I was following this question, but the tone that I try to play with an AVAudioPCMBuffer does not play. The code is pretty simple:
class Player: NSObject {
var engine = AVAudioEngine()
var player = AVAudioPlayerNode()
var mixer: AVAudioMixerNode!
var buffer: AVAudioPCMBuffer!
override init() {
mixer = engine.mainMixerNode
buffer = AVAudioPCMBuffer(pcmFormat: player.outputFormat(forBus: 0), frameCapacity: 100)
buffer.frameLength = 100
let sr = mixer.outputFormat(forBus: 0).sampleRate
let nChannels = mixer.outputFormat(forBus: 0).channelCount
var i = 0
while i < Int(buffer.frameLength) {
let val = sin(441 * Double(i) * Double.pi / sr)
buffer.floatChannelData?.pointee[i] = Float(val * 0.5)
i += Int(nChannels)
}
engine.attach(player)
engine.connect(player, to: mixer, format: player.outputFormat(forBus: 0))
engine.prepare()
}
func play() {
do {
try engine.start()
} catch {
print(error)
}
player.scheduleBuffer(buffer, at: nil, options: .loops) {
print("Played!")
}
player.play()
}
}
For some reason, though, the iPhone does not make any sound. In my ViewController, I have this:
class ViewController: UIViewController {
var player = Player()
override func viewDidAppear(_ animated: Bool) {
player.play()
}
}
As you can see, player is a class variable, so it should not be deallocated from memory.
When I run the app on my physical device (iPhone 6s iOS 11), it does not work, yet it does work on the simulator. Why is this not making any sound, and how can I fix it?
Thanks in advance!
Make sure your device is not on silent mode.
I just created a project that you can download for testing by yourself: https://github.com/mugx/TestSound
Related
I implemented the AudioKit "MICROPHONE ANALYSIS" example https://audiokit.io/examples/MicrophoneAnalysis/ in my App.
I want to analyze the microphone input frequency and then play the correct note which is near the frequency which was determined.
Normally the sound output is the speaker or a Bluetooth device connected to my iPhone but after implementing the "MICROPHONE ANALYSIS" example the sound output changed to the tiny little speaker on the top of the iPhone which is normally used when you get a call.
How can I switch to the "normal" speaker or to the connected Bluetooth device like before?
var mic: AKMicrophone!
var tracker: AKFrequencyTracker!
var silence: AKBooster!
func initFrequencyTracker() {
AKSettings.audioInputEnabled = true
mic = AKMicrophone()
tracker = AKFrequencyTracker(mic)
silence = AKBooster(tracker, gain: 0)
}
func deinitFrequencyTracker() {
plotTimer.invalidate()
do {
try AudioKit.stop()
AudioKit.output = nil
} catch {
print(error)
}
}
func initPlotTimer() {
AudioKit.output = silence
do {
try AudioKit.start()
} catch {
AKLog("AudioKit did not start!")
