100e4 0x108615140 0x189abd4d0) I've logged a feedback (FB19554461) but it's getting awfully late in the dev cycle. So I've been trying to think of a workaround. The problem is that customInfoViewControllers is pretty declarative in nature. There are no properties or delegate methods I am aware of that let me know when they are displaying or not. One trick I came up with was seeing if my custom info view controller's view was "visible" or not - I put that in quotes because it turns out it can be visible even when I think it's not, as when the transport bar is scrolled to the top my custom VC still has its top pixels showing, so it gets a viewDidAppear call. So, I tried to see if my view controllers view is completely visible, ie based on the results of the GGRect contains method. And that works! But the problem is it only accounts for my own custom info view controllers, and not the standard one that Apple provides. I can't think of a way at all to know whether that is showing. Any ideas?

I am would like to look at AVMetricEvent data during video playback, so I have added this code to a test video player app: let playerItem: AVPlayerItem = ... let allMetrics = playerItem.allMetrics() Task.init { print("metrics task started") do { for try await metricEvent in allMetrics { print("metric event: \(metricEvent.description)") } } catch { print("unexpected metric iterator error \(error)") } } Running this in Simulator on iPhone 16 Pro (18.0) does not result in any "metric event" diagnostic messages being printed when the video associated with this AVPlayerItem is playing. Only the "metric task started" diagnostic message is seen. What am I doing wrong that prevents metric data being received?

I'm working on an application that uses the iPhone camera for scientific purposes - and, as a result would like to receive video in as unprocessed format as possible. In particular, I'm interested in getting pixel buffers that contain pretty much the bayer data as the sensor sees it - with the minimum processing of color possible. Currently we configure the AVCaptureDevice to fix the focus and exposure, use a low ISO with no gain and set the white balance gains to 1. AVCaptureVideoDataOutput is using 32BGRA. What I'd like to do is remove any additional color and brightness processing such that the data is effectively processed with a linear transfer function (i.e. gamma function is 1). I thought that this might be down to using the AVCaptureDevice activeColorSpace - we currently use P3_D65 for this. But there only seems to be a few choices (e.g. sRGB, HLG_BT2020) all of which I think affect the gamma. So: is it possible to control or specify the gamma / transfer function when using CaptureVideoDelegate? if not, does one of the color space settings have a defined gamma function that I can effectively reverse it from the pixel data without losing too much information? or is there a better way to capture video-ish speed images (15-30fps) from the camera sensor that skips processing like this? Many thanks for any suggestions.

I am using Apple's original Lightning Digital AV-adapter (Lightning-to-HDMI dongle) to connect my iPhone to an external display via a HDMI cable. I need to synchronize rendering with the external display's refresh rate, so I create a new CADisplayLink tied to the external display's UIScreen: UIScreen.screens[externalDisplayIdx].displayLink(withTarget:, selector:). The callback is being called regularly, but with increasing delay relative to the CADisplayLink.timestamp, so the next time the callback is called, I have less and less time to draw the next frame (see the snippet below). Assuming 60 FPS, the value of secondsTillDeadline starts at an arbitrary value in the range of approx -0.0001 to 0.0166667, and then it slowly decreases towards zero (and for a brief period it goes into small negative numbers). Once it reaches zero, it flips back to 0.0166667 and continues to decrease again. This cycle repeats indefinitely. Changing the external display's resolution (UIScreen's mode) or the CADisplayLink's preferredFrameRateRange to a lower FPS does not seem to have any effect on the temporal drifting (even the rate of change seem to be the same). When I create a new CADisplayLink for the iPhone's main screen, the value of secondsTillDeadline is stable, it does not drift and it is very close to 0.0166667, as expected. Is this drift caused by the external monitor or by Apple's Lightning-to-HDMI dongle ...or is the problem somewhere else? Can the drifting be stopped? func onDisplayLinkUpdate(displayLink: CADisplayLink) { // Gradually decreases from 0.01667 to -0.0001, then flips back to 0.01667 and continues to decrease let secondsTillDeadline = displayLink.targetTimestamp - CACurrentMediaTime() }

If I want to edit image in preview app. But there is only option to rotate left and right 90degree rotations. No option to rorate in any prticular angle. So Please look into this and provide option in next update

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.

