I'm using AVFoundation to make a multi-track editor app, which can insert multiple track and clip, including scale some clip to change the speed of the clip, (also I'm not sure whether AVFoundation the best choice for me) but after making the scale with scaleTimeRange API, there is some short noise sound in play back. Also, sometimes it's fine when play AVMutableCompostion using AVPlayer with AVPlayerItem, but after exporting with AVAssetReader, will catch some short noise sounds in result file.... Not sure why.
Here is the example project, which can build and run directly. https://github.com/luckysmg/daily_images/raw/refs/heads/main/TestDemo.zip
Audio
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I'm working on a project to support spatial audio editing, using this sample project as a reference: https://developer.apple.com/documentation/Cinematic/editing-spatial-audio-with-an-audio-mix
This sample works well on an unedited capture, but does not work for a capture that has already been edited.
The failure is occurring at "let audioInfo = try await CNAssetSpatialAudioInfo(asset: myAsset)", which is throwing "no eligible audio tracks in asset".
I also find that for already edited captures, if i use CNAssetSpatialAudioInfo.assetContainsSpatialAudio, it returns false.
What i mean by "already edited" is that if I take a spatial capture with my iPhone 16, and then edit that capture in the Photos app using the Cinematic effect, and then save the edited output (e.g. edited_capture.mov), I can't import that edited_capture.mov into my project as a spatial audio asset.
Is this intentional behavior or a bug?
If it's intentional, can you describe why?
Topic:
Media Technologies
SubTopic:
Audio
Hello,
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? Do i need to create a custom catalog?
I bought two "Apple USB-C to Headphone Jack Adapters". Upon closer inspection, they seems to be of different generations:
The one with product ID 0x110a on top is working fine. The one with product ID 0x110b has two issues:
There is a short but loud click noise on the headphone when I connect it to the iPad.
When I play audio using AVAudioPlayer the first half of a second or so is cut off.
Here's how I'm playing the audio:
audioPlayer = try AVAudioPlayer(contentsOf: url)
audioPlayer?.delegate = self
audioPlayer?.prepareToPlay()
audioPlayer?.play()
Is this a known issue? Am I doing something wrong?
Is there any way for me to use an AutoMix api in my IOS apps, I would play tracks using the Apple Music api and use AutoMix to attempt to merge tracks.
Is this feature/api available to developers.
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.
Hello,
I'm evaluating the Apple Music Feed dataset and I noticed that the total number of songs available in the feed is too small. As of today, the number of objects returned in each feed is:
51,198,712 albums
23,093,698 artists
173,235,315 songs
This gives an average of 3.38 songs per album which is quite low. Also, iterating on the data I see that there are albums referencing songs that don't exist in the songs feed. I would like to know:
Is the feed data incomplete?
If so, in what situations an object may be missing from the feed?
Thank you in advance!
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()
}
}
}
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.
Environment
Device: iPhone 16e
iOS Version: 18.4.1 - 18.7.1
Framework: AVFoundation (AVAudioEngine)
Problem Summary
On iPhone 16e (iOS 18.4.1-18.7.1), the installTap callback stops being invoked after resuming from a phone call interruption. This issue is specific to phone call interruptions and does not occur on iPhone 14, iPhone SE 3, or earlier devices.
Expected Behavior
After a phone call interruption ends and audioEngine.start() is called, the previously installed tap should continue receiving audio buffers.
Actual Behavior
After resuming from phone call interruption:
Tap callback is no longer invoked
No audio data is captured
No errors are thrown
Engine appears to be running normally
Note: Normal pause/resume (without phone call interruption) works correctly.
Steps to Reproduce
Start audio recording on iPhone 16e
Receive or make a phone call (triggers AVAudioSession interruption)
End the phone call
Resume recording with audioEngine.start()
Result: Tap callback is not invoked
Tested devices:
iPhone 16e (iOS 18.4.1-18.7.1): Issue reproduces ✗
iPhone 14 (iOS 18.x): Works correctly ✓
iPhone SE 3 (iOS 18.x): Works correctly ✓
Code
Initial Setup (Works)
let inputNode = audioEngine.inputNode
inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in
self.processAudioBuffer(buffer, at: time)
}
audioEngine.prepare()
try audioEngine.start()
Interruption Handling
NotificationCenter.default.addObserver(
forName: AVAudioSession.interruptionNotification,
object: AVAudioSession.sharedInstance(),
queue: nil
) { notification in
guard let userInfo = notification.userInfo,
let typeValue = userInfo[AVAudioSessionInterruptionTypeKey] as? UInt,
let type = AVAudioSession.InterruptionType(rawValue: typeValue) else {
return
}
if type == .began {
self.audioEngine.pause()
} else if type == .ended {
try? self.audioSession.setActive(true)
try? self.audioEngine.start()
// Tap callback doesn't work after this on iPhone 16e
}
}
Workaround
Full engine restart is required on iPhone 16e:
func resumeAfterInterruption() {
audioEngine.stop()
inputNode.removeTap(onBus: 0)
inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil) { buffer, time in
self.processAudioBuffer(buffer, at: time)
}
audioEngine.prepare()
try audioSession.setActive(true)
try audioEngine.start()
}
This works but adds latency and complexity compared to simple resume.
