I have integrated the ShazamKit SDK into my iOS app and would like to implement the same functionality in my Android app. My question is: Can I use the Android version of the ShazamKit SDK for commercial purposes? After extensive research, I could not find any official information regarding the license of the Android version of the ShazamKit SDK. Could you please provide a formal license statement?

Hi all! I have been experiencing some issues when using the AVAudioEngine to play audio and record input while doing a voice chat (through the PTT Interface). I noticed if I connect any players to the AudioGraph OR call start that the audio session becomes active (this is on iOS). I don't see anything in the docs or the header files in the AVFoundation, but is it possible that calling the stop method on an engine deactivates the audio session too? In a normal app this behavior seems logical, but when using PTT all activation and deactivation of the audio session must go through the framework and its delegate methods. The issue I am debugging is that when the engine with the input node tapped gets stopped, and there is a gap between the input and when the server replies with inbound audio to be played and something seems to be getting the hardware/audio session into a jammed state. Thanks for any feedback and/or confirmation on this behavior!

Hello, I'm wondering how to capture 24MP photos. I'm currently testing on an iPhone 16 Pro Max. By default, the device's activeFormat supports 24MP (photo dimensions: {4032x3024, 5712x4284}). For the photoOutput, I'm setting the maxPhotoDimensions to videoDevice.activeFormat.supportedMaxPhotoDimensions.lastObject, and setting MaxPhotoQualityPrioritization to quality. When capturing, I'm applying the same maxPhotoDimensions and photoQualityPrioritization settings from the photoOutput directly to the AVCapturePhotoSettings. What could be the issue? // Objective-C // setup [self.photoOutput setMaxPhotoQualityPrioritization:AVCapturePhotoQualityPrioritizationQuality]; CMVideoDimensions maxPhotoDimensions = [(NSValue *)videoDevice.activeFormat.supportedMaxPhotoDimensions.lastObject CMVideoDimensionsValue]; [self.photoOutput setMaxPhotoDimensions:maxPhotoDimensions]; // capturing AVCapturePhotoSettings *photoSettings = [AVCapturePhotoSettings photoSettings]; photoSettings.maxPhotoDimensions = self.photoOutput.maxPhotoDimensions; photoSettings.photoQualityPrioritization = self.photoOutput.maxPhotoQualityPrioritization; [self.photoOutput capturePhotoWithSettings:photoSettings delegate:photoCaptureDelegate]; ...

I am trying to use AVAudioEngine for recording and playing for a voice chat kind of app, but when the speaker plays any audio while recording, the recording take the speaker audio as input. I want to filter that out. Are there any suggestions for the swift code

I need to implement a solution through an API or custom driver to completely block out the built-in speakers and microphone of Mac, because I need other apps to use specified external devices as audio input and output. Is there a way to achieve this requirement? What I mean is that even in system preferences, it should not be possible to choose the built-in microphone and speakers; only my external device can be used.

I'm able to get text to speech to audio file using the following code for iOS 12 iPhone 8 to create a car file: audioFile = try AVAudioFile( forWriting: saveToURL, settings: pcmBuffer.format.settings, commonFormat: .pcmFormatInt16, interleaved: false) where pcmBuffer.format.settings is: [AVAudioFileTypeKey: kAudioFileMP3Type, AVSampleRateKey: 48000, AVEncoderBitRateKey: 128000, AVNumberOfChannelsKey: 2, AVFormatIDKey: kAudioFormatLinearPCM] However, this code does not work when I run the app in iOS 18 on iPhone 13 Pro Max. The audio file is created, but it doesn't sound right. It has a lot of static and it seems the speech is very low pitch. Can anyone give me a hint or an answer?

