Right now I'm using AVAudioEngine, with AVAudioPlayer, AVAudioFile, AVAudioPCMBuffer to play couple a compressed soundtrack (m4a). My problem is that if the soundtrack is 40MB uncompressed and 1.8 in m4a when I load the sound in the buffer, the memory usage jump by 40MB (the uncompressed size of the file). How can I optimise that to use as little memory as possible?
Thanks.
let loopingBuffer : AVAudioPCMBuffer!
do{ let loopingFile = try AVAudioFile(forReading: fileURL)
loopingBuffer = AVAudioPCMBuffer(pcmFormat: loopingFile.processingFormat, frameCapacity: UInt32(loopingFile.length))!
do {
try loopingFile.read(into: loopingBuffer)
} catch
{
print(error)
}
} catch
{
print(error)
}
// player is AVAudioPlayerNode
player.scheduleBuffer(loopingBuffer, at: nil, options: [.loops])
Well, as a workaround, I decided to create a wrapper to split the audio into chunk of few second and playing and buffering them one at the time into the AVAudioPlayerNode.
As a result only a few seconds are RAM (twice that when buffering) at any time.
It brung the memory usage for my use case from 350Mo to less than 50Mo.
Here is the code, don't hesitate to use it or improve it (it's a first version). Any comments are welcome!
import Foundation
import AVFoundation
public class AVAudioStreamPCMPlayerWrapper
{
public var player: AVAudioPlayerNode
public let audioFile: AVAudioFile
public let bufferSize: TimeInterval
public let url: URL
public private(set) var loopingCount: Int = 0
/// Equal to the repeatingTimes passed in the initialiser.
public let numberOfLoops: Int
/// The time passed in the initialisation parameter for which the player will preload the next buffer to have a smooth transition.
/// The default value is 1s.
/// Note : better not go under 1s since the buffering mecanism can be triggered with a relative precision.
public let preloadTime: TimeInterval
public private(set) var scheduled: Bool = false
private let framePerBuffer: AVAudioFrameCount
/// To identify the the schedule cycle we are executed
/// Since the thread work can't be stopped when they are scheduled
/// we need to be sure that the execution of the work is done for the current playing cycle.
/// For exemple if the player has been stopped and restart before the async call has executed.
private var scheduledId: Int = 0
/// the time since the track started.
private var startingDate: Date = Date()
/// The date used to measure the difference between the moment the buffering should have occure and the actual moment it did.
/// Hence, we can adjust the next trigger of the buffering time to prevent the delay to accumulate.
private var lastBufferingDate = Date()
/// This class allow us to play a sound, once or multiple time without overloading the RAM.
/// Instead of loading the full sound into memory it only reads a segment of it at a time, preloading the next segment to avoid stutter.
/// - Parameters:
/// - url: The URL of the sound to be played.
/// - bufferSize: The size of the segment of the sound being played. Must be greater than preloadTime.
/// - repeatingTimes: How many time the sound must loop (0 it's played only once 1 it's played twice : repeating once)
/// -1 repeating indéfinitly.
/// - preloadTime: 1 should be the minimum value since the preloading mecanism can be triggered not precesily on time.
/// - Throws: Throws the error the AVAudioFile would throw if it couldn't be created with the URL passed in parameter.
public init(url: URL, bufferSize: TimeInterval, isLooping: Bool, repeatingTimes: Int = -1, preloadTime: TimeInterval = 1)throws
{
self.url = url
self.player = AVAudioPlayerNode()
self.bufferSize = bufferSize
self.numberOfLoops = repeatingTimes
self.preloadTime = preloadTime
try self.audioFile = AVAudioFile(forReading: url)
framePerBuffer = AVAudioFrameCount(audioFile.fileFormat.sampleRate*bufferSize)
}
public func scheduleBuffer()
{
scheduled = true
scheduledId += 1
scheduleNextBuffer(offset: preloadTime)
}
public func play()
{
player.play()
startingDate = Date()
scheduleNextBuffer(offset: preloadTime)
}
public func stop()
{
reset()
scheduleBuffer()
}
public func reset()
{
player.stop()
player.reset()
scheduled = false
audioFile.framePosition = 0
}
/// The first time this method is called the timer is offset by the preload time, then since the timer is repeating and has already been offset
/// we don't need to offset it again the second call.
