hi,
i need to read wether the transport is playing or stopped but my current method that works for vst does not work for au.
is there a lpx resource available for developers anywhere?
if (auto* playHead = processor->getPlayHead())
{
juce::AudioPlayHead::CurrentPositionInfo posInfo;
if (playHead->getCurrentPosition(posInfo))
{
bool isCurrentlyPlaying = posInfo.isPlaying;
if (isCurrentlyPlaying != wasTransportPlaying)
{
if (isCurrentlyPlaying)
{
wasTransportPlaying = isCurrentlyPlaying;
startAllTimers();
}
else
{
wasTransportPlaying = isCurrentlyPlaying;
stopAllTimers();
}
}
}
}
thanks :)
Audio
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The presentation "create audio drivers with DriverKit" from WWDC 2021 demonstrates how to use a dext to implement a virtual audio driver. It also says " If a virtual audio driver or device is all that is needed, the audio server plug-in driver model should continue to be used".
Indeed, in AudioDriverKit/AudioDriverKitTypes.h, there is no IOUserAudioTransportType Virtual, although CoreAudio/AudioHardwareBase.h includes kAudioDeviceTransportTypeVirtual.
For one of our products, we require virtual devices to implement a software loopback "cable". We've implemented this using the "traditional" HAL plugin, and as a proof-of-concept, also using a dext. In the dext, I tried setting the transport type to 'virt', which seems to only have the effect of changing the icon shown in Audio Midi Setup.
HAL plugins require an installer, and the installer has to kill coreaudiod in a post-install script. You have to turn off SIP to debug them. Just like AudioDriverKit drivers, they are out-of-process and run in a process not owned by the hosting app. Our HAL plugin's interface is property based; we had to write a lot of boiler-plate code to implement required properties. Writing an AudioDriverKit driver is in most respects easier - a lot of the scaffolding is implemented in the base driver, which we only alter where required. Debugging and installation is much easier.
The dext works just fine, as far as we can ascertain, just as well as a HAL plugin.
So, my question is - is the advice to use a HAL plugin for a virtual device still correct in 2025? And if so, what's the objection? We'd really prefer to ship the AudioDriverKit virtual audio device.
Let's consider the following code.
I've created an actor that loads a list of .mp3 files from a Bundle and then makes it available for audio reproduction.
Unfortunately, I'm experiencing a memory leak.
At the play method.
player.play()
From Instruments I get
_malloc_type_malloc_outlined libsystem_malloc.dylib
start_wqthread libsystem_pthread.dylib
private actor AudioActor {
enum Failure: Error {
case soundsNotLoaded([AudioPlayerClient.Sound: Error])
}
enum Player {
case music(AVAudioPlayer)
}
var players: [Sound: Player] = [:]
let bundles: [Bundle]
init(bundles: UncheckedSendable<[Bundle]>) {
self.bundles = bundles.wrappedValue
}
func load(sounds: [Sound]) throws {
try AVAudioSession.sharedInstance().setActive(true, options: [])
var errors: [Sound: Error] = [:]
for sound in sounds {
guard let url = bundle.url(forResource: sound.name, withExtension: "mp3")
else { continue }
do {
self.players[sound] = try .music(AVAudioPlayer(contentsOf: url))
} catch {
errors[sound] = error
}
}
guard errors.isEmpty
else { throw Failure.soundsNotLoaded(errors) }
}
func play(sound: Sound, loops: Int?) throws {
guard let player = self.players[sound]
else { return }
switch player {
case let .music(player):
player.numberOfLoops = loops ?? -1
player.play()
}
}
func stop(sound: Sound) throws {
guard let player = self.players[sound]
else { throw Failure.soundsNotLoaded([:]) }
switch player {
case let .music(player):
player.stop()
}
}
}
Feature Request: Long-Lived Access to Personal Apple Music Data
Use Case Summary
I'm developing a personal portfolio website (using Nuxt) and want to display information from my own Apple Music library - showcasing personal playlists, recently played tracks, or a read-only "now playing" widget. This is purely for personal use on my website and doesn't require other users to log in.
With Spotify's API, implementing this was straightforward thanks to automatic token refresh. I want a similarly seamless integration with Apple Music.
Challenge with MusicKit and Music User Tokens
Apple Music API requirements
Apple's Music API requires a valid Music User Token (MUT) for requests involving personal library data. Beyond the Apple Developer Token, you must obtain a user-specific token via MusicKit authentication to access your own library playlists, play history, or current playback status.
Token expiration and manual renewal
Music User Tokens expire after approximately 6 months without any mechanism to automatically refresh or renew them - unlike typical OAuth flows that provide refresh tokens. Apple's guidance suggests the device (e.g., iPhone) is responsible for obtaining new user tokens when old ones expire. This works for interactive apps on Apple devices but fails in server-side or long-lived web contexts like a personal website widget.