}
setupPlot()
plotTimer = Timer.scheduledTimer(timeInterval: 0.1, target: self, selector: #selector(updatePlotUI), userInfo: nil, repeats: true)
}
func setupPlot() {
let plot = AKNodeOutputPlot(mic, frame: audioInputPlot.bounds)
plot.translatesAutoresizingMaskIntoConstraints = false
plot.alpha = 0.3
plot.plotType = .rolling
plot.shouldFill = true
plot.shouldCenterYAxis = false
plot.shouldMirror = true
plot.color = UIColor(named: uiFarbe)
audioInputPlot.addSubview(plot)
// Pin the AKNodeOutputPlot to the audioInputPlot
var constraints = [plot.leadingAnchor.constraint(equalTo: audioInputPlot.leadingAnchor)]
constraints.append(plot.trailingAnchor.constraint(equalTo: audioInputPlot.trailingAnchor))
constraints.append(plot.topAnchor.constraint(equalTo: audioInputPlot.topAnchor))
constraints.append(plot.bottomAnchor.constraint(equalTo: audioInputPlot.bottomAnchor))
constraints.forEach { $0.isActive = true }
}
#objc func updatePlotUI() {
if tracker.amplitude > 0.1 {
let trackerFrequency = Float(tracker.frequency)
guard trackerFrequency < 7_000 else {
// This is a bit of hack because of modern Macbooks giving super high frequencies
return
}
var frequency = trackerFrequency
while frequency > Float(noteFrequencies[noteFrequencies.count - 1]) {
frequency /= 2.0
}
while frequency < Float(noteFrequencies[0]) {
frequency *= 2.0
}
var minDistance: Float = 10_000.0
var index = 0
for i in 0..<noteFrequencies.count {
let distance = fabsf(Float(noteFrequencies[i]) - frequency)
if distance < minDistance {
index = i
minDistance = distance
}
}
// let octave = Int(log2f(trackerFrequency / frequency))
frequencyLabel.text = String(format: "%0.1f", tracker.frequency)
if frequencyTranspose(note: notesToTanspose[index]) != droneLabel.text {
note = frequencyTranspose(note: notesToTanspose[index])
droneLabel.text = note
DispatchQueue.main.asyncAfter(deadline: .now() + 0.03, execute: {
self.prepareSinglePlayerFirstForStart(note: self.note)
self.startSinglePlayer()
})
}
}
}
func frequencyTranspose(note: String) -> String {
var indexNote = notesToTanspose.firstIndex(of: note)!
let chosenInstrument = UserDefaults.standard.object(forKey: "whichInstrument") as! String
if chosenInstrument == "Bb" {
if indexNote + 2 >= notesToTanspose.count {
indexNote -= 12
}
return notesToTanspose[indexNote + 2]
} else if chosenInstrument == "Eb" {
if indexNote - 3 < 0 {
indexNote += 12
}
return notesToTanspose[indexNote - 3]
} else {
return note
}
}
It's a good practice to control the session settings, so start by creating a method in your application to take care of that during initialisation.
Following up, there's an example where I set a category and the desired options:
func start() {
do {
let session = AVAudioSession.sharedInstance()
try session.setCategory(.playAndRecord, options: .defaultToSpeaker)
try session.setActive(true, options: .notifyOthersOnDeactivation)
try session.overrideOutputAudioPort(AVAudioSession.PortOverride.speaker)
try AudioKit.start()
} catch {
// your error handler
}
}
You can call the method start where you make the call to AudioKit.Start() in initPlotTimer.
The example above is using the AVAudioSession, which I believe is what AKSettings wraps (please feel free to edit my answer to not mislead future readers, as I'm not looking at the AudioKit source-code at the moment).
Now that AVAudioSession is exposed, let's stick with the method offered by AudioKit since that's what you're dealing with.
Here's another example using AKSettings:
func start() {
do {
AKSettings.channelCount = 2
AKSettings.ioBufferDuration = 0.002
AKSettings.audioInputEnabled = true
AKSettings.bufferLength = .medium
AKSettings.defaultToSpeaker = true
// check docs for other options and settings
try AKSettings.setSession(category: .playAndRecord, with: [.defaultToSpeaker, .allowBluetooth])
try AudioKit.start()
} catch {
// your handler
}
}
Have in mind that you don't necessarily have to call it start, or run AudioKit's start method, I'm just exposing the initialisation phase, to make it readable to you and other use-cases.
Reference:
https://developer.apple.com/documentation/avfoundation/avaudiosession/categoryoptions
https://audiokit.io/docs/Classes/AKSettings.html
When I add a bunch (20-40) samples playing and overlapping eachother simultaneously sometimes it starts getting distorted and then some waving, oscillating, and clicking begins to happen. A similar sound happens when the samples are playing the the app crashes - sounds like an abrupt, crunchy halt.
Notice the waviness begins between 0:05 and 0:10; nasty clicks start around 0:15.
Listen Here
How can I make it smoother? I am spawning AKPlayer objects (from 4.1) that play 4-8 second .wav files. Those go into AKBoosters which go into AKMixers which go into the final AKMixer for output.
Edit:
Many PenAudioNodes get plugged into the mixer of the AudioReceiver singleton.