private func scheduleNextBuffer(offset: TimeInterval)
{
guard scheduled else {return}
if audioFile.length == audioFile.framePosition
{
guard numberOfLoops == -1 || loopingCount < numberOfLoops else {return}
audioFile.framePosition = 0
loopingCount += 1
}
let buffer = AVAudioPCMBuffer(pcmFormat: audioFile.processingFormat, frameCapacity: framePerBuffer)!
let frameCount = min(framePerBuffer, AVAudioFrameCount(audioFile.length - audioFile.framePosition))
print("\(audioFile.framePosition/48000) \(url.relativeString)")
do
{
try audioFile.read(into: buffer, frameCount: frameCount)
DispatchQueue.global().async(group: nil, qos: DispatchQoS.userInteractive, flags: .enforceQoS) { [weak self] in
self?.player.scheduleBuffer(buffer, at: nil, options: .interruptsAtLoop)
self?.player.prepare(withFrameCount: frameCount)
}
let nextCallTime = max(TimeInterval( Double(frameCount) / audioFile.fileFormat.sampleRate) - offset, 0)
planNextPreloading(nextCallTime: nextCallTime)
} catch
{
print("audio file read error : \(error)")
}
}
private func planNextPreloading(nextCallTime: TimeInterval)
{
guard self.player.isPlaying else {return}
let id = scheduledId
lastBufferingDate = Date()
DispatchQueue.global().asyncAfter(deadline: .now() + nextCallTime, qos: DispatchQoS.userInteractive) { [weak self] in
guard let self = self else {return}
guard id == self.scheduledId else {return}
let delta = -(nextCallTime + self.lastBufferingDate.timeIntervalSinceNow)
self.scheduleNextBuffer(offset: delta)
}
}
}
Related
I'm subclassing InputStream from iOS Foundation SDK for my needs. I need to implement functionality that worker thread can sleep until data appear in the stream. The test I'm using to cover the functionality is below:
func testStreamWithRunLoop() {
let inputStream = BLEInputStream() // custom input stream subclass
inputStream.delegate = self
let len = Int.random(in: 0..<100)
let randomData = randData(length: len) // random data generation
let tenSeconds = Double(10)
let oneSecond = TimeInterval(1)
runOnBackgroundQueueAfter(oneSecond) {
inputStream.accept(randomData) // input stream receives the data
}
let dateInFuture = Date(timeIntervalSinceNow: tenSeconds) // time in 10 sec
inputStream.schedule(in: .current, forMode: RunLoop.Mode.default) //
RunLoop.current.run(until: dateInFuture) // wait for data appear in input stream
XCTAssertTrue(dateInFuture.timeIntervalSinceNow > 0, "Timeout. RunLoop didn't exit in 1 sec. ")
}
Here the overriden methods of InputStream
public override func schedule(in aRunLoop: RunLoop, forMode mode: RunLoop.Mode) {
self.runLoop = aRunLoop // save RunLoop object
var context = CFRunLoopSourceContext() // make context
self.runLoopSource = CFRunLoopSourceCreate(nil, 0, &context) // make source
let cfloopMode: CFRunLoopMode = CFRunLoopMode(mode as CFString)
CFRunLoopAddSource(aRunLoop.getCFRunLoop(), self.runLoopSource!, cfloopMode)
}
public func accept(_ data: Data) {
guard data.count > 0 else { return }
self.data += data
delegate?.stream?(self, handle: .hasBytesAvailable)
if let runLoopSource {
CFRunLoopSourceSignal(runLoopSource)
}
if let runLoop {
CFRunLoopWakeUp(runLoop.getCFRunLoop())
}
}
But calling CFRunLoopSourceSignal(runLoopSource) and CFRunLoopWakeUp(runLoop.getCFRunLoop()) not get exit from runLoop.
Does anybody know where I'm mistaking ?
Thanks all!
PS: Here the Xcode project on GitHub
Finally I figured out some issues with my code.
First of all I need to remove CFRunLoopSource object from run loop CFRunLoopRemoveSource(). In according with documentation if RunLoop has no input sources then it exits immediately.
public func accept(_ data: Data) {
guard data.count > 0 else { return }
self.data += data
delegate?.stream?(self, handle: .hasBytesAvailable)
if let runLoopSource, let runLoop, let runLoopMode {
CFRunLoopRemoveSource(runLoop.getCFRunLoop(), runLoopSource, runLoopMode)
}
if let runLoop {
CFRunLoopWakeUp(runLoop.getCFRunLoop())
}
}
Second issue is related that I used XCTest environment and it's RunLoop didn't exit for some reasons (Ask the community for help).
