I'm using AVAudioEngineto play sound files and record the output to a file. I have lots of sound effect files, each played by tapping a button. To play the file I create an AVAudioPlayerNode and connects it to the engine. After the file has played I try to disconnect/detach the node in the completionHandler closure, to free up memory. If I don't remove nodes I will just keep adding new nodes and new connections to the engine.
However, the next time I try to play a file, the app just freezes. Here's my code:
let url = NSBundle.mainBundle().URLForResource(name, withExtension: "wav")!
let audioPlayerFile = try AVAudioFile(forReading: url)
let playerNode = AVAudioPlayerNode()
self.engine.attachNode(playerNode)
self.engine.connect(playerNode, to: self.engine.mainMixerNode, format: audioPlayerFile.processingFormat)
playerNode.scheduleFile(audioPlayerFile, atTime: nil, completionHandler: { () -> Void in
self.engine.disconnectNodeOutput(playerNode)
self.engine.detachNode(playerNode)
})
playerNode.play()
What's the best way to implement a functionality like this, i.e. playing multiple files, potentially overlapping each other, while recording?
I've had a similar problem. I've found that detaching the node on a background thread keeps the app from freezing.
playerNode.scheduleBuffer(buffer, atTime: nil, options: nil) { () -> Void in
self.detach(playerNode)
}
And here's the method:
func detach(player: AVAudioPlayerNode) {
let priority = DISPATCH_QUEUE_PRIORITY_DEFAULT
dispatch_async(dispatch_get_global_queue(priority, 0)) { () -> Void in
self.engine.detachNode(player)
}
However, I'm still unsure this is a solution per se, because these background threads might be spinning infinitely without my knowledge.
Related
i'm currently working a musician app. In my app notes should be played with a specific duration. I don't get into detail when the notes are played. Basically there is a ui view (a vertical line) which is moving and when this hits my other ui views (rectangle) it should be played a note. Important here: the note should be played until the line is not hitting the rectangle anymore.
The note playing is no problem but I don't find any duration. Also it should be possible to play the same note multiple times with a delay.
So I tried to make this work with AudioKit cause it's seems like the best greatest solution for audio. But it has so much stuff. I took a look into their examples and found this:
let bundlePath = Bundle.main.bundlePath
let soundPath = ("\(bundlePath)/sounds")
let akSampler = AKAppleSampler()
let mixer = AKMixer(akSampler)
try! akSampler.loadSoundFont(soundPath, preset: 0, bank: 0)
mixer.start()
AudioKit.output = mixer
do {
_ = try AudioKit.engine.start()
} catch {
print("AudioKit wouldn't start!")
}
do {
try akSampler.play(noteNumber: myNote.rawValue, velocity: 100, channel: 1)
} catch let e{
print(e)
}
Unfortunately I can't pass any duration and when I call akSampler.stop(noteNumber: myNote.rawValue) it also stops the other notes with the same type.
I tried to find a solution with AVFoundation like so:
engine = AVAudioEngine()
sampler = AVAudioUnitSampler()
engine.attach(sampler)
engine.connect(sampler, to: engine.mainMixerNode, format: nil)
guard let bankURL = Bundle.main.url(forResource: "sounds", withExtension: "SF2") else {
print("could not load sound font")
return
}
... init engine
sampler.startNote(60, withVelocity: 64, onChannel: 0)
But same result. Also the same case that I can't pass any duration.
I also digged into MIDISequencer's but it seems that they generating a sequence which I can play but this does not fit on my problem.
Does someone has a solution here?
The laziest solution would be to just schedule a stop with asyncAfter when you trigger the note, e.g.,
func makeNote(note: MIDINoteNumber, dur: Double) {
sampler.play(noteNumber: note, velocity: 100, channel: 0)
DispatchQueue.main.asyncAfter(deadline: .now() + dur) {
self.sampler.stop(noteNumber: note)
}
}
A better solution would probably use either AKSequencer or AKAppleSequencer. Both allow you to create sequences on the fly by adding individual notes with a specified duration (in musical time, i.e., number of beats). AKSequencer is considerably more accurate, but AKAppleSequencer has more readily available code examples on the web. A little confusingly, the current AKAppleSequencer used to also be called AKSequencer, but their interfaces are sufficiently different that a quick look at the docs for the two classes will tell you which you're looking at.