Impact on personal projects
Displaying Apple Music data on a public-facing site becomes difficult. I would need to periodically re-authenticate through the MusicKit JS flow every few months just to keep a widget alive. Embedding credentials in a public site is insecure, and manual token refreshing is cumbersome and easy to forget.
Comparison to Spotify's Token Model
Spotify's API offers a developer-friendly authentication model. Their OAuth flow provides a Refresh Token that applications can use to obtain new access tokens automatically without requiring user re-authorization. This means a personal app can maintain continuous access to a user's Spotify data for extended periods until access is revoked.
When building a similar feature with Spotify, this automatic token renewal was crucial. I could safely store the refresh token on my server and have my app periodically update the access token. Many developers have created public-facing widgets showing currently playing tracks on blogs or GitHub profiles using this model. Unfortunately, Apple Music's API lacks an equivalent capability, putting it at a disadvantage for personal projects.
Proposed Solutions
I request Apple's consideration for one of these enhancements:
Provide a mechanism to refresh or extend a Music User Token programmatically for server-side applications. This could be an OAuth-style refresh token issued alongside the MUT, or a dedicated endpoint to exchange an expired MUT for a new one. This would enable renewal without a full user re-auth/login each time.
Allow developers to access their own Apple Music library data with just the long-lived Developer Token. Apple could permit GET requests to personal library endpoints using the Developer Token alone, or a special token tied to the developer's Apple ID. This access would be read-only - no ability to modify the library, purely for retrieving data. It could be an opt-in feature in the Apple Developer account settings.
Either solution would significantly improve the developer experience for Apple Music API in personal projects.
Security and Privacy Considerations
This request is not about accessing others' data or creating privacy loopholes - it's about empowering an Apple Music subscriber to access their own information more conveniently. The proposed options respect privacy principles:
The data accessed is only what the user already has access to - their own playlists, library items, or playback status.
An automatic token refresh can be designed securely (revocable tokens bound to a single account with no increase in permissions).
Read-only developer token access could be restricted to non-sensitive data and require explicit opt-in.
Conclusion
I request an improvement to Apple Music's developer experience through either (1) an automatic Music User Token refresh mechanism, or (2) a provision for read-only personal library access using a Developer Token. This would bring Apple Music integration capabilities closer to parity with services like Spotify for personal projects.
I ask Apple's Developer Relations and the Apple Music API team to consider this feature request. If there are existing best practices or workarounds with current APIs, I would appreciate guidance.
I invite feedback from Apple or other developers. Are there known patterns for maintaining an Apple Music user token for server-side applications, or any plans to support non-interactive use cases? Any advice is welcome.
Thank you for your consideration. I look forward to integrating Apple Music into my personal site as smoothly as with other services, and believe many developers would benefit from this added flexibility.
Sources:
User Authentication for MusicKit - Requirements for Music User Tokens
StackOverflow: Do Apple Music User Tokens expire? - Confirmation of 6-month expiration
MetaBrainz GSoC Blog - Documentation of MusicKit authentication limitations
Apple Developer Forums - Information on token renewal behavior
Spotify for Developers - Documentation on refresh token mechanism
Topic:
Media Technologies
SubTopic:
Audio
Tags:
Apple Music API
MusicKit
MusicKit JS
Apple Music Feed
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!
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 am trying to stream audio from local filesystem.
For that, I am trying to use an AVAssetResourceLoaderDelegate for an AVURLAsset. However, Content-Length is not known at the start. To overcome this, I tried several methods:
Set content length as nil, in the AVAssetResourceLoadingContentInformationRequest
Set content length to -1, in the ContentInformationRequest
Both of these cause the AVPlayerItem to fail with an error.
I also tried setting Content-Length as INT_MAX, and setting a renewalDate = Date(timeIntervalSinceNow: 5). However, that seems to be buggy. Even after updating the Content-Length to the correct value (e.g. X bytes) and finishing that loading request, the resource loader keeps getting requests with requestedOffset = X with dataRequest.requestsAllDataToEndOfResource = true. These requests keep coming indefinitely, and as a result it seems that the next item in the queue does not get played. Also, .AVPlayerItemDidPlayToEndTime notification does not get called.
I wanted to check if this is an expected behavior or is there a bug in this implementation. Also, what is the recommended way to stream audio of unknown initial length from local file system?
Thanks!
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
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?
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
iPhoneやiPadにおいて、画面上のボタンなどをタップした際に、特定の楽器音を発音させる方法をご存知の方いらっしゃいませんか?