I used real application environment and created Thread subclass to check my implementation. The thread by default has run loop without any input sources attached to it. I added input stream to it. And using main thread emulated that stream received data.
Here the Custom Thread implement that runs and sleep until it receive signal from BLEInputStream
class StreamThread: Thread, StreamDelegate {
let stream: BLEInputStream
init(stream: BLEInputStream) {
self.stream = stream
}
override func main() {
stream.delegate = self
stream.schedule(in: .current, forMode: RunLoop.Mode.default)
print("start()")
let tenSeconds = Double(10)
let dateInFuture = Date(timeIntervalSinceNow: tenSeconds)
RunLoop.current.run(until: dateInFuture)
print("after 10 seconds")
}
override func start() {
super.start()
}
func stream(_ aStream: Stream, handle eventCode: Stream.Event) {
if eventCode == .errorOccurred {
print("eventCode == .errorOccurred")
}
else if eventCode == .hasBytesAvailable {
print("eventCode == .hasBytesAvailable")
}
}
}
Here the some UIViewController methods which runs from main thread
override func viewDidAppear(_ animated: Bool) {
super.viewDidAppear(animated)
let baseDate = Date.now
let thread = StreamThread(stream: stream, baseDate: baseDate)
thread.start()
print("main thread pauses at \(Date.now.timeIntervalSince(baseDate))")
Thread.sleep(forTimeInterval: 2)
print("stream accepts Data \(Date.now.timeIntervalSince(baseDate))")
stream.accept(Data([1,2,3]))
}
Here the result:
Everything works as expected - the thread sleeps until input stream receive data. No processor resources consuming.
Although it's allowed to subclass InputStream, there is no good explanation in the documentation how to correctly implement custom InputStream
I am trying to create a spectrogram, like the one in the image, from an audio file using Swift for a macOS app. I am using AppKit but could implement SwiftUI as well. I cam across audio kit and it seems like the perfect library to use for this type of thing, but I have not been able to find any examples of what I am looking for in an of the audio kit repositories, audio kit UI nor the cookbook. Is this something that is possible with audio kit? If so, can anyone help me with this?
Thanks so much!
I have previously tried using apple's example project and changed the code in the AudioSpectrogram + AVCaptureAudioDataOutputSampleBufferDelegate file. The original code is as follows:
extension AudioSpectrogram: AVCaptureAudioDataOutputSampleBufferDelegate {
public func captureOutput(_ output: AVCaptureOutput,
didOutput sampleBuffer: CMSampleBuffer,
from connection: AVCaptureConnection) {
var audioBufferList = AudioBufferList()
var blockBuffer: CMBlockBuffer?
CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(
sampleBuffer,
bufferListSizeNeededOut: nil,
bufferListOut: &audioBufferList,
bufferListSize: MemoryLayout.stride(ofValue: audioBufferList),
blockBufferAllocator: nil,
blockBufferMemoryAllocator: nil,
flags: kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment,
blockBufferOut: &blockBuffer)
guard let data = audioBufferList.mBuffers.mData else {
return
}
/// The _Nyquist frequency_ is the highest frequency that a sampled system can properly
/// reproduce and is half the sampling rate of such a system. Although this app doesn't use
/// `nyquistFrequency` you may find this code useful to add an overlay to the user interface.
if nyquistFrequency == nil {
let duration = Float(CMSampleBufferGetDuration(sampleBuffer).value)
let timescale = Float(CMSampleBufferGetDuration(sampleBuffer).timescale)
let numsamples = Float(CMSampleBufferGetNumSamples(sampleBuffer))
nyquistFrequency = 0.5 / (duration / timescale / numsamples)
}
if self.rawAudioData.count < AudioSpectrogram.sampleCount * 2 {
let actualSampleCount = CMSampleBufferGetNumSamples(sampleBuffer)
let ptr = data.bindMemory(to: Int16.self, capacity: actualSampleCount)
let buf = UnsafeBufferPointer(start: ptr, count: actualSampleCount)
rawAudioData.append(contentsOf: Array(buf))
}
while self.rawAudioData.count >= AudioSpectrogram.sampleCount {
let dataToProcess = Array(self.rawAudioData[0 ..< AudioSpectrogram.sampleCount])
self.rawAudioData.removeFirst(AudioSpectrogram.hopCount)
self.processData(values: dataToProcess)
}
createAudioSpectrogram()
}
func configureCaptureSession() {
// Also note that:
//
// When running in iOS, you must add a "Privacy - Microphone Usage
// Description" entry.