Your question is asking about how to schedule MIDI events which is precisely what these classes are designed to do. You haven't really given a clear reason why generating a sequence doesn't fit your problem.
I am making a basic music app for iOS, where pressing notes causes the corresponding sound to play. I am trying to get multiple sounds stored in buffers to play simultaneously with minimal latency. However, I can only get one sound to play at any time.
I initially set up my sounds using multiple AVAudioPlayer objects, assigning a sound to each player. While it did play multiple sounds simultaneously, it didn't seem like it was capable of starting two sounds at the same time (it seemed like it would delay the second sound just slightly after the first sound was started). Furthermore, if I pressed notes at a very fast rate, it seemed like the engine couldn't keep up, and later sounds would start well after I had pressed the later notes.
I am trying to solve this problem, and from the research I have done, it seems like using the AVAudioEngine to play sounds would be the best method, where I can set up the sounds in an array of buffers, and then have them play back from those buffers.
class ViewController: UIViewController
{
// Main Audio Engine and it's corresponding mixer
var audioEngine: AVAudioEngine = AVAudioEngine()
var mainMixer = AVAudioMixerNode()
// One AVAudioPlayerNode per note
var audioFilePlayer: [AVAudioPlayerNode] = Array(repeating: AVAudioPlayerNode(), count: 7)
// Array of filepaths
let noteFilePath: [String] = [
Bundle.main.path(forResource: "note1", ofType: "wav")!,
Bundle.main.path(forResource: "note2", ofType: "wav")!,
Bundle.main.path(forResource: "note3", ofType: "wav")!]
// Array to store the note URLs
var noteFileURL = [URL]()
// One audio file per note
var noteAudioFile = [AVAudioFile]()
// One audio buffer per note
var noteAudioFileBuffer = [AVAudioPCMBuffer]()
override func viewDidLoad()
{
super.viewDidLoad()
do
{
// For each note, read the note URL into an AVAudioFile,
// setup the AVAudioPCMBuffer using data read from the file,
// and read the AVAudioFile into the corresponding buffer
for i in 0...2
{
noteFileURL.append(URL(fileURLWithPath: noteFilePath[i]))
// Read the corresponding url into the audio file
try noteAudioFile.append(AVAudioFile(forReading: noteFileURL[i]))
// Read data from the audio file, and store it in the correct buffer
let noteAudioFormat = noteAudioFile[i].processingFormat
let noteAudioFrameCount = UInt32(noteAudioFile[i].length)
noteAudioFileBuffer.append(AVAudioPCMBuffer(pcmFormat: noteAudioFormat, frameCapacity: noteAudioFrameCount)!)
// Read the audio file into the buffer
try noteAudioFile[i].read(into: noteAudioFileBuffer[i])
}
mainMixer = audioEngine.mainMixerNode
// For each note, attach the corresponding node to the audioEngine, and connect the node to the audioEngine's mixer.
for i in 0...2
{
audioEngine.attach(audioFilePlayer[i])
audioEngine.connect(audioFilePlayer[i], to: mainMixer, fromBus: 0, toBus: i, format: noteAudioFileBuffer[i].format)
}
// Start the audio engine
try audioEngine.start()
// Setup the audio session to play sound in the app, and activate the audio session
try AVAudioSession.sharedInstance().setCategory(AVAudioSession.Category.soloAmbient)
try AVAudioSession.sharedInstance().setMode(AVAudioSession.Mode.default)
try AVAudioSession.sharedInstance().setActive(true)
}
catch let error
{
print(error.localizedDescription)
}
}
func playSound(senderTag: Int)
{
let sound: Int = senderTag - 1
// Set up the corresponding audio player to play its sound.
audioFilePlayer[sound].scheduleBuffer(noteAudioFileBuffer[sound], at: nil, options: .interrupts, completionHandler: nil)
audioFilePlayer[sound].play()
}
Each sound should be playing without interrupting the other sounds, only interrupting its own sound when the sounds is played again. However, despite setting up multiple buffers and players, and assigning each one to its own Bus on the audioEngine's mixer, playing one sound still stops any other sounds from playing.
Furthermore, while leaving out .interrupts does prevent sounds from stopping other sounds, these sounds won't play until the sound that is currently playing completes. This means that if I play note1, then note2, then note3, note1 will play, while note2 will only play after note1 finishes, and note3 will only play after note2 finishes.