Hi, I am looking for a good way to play sounds at a high frequency. At the moment I am using the AVAudioEngine, and create a couple AVAudioPlayerNode and for each sound I need to play I create a AVAudioPCMBuffer. When the app needs to play a sound, I get the correct AVAudioPCMBuffer for the sound and use the first available AVAudioPlayerNode and feed it to the buffer. The timing for a metronome app has to be very precise because if it's of by about 16ms the user can hear that it is not playing had the right interval. For low speeds this is working without any problems, but at high speeds it is getting worse. Maybe anyone has an idea on how I can improve my method. Its a Plugin for Flutter. import AVFoundation class FastSoundPlayer { private var audioPlayers: [SoundPlayer?] = [] private var sounds: [String: Sound] = [:] private var engine = AVAudioEngine() let session = AVAudioSession.sharedInstance() init() { do { try session.setCategory(AVAudioSession.Category.playback, mode: AVAudioSession.Mode.default, options: [AVAudioSession.CategoryOptions.mixWithOthers]) try session.setActive(true) createSoundPlayers(count: 20) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } // Selector method to handle applicationDidBecomeActiveNotification func applicationDidBecomeActive() { // Reinitialize AVAudioEngine and reattach all nodes do { engine.reset() objc_sync_enter(audioPlayers) audioPlayers.removeAll() createSoundPlayers(count: 20) objc_sync_exit(audioPlayers) try engine.start() } catch { print("Error starting audio engine: \(error.localizedDescription)") } } func createSoundPlayers(count: Int) { for _ in 0..<count { let player = SoundPlayer() engine.attach(player.player) engine.connect(player.player, to: engine.mainMixerNode, format: nil) audioPlayers.append(player) } } func load(sound: Data, name: String) { let sound = Sound(soundData: sound) sounds[name] = sound } func play(name: String) { if !engine.isRunning { applicationDidBecomeActive() } guard let sound = sounds[name] else { print("Sound not found") return } if let player = getAvailablePlayer() { player.play(sound: sound) } } func getAvailablePlayer() -> SoundPlayer? { for player in audioPlayers { if !player!.isPlaying { return player } } return nil } } class SoundPlayer { let player = AVAudioPlayerNode() var isPlaying = false init() { player.volume = 1.0 } func play(sound: Sound) { player.scheduleBuffer(sound.sound!, at: nil, options: .interrupts, completionCallbackType: .dataPlayedBack) { _ in self.complete() } if (player.engine != nil && player.engine!.isRunning) { player.play() isPlaying = true } } func complete() { isPlaying = false } } class Sound { var sound: AVAudioPCMBuffer? init(soundData: Data) { do { let temporaryURL = FileManager.default.temporaryDirectory.appendingPathComponent("tempSound.wav") try soundData.write(to: temporaryURL) // Create AVAudioFile from the temporary file URL let audioFile = try AVAudioFile(forReading: temporaryURL) // Define the format for the PCM buffer (44100Hz, stereo) let format = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 44100, channels: 2, interleaved: false) // Create AVAudioPCMBuffer guard let pcmBuffer = AVAudioPCMBuffer(pcmFormat: format!, frameCapacity: AVAudioFrameCount(audioFile.length)) else { // Failed to create PCM buffer self.sound = nil return } // Read audio file into PCM buffer try audioFile.read(into: pcmBuffer) // Assign the created AVAudioPCMBuffer to the sound property self.sound = pcmBuffer } catch { print("Error loading sound file: \(error.localizedDescription)") self.sound = nil } } } Thanks!

