Hello, Using ShazamKit, based on a shazam catalog result, would it be possible to detect the audio-recorded FPS (speed)? I'm thinking that the shazam catalog which was created from an audio file can be used to compare the speed of a live recorded audio. Thank you!

iOS (Official) Photos app can display some EXIF-related metadata (e.g. camera and lens info, ISO, shutter speed, F-number) even when photos are offloaded to iCloud and the device is not connected to internet (e.g. airplane mode). However, with the Photos.framework, we need to download photos to retrive those metadata (which means it will not work with airplane mode). I tried the following methods, but none of those worked when photos were offloaded to iCloud and the device was in airplane mode: Requesting data with PHImageManager.default().requestImageDataAndOrientation Result: It does not return Data if the photo is not stored locally on the device, even with options.deliveryMode = .fastFormat Converting PHAsset#localIdentifier to an AssetsLibrary.framework URL (assets-library://asset/...) (I am aware that AssetsLibrary.framework is deprecated, but this was just a test.) Result: If PHImageManager does not returns Data, ALAsset#defaultRepresentation().metadata() returns an empty NSDictionary

I use the iTunes Library framework in one of my apps, starting with macOS Sequoia 15.1 i can't create the ITLibrary object anymore with the following error: Connection to amplibraryd was interrupted. clientName:iTunesLibrary(ITLibraryLoader) Error connecting to the server via proxy object Error Domain=NSCocoaErrorDomain Code=4097 "connection to service named com.apple.amp.library.framework" UserInfo={NSDebugDescription=connection to service named com.apple.amp.library.framework} configure failed: Error Domain=NSCocoaErrorDomain Code=4097 "connection to service named com.apple.amp.library.framework" UserInfo={NSDebugDescription=connection to service named com.apple.amp.library.framework} I created a new sandboxed macOS app, added the music folder read permission and it reproduced the error: import SwiftUI import iTunesLibrary @main struct ITLibraryLoaderApp: App { var body: some Scene { WindowGroup { ContentView() } } } struct ContentView: View { var body: some View { Button("Load ITLibrary") { loadLibrary() } } func loadLibrary() { do { let _ = try ITLibrary(apiVersion: "1.0", options: .none) } catch { print(error) } } } I restarted my developer machine and the music app with no luck.

After an Album, Playlist, or collection of songs have been added to the ApplicationMusicPlayer queue, clearing the queue can be easily accomplished with: ApplicationMusicPlayer.shared.queue.entries = [] This transitions the player to a paused state with an empty queue. After queueing a Station, the same code cannot be used to clear the queue. Instead, it causes the queue to be refilled with a current and next MusicItem from the Station. What's the correct way to detect that the ApplicationMusicPlayer is in the state where it's being refilled by a Station and clear it? I've tried the following approaches with no luck: # Reinitialize queue ApplicationMusicPlayer.shared.queue = ApplicationMusicPlayer.Queue() # Create empty Queue let songs: [Song] = [] let emptyQueue = ApplicationMusicPlayer.Queue(for: songs) ApplicationMusicPlayer.shared.queue = emptyQueue

We have a universal iOS/tvOS app that also supports iOS App on Mac. In our AVPlayer-based video player we support AirPlay with AVRouteDetector and AVRoutePickerView. We play HLS streams. When we try to AirPlay from an iOS device to an Apple TV or a Mac that has our app installed, it doesn't work. The receiver is marked as active in the route picker UI but the video doesn't show up on the receiver and playback stops. When our app isn't installed on the receiver device, everything works as expected. Has anyone encountered the same issue? Any solutions available for this?

