问题描述
我有 recorder.js ,它将记录音频并接受缓冲区输入,但是我想对音频缓冲区进行降采样,但是我很困惑在哪里调用它我已经写了.请检查我的函数,并在可能的情况下建议在何处调用它.
I have the recorder.js which will record the audio and takes buffer inputs but I want to downsample the audio buffers but I am lot confused where to call it though I have written it. Please check my function and if possible please suggest where to call it.
import InlineWorker from 'inline-worker';
export class Recorder {
config = {
bufferLen: 4096,
numChannels: 2,
mimeType: 'audio/mp3'
};
recording = false;
callbacks = {
getBuffer: [],
exportWAV: []
};
constructor(source, cfg) {
Object.assign(this.config, cfg);
this.context = source.context;
this.node = (this.context.createScriptProcessor ||
this.context.createJavaScriptNode).call(this.context,
this.config.bufferLen, this.config.numChannels, this.config.numChannels);
this.node.onaudioprocess = (e) => {
if (!this.recording) return;
var buffer = [];
for (var channel = 0; channel < this.config.numChannels; channel++) {
buffer.push(e.inputBuffer.getChannelData(channel));
}
this.worker.postMessage({
command: 'record',
buffer: buffer
});
};
source.connect(this.node);
this.node.connect(this.context.destination); //this should not be necessary
let self = {};
this.worker = new InlineWorker(function () {
let recLength = 0,
recBuffers = [],
sampleRate,
numChannels;
this.onmessage = function (e) {
switch (e.data.command) {
case 'init':
init(e.data.config);
break;
case 'record':
record(e.data.buffer);
break;
case 'exportWAV':
exportWAV(e.data.type);
break;
case 'getBuffer':
getBuffer();
break;
case 'clear':
clear();
break;
}
};
function init(config) {
sampleRate = config.sampleRate;
numChannels = config.numChannels;
initBuffers();
}
function record(inputBuffer) {
for (var channel = 0; channel < numChannels; channel++) {
recBuffers[channel].push(inputBuffer[channel]);
}
recLength += inputBuffer[0].length;
}
function exportWAV(type) {
let buffers = [];
for (let channel = 0; channel < numChannels; channel++) {
buffers.push(mergeBuffers(recBuffers[channel], recLength));
}
let interleaved;
if (numChannels === 2) {
interleaved = interleave(downsampleBuffer(buffers[0]), downsampleBuffer(buffers[1]));
} else {
interleaved = buffers[0];
}
let dataview = encodeWAV(interleaved);
let audioBlob = new Blob([dataview], {type: type});
this.postMessage({command: 'exportWAV', data: audioBlob});
}
function downsampleBuffer(buffer) {
if (16000 === sampleRate) {
return buffer;
}
var sampleRateRatio = sampleRate / 16000;
var newLength = Math.round(buffer.length / sampleRateRatio);
var result = new Float32Array(newLength);
var offsetResult = 0;
var offsetBuffer = 0;
while (offsetResult < result.length) {
var nextOffsetBuffer = Math.round((offsetResult + 1) * sampleRateRatio);
var accum = 0,
count = 0;
for (var i = offsetBuffer; i < nextOffsetBuffer && i < buffer.length; i++) {
accum += buffer[i];
count++;
}
result[offsetResult] = accum / count;
offsetResult++;
offsetBuffer = nextOffsetBuffer;
}
return result;
}
function getBuffer() {
let buffers = [];
for (let channel = 0; channel < numChannels; channel++) {
buffers.push(mergeBuffers(recBuffers[channel], recLength));
}
this.postMessage({command: 'getBuffer', data: buffers});
}
function clear() {
recLength = 0;
recBuffers = [];
initBuffers();
}
function initBuffers() {
for (let channel = 0; channel < numChannels; channel++) {
recBuffers[channel] = [];
}
}
function mergeBuffers(recBuffers, recLength) {
let result = new Float32Array(recLength);
let offset = 0;
for (let i = 0; i < recBuffers.length; i++) {
result.set(recBuffers[i], offset);
offset += recBuffers[i].length;
}
return result;
}
function interleave(inputL, inputR) {
let length = inputL.length + inputR.length;
let result = new Float32Array(length);
let index = 0,
inputIndex = 0;
while (index < length) {
result[index++] = inputL[inputIndex];
result[index++] = inputR[inputIndex];
inputIndex++;
}
return result;
}
function floatTo16BitPCM(output, offset, input) {
for (let i = 0; i < input.length; i++, offset += 2) {
let 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 (let i = 0; i < string.length; i++) {
view.setUint8(offset + i, string.charCodeAt(i));
}