Edit: I was able to get the audioFilePlayer to reset to the beginning again without using interrupt with the following code in the playSound function.
if audioFilePlayer[sound].isPlaying == true
{
audioFilePlayer[sound].stop()
}
audioFilePlayer[sound].scheduleBuffer(noteAudioFileBuffer[sound], at: nil, completionHandler: nil)
audioFilePlayer[sound].play()
This still leaves me with figuring out how to play these sounds simultaneously, since playing another sound will still stop the currently playing sound.
Edit 2: I found the solution to my problem. My answer is below.
It turns out that having the .interrupt option wasn't the issue (in fact, this actually turned out to be the best way to restart the sound that was playing in my experience, as there was no noticeable pause during the restart, unlike the stop() function). The actual problem that was preventing multiple sounds from playing simultaneously was this particular line of code.
// One AVAudioPlayerNode per note
var audioFilePlayer: [AVAudioPlayerNode] = Array(repeating: AVAudioPlayerNode(), count: 7)
What happened here was that each item of the array was being assigned the exact same AVAudioPlayerNode value, so they were all effectively sharing the same AVAudioPlayerNode. As a result, the AVAudioPlayerNode functions were affecting all of the items in the array, instead of just the specified item. To fix this and give each item a different AVAudioPlayerNode value, I ended up changing the above line so that it starts as an empty array of type AVAudioPlayerNode instead.
// One AVAudioPlayerNode per note
var audioFilePlayer = [AVAudioPlayerNode]()
I then added a new line to append to this array a new AVAudioPlayerNode at the beginning inside of the second for-loop of the viewDidLoad() function.
// For each note, attach the corresponding node to the audioEngine, and connect the node to the audioEngine's mixer.
for i in 0...6
{
audioFilePlayer.append(AVAudioPlayerNode())
// audioEngine code
}
This gave each item in the array a different AVAudioPlayerNode value. Playing a sound or restarting a sound no longer interrupts the other sounds that are currently being played. I can now play any of the notes simultaneously and without any noticeable latency between note press and playback.
I am having a really difficult time with playing audio in the background of my app. The app is a timer that is counting down and plays bells, and everything worked using the timer originally. Since you cannot run a timer over 3 minutes in the background, I need to play the bells another way.
The user has the ability to choose bells and set the time for these bells to play (e.g. play bell immediately, after 5 minutes, repeat another bell every 10 minutes, etc).
So far I have tried using notifications using DispatchQueue.main and this will work fine if the user does not pause the timer. If they re-enter the app though and pause, I cannot seem to cancel this queue or pause it in anyway.
Next I tried using AVAudioEngine, and created a set of nodes. These will play while the app is in the foreground but seem to stop upon backgrounding. Additionally when I pause the engine and resume later, it won't pause the sequence properly. It will squish the bells into playing one after the other or not at all.
If anyone has any ideas of how to solve my issue that would be great. Technically I could try remove everything from the engine and recreate it from the paused time when the user pauses/resumes, but this seems quite costly. It also doesn't solve the problem of the audio stopping in the background. I have the required background mode 'App plays audio or streams audio/video using Airplay', and it is also checked under the background modes in capabilities.
Below is a sample of how I tried to set up the audio engine. The registerAndPlaySound method is called several other times to create the chain of nodes (or is this done incorrectly?). The code is kinda messy at the moment because I have been trying many ways trying to get this to work.
func setupSounds{
if (attached){
engine.detach(player)
}
engine.attach(player)
attached = true
let mixer = engine.mainMixerNode
engine.connect(player, to: mixer, format: mixer.outputFormat(forBus: 0))
var bell = ""
do {
try engine.start()
} catch {
return
}
if (currentSession.bellObject?.startBell != nil){
bell = (currentSession.bellObject?.startBell)!
guard let url = Bundle.main.url(forResource: bell, withExtension: "mp3") else {
return
}
registerAndPlaySound(url: url, delay: warmUpTime)
}
}
func registerAndPlaySound(url: URL, delay: Double) {
do {
let file = try AVAudioFile(forReading: url)
let format = file.processingFormat
let capacity = file.length
let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: AVAudioFrameCount(capacity))
do {
try file.read(into: buffer)
}catch {
return
}
let sampleRate = buffer.format.sampleRate
let sampleTime = sampleRate*delay
let futureTime = AVAudioTime(sampleTime: AVAudioFramePosition(sampleTime), atRate: sampleRate)
player.scheduleBuffer(buffer, at: futureTime, options: AVAudioPlayerNodeBufferOptions(rawValue: 0), completionHandler: nil)
player.play()
} catch {
return
}
}
To explain my situation a little better I'm trying to make an app which will play a ping noise when a button is pressed and then proceed to record and transcribe the user's voice immediately after.