//
// When running in macOS, you must add a "Privacy - Microphone Usage
// Description" entry to `Info.plist`, and check "audio input" and
// "camera access" under the "Resource Access" category of "Hardened
// Runtime".
switch AVCaptureDevice.authorizationStatus(for: .audio) {
case .authorized:
break
case .notDetermined:
sessionQueue.suspend()
AVCaptureDevice.requestAccess(for: .audio,
completionHandler: { granted in
if !granted {
fatalError("App requires microphone access.")
} else {
self.configureCaptureSession()
self.sessionQueue.resume()
}
})
return
default:
// Users can add authorization in "Settings > Privacy > Microphone"
// on an iOS device, or "System Preferences > Security & Privacy >
// Microphone" on a macOS device.
fatalError("App requires microphone access.")
}
captureSession.beginConfiguration()
#if os(macOS)
// Note than in macOS, you can change the sample rate, for example to
// `AVSampleRateKey: 22050`. This reduces the Nyquist frequency and
// increases the resolution at lower frequencies.
audioOutput.audioSettings = [
AVFormatIDKey: kAudioFormatLinearPCM,
AVLinearPCMIsFloatKey: false,
AVLinearPCMBitDepthKey: 16,
AVNumberOfChannelsKey: 1]
#endif
if captureSession.canAddOutput(audioOutput) {
captureSession.addOutput(audioOutput)
} else {
fatalError("Can't add `audioOutput`.")
}
guard
let microphone = AVCaptureDevice.default(.builtInMicrophone,
for: .audio,
position: .unspecified),
let microphoneInput = try? AVCaptureDeviceInput(device: microphone) else {
fatalError("Can't create microphone.")
}
if captureSession.canAddInput(microphoneInput) {
captureSession.addInput(microphoneInput)
}
captureSession.commitConfiguration()
}
/// Starts the audio spectrogram.
func startRunning() {
sessionQueue.async {
if AVCaptureDevice.authorizationStatus(for: .audio) == .authorized {
self.captureSession.startRunning()
}
}
}
}
I got rid of the configureCaptureSession function and replaced the rest of the code to get the following code:
public func captureBuffer() {
var samplesArray:[Int16] = []
let asset = AVAsset(url: audioFileUrl)
let reader = try! AVAssetReader(asset: asset)
let track = asset.tracks(withMediaType: AVMediaType.audio)[0]
let settings = [
AVFormatIDKey : kAudioFormatLinearPCM
]
let readerOutput = AVAssetReaderTrackOutput(track: track, outputSettings: settings)
reader.add(readerOutput)
reader.startReading()
while let buffer = readerOutput.copyNextSampleBuffer() {
var audioBufferList = AudioBufferList(mNumberBuffers: 1, mBuffers: AudioBuffer(mNumberChannels: 1, mDataByteSize: 0, mData: nil))
var blockBuffer: CMBlockBuffer?
CMSampleBufferGetAudioBufferListWithRetainedBlockBuffer(
buffer,
bufferListSizeNeededOut: nil,
bufferListOut: &audioBufferList,
bufferListSize: MemoryLayout<AudioBufferList>.size,
blockBufferAllocator: nil,
blockBufferMemoryAllocator: nil,
flags: kCMSampleBufferFlag_AudioBufferList_Assure16ByteAlignment,
blockBufferOut: &blockBuffer
);
let buffers = UnsafeBufferPointer<AudioBuffer>(start: &audioBufferList.mBuffers, count: Int(audioBufferList.mNumberBuffers))
for buffer in buffers {
let samplesCount = Int(buffer.mDataByteSize) / MemoryLayout<Int16>.size
let samplesPointer = audioBufferList.mBuffers.mData!.bindMemory(to: Int16.self, capacity: samplesCount)
let samples = UnsafeMutableBufferPointer<Int16>(start: samplesPointer, count: samplesCount)
for sample in samples {
//do something with you sample (which is Int16 amplitude value)
samplesArray.append(sample)
}
}
guard let data = audioBufferList.mBuffers.mData else {
return
}
/// The _Nyquist frequency_ is the highest frequency that a sampled system can properly
/// reproduce and is half the sampling rate of such a system. Although this app doesn't use
/// `nyquistFrequency` you may find this code useful to add an overlay to the user interface.