}
function encodeWAV(samples) {
let buffer = new ArrayBuffer(44 + samples.length * 2);
let 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, numChannels, 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, numChannels * 2, 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;
}
}, self);
this.worker.postMessage({
command: 'init',
config: {
sampleRate: this.context.sampleRate,
numChannels: this.config.numChannels
}
});
this.worker.onmessage = (e) => {
let cb = this.callbacks[e.data.command].pop();
if (typeof cb == 'function') {
cb(e.data.data);
}
};
}
record() {
this.recording = true;
}
stop() {
this.recording = false;
}
clear() {
this.worker.postMessage({command: 'clear'});
}
getBuffer(cb) {
cb = cb || this.config.callback;
if (!cb) throw new Error('Callback not set');
this.callbacks.getBuffer.push(cb);
this.worker.postMessage({command: 'getBuffer'});
}
exportWAV(cb, mimeType) {
mimeType = mimeType || this.config.mimeType;
cb = cb || this.config.callback;
if (!cb) throw new Error('Callback not set');
this.callbacks.exportWAV.push(cb);
this.worker.postMessage({
command: 'exportWAV',
type: mimeType
});
}
static
forceDownload(blob, filename) {
let url = (window.URL || window.webkitURL).createObjectURL(blob);
let link = window.document.createElement('a');
link.href = url;
link.download = filename || 'output.wav';
let click = document.createEvent("Event");
click.initEvent("click", true, true);
link.dispatchEvent(click);
}
}
export default Recorder;
这是我从github存储库中获取的代码,但采样率为 48000
It's a code which I took from github repository but the sample rate is 48000
在对左缓冲区和右缓冲区进行降采样后,文件被上传此处
File after downsampling left and right buffers is uploaded here
从我的组件类中调用
recorder && recorder.exportWAV(function(blob) {
var formData=new FormData();
formData.append("event_name",fileName);
formData.append("file",new File([blob], fileName, {type: 'audio/mpeg;', lastModified: Date.now()}));
formData.append("fileExtension", "mp3");
fetch('http://localhost:6020/uploadFile', {
method: "POST", // *GET, POST, PUT, DELETE, etc.
cache: "no-cache", // *default, no-cache, reload, force-cache, only-if-cached
redirect: "follow", // manual, *follow, error
referrer: "no-referrer", // no-referrer, *client
body: formData, // body data type must match "Content-Type" header
})
.then(response => response.json())
.then(function(data){ console.log( JSON.stringify( data ) ) });
},"audio/mp3");
推荐答案
我成功地使用recorder.js
库对音频剪辑(wav文件)进行了下采样,并引用了ilikerei给出的解决方案. github.com/mattdiamond/Recorderjs/issues/186"rel =" nofollow noreferrer>此线程.
I downsampled my audio clips (wav files) successfully using recorder.js
library referring the solution given by ilikerei
in this thread.
在调用encodeWav()方法之前,将此方法添加到recorder.js中以对缓冲区重新采样以更改采样率.
Add this method to recorder.js to resample the buffer to change sample rate, before calling encodeWav() method.
function downsampleBuffer(buffer, rate) {
if (rate == sampleRate) {
return buffer;
}
if (rate > sampleRate) {
throw "downsampling rate show be smaller than original sample rate";
}
var sampleRateRatio = sampleRate / rate;
var newLength = Math.round(buffer.length / sampleRateRatio);
var result = new Float32Array(newLength);
var offsetResult = 0;
var offsetBuffer = 0;
while (offsetResult < result.length) {
var nextOffsetBuffer = Math.round((offsetResult + 1) * sampleRateRatio);
// Use average value of skipped samples
var accum = 0, count = 0;
for (var i = offsetBuffer; i < nextOffsetBuffer && i < buffer.length; i++) {
accum += buffer[i];
count++;
}
result[offsetResult] = accum / count;
// Or you can simply get rid of the skipped samples:
// result[offsetResult] = buffer[nextOffsetBuffer];
offsetResult++;
offsetBuffer = nextOffsetBuffer;
}
return result;
}
然后使用新的样本缓冲区调用encodeWav.
Then call encodeWav with the new sample buffer.
var downsampledBuffer = downsampleBuffer(interleaved, targetRate);
var dataview = encodeWAV(downsampledBuffer);
现在使用新的采样率进行编码.
Now use the new sample rate for encoding.
/* sample rate */
view.setUint32(24, newRate, true);
/* byte rate (sample rate * block align) */
view.setUint32(28, newRate * 4, true);
这篇关于将来自缓冲输入的音频从48000降采样为16000的文章就介绍到这了,希望我们推荐的答案对大家有所帮助,也希望大家多多支持!