We're integrating a web based group calling application within a native iOS application and finding that every time a CallKit session gets fully established the web based media streams break, rendering as gray with no audio. Up to iOS 18 we worked around it by not fulfilling the call start action but that's no longer an option as the audio stopped getting automatically redirected to the speakers. We would now need the CXProvider's didActivateAudioSession callback but that would break the video. The sample project loads up a simple webpage in a WKWebView which contains a video tag streaming the media from the device's camera. At the same time it sets up a new CallKit session by requesting and fulfilling a CXStartCallAction transaction. You will notice that the media doesn't render and, if you are to follow the warnings we left, you will find that not fulfilling the CXStartCallAction fixes it. Unfortunately that's not a workaround we can use as we need the CXProvider delegate to inform us about audio session changes so we can redirect the audio to the speaker (so the proximity sensor doesn't activate and locking the screen doesn't end the call) Any insights or workarounds would be greatly appreciated.

https://api.media.apple.com/v1/feed/exports/song_2024-11-02T16-02/parts?limit=200&offset=400 This is the api used to get parquet file urls. I need all the urls in one api hit, right now if I don't provide the limit then default it is taking 100 and max is 200. How to get all the records in one hit? Or the count of parquet records in one hit?

Hi, I am in need to get the total number of parquet files that are present in the apple music feed api for songs, artists. As there is option for limit and offset. But limit is limited to 200 records and offset is uncertain. How to get total number of parquet files number without quering apple music feed api mulitple times? Need help regarding this. Thanks!

I am attempting to do batch Transcription of audio files exported from Voice Memos, and I am running into an interesting issue. If I only transcribe a single file it works every time, but if I try to batch it, only the last one works, and the others fail with No speech detected. I assumed it must be something about concurrency, so I implemented what I think should remove any chance of transcriptions running in parallel. And with a mocked up unit of work, everything looked good. So I added the transcription back in, and 1: It still fails on all but the last file. This happens if I am processing 10 files or just 2. 2: It no longer processes in order, any file can be the last one that succeeds. And it seems to not be related to file size. I have had paragraph sized notes finish last, but also a single short sentence that finishes last. I left the mocked processFiles() for reference. Any insights would be greatly appreciated. import Speech import SwiftUI struct ContentView: View { @State private var processing: Bool = false @State private var fileNumber: String? @State private var fileName: String? @State private var files: [URL] = [] let locale = Locale(identifier: "en-US") let recognizer: SFSpeechRecognizer? init() { self.recognizer = SFSpeechRecognizer(locale: self.locale) } var body: some View { VStack { if files.count > 0 { ZStack { ProgressView() Text(fileNumber ?? "-") .bold() } Text(fileName ?? "-") } else { Image(systemName: "folder.badge.minus") Text("No audio files found") } } .onAppear { files = getFiles() Task { await processFiles() } } } private func getFiles() -> [URL] { do { let documentsURL = FileManager.default.urls(for: .documentDirectory, in: .userDomainMask).first! let path = documentsURL.appendingPathComponent("Voice Memos").absoluteURL let contents = try FileManager.default.contentsOfDirectory(at: path, includingPropertiesForKeys: nil, options: []) let files = (contents.filter {$0.pathExtension == "m4a"}).sorted { url1, url2 in url1.path < url2.path } return files } catch { print(error.localizedDescription) return [] } } private func processFiles() async { var fileCount = files.count for file in files { fileNumber = String(fileCount) fileName = file.lastPathComponent await processFile(file) fileCount -= 1 } } // private func processFile(_ url: URL) async { // let seconds = Double.random(in: 2.0...10.0) // await withCheckedContinuation { continuation in // DispatchQueue.main.asyncAfter(deadline: .now() + seconds) { // continuation.resume() // print("\(url.lastPathComponent) \(seconds)") // } // } // } private func processFile(_ url: URL) async { let recognitionRequest = SFSpeechURLRecognitionRequest(url: url) recognitionRequest.requiresOnDeviceRecognition = false recognitionRequest.shouldReportPartialResults = false await withCheckedContinuation { continuation in recognizer?.recognitionTask(with: recognitionRequest) { (transcriptionResult, error) in guard transcriptionResult != nil else { print("\(url.lastPathComponent.uppercased())") print(error?.localizedDescription ?? "") return } if ((transcriptionResult?.isFinal) == true) { if let finalText: String = transcriptionResult?.bestTranscription.formattedString { print("\(url.lastPathComponent.uppercased())") print(finalText) } } } continuation.resume() } } }

We are experiencing an issue with our HLS MPEG-TS streams on Apple devices, where the AVPlayer in our iOS app and Safari jumps back to the start when the player automatically changes quality. This occurs despite the stream still indicating that it is live and there is no change in the seekbar. After testing our streams with the Apple HLS Validator, the only problem that occured was an "Measured peak bitrate compared to multivariant playlist declared value exceeds error tolerance"-Error. On Chrome and on our Android-App this playback bug does not happen. Has someone else experienced similar issues with the AVPlayer?