Here's my AudioReceiver singleton:
class AudioReceiver {
static var sharedInstance = AudioReceiver()
private var audioNodes = [UUID : AudioNode]()
private let mixer = AKMixer()
private let queue = DispatchQueue(label: "audio-queue")
//MARK: - Setup & Teardown
init() {
AudioKit.output = mixer //peakLimiter
AudioKit.start()
}
//MARK: - Public
func audioNodeBegan(_ message : AudioNodeMessage) {
queue.async {
var audioNode: AudioNode?
switch message.senderType {
case .pen:
audioNode = PenAudioNode()
case .home:
audioNode = LoopingAudioNode(with: AudioHelper.homeLoopFile())
default:
break
}
if let audioNode = audioNode {
self.audioNodes[message.senderId] = audioNode
self.mixer.connect(input: audioNode.output)
audioNode.start(message)
}
}
}
func audioNodeMoved(_ message : AudioNodeMessage) {
queue.async {
if let audioNode = self.audioNodes[message.senderId] {
audioNode.update(message)
}
}
}
func audioNodeEnded(_ message : AudioNodeMessage) {
queue.async {
if let audioNode = self.audioNodes[message.senderId] {
audioNode.stop(message)
}
self.audioNodes[message.senderId] = nil
}
}
}
Here's my PenAudioNode:
class PenAudioNode {
fileprivate var mixer: AKMixer?
fileprivate var playersBoosters = [AKPlayer : AKBooster]()
fileprivate var finalOutput: AKNode?
fileprivate let file: AKAudioFile = AudioHelper.randomBellSampleFile()
//MARK: - Setup & Teardown
init() {
mixer = AKMixer()
finalOutput = mixer!
}
}
extension PenAudioNode: AudioNode {
var output: AKNode {
return finalOutput!
}
func start(_ message: AudioNodeMessage) {
}
func update(_ message: AudioNodeMessage) {
if let velocity = message.velocity {
let newVolume = Swift.min((velocity / 50) + 0.1, 1)
mixer!.volume = newVolume
}
if let isClimactic = message.isClimactic, isClimactic {
let player = AKPlayer(audioFile: file)
player.completionHandler = { [weak self] in
self?.playerCompleted(player)
}
let booster = AKBooster(player)
playersBoosters[player] = booster
booster.rampTime = 1
booster.gain = 0
mixer!.connect(input: booster)
player.play()
booster.gain = 1
}
}
func stop(_ message: AudioNodeMessage) {
for (_, booster) in playersBoosters {
booster.gain = 0
}
DispatchQueue.global().asyncAfter(deadline: DispatchTime.now() + 1) {
self.mixer!.stop()
self.output.disconnectOutput()
}
}
private func playerCompleted(_ player: AKPlayer) {
playersBoosters.removeValue(forKey: player)
}
}
This sounds like you are not releasing objects and you are eventually overloading the audio engine with too many instances of processing nodes connected in the graph. In particular not releasing AKBoosters will cause an issue like this. I can't really tell what your code is doing, but if you are spawning objects without releasing them properly, it will lead to garbled audio.
You want to conserve objects as much as possible and make sure you are using the absolute minimum amount of AKNode based processing.
There are various ways to debug this, but you can start by printing out the current state of the AVAudioEngine:
AudioKit.engine.description
That will show how many nodes you have connected in the graph at any given moment.
I want to allow background audio while the app is not in focus. I currently have this code, which should allow that:
do {
try AKSettings.setSession(category: .playback, with: .mixWithOthers)
} catch {
print("error")
}
AKSettings.playbackWhileMuted = true
I also have the setting 'Audio, Airplay and Picture in Picture' enabled in capabilities settings. However, when I press the home button on my device the audio doesn't keep playing. What am I doing wrong? I am using AudioKit to produce sounds if that matters.
