Since many users like me use Apple Music on Android, the app is almost as feature-rich as iOS. It would be fantastic if the developers could add the new iOS 26 features to the Android app, along with a minor UI change. I know it’s challenging to implement liquid glass on Android hardware or design, but features like auto-mix, pronunciation, and translation could be added.
kindly consider this request !!!!
Audio
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I started playing which transcription of audio files on macOS today, latest beta of Xcode and latest beta of Tahoe. Transcription itself works really well, but for some reason the majority of the results contain no audioTimeRange. I got 22 single-word results with time ranges, spread out all over total file of 53 minutes.
Is there something I can do to improve this? To my understanding, I have followed sample code and instructions very closely, but the SwiftTranscriptionSampleApp and other examples I've seen lead me to believe I should be getting a lot more time ranges than I actually do.
Hi all,
I'm working on an audio visualizer app that plays files from the user's music library utilizing MediaPlayer and AVAudioEngine. I'm working on getting the music library functionality working before the visualizer aspect.
After setting up the engine for file playback, my app inexplicably crashes with an EXC_BREAKPOINT with code = 1. Usually this means I'm unwrapping a nil value, but I think I'm handling the optionals correctly with guard statements. I'm not able to pinpoint where it's crashing. I think it's either in the play function or the setupAudioEngine function. I removed the processAudioBuffer function and my code still crashes the same way, so it's not that. The device that I'm testing this on is running iOS 26 beta 3, although my app is designed for iOS 18 and above.
After commenting out code, it seems that the app crashes at the scheduleFile call in the play function, but I'm not fully sure.
Here is the setupAudioEngine function:
private func setupAudioEngine() {
do {
try AVAudioSession.sharedInstance().setCategory(.playback, mode: .default)
try AVAudioSession.sharedInstance().setActive(true)
} catch {
print("Audio session error: \(error)")
}
engine.attach(playerNode)
engine.attach(analyzer)
engine.connect(playerNode, to: analyzer, format: nil)
engine.connect(analyzer, to: engine.mainMixerNode, format: nil)
analyzer.installTap(onBus: 0, bufferSize: 1024, format: nil) { [weak self] buffer, _ in
self?.processAudioBuffer(buffer)
}
}
Here is the play function:
func play(_ mediaItem: MPMediaItem) {
guard let assetURL = mediaItem.assetURL else {
print("No asset URL for media item")
return
}
stop()
do {
audioFile = try AVAudioFile(forReading: assetURL)
guard let audioFile else {
print("Failed to create audio file")
return
}
duration = Double(audioFile.length) / audioFile.fileFormat.sampleRate
if !engine.isRunning {
try engine.start()
}
playerNode.scheduleFile(audioFile, at: nil)
playerNode.play()
DispatchQueue.main.async { [weak self] in
self?.isPlaying = true
self?.startDisplayLink()
}
} catch {
print("Error playing audio: \(error)")
DispatchQueue.main.async { [weak self] in
self?.isPlaying = false
self?.stopDisplayLink()
}
}
}
Here is a link to my test project if you want to try it out for yourself:
https://github.com/aabagdi/VisualMan-example
Thanks!
I work on an iOS app that records video and audio. We've been getting reports for a while from users who are experiencing their video recordings being cut off. After investigating, I found that many users are receiving the AVAudioSessionMediaServicesWereResetNotification (.mediaServicesWereResetNotification) notification while recording. It's associated with the AVFoundationErrorDomain[-11819] error, which seems to indicate that the system audio daemon crashed. We have a handler registered to end the recording, show the user a prompt, and restart our AV sessions. However, from our logs this looks to be happening to hundreds of users every day and it's not an ideal user experience, so I would like to figure out why this is happening and if it's due to something that we're doing wrong.
The debug menu option to trigger the audio session reset is not of much use, because it can't be triggered unless you leave the app and go to system settings. So our app can't be recording video when the debug reset is triggered. So far I haven't found a way to reproduced the issue locally, but I can see that it's happening to users from logs.
I've found some posts online from developers experiencing similar issues, but none of them seem to directly address our issue. The system error doesn't include a userInfo dictionary, and as far as I can tell it's a system daemon crash so any logs would need to be captured from the OS.
Is there any way that I could get more information about what may be causing this error that I may have missed?
Topic:
Media Technologies
SubTopic:
Audio
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 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?
Topic:
Media Technologies
SubTopic:
Audio
Environment→ ・Device: iPad 10th generation ・OS:**iOS18.3.2
I'm using AVAudioSession to record sound in my application. But I recently came to realize that when the app starts a recording session on a tablet, OS automatically sets the tablet volume to 50% and when after recording ends, it doesn't change back to the previous volume level before starting the recording. So I would like to know whether this is an OS default behavior or a bug?
If it's a default behavior, I much appreciate if I can get a link to the documentation.
I have an AUv3 that passes all validation and can be loaded into Logic Pro without issue. The UI for the plug in can be any aspect ratio but Logic insists on presenting it in a view with a fixed aspect ratio. That is when resizing, both the height and width are resized. I have never managed to work out what it is I need to do specify to Logic to allow the user to resize width or height independently of each other.
Can anyone tell me what I need to specify in the AU code that will inform Logic that the view can be resized from any side of the window/panel?
iPadOS 18.3 beta 3 (22D5055b) fixed the issue for me and my 7th generation iPad.
Topic:
Media Technologies
SubTopic:
Audio
Hello,
I'm observing an intermittent memory leak being reported in the iOS Simulator when initializing and starting an AVAudioEngine. Even with minimal setup—just attaching a single AVAudioPlayerNode and connecting it to the mainMixerNode—Xcode's memory diagnostics and Instruments sometimes flag a leak.
