The AudioBuffer interface represents a short audio asset residing in memory, created from an audio file using the AudioContext.decodeAudioData() method, or from raw data using AudioContext.createBuffer(). Once put into an AudioBuffer, the audio can then be played by being passed into an AudioBufferSourceNode.

Objects of these types are designed to hold small audio snippets, typically less than 45 s. For longer sounds, objects implementing the MediaElementAudioSourceNode are more suitable. The buffer contains the audio signal waveform encoded as a series of amplitudes in the following format: non-interleaved IEEE754 32-bit linear PCM with a nominal range between -1 and +1, that is, a 32-bit floating point buffer, with each sample between -1.0 and 1.0. If the AudioBuffer has multiple channels, they are stored in separate buffers.

AudioBuffer()

Creates and returns a new AudioBuffer object instance.

AudioBuffer.sampleRate Read only

Returns a float representing the sample rate, in samples per second, of the PCM data stored in the buffer.

AudioBuffer.length Read only

Returns an integer representing the length, in sample-frames, of the PCM data stored in the buffer.

AudioBuffer.duration Read only

Returns a double representing the duration, in seconds, of the PCM data stored in the buffer.

AudioBuffer.numberOfChannels Read only

Returns an integer representing the number of discrete audio channels described by the PCM data stored in the buffer.

AudioBuffer.getChannelData()

Returns a Float32Array containing the PCM data associated with the channel, defined by the channel parameter (with 0 representing the first channel).

AudioBuffer.copyFromChannel()

Copies the samples from the specified channel of the AudioBuffer to the destination array.

AudioBuffer.copyToChannel()

Copies the samples to the specified channel of the AudioBuffer, from the source array.

The following simple example shows how to create an AudioBuffer and fill it with random white noise. You can find the full source code at our webaudio-examples repository; a running live version is also available.

const audioCtx = new (window.AudioContext || window.webkitAudioContext)();

// Create an empty three-second stereo buffer at the sample rate of the AudioContext
const myArrayBuffer = audioCtx.createBuffer(
  2,
  audioCtx.sampleRate * 3,
  audioCtx.sampleRate,
);

// Fill the buffer with white noise;
// just random values between -1.0 and 1.0
for (let channel = 0; channel < myArrayBuffer.numberOfChannels; channel++) {
  // This gives us the actual array that contains the data
  const nowBuffering = myArrayBuffer.getChannelData(channel);
  for (let i = 0; i < myArrayBuffer.length; 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
const source = audioCtx.createBufferSource();

// set the buffer in the AudioBufferSourceNode
source.buffer = myArrayBuffer;

// connect the AudioBufferSourceNode to the
// destination so we can hear the sound
source.connect(audioCtx.destination);

// start the source playing
source.start();

• Using the Web Audio API