I am using a singleton to house all of the AudioKit components which I named AudioPlayer.swift. Here is what I have in my AudioPlayer.swift singleton file:
class AudioPlayer: NSObject {
var currentFrequency = String()
var soundIsPlaying = false
var leftOscillator = AKOscillator()
var rightOscillator = AKOscillator()
var rain = try! AKAudioFile()
var rainPlayer: AKAudioPlayer!
var envelope = AKAmplitudeEnvelope()
override init() {
super.init()
do {
try AKSettings.setSession(category: .playback, with: .mixWithOthers)
} catch {
print("error")
}
AKSettings.playbackWhileMuted = true
AudioKit.output = envelope
AudioKit.start()
}
func setupFrequency(left: AKOscillator, right: AKOscillator, frequency: String) {
currentFrequency = frequency
leftOscillator = left
rightOscillator = right
let leftPanner = AKPanner(leftOscillator)
leftPanner.pan = -1
let rightPanner = AKPanner(rightOscillator)
rightPanner.pan = 1
//Set up rain and rainPlayer
do {
rain = try AKAudioFile(readFileName: "rain.wav")
rainPlayer = try AKAudioPlayer(file: rain, looping: true, deferBuffering: false, completionHandler: nil)
} catch { print(error) }
let mixer = AKMixer(leftPanner, rightPanner, rainPlayer)
//Put mixer in sound envelope
envelope = AKAmplitudeEnvelope(mixer)
envelope.attackDuration = 2.0
envelope.decayDuration = 0
envelope.sustainLevel = 1
envelope.releaseDuration = 0.2
//Start AudioKit stuff
AudioKit.output = envelope
AudioKit.start()
leftOscillator.start()
rightOscillator.start()
rainPlayer.start()
envelope.start()
soundIsPlaying = true
}
}
And here is an example of one of my sound effect view controllers, which reference the AudioKit singleton to send it a certain frequency (I have about a dozen of these view controllers, each with its own frequency settings):
class CalmView: UIViewController {
let leftOscillator = AKOscillator()
let rightOscillator = AKOscillator()
override func viewDidLoad() {
super.viewDidLoad()
leftOscillator.amplitude = 0.3
leftOscillator.frequency = 220
rightOscillator.amplitude = 0.3
rightOscillator.frequency = 230
}
#IBAction func playSound(_ sender: Any) {
if shared.soundIsPlaying == false {
AudioKit.stop()
shared.setupFrequency(left: leftOscillator, right: rightOscillator, frequency: "Calm")
} else if shared.soundIsPlaying == true && shared.currentFrequency != "Calm" {
AudioKit.stop()
shared.leftOscillator.stop()
shared.rightOscillator.stop()
shared.rainPlayer.stop()
shared.envelope.stop()
shared.setupFrequency(left: leftOscillator, right: rightOscillator, frequency: "Calm")
} else {
shared.soundIsPlaying = false
shared.envelope.stop()
}
}
}
I instantiated the AudioPlayer singleton in my ViewController.swift file.
It depends on when you are doing your configuration in relation to when AudioKit is started. If you're using AudioKit you should be using its AKSettings to manage your session category. Basically not only the playback category but also mixWithOthers. By default, does this:
/// Set the audio session type
#objc open static func setSession(category: SessionCategory,
with options: AVAudioSessionCategoryOptions = [.mixWithOthers]) throws {
So you'd do something like this in your ViewController:
do {
if #available(iOS 10.0, *) {
try AKSettings.setSession(category: .playAndRecord, with: [.defaultToSpeaker, .allowBluetooth, .allowBluetoothA2DP])
} else {
// Fallback on earlier versions
}
} catch {
print("Errored setting category.")
}
So I think its a matter of getting that straight. It might also help to have inter-app audio set up. If you still have trouble and provide more information, I can help more, but this is as good an answer as I can muster based on the info you've given so far.
I'm trying to make my iphone play a tune without using prerecorded files. What are my options here? AVAudioEngine, AudioKit? I've looked at them, but the learning curve is relatively steep for something I'm hoping is easy. They also seem like tools for creating sound effect given a PCM buffer window.
I'd like to be able to do something like
pitchCreator.play(["C4", "E4", "G4"], durations: [1, 1, 1])
Preferrably sounding like an instrument or at least not like a pure sine wave.
