Does an artist similarity station broaden selection variety compared to a song similarity station? You don't have to answer if it is against nondisclosure terms.

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?

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?

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.

I found that when the development tool above Xcode16 ran my app, I opened the suspended inscription function, and then opened the system camera, the content in the suspended window would not be displayed, and the suspended window would have a black screen. However, this phenomenon does not appear on Xcode15.4 development tools, it is the same code, I do not know why

Hi I'm working on a project that require video frame PTS to be consistent between original video and a transcoded one. It's working fairly well on regular mp4, however if I set preferredOutputSegmentInterval to have generate a fMP4 output, even I specified the initialSegmentStartTime as 0, it always add one frame pts offset to all frames. For example: if I use the code sample provided by Apple: https://developer.apple.com/videos/play/wwdc2020/10011/?time=406, useffprobe -select_streams v:0 -show_entries packet=pts_time -of csv ~/Downloads/fmp4/prog_index.m3u8 to display the pts of the output, it doesn't start from 0, but has some one frame pts offset. I also tried open with MP4Box, it also shows the first frames dts and cts are not start from 0. However, if I use AVAssetReader to read the same output video, and get the PTS from 1st frame, it's returning 0. So I can't use it to calculate the pts difference between 2 videos neither. Can I get some help to understand why there is difference between AVAssetWriter/Reader fMP4's pts and others like ffprobe?

We have the necessary background recording entitlements, and for many users... do not run into any issues. However, there is a subset of users that routinely get recordings ending.. we have narrowed this down and believe it to be the work of the watch dog. First we removed the entire view hierarchy when app is backgrounded. There is just 'Text("Recording")' This got the CPU usage in profiler down to 0%. We saw massive improvements to recording success rate. We walked away assuming that was enough. However we are still seeing the same sort of crashes. All in the background. We're using Observation to drive audio state changes to a Live Activity. Are those Observations causing the problem? Why doesn't apple provide a better API to background audio? The internet is full of weird issues https://stackoverflow.com/questions/76010213/why-is-my-react-native-app-sometimes-terminated-in-the-background-while-tracking https://stackoverflow.com/questions/71656047/why-is-my-react-native-app-terminating-in-the-background-while-recording-ios-r https://github.com/expo/expo/issues/16807 This is such a terrible user experience. And we have very little visibility into what is happening and why. No where in apple documentation states that in order for background recording to work, the app can only be 'Text("Recording")' It does not outline a CPU or memory threshold. It just kills us.

I am using an AVAudioPlayer to play a "tick" sound once per second in a SwiftUI app. When running the app on an iPhone 16 (18.2.1) the tick sounds increase in volume after a few seconds. This does not happen in the simulator nor on an iPhone SE 2020 (18.1.1).

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)

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!

In our Apple TV application, we are using the native AVPlayer for live playback functionality. During live restart playback, we intermittently encounter an error when the playback timeline approaches the actual live event end time. Error: The operation couldn’t be completed. (CoreMediaErrorDomain error -16839 - Unable to get playlist before long download timer) / Failure reason: Scenario: The live event is scheduled from 7:00 AM to 8:00 AM. Restart playback begins at 7:20 AM, allowing the user to watch the event from the start while the live stream continues in real-time. As the restart playback timeline approaches the actual event end time (8:00 AM), AVPlayer displays an error, and playback continues in the background.

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 prefer to use the album fetched from the library instead of the catalog since this is faster. If doing so, how can I check if all tracks of an album are added to the library. In this case I'd like to fetch the catalog version or throw an error (for example when offline). Using .with(.tracks) on the library album fetches the tracks added to the library. The trackCount property is referring to the tracks that can be fetched from the library. The isComplete property is always nil when fetching from the library. One possible way is checking the trackNumber and discCount properties. However this only detects that not all tracks of an album are added to the library if there is a song not added ahead of one that is. I'd like to be able to handle this edge case as well. Is there currently a way to do this? I'd prefer to not rely on the apple music catalog for this since this is supposed to work offline as well. Fetching and storing all trackIDs when connected and later comparing against these would work, but this would potentially mean storing tens of thousands of track ids. Thank you