Somehow I have a corrupted audio plugin authentication problem. I’m on a silicon Mac M1 and two audio plugins that were installed and working will now not authenticate. The vendors both are unable to troubleshoot and I think the issue is a corrupted low level file. One product authenticates correctly when I created a new user but another plugin only authenticates on the original user account and not on the newly created user. Reinstalling the plugins and the Mac OS does not fix the issue. Any thoughts?

How to capture the audio being played by apple music in ios and combine it with fft to achieve audio visualization?

I am experiencing a bug when using a AVCapturePhotoBracketSettings object to capture a bracketed photo sequence on iPhone 16 Pro. Specifically, when I pass in an array of exposure values: [-x, 0, +x], where x >= 3. Specifically, the high exposure photo capture returns a black image. STEPS TO REPRODUCE Run the sample app I have provided on an iPhone 16 Pro Notice that bracketed images captured where the eV is set to [-3,0,+3], [-4,0,+4], or [-5,0,+5] return a black image for the high exposure photo. Notice that on other iOS devices (like iPhone 13 Pro), the high exposure photo is returned as high brightness as expected. I have also added two folders in the sample project that show screenshots of the bug: iPhone13Pro & iPhone16Pro Sample Project: https://www.icloud.com/iclouddrive/090O_68Z0Nh2UOxmPRwu56Tmw#Focused16ProBracketedCaptureBug

Hey, I am fairly new to working with AVFoundation etc. As far as I could research on my own, if I want to get metadata from let's say a .m4a audio file, I have to get the data and then create an AVAsset. My files are all on local servers and therefore I would not be able to just pass in the URL. The extraction of the metadata works fine - however those AVAssets create a huge overhead in storage consumption. To my knowledge the data instances of each audio file and AVAsset should only live inside the function I call to extract the metadata, however those data/AVAsset instances still live on on storage as I can clearly see that the app's file size increases by multiple Gigabytes (equal to the library size I test with). However, the only data that I purposefully save with SwiftData is the album artwork. Is this normal behavior for AVAssets or am I missing some detail? PS. If I forgot to mention something important, please ask. This is my first ever post, so I'm not too sure what is worth mentioning. Thank you in advance! Denis

The media services used for HLS streaming in an AVPlayer seem to crash if your segments are too large. Anything over 20Mbps seems to cause a crash. I have tried adjusting the segment length to 1 second also and it didn't help. I am remuxing Dolby Vision and HDR video and want to avoid transcoding and losing any metadata. However the segments are too large. Is there a workaround for this? Otherwise it seems AVFoundation is not suited to high bitrate HLS and I should be using MPV or similar.