現在音楽学習アプリを作成途中で、画面上の鍵盤や指板のボタン状のframeに、単音又は和音を割当て発音させる事を考えております
SwiftUIのcodeのみで実現できないでしょうか
嘗て、MIDIのlevel1の楽器の発音機能があった様に記憶していますが、現在のOS上では同様の機能を実装してないように思えます
皆様のお知恵をお貸しください
Hello,
I have an iOS app that is recording audio that is working fine on iPads/iPhones. It asks for microphone permission and after that recording works.
I installed the same app on my M3 MacBook via TestFlight since iPad apps are supposed to work without a change that way. The app starts fine and everything, but it never asks for Microphone permission, so I can't record.
Do I need to do something to make this happen (this is not macCatalyst, its running the arm64 iPhone binary on macOS)
thanks
I'm encountering errors while using AVAudioEngine with voice processing enabled (setVoiceProcessingEnabled(true)) in scenarios where the input and output audio devices are not the same. This issue arises specifically with mismatched devices, preventing the application from functioning as expected.
Works: Paired devices (e.g., MacBook Pro mic → MacBook Pro speakers)
Fails: Mismatched devices (e.g., AirPods mic → MacBook Pro speakers)
When using paired input and output devices:
The setup works as expected.
Example: MacBook Pro microphone → MacBook Pro speakers.
When using mismatched devices:
AVAudioEngine setup fails during aggregate device construction.
Example: AirPods microphone → MacBook Pro speakers.
Error logs indicate a channel count mismatch.
Here are the partial logs. Due to the content limit, I cannot post the entire logs.
AUVPAggregate.cpp:1000 client-side input and output formats do not match (err=-10875)
AUVPAggregate.cpp:1036 err=-10875
AVAEInternal.h:109 [AVAudioEngineGraph.mm:1344:Initialize: (err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)): error -10875
AggregateDevice.mm:329 Failed expectation of constructed aggregate (312): mInput.streamChannelCounts == inputStreamChannelCounts
AggregateDevice.mm:331 Failed expectation of constructed aggregate (312): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U)
AggregateDevice.mm:182 error fetching default pair
AggregateDevice.mm:329 Failed expectation of constructed aggregate (336): mInput.streamChannelCounts == inputStreamChannelCounts
AggregateDevice.mm:331 Failed expectation of constructed aggregate (336): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U)
AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702]
AudioHardware-mac-imp.cpp:3484 AudioDeviceSetProperty: no device with given ID
AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702]
AggregateDevice.mm:182 error fetching default pair
AggregateDevice.mm:329 Failed expectation of constructed aggregate (348): mInput.streamChannelCounts == inputStreamChannelCounts
AggregateDevice.mm:331 Failed expectation of constructed aggregate (348): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U)
Is it possible to use voice processing with different input/output devices?
If yes, are there any specific configurations required to handle mismatched devices?
How can we resolve channel count mismatch errors during aggregate device construction?
Are there settings or API adjustments to enforce compatibility between input/output devices?
Are there any workarounds or alternative approaches to achieve voice processing functionality with mismatched devices?
For instance, can we force an intermediate channel configuration or downmix input/output formats?
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.
I have a SwiftUI app - (https://youtu.be/VbAfUk_eYl0?si=JxUBh0Bpb-vc1E1U) - which I thought was almost ready for release - a manager for airdropped audio files from Logic Pro or other music creation applications. It uses AVAudioEngine and AVAudioPlayerNode to play audio, and the MediaPlayer API to integrate with car audio and similar, all of which works well.
It does not currently have an explicit CarPlay integration (and I'm slightly horrified at the amount of work that is going to require).
I had the good or bad luck of getting a loaner car with carplay while mine is being repaired yesterday, and lo and behold, when connected to the vehicle via CarPlay, there is no audio output in the vehicle at all. The now playing panel correctly shows the information my app provides about the currently playing song; the player node believes it is playing, the AVAudioSession is configured as it should be. But there is no sound.
Obviously I cannot ship it in this state.
I've tried fiddling with the parameters the AVAudioSession is configured with, in case there was some parameter that was preventing audio output, to no avail - currently:
var options = AVAudioSession.CategoryOptions()
options.insert(.allowAirPlay)
options.insert(.allowBluetooth)
options.insert(.allowBluetoothA2DP)
try session.setCategory(.playback, mode: .default, options: options)
try? session.setPreferredIOBufferDuration(0.002) // ~96 samples at 44.1kHz
try? session.setPrefersNoInterruptionsFromSystemAlerts(true)
try? session.setPrefersInterruptionOnRouteDisconnect(false)
try session.setActive(true, options: [.notifyOthersOnDeactivation])
All diagnostics within the app show the player operating correctly - files are played and flushed; AVAudioPlayerNodeCompletionCallbacks are called when they should be. But the output is not audible in the vehicle.
