在网络上录制音频，预设：16000Hz 16bit

function floatTo16BitPCM(output, offset, input){
  for (var i = 0; i < input.length; i++, offset+=2){
    var s = Math.max(-1, Math.min(1, input[i]));
    output.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7FFF, true);
  }
}

function writeString(view, offset, string){
  for (var i = 0; i < string.length; i++){
    view.setUint8(offset + i, string.charCodeAt(i));
  }
}

function encodeWAV(samples){
  var buffer = new ArrayBuffer(44 + samples.length * 2);
  var view = new DataView(buffer);

  /* RIFF identifier */
  writeString(view, 0, 'RIFF');
  /* RIFF chunk length */
  view.setUint32(4, 36 + samples.length * 2, true);
  /* RIFF type */
  writeString(view, 8, 'WAVE');
  /* format chunk identifier */
  writeString(view, 12, 'fmt ');
  /* format chunk length */
  view.setUint32(16, 16, true);
  /* sample format (raw) */
  view.setUint16(20, 1, true);
  /* channel count */
  view.setUint16(22, 2, true);
  /* sample rate */
  view.setUint32(24, sampleRate, true);
  /* byte rate (sample rate * block align) */
  view.setUint32(28, sampleRate * 4, true);
  /* block align (channel count * bytes per sample) */
  view.setUint16(32, 4, true);
  /* bits per sample */
  view.setUint16(34, 16, true);
  /* data chunk identifier */
  writeString(view, 36, 'data');
  /* data chunk length */
  view.setUint32(40, samples.length * 2, true);

  floatTo16BitPCM(view, 44, samples);

  return view;
}

您好，我正在使用此源代码为我的学校考试录制音频。它以 44100Hz 和 16 位录制音频。我想更改录制设置以以 16000Hz 和 16 位录制音频。 我尝试将函数encodeWAV中的44修改为16，但它不起作用。

function encodeWAV(samples){
  var buffer = new ArrayBuffer(44 + samples.length * 2);
  var view = new DataView(buffer)

我也尝试过更改 floadRToBitPCM。我尝试将44更改为16，但也没有成功。

floatTo16BitPCM(view, 44, samples);

你能帮我解决这个问题吗？我不知道如何修改这个源代码。

Edit:

另一种选择（在我看来更好）就是使用 HTML 的 MediaRecorder 并记录在.ogg格式，demo，它是git repo

我假设你正在使用this正如消息来源，就像杰克特所说，这行floatTo16BitPCM(view, 44, samples);和采样率没有关系...

如果你想重新采样数据，你必须修改：

function exportWAV(type){
    var buffers = [];
    for (var channel = 0; channel < numChannels; channel++){
        buffers.push(mergeBuffers(recBuffers[channel], recLength));
    }
    if (numChannels === 2){
        var interleaved = interleave(buffers[0], buffers[1]);
    } else {
        var interleaved = buffers[0];
    }
    var dataview = encodeWAV(interleaved);
    var audioBlob = new Blob([dataview], { type: type });
    this.postMessage(audioBlob);
}

进入这个：

function exportWAV(type, desiredSamplingRate){
    var buffers = [];
    for (var channel = 0; channel < numChannels; channel++){
        var buffer = mergeBuffers(recBuffers[channel], recLength);
        buffer = interpolateArray(buffer, desiredSamplingRate, sampleRate);
        buffers.push(buffer);
    }
    sampleRate = desiredSamplingRate;
    if (numChannels === 2){
        var interleaved = interleave(buffers[0], buffers[1]);
    } else {
        var interleaved = buffers[0];
    }
    var dataview = encodeWAV(interleaved);
    var audioBlob = new Blob([dataview], { type: type });
    this.postMessage(audioBlob);
}

重新采样数据的代码，

// for changing the sampling rate, data,
function interpolateArray(data, newSampleRate, oldSampleRate) {
    var fitCount = Math.round(data.length*(newSampleRate/oldSampleRate));
    var newData = new Array();
    var springFactor = new Number((data.length - 1) / (fitCount - 1));
    newData[0] = data[0]; // for new allocation
    for ( var i = 1; i < fitCount - 1; i++) {
    var tmp = i * springFactor;
    var before = new Number(Math.floor(tmp)).toFixed();
    var after = new Number(Math.ceil(tmp)).toFixed();
    var atPoint = tmp - before;
    newData[i] = this.linearInterpolate(data[before], data[after], atPoint);
    }
    newData[fitCount - 1] = data[data.length - 1]; // for new allocation
    return newData;
};
function linearInterpolate(before, after, atPoint) {
    return before + (after - before) * atPoint;
};

Edit:如果你不打算改变太多，你可以将其硬编码为

function exportWAV(type){
    var buffers = [], desiredSamplingRate = 16000;