I have spent the past 2 weeks diving into CoreAudio and have seemingly run into a wall... For my initial test, I am simply trying to create an AUGraph for monitoring input levels from a user chosen Audio Input Device (multi-channel in my case). I was not able to find any way to monitor input levels of a single AUHAL input device - so I decided to create a simple AUGraph for input level monitoring. Graph looks like: [AUHAL Input Device] -> [B1] -> [MatrixMixerAU] -> [B2] -> [AUHAL Output Device] Where B1 is an audio stream consisting of all the input channels available from the input device. The MatrixMixer has metering mode turned on, and level meters are read from the each submix of the MatrixMixer using kMatrixMixerParam_PostAveragePower. B2 is a stereo (2 channel) stream from the MatrixMixerAU to the default audio device - however, since I don't really want to pass audio through to an actual output, I have the volume muted on the MatrixMixerAU output channel. I tried using a GenericOutputAU instread of the default system output, however, the GenericOutputAU never seems to pull date from the ringBuffer (the graph renderProc is never called if a GenericOutputAU is used instead of AUHAL default output device). I have not been able to get this simple graph to work. I do not see any errors when creating the graph and initializing the graph, and I have verified that the inputProc is being called for filling up the ringBuffer - but when I read the level of the MatrixMixer, the levels are always -758 (silence). I have posted my demo project on github in hopes I can find someone with CoreAudio expertise to help with this problem. I am willing to move this to DTS Code Level support if there is someone in DTS with CoreAudio experience. Notes: My App is not sandboxed in this test I have tried with and without hardened runtime with Audio Input checked The multichannel audio device I am using for testing is the Audient iD14 USB-C audio device. It supports 12 input channels and 6 output channels. All input channels have been tested and are working in Ableton Live and Logic Pro. Of particular interest, is that I can't even get the Apple CAPlayThrough demo to work on my system. I see no errors when creating the graph, but all I hear is silence. The MatrixMixerTest from the Apple documentation archives does work - but note, that that demo does not use Audio Input devices - it reads audio into the graph from an audio file. Link to github project page. Diagram of AUGraph for initial test (code that is on github) Once I get audio input level metering to work, my plan is to implement something like in Phase 2 below - with the purpose of capturing a stereo input stream, mixing to mono, and sending to lowpass, bandpass, hipass AUs - and will again use MatrixMixer for level monitoring of the levels out of each filter. I have no plans on passthough audio (sending actual audio out to devices) - I am simple monitoring input levels Diagram of ultimate scope - rendering audio levels of a stereo to mono stream after passing through various filters

I created a locked camera capture extension as explained in Apple's documentation. I'm trying to explore the possibilities of using a bluetooth peripheral from that extension - anybody knows if this is possible? The CBCentralManagerDelegate reports .unsupported in func centralManagerDidUpdateState, even if I have provided all the permissions in Info.plist.

Hello, I have a problem reading a 2D data matrix type code with a camera. In the application, I use AVFoundation to operate the camera and work with 2D codes, and in the vast majority there is no problem with loading. Nothing special. I originally thought it might be a problem in my code, but I got the same result when I tried with the Camera app integrated in IOS. It can be seen that only the LiveText API for text recognition worked. But I am attaching the code with which the camera has a problem, even though the code looks perfectly fine at first glance. A classic handheld 2D code reader will read the code just fine. Can someone please explain to me why the camera, which normally reads these codes at the speed of light, sometimes has a problem with the codes? Thank you [Personal Information Edited by Moderator]

My Camera app is repeatedly opening even though I am not taking any action to open it when I use iPhone. Today during a FaceTime call, the Camera app opened while the phone was unlocked without me touching anything. It didn’t end the FaceTime call, but just put the video on pause for the person I was speaking with. I force closed the Camera app, then it happened again a few minutes later. This has happened while using Google Maps and other apps as well, while the phone is unlocked. This is also happening while the phone is locked, just sitting on a table. All the sudden I look over and the screen is active showing the camera view. Today this has happened at least 20 times. I need to know how to stop it. I am on iOS 18.1 and enrolled in iOS 18 Public Beta. There are no pending software updates.