For the ping sound I'm using System Sound Services, to record the audio I'm using AudioToolbox, and to transcribe it I'm using Speech kit.
I believe the crux of my problem lies in the timing of the asynchronous System sound services play function:
//Button pressed function
let audiosession = AVAudioSession.sharedInstance()
let filename = "Ping"
let ext = "wav"
if let soundUrl = Bundle.main.url(forResource: filename, withExtension: ext){
var soundId: SystemSoundID = 0
AudioServicesCreateSystemSoundID(soundUrl as CFURL, &soundId)
AudioServicesAddSystemSoundCompletion(soundId, nil, nil, {(soundid,_) -> Void in
AudioServicesDisposeSystemSoundID(soundid)
print("Sound played!")}, nil)
AudioServicesPlaySystemSound(soundId)
}
do{
try audiosession.setCategory(AVAudioSessionCategoryRecord)
try audiosession.setMode(AVAudioSessionModeMeasurement)
try audiosession.setActive(true, with: .notifyOthersOnDeactivation)
print("Changing modes!")
}catch{
print("error with audio session")
}
recognitionRequest = SFSpeechAudioBufferRecognitionRequest()
guard let inputNode = audioEngine.inputNode else{
fatalError("Audio engine has no input node!")
}
guard let recognitionRequest = recognitionRequest else{
fatalError("Unable to create a speech audio buffer recognition request object")
}
recognitionRequest.shouldReportPartialResults = true
recognitionTask = speechRecognizer?.recognitionTask(with: recognitionRequest, delegate: self)
let recordingFormat = inputNode.outputFormat(forBus: 0)
inputNode.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) { (buffer, when) in
self.recognitionRequest?.append(buffer)
}
audioEngine.prepare()
do{
try audioEngine.start()
delegate?.didStartRecording()
}catch{
print("audioEngine couldn't start because of an error")
}
What happens when I run this code is that it records the voice and transcribes it successfully. However the ping is never played. The two(non-error) print statements I have in there fire in the order:
Changing modes!
Sound played!
So to my understanding, the reason the ping sound isn't being played is because by the time it actually completes I've already changed the audio session category from playback to record. Just to verify this is true, I tried removing everything but the sound services ping and it plays the sound as expected.
So my question is what is the best way to bypass the asynchronous nature of the AudioServicesPlaySystemSound call? I've experimented with trying to pass self into the completion function so I could have it trigger a function in my class which then runs the recording chunk. However I haven't been able to figure out how one actually goes about converting self to an UnsafeMutableRawPointer so it can be passed as clientData. Furthermore, even if I DID know how to do that, I'm not sure if it's even a good idea or the intended use of that parameter.
Alternatively, I could probably solve this problem by relying on something like notification center. But once again that just seems like a very clunky way of solving the problem that I'm going to end up regretting later.
Does anyone know what the correct way to handle this type of situation is?
Update:
As per Gruntcake's request, here is my attempt to access self in the completion block.
First I create a userData constant which is an UnsafeMutableRawPointer to self:
var me = self
let userData = withUnsafePointer(to: &me) { ptr in
return unsafeBitCast(ptr, to: UnsafeMutableRawPointer.self)
Next I use that constant in my callback block, and attempt to access self from it:
AudioServicesAddSystemSoundCompletion(soundId, nil, nil, {(sounded,me) -> Void in
AudioServicesDisposeSystemSoundID(sounded)
let myself = Unmanaged<myclassname>.fromOpaque(me!).takeRetainedValue()
myself.doOtherStuff()
print("Sound played!")}, userData)
Your attempt to call doOtherStuff() in the completion block is a correct approach (the only other one is notifications, those are the only two options)
What is complicating it in this case is the bridging from Obj-C to Swift that is necessary. Code to do that is:
let myData = unsafeBitCast(self, UnsafeMutablePointer<Void>.self)
AudioServicesAddSystemSoundCompletion(YOUR_SOUND_ID, CFRunLoopGetMain(), kCFRunLoopDefaultMode,{ (mSound, mVoid) in
let me = unsafeBitCast(mVoid, YOURCURRENTCLASS.self)
//me it is your current object so if yo have a variable like
// var someVar you can do
print(me.someVar)
}, myData)
Credit: This code was taken from an answer to this question, though it is not the accepted answer:
How do I implement AudioServicesSystemSoundCompletionProc in Swift?