Context: I am currently developing an app using the Push-to-Talk (PTT) framework. I have reviewed both the PTT framework documentation and the CallKit demo project to better understand how to properly manage audio session activation and AVAudioEngine setup. I am not activating the audio session manually. The audio session configuration is handled in the incomingPushResult or didBeginTransmitting callbacks from the PTChannelManagerDelegate. I am using a single AVAudioEngine instance for both input and playback. The engine is started in the didActivate callback from the PTChannelManagerDelegate. When I receive a push in full duplex mode, I set the active participant to the user who is speaking. Issue When I attempt to talk while the other participant is already speaking, my input tap on the input node takes a few seconds to return valid PCM audio data. Initially, it returns an empty PCM audio block. Details: The audio session is already active and configured with .playAndRecord. The input tap is already installed when the engine is started. When I talk from a neutral state (no one is speaking), the system plays the standard "microphone activation" tone, which covers this initial delay. However, this does not happen when I am already receiving audio. Assumptions / Current Setup Because the audio session is active in play and record, I assumed that microphone input would be available immediately, even while receiving audio. However, there seems to be a delay before valid input is delivered to the tap, only occurring when switching from a receive state to simultaneously talking. Questions Is this expected behavior when using the PTT framework in full duplex mode with a shared AVAudioEngine? Should I be restarting or reconfiguring the engine or audio session when beginning to talk while receiving audio? Is there a recommended pattern for managing microphone readiness in this scenario to avoid the initial empty PCM buffer? Would using separate engines for input and output improve responsiveness? I would like to confirm the correct approach to handling simultaneous talk and receive in full duplex mode using PTT framework and AVAudioEngine. Specifically, I need guidance on ensuring the microphone is ready to capture audio immediately without the delay seen in my current implementation. Relevant Code Snippets Engine Setup func setup() { let input = audioEngine.inputNode do { try input.setVoiceProcessingEnabled(true) } catch { print("Could not enable voice processing \(error)") return } input.isVoiceProcessingAGCEnabled = false let output = audioEngine.outputNode let mainMixer = audioEngine.mainMixerNode audioEngine.connect(pttPlayerNode, to: mainMixer, format: outputFormat) audioEngine.connect(beepNode, to: mainMixer, format: outputFormat) audioEngine.connect(mainMixer, to: output, format: outputFormat) // Initialize converters converter = AVAudioConverter(from: inputFormat, to: outputFormat)! f32ToInt16Converter = AVAudioConverter(from: outputFormat, to: inputFormat)! audioEngine.prepare() } Input Tap Installation func installTap() { guard AudioHandler.shared.checkMicrophonePermission() else { print("Microphone not granted for recording") return } guard !isInputTapped else { print("[AudioEngine] Input is already tapped!") return } let input = audioEngine.inputNode let microphoneFormat = input.inputFormat(forBus: 0) let microphoneDownsampler = AVAudioConverter(from: microphoneFormat, to: outputFormat)! let desiredFormat = outputFormat let inputFramesNeeded = AVAudioFrameCount((Double(OpusCodec.DECODED_PACKET_NUM_SAMPLES) * microphoneFormat.sampleRate) / desiredFormat.sampleRate) input.installTap(onBus: 0, bufferSize: inputFramesNeeded, format: input.inputFormat(forBus: 0)) { [weak self] buffer, when in guard let self = self else { return } // Output buffer: 1920 frames at 16kHz guard let outputBuffer = AVAudioPCMBuffer(pcmFormat: desiredFormat, frameCapacity: AVAudioFrameCount(OpusCodec.DECODED_PACKET_NUM_SAMPLES)) else { return } outputBuffer.frameLength = outputBuffer.frameCapacity let inputBlock: AVAudioConverterInputBlock = { inNumPackets, outStatus in outStatus.pointee = .haveData return buffer } var error: NSError? let converterResult = microphoneDownsampler.convert(to: outputBuffer, error: &error, withInputFrom: inputBlock) if converterResult != .haveData { DebugLogger.shared.print("Downsample error \(converterResult)") } else { self.handleDownsampledBuffer(outputBuffer) } } isInputTapped = true }

FairPlay-Protected HLS Files Not Transferred via Quick Start I have an iOS app that downloads HLS files, which are protected by FairPlay. These files are stored locally, and their locations are managed using Core Data. When playing these tracks, I use AVURLAsset to access the stored file paths. Recently, a client upgraded to a new iPhone and used Quick Start to transfer data from his old device. While all other app data was successfully transferred, including Core Data records and UserDefaults, the actual HLS files were missing. As a result, the app retained metadata about the downloaded content, but the files themselves were gone, causing playback failures. Does Quick Start exclude certain types of locally stored files, especially DRM-protected HLS downloads, or is the issue related to how FairPlay-protected content is handled during the transfer of locally stored files?