There are different microphones that can be connected via a 3.5-inch jack or via USB or via Bluetooth, the behavior is the same. There is a code that gets access to the microphone (connected to the 3.5-inch audio jack) and starts an audio capture session. At the same time, the microphone use icon starts to be displayed. The capture of the audio device (microphone) continues for a few seconds, then the session stops, the microphone use icon disappears, then there is a pause of a few seconds, and then a second attempt is made to access the same microphone and start an audio capture session. At the same time, the microphone use icon is displayed again. After a few seconds, access to the microphone stops and the audio capture session stops, after which the microphone access icon disappears. Next, we will try to perform the same actions, but after the first stop of access to the microphone, we will try to pull the microphone plug out of the connector and insert it back before trying to start the second session. In this case, the second attempt to access begins, the running part of the program does not return errors, but the microphone access icon is not displayed, and this is the problem. After the program is completed and restarted, this icon is displayed again. This problem is only the tip of the iceberg, since it manifests itself in the fact that it is not possible to record sound from the audio microphone after reconnecting the microphone until the program is restarted. Is this normal behavior of the AVFoundation framework? Is it possible to somehow make it so that after reconnecting the microphone, access to it occurs correctly and the usage indicator is displayed? What additional actions should the programmer perform in this case? Is there a description of this behavior somewhere in the documentation? Below is the code to demonstrate the described behavior. I am also attaching an example of the microphone usage indicator icon. Computer description: MacBook Pro 13-inch 2020 Intel Core i7 macOS Sequoia 15.1. #include <chrono> #include <condition_variable> #include <iostream> #include <mutex> #include <thread> #include <AVFoundation/AVFoundation.h> #include <Foundation/NSString.h> #include <Foundation/NSURL.h> AVCaptureSession* m_captureSession = nullptr; AVCaptureDeviceInput* m_audioInput = nullptr; AVCaptureAudioDataOutput* m_audioOutput = nullptr; std::condition_variable conditionVariable; std::mutex mutex; bool responseToAccessRequestReceived = false; void receiveResponse() { std::lock_guard<std::mutex> lock(mutex); responseToAccessRequestReceived = true; conditionVariable.notify_one(); } void waitForResponse() { std::unique_lock<std::mutex> lock(mutex); conditionVariable.wait(lock, [] { return responseToAccessRequestReceived; }); } void requestPermissions() { responseToAccessRequestReceived = false; [AVCaptureDevice requestAccessForMediaType:AVMediaTypeAudio completionHandler:^(BOOL granted) { const auto status = [AVCaptureDevice authorizationStatusForMediaType:AVMediaTypeAudio]; std::cout << "Request completion handler granted: " << (int)granted << ", status: " << status << std::endl; receiveResponse(); }]; waitForResponse(); } void timer(int timeSec) { for (auto timeRemaining = timeSec; timeRemaining > 0; --timeRemaining) { std::cout << "Timer, remaining time: " << timeRemaining << "s" << std::endl; std::this_thread::sleep_for(std::chrono::seconds(1)); } } bool updateAudioInput() { [m_captureSession beginConfiguration]; if (m_audioOutput) { AVCaptureConnection *lastConnection = [m_audioOutput connectionWithMediaType:AVMediaTypeAudio]; [m_captureSession removeConnection:lastConnection]; } if (m_audioInput) { [m_captureSession removeInput:m_audioInput]; [m_audioInput release]; m_audioInput = nullptr; } AVCaptureDevice* audioInputDevice = [AVCaptureDevice deviceWithUniqueID: [NSString stringWithUTF8String: "BuiltInHeadphoneInputDevice"]]; if (!audioInputDevice) { std::cout << "Error input audio device creating" << std::endl; return false; } // m_audioInput = [AVCaptureDeviceInput deviceInputWithDevice:audioInputDevice error:nil]; // NSError *error = nil; NSError *error = [[NSError alloc] init]; m_audioInput = [AVCaptureDeviceInput deviceInputWithDevice:audioInputDevice error:&error]; if (error) { const auto code = [error code]; const auto domain = [error domain]; const char* domainC = domain ? [domain UTF8String] : nullptr; std::cout << code << " " << domainC << std::endl; } if (m_audioInput && [m_captureSession canAddInput:m_audioInput]) { [m_audioInput retain]; [m_captureSession addInput:m_audioInput]; } else { std::cout << "Failed to create audio device input" << std::endl; return false; } if (!m_audioOutput) { m_audioOutput = [[AVCaptureAudioDataOutput alloc] init]; if (m_audioOutput && [m_captureSession canAddOutput:m_audioOutput]) { [m_captureSession addOutput:m_audioOutput]; } else { std::cout << "Failed to add audio output" << std::endl; return false; } } [m_captureSession commitConfiguration]; return true; } void start() { std::cout << "Starting..." << std::endl; const bool updatingResult = updateAudioInput(); if (!updatingResult) { std::cout << "Error, while updating audio input" << std::endl; return; } [m_captureSession startRunning]; } void stop() { std::cout << "Stopping..." << std::endl; [m_captureSession stopRunning]; } int main() { requestPermissions(); m_captureSession = [[AVCaptureSession alloc] init]; start(); timer(5); stop(); timer(10); start(); timer(5); stop(); }