Questions
Is this expected behavior on iPhone 16e?
What is the recommended way to handle phone call interruptions?
Why does this only affect iPhone 16e and not iPhone 14 or SE 3?
Any guidance would be appreciated!
Since MacOS 26 Apple Music has inconsitent drops to the Quality of some Tracks indiscrimantly. I don't know if others Expereinced it. It doesn't happen on the Speakers or connected via Bluetooth, but the AUX I/O has it quite often. It is more noticable on Headphones with 48kHz and higher Frequency Bandwidth.
Here is the FB18062589
Hi everyone!
I’ve developed a location-based Audio AR app in Unity with FMOD & Resonance Audio and AirPods Pro Head-Tracking to create a ubiquitous augmented soundscape experience. Think of it as an audio version of Pokémon Go, but with a more precise location requirement to ensure spatial audio is placed correctly.
I want this experience to run in the background on iOS, but from what I’ve gathered, it seems Unity doesn’t support this well. So, I’m considering developing a Swift version instead.
Since this is primarily for research purposes, privacy concerns are not a major issue in my case. However, I’ve come across some potential challenges:
Real-time precise location updates – Can iOS provide fully instantaneous, high-accuracy location updates in the background?
Continuous real-time data processing – Can an app continuously process spatial audio, head-tracking, and location data while running in the background?
I’m not sure if newer iOS versions have improved in these areas or if there are workarounds to achieve this.
Would this kind of experience be feasible to run in the background on iOS? Any insights or pointers would be greatly appreciated!
I’m very new to iOS development, so apologies if this is a basic question. Thanks in advance!
Description: I have identified a specific issue when recording acoustic guitar and other instruments on the iPhone 17 Pro Max using native applications (Voice Memos, Camera). The recordings contain an unnatural metallic resonance (ringing artifacts) that should not be present.
Testing and Methodology:
Hardware Verification: Initially, I suspected a hardware defect in the audio chip or microphone. However, extensive testing with third-party software suggests this is likely a software-level issue.
AudioShare Test: I conducted a test using the AudioShare app in "Measurement Mode" (which bypasses standard iOS system-wide audio processing). In this mode, the audio remains perfectly clean, and the metallic ringing disappears entirely.
Conclusion: The issue is rooted in the DSP (Digital Signal Processing) algorithms that iOS applies for noise suppression or voice enhancement. These algorithms appear to misinterpret the high-frequency overtones of acoustic instruments as background noise and attempt to "filter" them, resulting in audible digital artifacts.
Comparison Results: This issue has not been observed on devices from other brands or on older iPhone models (preliminary tests suggest older versions handle this better). Notably, the problem persists even in GarageBand, as the app still utilizes certain system-level processing layers.
Proposed Solution: I suggest adding a "Raw Audio" or "Instrument Mode" toggle within the Microphone/Audio settings for native apps. This mode should disable aggressive DSP processing, similar to how the AVAudioSession.Mode.measurement works in specialized apps.
Attachments: I am attaching 4 archives, including a final "Measurement Mode" folder with comparative samples (Measurement Mode vs. Standard Mode). The artifacts are most prominent when monitored through headphones.
Hi,
I'm still stuck getting a basic record-with-playthrouh pipeline to work.
Has anyone a sample of setting up a AVAudioEngine pipeline for recording with playthrough?
Plkaythrough works with AVPlayerNode as input but not with any microphone input. The docs mention the "enabled state" of the outputNode of the engine without explaining the concept, i.e. how to enable an output.
When the engine renders to and from an audio device, the AVAudioSession category and the availability of hardware determines whether an app performs output. Check the output node’s output format (specifically, the hardware format) for a nonzero sample rate and channel count to see if output is in an enabled state.
Well, in my setup the output is NOT enabled, and any attempt to switch (e.g. audioEngine.outputNode.auAudioUnit.setDeviceID(deviceID) )/ attach a dedicated device / ... results in exceptions / errors
Hello.
My team and I think we have an issue where our app is asked to gracefully shutdown with a following SIGTERM. As we’ve learned, this is normally not an issue. However, it seems to also be happening while our app (an audio streamer) is actively playing in the background.
From our perspective, starting playback is indicating strong user intent. We understand that there can be extreme circumstances where the background audio needs to be killed, but should it be considered part of normal operation? We hope that’s not the case.
All we see in the logs is the graceful shutdown request. We can say with high certainty that it’s happening though, as we know that playback is running within 0.5 seconds of the crash, without any other tracked user interaction.