if nyquistFrequency == nil {
let duration = Float(CMSampleBufferGetDuration(buffer).value)
let timescale = Float(CMSampleBufferGetDuration(buffer).timescale)
let numsamples = Float(CMSampleBufferGetNumSamples(buffer))
nyquistFrequency = 0.5 / (duration / timescale / numsamples)
}
if self.rawAudioData.count < AudioSpectrogram.sampleCount * 2 {
let actualSampleCount = CMSampleBufferGetNumSamples(buffer)
let ptr = data.bindMemory(to: Int16.self, capacity: actualSampleCount)
let buf = UnsafeBufferPointer(start: ptr, count: actualSampleCount)
rawAudioData.append(contentsOf: Array(buf))
}
while self.rawAudioData.count >= AudioSpectrogram.sampleCount {
let dataToProcess = Array(self.rawAudioData[0 ..< AudioSpectrogram.sampleCount])
self.rawAudioData.removeFirst(AudioSpectrogram.hopCount)
self.processData(values: dataToProcess)
}
createAudioSpectrogram()
}
}
In AudioSpectrogram: CALayer file, I changed the original lines 10-30 from
public class AudioSpectrogram: CALayer {
// MARK: Initialization
override init() {
super.init()
contentsGravity = .resize
configureCaptureSession()
audioOutput.setSampleBufferDelegate(self,
queue: captureQueue)
}
required init?(coder: NSCoder) {
fatalError("init(coder:) has not been implemented")
}
override public init(layer: Any) {
super.init(layer: layer)
}
to the following:
public class AudioSpectrogram: CALayer {
#objc var audioFileUrl: URL
// MARK: Initialization
override init() {
self.audioFileUrl = selectedTrackUrl!
super.init()
contentsGravity = .resize
captureBuffer()
}
required init?(coder: NSCoder) {
fatalError("init(coder:) has not been implemented")
}
override public init(layer: Any) {
self.audioFileUrl = selectedTrackUrl!
super.init(layer: layer)
}
The changed code allows me to specify the audio file to use when the Spectrogram is called from another area in my app.
The following is an example of what I am trying to achieve. It was done using FFMPEG.
Example Spectrogram
This is the output I get from my code:
Output Image
AudioKit is not the tool you want for this. You want AVFoundation. Apple has an example project of exactly what you're describing.
The tool at the heart of this is a DCT (discrete cosine transform) to convert windows of samples into a collection of component frequencies you can visualize. AVFoundation is the tool you use to turn your audio file or live recording into a buffer of audio samples so you can apply the DCT.
There actually is a Spectrogram in the AudioKitUI Swift package: https://github.com/AudioKit/AudioKitUI/blob/main/Sources/AudioKitUI/Visualizations/SpectrogramView.swift
You would need to pass it an AudioKit Node but it should be interchangeable with the other visualizers in the Cookbook.
I found similar questions here, here, and here but with none of the answers I have been able to solve the problem. Simply put, the audio does not jump at a specific moment but instead starts from scratch.
func seekTo(time: Double) {
player.stop()
let startSample = Double(time * audioSampleRate)
let lengthSamples: AVAudioFramePosition = AVAudioFramePosition(Double(audioLengthSamples) - startSample)
let frameCount = AVAudioFrameCount(audioLengthSamples - lengthSamples)
if currentPosition < audioLengthSamples {
player.scheduleSegment(audioFile!, startingFrame: AVAudioFramePosition(startSample), frameCount: AVAudioFrameCount(frameCount), at: nil, completionHandler: nil)
let wasPlaying = player.isPlaying
if wasPlaying {
player.play()
}
}
}
The time variable is where it receives the specific time to which the audio should skip.
Any help?
I have AVAudioPCMBuffer, I set it as a source to the AVAudioPlayerNode
internal func play() {
guard let buf = pcmBuf else {
logger?.log(severity: .error, msg: "Sound pcmBuf is nil.")
return
}
if !isPlaying {
player.scheduleBuffer(buf, at: nil, options: .loops)
player.play()
}
}
It is possible that the user seeks forward, in order to do it I need to offset the buffer and set it again, like this
internal func play() {
guard let buf = pcmBuf else {
logger?.log(severity: .error, msg: "Sound pcmBuf is nil.")
return
}
buf.offset = 10 // Eg: in sec
if !isPlaying {
player.scheduleBuffer(buf, at: nil, options: .loops)
player.play()
}
}
So, it will look like I set a new buffer, but with the needed offset, and the playback will start from the required point.