On some devices, loadFileRepresentation(forTypeIdentifier: completionHandler) take a loong time(about two minute) to callback result for some large video(about 200 MB, take by device camera). environment: Model: iPhone 12 Model Number: MGGM3CH/A iOS Version: 18.3.2 PHPickerResult.NSItemProvider.loadFileRepresentation() // import PhotosUI func picker(_ picker: PHPickerViewController, didFinishPicking results: [PHPickerResult]) { picker.dismiss(animated: true, completion: nil) guard let provider = results.last?.itemProvider else { return } guard provider.hasItemConformingToTypeIdentifier(UTType.movie.identifier) else { return } Task { provider.loadFileRepresentation(forTypeIdentifier: UTType.movie.identifier) { url, error in guard let url = url else { return } // Do some stuff... } } } ps: I also try some other function, eg: provide.loadItem(forTypeIdentifier:), but not work too.

Hello All, It seems that it's "very easy" (😬) to implement a little Swift code inside the prepared AU using Xcode 16.2 on Sequoia 15.1.1 and a Mac Studio M1 Ultra, but my issue is that I finally don't know... where. The documentation says that I've to find the AudioUnitViewController.swift file and then modify the render block : audioUnit.renderBlock = { (numFrames, ioData) in // Process audio here } in the Xcode project automatically generated, but I didn't find such a file... If somebody can help me in showing where is the file to be modified, I'll be very grateful ! Thank you very much. J

On some devices, when i select the same media multiple times, the data by` loadFileRepresentation(forTypeIdentifier: completionHandler) ` returned is different(data.count is not equal). environment: * Model: iPhone 12 * Model Number: MGGM3CH/A * iOS Version: 18.3.2 ```Swift // import PhotosUI func picker(_ picker: PHPickerViewController, didFinishPicking results: [PHPickerResult]) { picker.dismiss(animated: true, completion: nil) guard let provider = results.last?.itemProvider else { return } guard provider.hasItemConformingToTypeIdentifier(UTType.movie.identifier) else { return } Task { provider.loadFileRepresentation(forTypeIdentifier: UTType.movie.identifier) { url, error in guard let url = url else { return } if let data = try? Data(contentsOf: url) { print("data count is: \(data.count)") } } } } ``` ps: I also try some other function, eg: ` provide.loadItem(forTypeIdentifier:)`, but not work too.

I’ve been researching how to achieve a recording playback effect in iOS similar to the hands-free calling effect in the system’s phone app. How can this be implemented? I tried using the voice chat recording method, but found that the volume of the speaker output is too low. How should this issue be addressed? I couldn’t find a suitable API. Could you provide me with some documentation or sample code? Thank you.

How can I use my RGB Curve points: let redCurve = [CIVector(x: 0, y: 0), CIVector(x: 0.235, y: 0.152), CIVector(x: 0.5, y: 0.5), CIVector(x: 1, y: 1)] let greenCurve = [CIVector(x: 0, y: 0), CIVector(x: 0.247, y: 0.196), CIVector(x: 0.5, y: 0.5), CIVector(x: 1, y: 1)] let blueCurve = [CIVector(x: 0, y: 0), CIVector(x: 0.235, y: 0.184), CIVector(x: 0.466, y: 0.466), CIVector(x: 1, y: 1)] in colorCurvesFilter which I've found in Apple Docs: func colorCurves(inputImage: CIImage) -> CIImage { let colorCurvesEffect = CIFilter.colorCurves() colorCurvesEffect.inputImage = inputImage colorCurvesEffect.curvesDomain = CIVector(x: 0, y: 1) colorCurvesEffect.curvesData = Data( bytes: [Float32]([ 0.0,0.0,0.0, 0.8,0.8,0.8, 1.0,1.0,1.0 ]), count: 36) colorCurvesEffect.colorSpace = CGColorSpaceCreateDeviceRGB() return colorCurvesEffect.outputImage! }