Can you verify if this is intended behavior, and if there’s something we can do about it from our end. From our logs it doesn’t look to be related to either memory usage within the app, or the system as a whole.
Best,
John
I've got a setup using AVAudioEngine with several tone generator nodes, each with a chain of processing nodes, the chains then mixed into the main output.
Generator ➡️ Effect ➡️... ➡️ .mainMixerNode ➡️ .outputNode).
Generator ➡️ Effect ➡️... ⤴️
...
Generator ➡️ Effect ➡️... ⤴️
The user should be able to mute any chain individually. I've found several potential approaches to muting, but not terribly happy with any of them.
Adjust the amplitudes directly in my tone generators. Issue: Consumes CPU even when completely muted. 4 generators adds ~15% cpu, even when all chains are muted.
Detach/attach chains that are muted/unmuted. Issue: Causes loud clicking/popping sounds whenever muted/unmuted.
Fade mixer output volume while detaching/attaching a chain (just cutting the volume immediately to 0 doesn't get rid of the clicking/popping). Issue: Causes all channels to fade during the transition, so not ideal.
The rest of these ideas are variations on making volume control+detatch/attach work for individual chains, since approach #3 worked well.
Add an AVAudioMixer to the end of each chain (just for volume control). Issue: Only the mixer on the final chain functions -- the others block all output. Not sure what's going on there.
Use matrix mixer (for multi-input volume control). Plus detach/attach to reduce CPU if necessary. Not yet attempted, due to perceived complexity and reports of fragility in order of wiring in. A bunch of effort before I even know if it's going to work.
Develop my own fader node to put on the end of each channel. Unlike the tone generator (simple AVSourceNode), developing an effect node seems complex and time consuming. Might not even fix CPU use.
I'm not completely averse to the learning curve of either 5 or 6, but would rather get some guidance on best approach before diving in. They both seem likely to take more effort than I'd like for the simple behavior I'm trying to achieve.
Hello everyone,
I am working on an app that allows you to review your own music using Apple Music. Currently I am running into an issue with the skipping forwards and backwards outside of the app.
How it should work: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song on an album should play and the information should change to reflect that in the app.
If you play a song in Apple Music, you can see a Now Playing view in the lock screen.
When you skip forward or backwards, it will do either action and it would reflect that when you see a little frequency icon on artwork image of a song.
What it's doing: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song is reflected outside of the app, but not in the app.
When skipping a song outside of the app, it works correctly to head to the next song.
But when I return to the app, it is not reflected
NOTE: I am not using MusicKit variables such as Track, Album to display the songs. Since I want to grab the songs and review them I need a rating so I created my own that grabs the MusicItemID, name, artist(s), etc.
NOTE: I am using ApplicationMusicPlayer.shared
Is there a way to get the song to reflect in my app?
(If its easier, a simple example of it would be nice. No need to create an entire xprod file)
Does anyone know how to pronounce the sound of a specific instrument when you tap a button on the screen on your iPhone or iPad? Now, in the middle of creating a music learning app, I'm thinking of assigning monotones or chords to the button-like frames on the keyboard and fingerboard on the screen. Can it be achieved with SwiftUI chords alone? Once upon a time, MIDI level 1 I remember that there was a pronunciation function of the instrument, but I don't think about implementing the same function in the current OS. Please lend me your wisdom.
Topic:
Media Technologies
SubTopic:
Audio
Since iOS 18, the system setting “Allow Audio Playback” (enabled by default) allows third-party app audio to continue playing while the user is recording video with the Camera app. This has created a problem for the app I’m developing.
➡️ The problem:
My app plays continuous audio in both foreground and background states. If the user starts recording video using the iOS Camera app, the app’s audio — still playing in the background — gets captured in the video — obviously an unintended behavior.
Yes, the user could stop the app manually before starting the video recording, but that can’t be guaranteed. As a developer, I need a way to stop the app’s audio before the video recording begins.
So far, I haven’t found a reliable way to detect when video recording starts if ‘Allow Audio Playback’ is ON.
➡️ What I’ve tried:
— AVAudioSession.interruptionNotification → doesn’t fire
— devicesChangedEventStream → not triggered
I don’t want to request mic permission (app doesn’t use mic). also, disabling the app from playing audio in the background isn’t an option as it is a crucial part of the user experience
➡️ What I need:
A reliable, supported way to detect when the Camera app begins video recording, without requiring mic access — so I can stop audio and avoid unintentional overlap with the user’s recordings.
Any official guidance, workarounds, or AVFoundation techniques would be greatly appreciated.
Thanks.
Hi everyone 👋
I’m building an iOS app in Swift where I want to do the following:
Record the user’s voice
Transcribe the spoken sentence (speech-to-text)
Auto-detect the spoken language
Translate it to another language selected by the user (e.g., English → Spanish or Hindi → English)
Speak back (text-to-speech) the translated text on the same device
Is this possible to record via phone mic and play the transcribe voice into headphone's audio?