The problem is that there is no offset method...
How to do it?
I don't think you need to offset the buffer. You should pass your play function a valid second parameter instead of nil.
https://developer.apple.com/documentation/avfaudio/avaudioplayernode/1388422-schedulebuffer
"Schedules the playing samples from an audio buffer at the time and playback options you specify."
So pass in a valid AVAudioTime.
https://developer.apple.com/documentation/avfaudio/avaudioplayernode#1669195
init(sampleTime: AVAudioFramePosition,
atRate sampleRate: Double)
I am guessing this is probably the initialiser you want.
I want to build a simple metronome app using AVAudioEngine with these features:
Solid timing (I know, I know, I should be using Audio Units, but I'm still struggling with Core Audio stuff / Obj-C wrappers etc.)
Two different sounds on the "1" and on beats "2"/"3"/"4" of the bar.
Some kind of visual feedback (at least a display of the current beat) which needs to be in sync with audio.
So I have created two short click sounds (26ms / 1150 samples # 16 bit / 44,1 kHz / stereo wav files) and load them into 2 buffers. Their lengths will be set to represent one period.
My UI setup is simple: A button to toggle start / pause and a label to display the current beat (my "counter" variable).
When using scheduleBuffer's loop property the timing is okay, but as I need to have 2 different sounds and a way to sync/update my UI while looping the clicks I cannot use this. I figured out to use the completionHandler instead which the restarts my playClickLoop() function - see my code attach below.
Unfortunately while implementing this I didn't really measure the accuracy of the timing. As it now turns out when setting bpm to 120, it plays the loop at only about 117,5 bpm - quite steadily but still way too slow. When bpm is set to 180, my app plays at about 172,3 bpm.
What's going on here? Is this delay introduced by using the completionHandler? Is there any way to improve the timing? Or is my whole approach wrong?
Thanks in advance!
Alex
import UIKit
import AVFoundation
class ViewController: UIViewController {
private let engine = AVAudioEngine()
private let player = AVAudioPlayerNode()
private let fileName1 = "sound1.wav"
private let fileName2 = "sound2.wav"
private var file1: AVAudioFile! = nil
private var file2: AVAudioFile! = nil
private var buffer1: AVAudioPCMBuffer! = nil
private var buffer2: AVAudioPCMBuffer! = nil
private let sampleRate: Double = 44100
private var bpm: Double = 180.0
private var periodLengthInSamples: Double { 60.0 / bpm * sampleRate }
private var counter: Int = 0
private enum MetronomeState {case run; case stop}
private var state: MetronomeState = .stop
#IBOutlet weak var label: UILabel!
override func viewDidLoad() {
super.viewDidLoad()
//
// MARK: Loading buffer1
//
let path1 = Bundle.main.path(forResource: fileName1, ofType: nil)!
let url1 = URL(fileURLWithPath: path1)
do {file1 = try AVAudioFile(forReading: url1)
buffer1 = AVAudioPCMBuffer(
pcmFormat: file1.processingFormat,
frameCapacity: AVAudioFrameCount(periodLengthInSamples))
try file1.read(into: buffer1!)
buffer1.frameLength = AVAudioFrameCount(periodLengthInSamples)
} catch { print("Error loading buffer1 \(error)") }
//
// MARK: Loading buffer2
//
let path2 = Bundle.main.path(forResource: fileName2, ofType: nil)!
let url2 = URL(fileURLWithPath: path2)
do {file2 = try AVAudioFile(forReading: url2)
buffer2 = AVAudioPCMBuffer(
pcmFormat: file2.processingFormat,
frameCapacity: AVAudioFrameCount(periodLengthInSamples))
try file2.read(into: buffer2!)
