BaseAudioContext.createBuffer()

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createBuffer() 方法在 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.

注意 : 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.

句法

var buffer = baseAudioContext.createBuffer(numOfchannels, length, sampleRate);

参数

注意 : 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
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 the length to use for a specific number of seconds of audio, use numSeconds * sampleRate .
sampleRate

AudioBuffer configured based on the specified options.

异常

NotSupportedError: One or more of the options are negative or otherwise has an invalid value (such as numberOfChannels being higher than supported, or a sampleRate outside the nominal range).
RangeError

范例

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.

注意 : 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 ，或 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();

规范

规范	状态	注释
Web 音频 API The definition of 'createBuffer()' in that specification.	工作草案

浏览器兼容性

The compatibility table on this page is generated from structured data. If you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. 更新 GitHub 上的兼容性数据

	桌面						移动
	Chrome	Edge	Firefox	Internet Explorer	Opera	Safari	Android webview	Chrome for Android	Firefox for Android	Opera for Android	Safari on iOS	Samsung Internet
`createBuffer`	Chrome 10 Prefixed 10 Prefixed Prefixed Implemented with the vendor prefix: webkit	Edge ≤18	Firefox 53 注意事项 53 注意事项 Originally implemented on `AudioContext` in Firefox 25.	IE 不支持 No	Opera 22 22 15 Prefixed Prefixed Implemented with the vendor prefix: webkit	Safari 6 Prefixed 6 Prefixed Prefixed Implemented with the vendor prefix: webkit	WebView Android Yes	Chrome Android 33	Firefox Android 53 注意事项 53 注意事项 Originally implemented on `AudioContext` in Firefox Android 26.	Opera Android 22 22 14 Prefixed Prefixed Implemented with the vendor prefix: webkit	Safari iOS 6 Prefixed 6 Prefixed Prefixed Implemented with the vendor prefix: webkit	Samsung Internet Android 2.0

图例

完整支持
不支持

要求使用供应商前缀或不同名称。

另请参阅

使用 Web 音频 API

元数据

最后修改： Nov 29, 2019

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句法

参数

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规范

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