No external cameras show up in the app on visionOS. We use this sample code as a basis for our tests: https://developer.apple.com/documentation/visionos/displaying-video-from-connected-devices We also received the needed entitlement from Apple, but every camera we tried so far does not show up on visionOS. We tried the following devices and hubs: Insta360 X4 Somikon Endoscope Camera: USB HD Endoscope Camera EMEET Full HD Webcam - C960 BENFEI Video/Audio Capture Card, 4K HDMI auf USB C/A Logitech C920 HD PRO Webcam, Anker PowerConf C200 Insta360 GO 3S Anker 341 USB-C Hub UGREEN Revodok Pro 10Gbps USB-C Hub All Vision Pro devices we tried run with visionOS 2.3. When trying the same code on iPad we can actually use external cameras. Steps to reproduce: Start the app on a Vision Pro device and connect an external camera. The connected camera does not show up in the dropdown. Development environment: Xcode 16.2, macOS 15.3 Run-time configuration: iOS 18.3, visionOS 2.3

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

So, I've been wondering how fast a an offline STT -> ML Prompt -> TTS roundtrip would be. Interestingly, for many tests, the SpeechTranscriber (STT) takes the bulk of the time, compared to generating a FoundationModel response and creating the Audio using TTS. E.g. InteractionStatistics: - listeningStarted: 21:24:23 4480 2423 - timeTillFirstAboveNoiseFloor: 01.794 - timeTillLastNoiseAboveFloor: 02.383 - timeTillFirstSpeechDetected: 02.399 - timeTillTranscriptFinalized: 04.510 - timeTillFirstMLModelResponse: 04.938 - timeTillMLModelResponse: 05.379 - timeTillTTSStarted: 04.962 - timeTillTTSFinished: 11.016 - speechLength: 06.054 - timeToResponse: 02.578 - transcript: This is a test. - mlModelResponse: Sure! I'm ready to help with your test. What do you need help with? Here, between my audio input ending and the Text-2-Speech starting top play (using AVSpeechUtterance) the total response time was 2.5s. Of that time, it took the SpeechAnalyzer 2.1s to get the transcript finalized, FoundationModel only took 0.4s to respond (and TTS started playing nearly instantly). I'm already using reportingOptions: [.volatileResults, .fastResults] so it's probably as fast as possible right now? I'm just surprised the STT takes so much longer compared to the other parts (all being CoreML based, aren't they?)

Hi I'm new to the forum, I'm planning an app just for Apple watch, I would like to use bluetooth audio in background, how can I do it? The messages I send via bluetooth stop as soon as the watch display turns off. Thank you! Nax

When building an application that can be built on iOS using macCatalyst, a link error like the one below will occur. Undefined symbol: OBJC_CLASS$_AVPlayerViewController The AVPlayerViewController documentation seems to support macCatalyst, but what is the reality? [AVPlayerViewController](https://developer.apple.com/documentation/avkit/avplayerviewcontroller? language=objc) Each version of the environment is as follows. Xcode 16.2 macOS deployment target: macOS 10.15 iOS deployment target: iOS 13.0 Thank you for your support.

I'm developing the VisionOS app. I want to know how to play spatial audio in addition to RealityKit? If it's iOS or macOS, how to play spatial audio in addition to RealityKit?

I started playing which transcription of audio files on macOS today, latest beta of Xcode and latest beta of Tahoe. Transcription itself works really well, but for some reason the majority of the results contain no audioTimeRange. I got 22 single-word results with time ranges, spread out all over total file of 53 minutes. Is there something I can do to improve this? To my understanding, I have followed sample code and instructions very closely, but the SwiftTranscriptionSampleApp and other examples I've seen lead me to believe I should be getting a lot more time ranges than I actually do.

Hello, I am having difficulties with configuring MusicKit correctly for my web app that I am building, seeking assistance with the issues I am having. Would greatly appreciate any help! After allowing access to the following, "Access Request media.mydomain.com would like to access Apple Music, media library, and listening activity for myemail'@icloud.com.", I get a popup error that states, "Authorization failed. Please try again.". Following is the information that is given in developer console: [Error] Failed to load resource: the server responded with a status of 403 () (webPlayerLogout, line 0) [Error] Authorization failed: AUTHORIZATION_ERROR: Unauthorized (anonymous function) (media.mydomain.com:398)