Here is a simplified version of the code I'm using:
// This function is called when the user taps a button in the view controller:
#import "ViewController.h"
@interface ViewController ()
@end
@implementation ViewController
- (void)viewDidLoad {
[super viewDidLoad];
}
- (IBAction)myButtonAction:(id)sender {
NSLog(@"Test");
soundCreate();
}
@end
// media.m
static AVAudioEngine *audioEngine = nil;
void soundCreate(void)
{
if (audioEngine != nil)
return;
[[AVAudioSession sharedInstance] setCategory:AVAudioSessionCategoryAmbient error:nil];
[[AVAudioSession sharedInstance] setActive:YES error:nil];
audioEngine = [[AVAudioEngine alloc] init];
AVAudioPlayerNode* playerNode = [[AVAudioPlayerNode alloc] init];
[audioEngine attachNode:playerNode];
[audioEngine connect:playerNode to:(AVAudioNode *)[audioEngine mainMixerNode] format:nil];
[audioEngine startAndReturnError:nil];
}
In the memory leak report, the following call stack is repeated, seemingly in a loop:
ListenerMap::InsertEvent(XAudioUnitEvent const&, ListenerBinding*) AudioToolboxCore
ListenerMap::AddParameter(AUListener*, void*, XAudioUnitEvent const&) AudioToolboxCore
AUListenerAddParameter AudioToolboxCore
addOrRemoveParameterListeners(OpaqueAudioComponentInstance*, AUListenerBase*, AUParameterTree*, bool) AudioToolboxCore
0x180178ddf
Hi everyone,
I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms.
Problem:
When the app is recording audio and an interruption occurs:
I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began).
On .ended, I check for .shouldResume and call audioRecorder?.record() again.
The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder.
Repro:
Start a recording with AVAudioRecorder
Simulate a system interruption (e.g., incoming call)
Resume recording after the interruption
Stop and inspect the output audio file
Expected: Full audio (before and after interruption) should be saved.
Actual: Only the audio after interruption is saved; the earlier part is missing
Notes:
According to the documentation, calling .record() after .pause() should resume recording into the same file.
I confirmed that the file URL does not change, and I do not recreate the recorder instance.
No error is thrown by the system during this process.
This behavior happens consistently when the app is interrupted and resumed.
Question:
Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen?
Thanks in advance!
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.
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)
According to the header file the outputVolume properties supported range is 0.0-1.0:
/*! @property outputVolume
@abstract The mixer's output volume.
@discussion
This accesses the mixer's output volume (0.0-1.0, inclusive).
@property (nonatomic) float outputVolume;
However when setting the volume to 2.0 the audio does indeed play louder. Is the header file out of date and if so, what is the supported range for outputVolume?
Thanks
使用AVSpeechUtterance实现iOS语音播报,选择语言为简体中文“zh-CN”,读取中文“袆”(hui 第一声)错误,读成了“祎”(yi 第一声),希望能优化。
I am developing an app that uses MusicKit to play music and then I need to have spoken words played to the user, while ducking the audio coming from MusicKit (application music player)
the built in Siri voices are not off sufficient quality so I am using an external service to create an mp3 file and then play this back using AVAudioSession
Sample code below
the problem I am having is that .duckOthers is not ducking the Application Music Player output
Is this a bug or am I doing this wrong?
// Configure audio session for system-wide ducking
try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, options: [.duckOthers, .mixWithOthers])
try AVAudioSession.sharedInstance().setActive(true)
// Set the ducking level to maximum
try AVAudioSession.sharedInstance().setPreferredIOBufferDuration(0.005)
// Create and configure audio player
self.audioPlayer = try AVAudioPlayer(data: audioData)
self.audioPlayer?.delegate = self
self.audioPlayer?.volume = 1.0 // Ensure full volume for speech
self.audioPlayer?.prepareToPlay()
// Set the audio player's settings for maximum clarity
self.audioPlayer?.enableRate = false
self.audioPlayer?.pan = 0.0 // Center the audio
self.audioPlayer?.play()
Hi, In my project I am using AVFoundation for recording the audio. We are using AVAudioMixerNode class below method to record the audio packet.
**func installTap(
onBus bus: AVAudioNodeBus,
bufferSize: AVAudioFrameCount,
format: AVAudioFormat?,
block tapBlock: @escaping AVAudioNodeTapBlock
)
**
It works perfectly fine.
But in production env some small percentage of the user we are facing issue like after recording few packets it stops automatically without stopping the audio engine. Can anyone help here that why this happens? I have also observed for mediaServicesWereResetNotification and added log on receiving this notification but when this issue happens I don't see any occurence of this log. Also is there any callback when the engine stops?
Hi. I am working on an audio app for iOS. I have added the CPNowPlayingPlaybackRateButton to my CPNowPlayingTemplate.
When the button is clicked, my handler changes the rate in the AVPlayer and updates the MPNowPlayingInfoCenter to the new rate, for example, 2.0.
Throughout, the Carplay button always displays "0x". I am wondering how to get this UI to accurately reflect the playback rate the user has selected, as always displaying 0x is a poor user experience.
You may suggest MPChangePlaybackRateCommand is relevant here, but I have not been able to get that to work either, and judging by posts online, not many other people have either. I have made a post about that here: https://developer.apple.com/forums/thread/773099
Is this a known Apple bug? Is there a way to get the UI to accurately reflect the playback rate of my audio?
Kind regards.
Topic:
Media Technologies
SubTopic:
Audio
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
Hello,
I'm trying to receive parquet files using the example that provided in documentation. I've done all required steps but receive constantly error 500 with "Upstream Service Error". By looking into the issues list, seems this error exists for months. Is it possible to get it working?