I would much prefer to ship this app without full-blown CarPlay integration, but with working audio when connected via CarPlay, and work on full CarPlay integration for the next release.
Is there some secret handshake I am just missing to make this work?
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?
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?
Topic:
Media Technologies
SubTopic:
Audio
Tags:
Apple Music API
MusicKit
MusicKit JS
Apple Music Feed
I have a memory leak, when using AVAudioPlayer. I managed to narrow down the issue into a very simple app, which code I paste in at the end.
The memory leak start immediately when I start playing sound, but only in the emylator. On the real iPhone there is no memory leak.
The memory leak on the Simulator looks like this:
import SwiftUI
import AVFoundation
struct ContentView_Audio: View {
var sound: AVAudioPlayer?
init() {
guard let path = Bundle.main.path(forResource: "cd201", ofType: "mp3") else { return }
let url = URL(fileURLWithPath: path)
do {
try AVAudioSession.sharedInstance().setCategory(.playback, mode: .default, options: [.mixWithOthers])
} catch {
return
}
do {
try AVAudioSession.sharedInstance().setActive(true)
} catch {
return
}
do {
sound = try AVAudioPlayer(contentsOf: url)
} catch {
return
}
}
var body: some View {
HStack {
Button {
playSound()
} label: {
ZStack {
Circle()
.fill(.mint.opacity(0.3))
.frame(width: 44, height: 44)
.shadow(radius: 8)
Image(systemName: "play.fill")
.resizable()
.frame(width: 20, height: 20)
}
}
.padding()
Button {
stopSound()
} label: {
ZStack {
Circle()
.fill(.mint.opacity(0.3))
.frame(width: 44, height: 44)
.shadow(radius: 8)
Image(systemName: "stop.fill")
.resizable()
.frame(width: 20, height: 20)
}
}
.padding()
}
}
private func playSound() {
guard sound != nil else { return }
sound?.volume = 1
// sound?.numberOfLoops = -1
sound?.play()
}
func stopSound() {
sound?.stop()
}
}
Hi all,
i can successfully match music using shazamkit on Apple using SwiftUI, a simple app that let user to load an audio file and exctracts the relative match, while i am unable to match music using shamzamkit on Android. I am trying to make the same simple app but i cannot match music as i get MATCH_ATTEMPT_FAILED every time i try to. I don't know what i am doing wrong but the shazam part in the kotlin Android code is in this method :
suspend fun processAudioFileInBackground(
filePath: String,
developerTokenProvider: DeveloperTokenProvider
) = withContext(Dispatchers.IO) {
val bufferSize = 1024 * 1024
val audioFile = FileInputStream(filePath)
val byteBuffer = ByteBuffer.allocate(bufferSize)
byteBuffer.order(ByteOrder.LITTLE_ENDIAN)
var bytesRead: Int
while (audioFile.read(byteBuffer.array()).also { bytesRead = it } != -1) {
val signatureGenerator = (ShazamKit.createSignatureGenerator(AudioSampleRateInHz.SAMPLE_RATE_44100) as ShazamKitResult.Success).data
signatureGenerator.append(byteBuffer.array(), bytesRead, System.currentTimeMillis())
val signature = signatureGenerator.generateSignature()
println("Signature: ${signature.durationInMs}")
val catalog = ShazamKit.createShazamCatalog(developerTokenProvider, Locale.ENGLISH)
val session = (ShazamKit.createSession(catalog) as ShazamKitResult.Success).data
val matchResult = session.match(signature)
println("MatchResult : $matchResult")
setMatchResult(matchResult)
byteBuffer.clear()
}
audioFile.close()
}
I noticed that changing Locale in catalog creation results in different result as i get NoMatch without exception. Can you please help me with this?
Topic:
Media Technologies
SubTopic:
Audio
According to the documentation (https://developer.apple.com/documentation/avfoundation/avplayeritem/externalmetadata), AVPlayerItem should have an externalMetadata property. However it does not appear to be visible to my app. When I try, I get:
Value of type 'AVPlayerItem' has no member 'externalMetadata'
Documentation states iOS 12.2+; I am building with a minimum deployment target of iOS 18.
Code snippet:
import Foundation
import AVFoundation
/// ... in function ...
// create metadata as described in https://developer.apple.com/videos/play/wwdc2022/110338
var title = AVMutableMetadataItem()
title.identifier = .commonIdentifierAlbumName
title.value = "My Title" as NSString?
title.extendedLanguageTag = "und"
var playerItem = await AVPlayerItem(asset: composition)
playerItem.externalMetadata = [ title ]