EDIT: The below code has been replaced by AudioKit
To anyone wondering this; I did make it work (kind of) using code similar to the one below.
class PitchCreator {
var engine: AVAudioEngine
var player: AVAudioPlayerNode
var mixer: AVAudioMixerNode
var buffer: AVAudioPCMBuffer
init() {
engine = AVAudioEngine()
player = AVAudioPlayerNode()
mixer = engine.mainMixerNode;
buffer = AVAudioPCMBuffer(PCMFormat: player.outputFormatForBus(0), frameCapacity: 100)
buffer.frameLength = 4096
engine.attachNode(player)
engine.connect(player, to: mixer, format: player.outputFormatForBus(0))
}
func play(frequency: Float) {
let signal = self.createSignal(frequency, amplitudes: [1.0, 0.5, 0.3, 0.1], bufferSize: Int(buffer.frameLength), sampleRate: Float(mixer.outputFormatForBus(0).sampleRate))
for i in 0 ..< signal.count {
buffer.floatChannelData.memory[i] = 0.5 * signal[i]
}
do {
try engine.start()
player.play()
player.scheduleBuffer(buffer, atTime: nil, options: .Loops, completionHandler: nil)
} catch {}
}
func stop() {
engine.stop()
player.stop()
}
func createSignal(frequency: Float, amplitudes: [Float], bufferSize: Int, sampleRate: Float) -> [Float] {
let π = Float(M_PI)
let T = sampleRate / frequency
var x = [Float](count: bufferSize, repeatedValue: 0.0)
for k in 0 ..< x.count {
for h in 0 ..< amplitudes.count {
x[k] += amplitudes[h] * sin(2.0 * π * Float(h + 1) * Float(k) / T)
}
}
return x
}
}
But it doesn't sound good enough so I've gone with sampling the notes I need and just use AVAudioPlayer instead to play them.
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I tried to use the super.init before initializing the objects and tried placing self in front of my objects. Can anyone tell me what I am doing wrong?
This is the code I currently have:
//
// RecordController.swift
// TuneUpV2
//
// Created by Don Nijssen on 05-05-15.
// Copyright (c) 2015 Don Nijssen. All rights reserved.
//
import UIKit
import AVFoundation
class RecordController: NSObject {
var audioEngine: AVAudioEngine
var audioInputNode : AVAudioInputNode
var audioPlayerNode: AVAudioPlayerNode
var audioMixerNode: AVAudioMixerNode
var audioBuffer: AVAudioPCMBuffer
override init(){
audioEngine = AVAudioEngine()
audioPlayerNode = AVAudioPlayerNode()
audioInputNode = AVAudioInputNode()
audioMixerNode = AVAudioMixerNode()
audioBuffer = AVAudioPCMBuffer()
audioMixerNode = audioEngine.mainMixerNode
let frameLength = UInt32(256)
audioBuffer = AVAudioPCMBuffer(PCMFormat: audioPlayerNode.outputFormatForBus(0), frameCapacity: frameLength)
audioBuffer.frameLength = frameLength
audioInputNode = audioEngine.inputNode
audioInputNode.installTapOnBus(0, bufferSize:frameLength, format: audioInputNode.outputFormatForBus(0), block: {(buffer, time) in
//let channels = UnsafeArray(start: buffer.floatChannelData, length: Int(buffer.format.channelCount))
//let floats = UnsafeArray(start: channels[0], length: Int(buffer.frameLength))
for var i = 0; i < Int(self.audioBuffer.frameLength); i+=Int(self.audioMixerNode.outputFormatForBus(0).channelCount)
{
// doing my real time stuff
//self.audioBuffer.floatChannelData.memory[i] = floats[i];
println(self.audioEngine.inputNode.rate);
}
})
// setup audio engine
audioEngine.attachNode(audioPlayerNode)
audioEngine.connect(audioPlayerNode, to: audioMixerNode, format: audioPlayerNode.outputFormatForBus(0))
}
func start()
{
audioEngine.startAndReturnError(nil)
// play player and buffer
audioPlayerNode.play()
audioPlayerNode.scheduleBuffer(audioBuffer, atTime: nil, options: .Loops, completionHandler: nil)
}
func stop()
{
audioEngine.stop();
audioPlayerNode.stop();
}
}
Where exactly did you call super.init()?
