AudioBuffer转换为WAVE音频资源

/**
 * 把AudioBuffer转换为WAVE音频资源
 * @param {AudioBuffer} abuffer - 音频数据
 * @param {number} len - 音频数据的总样本数 = 每秒钟的采样次数(audioBuffer.sampleRate) * 音频数据的总时长(audioBuffer.duration)
 * @returns {Blob} WAVE音频资源
 */
function bufferToWave(abuffer, len) {
  let numOfChan = abuffer.numberOfChannels,
    length = len * numOfChan * 2 + 44,
    buffer = new ArrayBuffer(length),
    view = new DataView(buffer),
    channels = [],
    i,
    sample,
    offset = 0,
    pos = 0;

  // write WAVE header
  // "RIFF"
  setUint32(0x46464952);
  // file length - 8
  setUint32(length - 8);
  // "WAVE"
  setUint32(0x45564157);
  // "fmt " chunk
  setUint32(0x20746d66);
  // length = 16
  setUint32(16);
  // PCM (uncompressed)
  setUint16(1);
  setUint16(numOfChan);
  setUint32(abuffer.sampleRate);
  // avg. bytes/sec
  setUint32(abuffer.sampleRate * 2 * numOfChan);
  // block-align
  setUint16(numOfChan * 2);
  // 16-bit (hardcoded in this demo)
  setUint16(16);
  // "data" - chunk
  setUint32(0x61746164);
  // chunk length
  setUint32(length - pos - 4);

  // write interleaved data
  for (i = 0; i < abuffer.numberOfChannels; i++) {
    channels.push(abuffer.getChannelData(i));
  }

  while (pos < length) {
    // interleave channels
    for (i = 0; i < numOfChan; i++) {
      // clamp
      sample = Math.max(-1, Math.min(1, channels[i][offset]));
      // scale to 16-bit signed int
      sample = (0.5 + sample < 0 ? sample * 32768 : sample * 32767) | 0;
      // write 16-bit sample
      view.setInt16(pos, sample, true);
      pos += 2;
    }
    // next source sample
    offset++;
  }

  // create Blob
  return new Blob([buffer], { type: 'audio/wav' });

  function setUint16(data) {
    view.setUint16(pos, data, true);
    pos += 2;
  }

  function setUint32(data) {
    view.setUint32(pos, data, true);
    pos += 4;
  }
}