buffer2.frameLength = AVAudioFrameCount(periodLengthInSamples)
} catch { print("Error loading buffer2 \(error)") }
//
// MARK: Configure + start engine
//
engine.attach(player)
engine.connect(player, to: engine.mainMixerNode, format: file1.processingFormat)
engine.prepare()
do { try engine.start() } catch { print(error) }
}
//
// MARK: Play / Pause toggle action
//
#IBAction func buttonPresed(_ sender: UIButton) {
sender.isSelected = !sender.isSelected
if player.isPlaying {
state = .stop
} else {
state = .run
try! engine.start()
player.play()
playClickLoop()
}
}
private func playClickLoop() {
//
// MARK: Completion handler
//
let scheduleBufferCompletionHandler = { [unowned self] /*(_: AVAudioPlayerNodeCompletionCallbackType)*/ in
DispatchQueue.main.async {
switch state {
case .run:
self.playClickLoop()
case .stop:
engine.stop()
player.stop()
counter = 0
}
}
}
//
// MARK: Schedule buffer + play
//
if engine.isRunning {
counter += 1; if counter > 4 {counter = 1} // Counting from 1 to 4 only
if counter == 1 {
//
// MARK: Playing sound1 on beat 1
//
player.scheduleBuffer(buffer1,
at: nil,
options: [.interruptsAtLoop],
//completionCallbackType: .dataPlayedBack,
completionHandler: scheduleBufferCompletionHandler)
} else {
//
// MARK: Playing sound2 on beats 2, 3 & 4
//
player.scheduleBuffer(buffer2,
at: nil,
options: [.interruptsAtLoop],
//completionCallbackType: .dataRendered,
completionHandler: scheduleBufferCompletionHandler)
}
//
// MARK: Display current beat on UILabel + to console
//
DispatchQueue.main.async {
self.label.text = String(self.counter)
print(self.counter)
}
}
}
}
As Phil Freihofner suggested above, here's the solution to my own problem:
The most important lesson I learned: The completionHandler callback provided by the scheduleBuffer command is not called early enough to trigger re-scheduling of another buffer while the first one is still playing. This will result in (inaudible) gaps between the sounds and mess up the timing. There must already be another buffer "in reserve", i.e. having been schdeduled before the current one has been scheduled.
Using the completionCallbackType parameter of scheduleBuffer didn't change much considering the time of the completion callback: When setting it to .dataRendered or .dataConsumed the callback was already too late to re-schedule another buffer. Using .dataPlayedback made things only worse :-)
So, to achieve seamless playback (with correct timing!) I simply activated a timer that triggers twice per period. All odd numbered timer events will re-schedule another buffer.
Sometimes the solution is so easy it's embarrassing... But sometimes you have to try almost every wrong approach first to find it ;-)
My complete working solution (including the two sound files and the UI) can be found here on GitHub:
https://github.com/Alexander-Nagel/Metronome-using-AVAudioEngine
import UIKit
import AVFoundation
private let DEBUGGING_OUTPUT = true
class ViewController: UIViewController{
private var engine = AVAudioEngine()
private var player = AVAudioPlayerNode()
private var mixer = AVAudioMixerNode()
private let fileName1 = "sound1.wav"
private let fileName2 = "sound2.wav"
private var file1: AVAudioFile! = nil
private var file2: AVAudioFile! = nil
private var buffer1: AVAudioPCMBuffer! = nil
private var buffer2: AVAudioPCMBuffer! = nil
private let sampleRate: Double = 44100
private var bpm: Double = 133.33
private var periodLengthInSamples: Double {
60.0 / bpm * sampleRate
}
private var timerEventCounter: Int = 1
private var currentBeat: Int = 1
private var timer: Timer! = nil
private enum MetronomeState {case running; case stopped}
private var state: MetronomeState = .stopped
#IBOutlet weak var beatLabel: UILabel!
#IBOutlet weak var bpmLabel: UILabel!
#IBOutlet weak var playPauseButton: UIButton!
override func viewDidLoad() {
super.viewDidLoad()
bpmLabel.text = "\(bpm) BPM"
setupAudio()
}
private func setupAudio() {
//
// MARK: Loading buffer1
//
let path1 = Bundle.main.path(forResource: fileName1, ofType: nil)!
let url1 = URL(fileURLWithPath: path1)
do {file1 = try AVAudioFile(forReading: url1)
buffer1 = AVAudioPCMBuffer(
pcmFormat: file1.processingFormat,
frameCapacity: AVAudioFrameCount(periodLengthInSamples))
try file1.read(into: buffer1!)
buffer1.frameLength = AVAudioFrameCount(periodLengthInSamples)
} catch { print("Error loading buffer1 \(error)") }
//
// MARK: Loading buffer2
//
let path2 = Bundle.main.path(forResource: fileName2, ofType: nil)!