I have used AVQueuePlayer in my music app to play sequence of audios from a remote server, this how I have defined things my player in my ViewModel Variables private var cancellables = Set() private let audioSession = AVAudioSession.sharedInstance() private var avQueuePlayer: AVQueuePlayer? @Published var playbackSpeed: Float = 1.0 before starting playback, I am making sure that audio session is set properly, the code snippet used for that is do { try audioSession.setCategory(.playback, mode: .default, options: []) try audioSession.setActive(true, options: []) } catch { return } and this is the function I am using to update playback speed func updatePlaybackSpeed(_ newSpeed: Float){ if newSpeed > 0.0, newSpeed <= 2.0{ playbackSpeed = newSpeed avQueuePlayer?.rate = newSpeed print("requested speed is (newSpeed) and actual speed is (String(describing: avQueuePlayer?.rate))") } } sometimes whatever speed is set, player seems to play at the same speed as it was set, e.g. Once I got "requested speed is 1.5 and actual speed is 1.5", and player also seemed to play at the speed of 1.5 but another time I got "requested speed is 2.0 and actual speed is 2.0", but player still seemed to play at the speed of 1.0 to observe changes in rate, I used this **private func observeRateChanges() { guard let avQueuePlayer = self.avQueuePlayer else { return } NotificationCenter.default.publisher(for: AVQueuePlayer.rateDidChangeNotification, object: avQueuePlayer) .compactMap { $0.userInfo?[AVPlayer.rateDidChangeReasonKey] as? AVPlayer.RateDidChangeReason } .sink { reason in switch reason { case .appBackgrounded: print("The app transitioned to the background.") case .audioSessionInterrupted: print("The system interrupts the app’s audio session.") case .setRateCalled: print("The app set the player’s rate.") case .setRateFailed: print("An attempt to change the player’s rate failed.") default: break } } .store(in: &cancellables) }** when rate was set properly, I got this "The app set the player’s rate." from the above function, but when it wasn't, I got this "An attempt to change the player’s rate failed.," now I am not able to understand why rate is not being set, and if it gave "requested speed is 2.0 and actual speed is 2.0" from updatePlaybackSpeed function, why does the player seems to play with the speed of 1.0?

We are developing an apple music app on phone, the developed web works fine on chrome, but when i load it on webivew on my phone, i can't play the first song, We doubt that the drm init, key exchange, session creation was on the music.play() function, while we trigger the play, the drm or session was not ok for play a real song, so it got an error so we may wanna know: what about the realative process of drm, key, session, etc in the play() function? are there some state detect function to show weather the drm is ok？

If new photo is added to library and app is not running in foreground or was not opened after the new photo was added but the app is having full access to gallery, can it access, read the new photo - If the app is not specifically a cloud syncing app, can it have this attached function, suppose it is a game app or beauty camera app?

Hi all, with my app ScreenFloat, you can record your screen, along with system- and microphone audio. Those two audio feeds are recorded into separate audio tracks in order to individually remove or edit them later on. Now, these recordings you create with ScreenFloat can be drag-and-dropped to other apps instantly. So far, so good, but some apps, like Slack, or VLC, or even websites like YouTube, do not play back multiple audio tracks, just one. So what I'm trying to do is, on dragging the video recording file out of ScreenFloat, instantly baking together the two individual audio tracks into one, and offering that new file as the drag and drop file, so that all audio is played in the target app. But it's slow. I mean, it's actually quite fast, but for drag and drop, it's slow. My approach is this: "Bake together" the two audio tracks into a one-track m4a audio file using AVMutableAudioMix and AVAssetExportSession Take the video track, add the new audio file as an audio track to it, and render that out using AVAssetExportSession For a quick benchmark, a 3'40'' movie, step 1 takes ~1.7 seconds, and step two adds another ~1.5 seconds, so we're at ~3.2 seconds. That's an eternity for a drag and drop, where the user might cancel if there's no immediate feedback. I could also do it in one step, but then I couldn't use the AV*Passthrough preset, and that makes it take around 32 seconds then, because I assume it touches the video data (which is unnecessary in this case, so I think the two-step approach here is the fastest). So, my question is, is there a faster way? The best idea I can come up with right now is, when initially recording the screen with system- and microphone audio as separate tracks, to also record both of them into a third, muted, "hidden" track I could use later on, basically eliminating the need for step one and just ripping the two single audio tracks out of the movie and only have the video and the "hidden" track (then unmuted), but I'd still have a ~1.5 second delay there. Also, there's the processing and data overhead (basically doubling the movie's audio data). All this would be great for an export operation (where one expects it to take a little time), but for a drag-and-drop operation, it's not ideal. I've discarded the idea of doing a promise file drag, because many apps do not accept those, and I want to keep wide compatibility with all sorts of apps. I'd appreciate any ideas or pointers. Thank you kindly, Matthias