This works in my playground:
override init(){
audioEngine = AVAudioEngine()
audioPlayerNode = AVAudioPlayerNode()
audioInputNode = AVAudioInputNode()
audioMixerNode = AVAudioMixerNode()
audioBuffer = AVAudioPCMBuffer()
// Call super.init() immediately after all subclass properties are initialized
super.init()
audioMixerNode = audioEngine.mainMixerNode
let frameLength = UInt32(256)
audioBuffer = AVAudioPCMBuffer(PCMFormat: audioPlayerNode.outputFormatForBus(0), frameCapacity: frameLength)
audioBuffer.frameLength = frameLength
audioInputNode = audioEngine.inputNode
audioInputNode.installTapOnBus(0, bufferSize:frameLength, format: audioInputNode.outputFormatForBus(0), block: {(buffer, time) in
//let channels = UnsafeArray(start: buffer.floatChannelData, length: Int(buffer.format.channelCount))
//let floats = UnsafeArray(start: channels[0], length: Int(buffer.frameLength))
for var i = 0; i < Int(self.audioBuffer.frameLength); i+=Int(self.audioMixerNode.outputFormatForBus(0).channelCount)
{
// doing my real time stuff
//self.audioBuffer.floatChannelData.memory[i] = floats[i];
println(self.audioEngine.inputNode.rate);
}
})
// setup audio engine
audioEngine.attachNode(audioPlayerNode)
audioEngine.connect(audioPlayerNode, to: audioMixerNode, format: audioPlayerNode.outputFormatForBus(0))
}
Hope this helps
If your class is a subclass and you have non optional properties then you have to have give initial value to all properties. Since it is a sub class of NSObject and you are overriding the init method from its parent class you have to provide super.init() call before initialising properties.
class RecordController: NSObject {
var audioEngine: AVAudioEngine
var audioInputNode : AVAudioInputNode
var audioPlayerNode: AVAudioPlayerNode
var audioMixerNode: AVAudioMixerNode
var audioBuffer: AVAudioPCMBuffer
override init(){
super.init()
self.audioEngine = AVAudioEngine()
self.audioPlayerNode = AVAudioPlayerNode()
self.audioInputNode = AVAudioInputNode()
self.audioMixerNode = AVAudioMixerNode()
self.audioBuffer = AVAudioPCMBuffer()
audioMixerNode = audioEngine.mainMixerNode
let frameLength = UInt32(256)
audioBuffer = AVAudioPCMBuffer(PCMFormat: audioPlayerNode.outputFormatForBus(0), frameCapacity: frameLength)
audioBuffer.frameLength = frameLength
audioInputNode = audioEngine.inputNode
audioInputNode.installTapOnBus(0, bufferSize:frameLength, format: audioInputNode.outputFormatForBus(0), block: {(buffer, time) in
//let channels = UnsafeArray(start: buffer.floatChannelData, length: Int(buffer.format.channelCount))
//let floats = UnsafeArray(start: channels[0], length: Int(buffer.frameLength))
for var i = 0; i < Int(self.audioBuffer.frameLength); i+=Int(self.audioMixerNode.outputFormatForBus(0).channelCount)
{
// doing my real time stuff
//self.audioBuffer.floatChannelData.memory[i] = floats[i];
println(self.audioEngine.inputNode.rate);
}
})
// setup audio engine
audioEngine.attachNode(audioPlayerNode)
audioEngine.connect(audioPlayerNode, to: audioMixerNode, format: audioPlayerNode.outputFormatForBus(0))
}
func start()
{
audioEngine.startAndReturnError(nil)
// play player and buffer
audioPlayerNode.play()
audioPlayerNode.scheduleBuffer(audioBuffer, atTime: nil, options: .Loops, completionHandler: nil)
}
func stop()
{
audioEngine.stop();
audioPlayerNode.stop();
}
}
As per Apple documentation.