AudioBuffer
===========

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.

Constructor
-----------

`AudioBuffer()`

Creates and returns a new `AudioBuffer` object instance.

Instance properties
-------------------

`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.

Instance methods
----------------

`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.

Example
-------

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.


```js
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();
```

Specifications
--------------

| Specification |
| --- |
| Web Audio API # AudioBuffer |

Browser compatibility
---------------------

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See also
--------

* Using the Web Audio API