I'm trying to implement Ambisonic B-Format audio playback on Vision Pro with head tracking. So far audio plays, head tracking works, and the sound appears to be stereo. The problem is that it is not a proper binaural playback when compared to playing back the audiofile with a DAW. Has anyone successfully implemented B-Format playback on Vision Pro? Any suggestions on my current implementation: func playAmbiAudioForum() async { do { try AVAudioSession.sharedInstance().setCategory(.playback) try AVAudioSession.sharedInstance().setActive(true) // AudioFile laoding/preperation guard let testFileURL = Bundle.main.url(forResource: "audiofile", withExtension: "wav") else { print("Test file not found") return } let audioFile = try AVAudioFile(forReading: testFileURL) let audioFileFormat = audioFile.fileFormat // create AVAudioFormat with Ambisonics B Format guard let layout = AVAudioChannelLayout(layoutTag: kAudioChannelLayoutTag_Ambisonic_B_Format) else { print("layout failed") return } let format = AVAudioFormat( commonFormat: audioFile.processingFormat.commonFormat, sampleRate: audioFile.fileFormat.sampleRate, interleaved: false, channelLayout: layout ) // write audiofile to buffer guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: UInt32(audioFile.length)) else { print("buffer failed") return } try audioFile.read(into: buffer) playerNode.renderingAlgorithm = .HRTF // connecting nodes audioEngine.attach(playerNode) audioEngine.connect(playerNode, to: audioEngine.outputNode, format: format) audioEngine.prepare() playerNode.scheduleBuffer(buffer, at: nil) { print("File finished playing") } try audioEngine.start() playerNode.play() } catch { print("Setup error:", error) } }

I am working on a project for macOS where I am taking an AVCaptureSession's CVPixelBuffer and I need to convert it into a MTLTexture for rendering. On macOS the pixel format is 2vuy, there does not seem to be a clear format conversion while converting to a metal texture. I have been able to convert it to a texture but the color space seems to be off as it is rendering distorted colors with a double image. I believe 2vuy is a single pane color space and I have tried to account for that, but I am unaware of what is off. I have attached The CVPixelBuffer and The distorted MTLTexture along with a laundry list of errors. On iOS my conversions are fine, it is only the macOS 2vuy pixel format that seems to have issues. My code for the conversion is also attached. If there are any suggestions or guidance on how to properly convert a 2vuy CVPixelBuffer to a MTLTexture I would greatly appreciate it. Many Thanks Conversion_Logs.txt ConversionCode.swift

I'm trying to load Music Kit on the server with solid js. I can confirm that my implementation has been sufficient to return authentication tokens and for MusicKit.isAuthorized to return true. My issue is that if I reload the page, it only succeeds intermittently (perhaps 25% of the time?). My question is - what is wrong with my implementation? Removing the async keyword ensures it loads every time but playing and queuing music no longer works. I'm currently assuming this is an SSR issue but the docs haven't explicitly specified this isn't possible. I have the following boilerplate: export default createHandler( () => ( <StartServer document={({ assets, children, scripts }) => { return ( <html lang="en"> <head> <meta name="apple-music-developer-token" content={authResult.token} /> <meta name="apple-music-app-name" content="app name" /> <meta name="apple-music-app-build" content="1978.4.1" /> {assets} <script src="https://js-cdn.music.apple.com/musickit/v3/musickit.js" async /> </head> <body> <div id="app">{children}</div> {scripts} </body> </html> ) }} /> )) When I first load my app, I'll encounter: musickit.js:13 Uncaught TypeError: Cannot read properties of undefined (reading 'node') at musickit.js:13:10194 at musickit.js:13:140 at musickit.js:13:209 The intermittence signals an issue relating to the async keyword. An expansion on this issue can be found here.

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?

Everytime I put my AirPods in and connect them to my phone or my Mac or my iPad since the iOS 18.3 update on my devices they’ve been disconnecting without reason, pausing songs I’m in the middle of playing, and only partially reconnecting in one pod and it’s getting really frustrating