BaseAudioContext.createBuffer()
The createBuffer()
method of the BaseAudioContext
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 specification,
because all the decoding was done on the main thread, so
createBuffer()
was blocking other code execution. The asynchronous method
decodeAudioData()
does the same thing — takes compressed audio, such as an
MP3 file, and directly gives you back an AudioBuffer
that you can
then play via an AudioBufferSourceNode
. For simple use cases
like playing an MP3, decodeAudioData()
is what you should be using.
Syntax
var buffer = baseAudioContext.createBuffer(numOfChannels, length, sampleRate);
Parameters
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.
numOfChannels
-
An integer representing the number of channels this buffer should have. The default value is 1, and all user agents must support at least 32 channels.
length
-
An integer representing the size of the buffer in sample-frames (where each sample-frame is the size of a sample in bytes multiplied by
numOfChannels
). To determine thelength
to use for a specific number of seconds of audio, usenumSeconds * sampleRate
. sampleRate
-
The sample rate of the linear audio data in sample-frames per second. All browsers must support sample rates in at least the range 8,000 Hz to 96,000 Hz.
Returns
An AudioBuffer
configured based on the specified options.
Exceptions
NotSupportedError
DOMException
-
Thrown if one or more of the options are negative or otherwise has an invalid value (such as
numberOfChannels
being higher than supported, or asampleRate
outside the nominal range). RangeError
DOMException
-
Thrown if there isn't enough memory available to allocate the buffer.
Examples
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 three-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)();
// Create an empty three-second stereo buffer at the sample rate of the AudioContext
var 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 (var channel = 0; channel < myArrayBuffer.numberOfChannels; channel++) {
// This gives us the actual ArrayBuffer that contains the data
var nowBuffering = myArrayBuffer.getChannelData(channel);
for (var 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
var 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 # dom-baseaudiocontext-createbuffer |
Browser compatibility
BCD tables only load in the browser