When using an AVAudioPlayerNode to schedule a short buffer to play immediately on a touch event ("Touch Up Inside"), I've noticed audible glitches / artifacts on playback while testing. The audio does not glitch at all in iOS simulator, however there is audible distortion on playback when I run the app on an actual iOS device. The audible distortion occurs randomly (the triggered sound will sometimes sound great, while other times it sounds distorted)
I've tried using different audio files, file formats, and preparing the buffer for playback using the prepareWithFrameCount method, but unfortunately the result is always the same and I'm stuck wondering what could be going wrong..
I've stripped the code down to globals for clarity and simplicity. Any help or insight would be greatly appreciated. This is my first attempt at developing an iOS app and my first question posted on Stack Overflow.
let filePath = NSBundle.mainBundle().pathForResource("BD_withSilence", ofType: "caf")!
let fileURL: NSURL = NSURL(fileURLWithPath: filePath)!
var error: NSError?
let file = AVAudioFile(forReading: fileURL, error: &error)
let fileFormat = file.processingFormat
let frameCount = UInt32(file.length)
let buffer = AVAudioPCMBuffer(PCMFormat: fileFormat, frameCapacity: frameCount)
let audioEngine = AVAudioEngine()
let playerNode = AVAudioPlayerNode()
func startEngine() {
var error: NSError?
file.readIntoBuffer(buffer, error: &error)
audioEngine.attachNode(playerNode)
audioEngine.connect(playerNode, to: audioEngine.mainMixerNode, format: buffer.format)
audioEngine.prepare()
func start() {
var error: NSError?
audioEngine.startAndReturnError(&error)
}
start()
}
startEngine()
let frameCapacity = AVAudioFramePosition(buffer.frameCapacity)
let frameLength = buffer.frameLength
let sampleRate: Double = 44100.0
func play() {
func scheduleBuffer() {
playerNode.scheduleBuffer(buffer, atTime: nil, options: AVAudioPlayerNodeBufferOptions.Interrupts, completionHandler: nil)
playerNode.prepareWithFrameCount(frameLength)
}
if playerNode.playing == false {
scheduleBuffer()
let time = AVAudioTime(sampleTime: frameCapacity, atRate: sampleRate)
playerNode.playAtTime(time)
}
else {
scheduleBuffer()
}
}
// triggered by a "Touch Up Inside" event on a UIButton in my ViewController
#IBAction func triggerPlay(sender: AnyObject) {
play()
}
Update:
Ok I think I've identified the source of the distortion: the volume of the node(s) is too great at output and causes clipping. By adding these two lines in my startEngine function, the distortion no longer occurred:
playerNode.volume = 0.8
audioEngine.mainMixerNode.volume = 0.8
However, I'm still don't know why I need to lower the output- my audio file itself does not clip. I'm guessing that it might be a result of the way that the AVAudioPlayerNodeBufferOptions.Interrupts is implemented. When a buffer interrupts another buffer, could there be an increase in output volume as a result of the interruption, causing output clipping? I'm still looking for a solid understanding as to why this occurs.. If anyone is willing/able to provide any clarification about this that would be fantastic!
Not sure if this is the problem you experienced in 2015, it may be the same issue that #suthar experienced in 2018.
I experienced a very similar problem and was due to the fact that the sampleRate on the device is different to the simulator. On macOS it is 44100 and on iOS Devices (late model ones) it is 48000.
So when you fill your buffer with 44100 samples on a 48000 device, you get 3900 samples of silence. When played back it doesn't sound like silence, it sounds like a glitch.
I used the mainMixer format when connecting my playerNode and also when creating my pcmBuffer. Don't refer to 48000 or 44100 anywhere in the code.
audioEngine.attach( playerNode)
audioEngine.connect( playerNode, to:mixerNode, format:mixerNode.outputFormat(forBus:0))
let pcmBuffer = AVAudioPCMBuffer( pcmFormat:SynthEngine.shared.audioEngine.mainMixerNode.outputFormat( forBus:0),
frameCapacity:AVAudioFrameCount(bufferSize))