Hello Apple Developer Community, We are developing a music management platform for restaurants and cafes in Saudi Arabia. Our app enables businesses to schedule playlists and allows visitors to request songs via barcodes. Music playback is powered by Apple Music, and users must have their own Apple Music subscriptions to access the music. Our service charges a monthly subscription fee for these management features, not for music access itself. Project Overview and MusicKit Role Our app integrates MusicKit to leverage Apple Music’s catalog and playback capabilities. Users log in with their Apple Music accounts, ensuring they have an active subscription for music playback. Our platform’s value lies in its tools—playlist scheduling and song requests—which are built on top of MusicKit’s APIs. We offer these features exclusively in Saudi Arabia. Legal Context in Saudi Arabia In Saudi Arabia, to our understanding, no special licenses are required for playing music in commercial venues like restaurants and cafes. This means our clients can use Apple Music subscriptions for playback without additional performance rights licenses. While this aligns with local laws, we recognize that Apple’s global policies may impose stricter requirements, prompting our need for clarification. Subscription Model and Monetization Concerns We charge a monthly subscription fee for access to our app’s features (e.g., scheduling playlists and managing song requests). This fee is separate from the Apple Music subscription, which users must maintain for playback. However, Apple’s MusicKit terms state: "You agree not to require payment for or indirectly monetize access to the Apple Music service." We’re concerned whether our subscription model might be interpreted as indirectly monetizing Apple Music access, given its reliance on MusicKit for functionality. Scheduling Feature and Synchronization Rights Our app allows businesses to schedule playlists for general time slots (e.g., “play this playlist from 6 PM to 8 PM”). It does not support precise scheduling, such as playing a specific song at an exact moment (e.g., “play this song at 7:30 PM”). Apple’s guidelines mention that “deeper or more complex music integration” may require additional licenses, like synchronization rights. We’re unsure if our general scheduling feature crosses this threshold or remains within MusicKit’s standard usage. Questions for Clarification We’d greatly appreciate expert input on the following: Monetization: Does our subscription fee for management features (scheduling and song requests) violate Apple’s policy against indirectly monetizing Apple Music access? Local Context: Given that Saudi Arabia requires no additional licenses for commercial music playback, does this impact our compliance with Apple’s global terms? Scheduling: Does our playlist scheduling for general time slots (not exact moments) fall within MusicKit’s permitted scope, or does it require further licensing? Thank you in advance for any insights or guidance to ensure our app aligns with Apple’s policies!

I have an app that has a WKWebView for watching YouTube videos. When the videos are windowed the audio seems fine, positionally as well. All perfectly. When I fullscreen the video and it goes into the native visionOS video player the audio messes up. It will suddenly sound like it is in your ears, or maybe even just one ear channel, or the position will be wrong. It might be fine for a moment but the second I touch the controls or move the window the sound jumps across the room, away from the window, or switches to stereo. Sometimes exiting windows entirely you will still hear the videos playing. Even if you open the window back up and go to another screen and open another video, now you hear 2 videos playing at the same time with no way to stop the first one in the background, requiring to force restart the app. It is all sorts of glitchy. I haven't the slightest clue what is happening here. I am strongly feeling this is a visionOS bug. I tried using AVAudioSession to change some of the sound settings, and that makes zero difference in behavior. Multiple testers have also reported this behavior and it has been seen on both visionOS 2.3 and 2.4 betas. Thanks for the help! This is driving me mad! It is extremely consistent behavior!

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

Hi there! We have a suite of AudioUnit v2 plugins that have been shipped for some time as aufx plugins, and we are looking into MIDI-related platform upgrades, so we need a way to update these plugins to request MIDI from Logic (and other AU hosts) but avoid changing our AU type and subtype so we don't break existing sessions. Any ideas on how we can do this?

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.

After 18.2 IOS update, videos are not playing in Netflix, Amazon Prime and youtube

I have spent a long time refactoring lots of older Swift code to compile without error in Swift 6. The app is a v3 audio unit host and audio unit. Having installed Sonoma and XCode 16 I compile the code using Swift 6 and it compiles and runs without any warnings or errors. My host will load my AU no problem. LOGIC PRO is still the ONLY audio unit host that will load native Mac V3 audio units and so I like to test my code using Logic. In Sonoma with XCode 16... My AU passes the most stringent AUVAL tests both in terminal and Logic pro. If I compile the AU source in Swift 5 Logic will see the AU, load it and run it without problems. But when I compile the AU in Swift 6 Logic sees the AU, will scan it and verify it passes the tests but will not load the AU. In XCode I see a log message that a "helper application failed to run" but the debugger never connects to the AU and I don't think Logic even gets as far as instantiating the AU. So... what is causing this? I'm stumped.. Developing AUv3 is a brain-aching maze of undocumented hurdles and I'm hoping someone might have found a solution for this one. Meanwhile I guess my only option is to continue using the Swift 5 compiler. (appending a little note just to mention that all the DSP code is written in C/C++, Swift is used mainly for the user interface and also does some offline thready work )