The createBuffer() method of the AudioContext Interface is used to create a new, empty AudioBuffer object, which can then be populated by data, and played via an AudioBufferSourceNode

For more details about audio buffers, check out the AudioBuffer reference page.

Note: createBuffer() used to be able to take compressed data and give back decoded samples, but this ability was removed from the spec, because all the decoding was done on the main thread, therefore createBuffer() was blocking other code execution. The asynchronous method decodeAudioData() does the same thing — takes compressed audio, say, an MP3 file, and directly gives you back an AudioBuffer that you can then set to play via in an AudioBufferSourceNode. For simple uses like playing an MP3, decodeAudioData() is what you should be using.


var audioCtx = new AudioContext();
var buffer = audioCtx.createBuffer(2, 22050, 44100);


Note: For an in-depth explanation of how audio buffers work, and what these parameters mean, read Audio buffers: frames, samples and channels from our Basic concepts guide.

An integer representing the number of channels this buffer should have. Implementations must support a minimum of 32 channels, and the value should be a minimum value of 1.
An integer representing the size of the buffer in sample-frames.
The sample-rate of the linear audio data in sample-frames per second. An implementation must support sample-rates in at least the range 22050 to 96000.


An AudioBuffer.


First, a couple of simple trivial examples, to help explain how the parameters are used:

var audioCtx = new AudioContext();
var buffer = audioCtx.createBuffer(2, 22050, 44100);

If you use this call, you will get a stereo buffer (two channels), that, when played back on an AudioContext running at 44100Hz (very common, most normal sound cards run at this rate), will last for 0.5 seconds: 22050 frames / 44100Hz = 0.5 seconds.

var audioCtx = new AudioContext();
var buffer = audioCtx.createBuffer(1, 22050, 22050);

If you use this call, you will get a mono buffer (one channel), that, when played back on an AudioContext running at 44100Hz, will be automatically *resampled* to 44100Hz (and therefore yield 44100 frames), and last for 1.0 second: 44100 frames / 44100Hz = 1 second.

Note: audio resampling is very similar to image resizing: say you've got a 16 x 16 image, but you want it to fill a 32x32 area: you resize (resample) it. the result has less quality (it can be blurry or edgy, depending on the resizing algorithm), but it works, and the resized image takes up less space. Resampled audio is exactly the same — you save space, but in practice you will be unable to properly reproduce high frequency content (treble sound).

Now let's look at a more complex createBuffer() example, in which we create a two second buffer, fill it with white noise, and then play it via an AudioBufferSourceNode. The comment should clearly explain what is going on. You can also run the code live, or view the source.

var audioCtx = new (window.AudioContext || window.webkitAudioContext)();
var button = document.querySelector('button');
var pre = document.querySelector('pre');
var myScript = document.querySelector('script');

pre.innerHTML = myScript.innerHTML;

// Stereo
var channels = 2;
// Create an empty two second stereo buffer at the
// sample rate of the AudioContext
var frameCount = audioCtx.sampleRate * 2.0;

var myAudioBuffer = audioCtx.createBuffer(channels, frameCount, audioCtx.sampleRate);

button.onclick = function() {
  // Fill the buffer with white noise;
  //just random values between -1.0 and 1.0
  for (var channel = 0; channel < channels; channel++) {
   // This gives us the actual ArrayBuffer that contains the data
   var nowBuffering = myAudioBuffer.getChannelData(channel);
   for (var i = 0; i < frameCount; i++) {
     // Math.random() is in [0; 1.0]
     // audio needs to be in [-1.0; 1.0]
     nowBuffering[i] = Math.random() * 2 - 1;

  // Get an AudioBufferSourceNode.
  // This is the AudioNode to use when we want to play an AudioBuffer
  var source = audioCtx.createBufferSource();
  // set the buffer in the AudioBufferSourceNode
  source.buffer = myAudioBuffer;
  // connect the AudioBufferSourceNode to the
  // destination so we can hear the sound
  // start the source playing


Specification Status Comment
Web Audio API
The definition of 'createBuffer()' in that specification.
Working Draft  

Browser compatibility

Feature Chrome Firefox (Gecko) Internet Explorer Opera Safari (WebKit)
Basic support 10.0webkit 25.0 (25.0)  No support 15.0 webkit
Feature Android Firefox Mobile (Gecko) Firefox OS IE Mobile Opera Mobile Safari Mobile Chrome for Android
Basic support ? 26.0 1.2 ? ? ? 33.0

See also


© 2016 Mozilla Contributors
Licensed under the Creative Commons Attribution-ShareAlike License v2.5 or later.

API AudioContext createBuffer Method Reference Référence Web Audio API