let url2 = URL(fileURLWithPath: path2)
do {file2 = try AVAudioFile(forReading: url2)
buffer2 = AVAudioPCMBuffer(
pcmFormat: file2.processingFormat,
frameCapacity: AVAudioFrameCount(periodLengthInSamples))
try file2.read(into: buffer2!)
buffer2.frameLength = AVAudioFrameCount(periodLengthInSamples)
} catch { print("Error loading buffer2 \(error)") }
//
// MARK: Configure + start engine
//
engine.attach(player)
engine.connect(player, to: engine.mainMixerNode, format: file1.processingFormat)
engine.prepare()
do { try engine.start() } catch { print(error) }
}
//
// MARK: Play / Pause toggle action
//
#IBAction func buttonPresed(_ sender: UIButton) {
sender.isSelected = !sender.isSelected
if state == .running {
//
// PAUSE: Stop timer and reset counters
//
state = .stopped
timer.invalidate()
timerEventCounter = 1
currentBeat = 1
} else {
//
// START: Pre-load first sound and start timer
//
state = .running
scheduleFirstBuffer()
startTimer()
}
}
private func startTimer() {
if DEBUGGING_OUTPUT {
print("# # # # # # # # # # # # # # # # # # # # # # # # # # # # # # # # # # ")
print()
}
//
// Compute interval for 2 events per period and set up timer
//
let timerIntervallInSamples = 0.5 * self.periodLengthInSamples / sampleRate
timer = Timer.scheduledTimer(withTimeInterval: timerIntervallInSamples, repeats: true) { timer in
//
// Only for debugging: Print counter values at start of timer event
//
// Values at begin of timer event
if DEBUGGING_OUTPUT {
print("timerEvent #\(self.timerEventCounter) at \(self.bpm) BPM")
print("Entering \ttimerEventCounter: \(self.timerEventCounter) \tcurrentBeat: \(self.currentBeat) ")
}
//
// Schedule next buffer at 1st, 3rd, 5th & 7th timerEvent
//
var bufferScheduled: String = "" // only needed for debugging / console output
switch self.timerEventCounter {
case 7:
//
// Schedule main sound
//
self.player.scheduleBuffer(self.buffer1, at:nil, options: [], completionHandler: nil)
bufferScheduled = "buffer1"
case 1, 3, 5:
//
// Schedule subdivision sound
//
self.player.scheduleBuffer(self.buffer2, at:nil, options: [], completionHandler: nil)
bufferScheduled = "buffer2"
default:
bufferScheduled = ""
}
//
// Display current beat & increase currentBeat (1...4) at 2nd, 4th, 6th & 8th timerEvent
//
if self.timerEventCounter % 2 == 0 {
DispatchQueue.main.async {
self.beatLabel.text = String(self.currentBeat)
}
self.currentBeat += 1; if self.currentBeat > 4 {self.currentBeat = 1}
}
//
// Increase timerEventCounter, two events per beat.
//
self.timerEventCounter += 1; if self.timerEventCounter > 8 {self.timerEventCounter = 1}
//
// Only for debugging: Print counter values at end of timer event
//
if DEBUGGING_OUTPUT {
print("Exiting \ttimerEventCounter: \(self.timerEventCounter) \tcurrentBeat: \(self.currentBeat) \tscheduling: \(bufferScheduled)")
print()
}
}
}
private func scheduleFirstBuffer() {
player.stop()
//
// pre-load accented main sound (for beat "1") before trigger starts
//
player.scheduleBuffer(buffer1, at: nil, options: [], completionHandler: nil)
player.play()
beatLabel.text = String(currentBeat)
}
}
Thanks so much for your help everyone! This is a wonderful community.
Alex
How accurate is the tool or process which you are using to get your measure?
I can't tell for sure that your files have the correct number of PCM frames as I am not a C programmer. It looks like data from the wav header is included when you load the files. This makes me wonder if maybe there is some latency incurred with the playbacks while the header information is processed repeatedly at the start of each play or loop.
I had good luck building a metronome in Java by using a plan of continuously outputting an endless stream derived from reading PCM frames. Timing is achieved by counting PCM frames and routing in either silence (PCM datapoint = 0) or the click's PCM data, based on the period of the chosen metronome setting and the length of the click in PCM frames.