I'm using the Google Cloud API for Speech-to-text, with a NodeJS back-end. The app needs to be able to listen for voice commands, and transmit them to the back-end as a buffer. For this, I need to send the buffer of the preceding audio when silence is detected.
Any help would be appreciated. Including the js code below
if (!navigator.getUserMedia)
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia || navigator.msGetUserMedia;
if (navigator.getUserMedia) {
navigator.getUserMedia({audio: true}, success, function (e) {
alert('Error capturing audio.');
});
} else alert('getUserMedia not supported in this browser.');
var recording = false;
window.startRecording = function () {
recording = true;
};
window.stopRecording = function () {
recording = false;
// window.Stream.end();
};
function success(e) {
audioContext = window.AudioContext || window.webkitAudioContext;
context = new audioContext();
// the sample rate is in context.sampleRate
audioInput = context.createMediaStreamSource(e);
var bufferSize = 4096;
recorder = context.createScriptProcessor(bufferSize, 1, 1);
recorder.onaudioprocess = function (e) {
if (!recording) return;
console.log('recording');
var left = e.inputBuffer.getChannelData(0);
console.log(convertoFloat32ToInt16(left));
};
audioInput.connect(recorder);
recorder.connect(context.destination);
}
I'm not too sure as to what exactly is being asked in the question, so this answer is only intended to give a way to detect silences in an AudioStream.
To detect silence in an AudioStream, you can use an AudioAnalyser node, on which you will call the getByteFrequencyData method at regular intervals, and check whether there were sounds higher than than your expected level for a given time.
You can set the threshold level directly with the minDecibels property of the AnalyserNode.
function detectSilence(
stream,
onSoundEnd = _=>{},
onSoundStart = _=>{},
silence_delay = 500,
min_decibels = -80
) {
const ctx = new AudioContext();
const analyser = ctx.createAnalyser();
const streamNode = ctx.createMediaStreamSource(stream);
streamNode.connect(analyser);
analyser.minDecibels = min_decibels;
const data = new Uint8Array(analyser.frequencyBinCount); // will hold our data
let silence_start = performance.now();
let triggered = false; // trigger only once per silence event
function loop(time) {
requestAnimationFrame(loop); // we'll loop every 60th of a second to check
analyser.getByteFrequencyData(data); // get current data
if (data.some(v => v)) { // if there is data above the given db limit
if(triggered){
triggered = false;
onSoundStart();
}
silence_start = time; // set it to now
}
if (!triggered && time - silence_start > silence_delay) {
onSoundEnd();
triggered = true;
}
}
loop();
}
function onSilence() {
console.log('silence');
}
function onSpeak() {
console.log('speaking');
}
navigator.mediaDevices.getUserMedia({
audio: true
})
.then(stream => {
detectSilence(stream, onSilence, onSpeak);
// do something else with the stream
})
.catch(console.error);And as a fiddle since stackSnippets may block gUM.
You can use SpeechRecognition result event to determine when a word or phrase has been recognized, for example, ls, cd, pwd or other commands, pass the .transcript of SpeechRecognitionAlternative to speechSynthesis.speak() where at attached start and end event of SpeechSynthesisUtterance call .start() or .resume() on MediaRecorder object where MediaStream is passed; convert the Blob at dataavailable event to an ArrayBuffer using FileReader or Response.arrayBuffer().
We could alternatively use audiostart or soundstart with audioend or soundend events of SpeechRecognition to record the users' actual voice, though the ends may not be fired consistently in relation to the actual start and end of audio captured by only a standard system microphone.
<!DOCTYPE html>
<html>
<head>
<title>Speech Recognition Recording</title>
</head>
<body>
<input type="button" value="Stop speech command recognition" id="stop">
<script>
navigator.mediaDevices.getUserMedia({
audio: true
})
.then(stream => {
const recorder = new MediaRecorder(stream);
const recognition = new webkitSpeechRecognition();
const synthesis = new SpeechSynthesisUtterance();
const handleResult = e => {
recognition.onresult = null;
console.log(e.results);
const result = e.results[e.results.length - 1];
if (result.isFinal) {
const [{transcript}] = result;
console.log(transcript);
synthesis.text = transcript;
window.speechSynthesis.speak(synthesis);
}
}
synthesis.onstart = () => {
if (recorder.state === "inactive") {
recorder.start()
} else {
if (recorder.state === "paused") {
recorder.resume();
}
}
}
synthesis.onend = () => {
recorder.pause();
recorder.requestData();
}
recorder.ondataavailable = async(e) => {
if (stream.active) {
try {
const blobURL = URL.createObjectURL(e.data);
const request = await fetch(blobURL);
const ab = await request.arrayBuffer();
console.log(blobURL, ab);
recognition.onresult = handleResult;
// URL.revokeObjectURL(blobURL);
} catch (err) {
throw err
}
}
}
recorder.onpause = e => {
console.log("recorder " + recorder.state);
}
recognition.continuous = true;
recognition.interimResults = false;
recognition.maxAlternatives = 1;
recognition.start();
recognition.onend = e => {
console.log("recognition ended, stream.active", stream.active);
if (stream.active) {
console.log(e);
// the service disconnects after a period of time
recognition.start();
}
}
recognition.onresult = handleResult;
stream.oninactive = () => {
console.log("stream ended");
}
document.getElementById("stop")
.onclick = () => {
console.log("stream.active:", stream.active);
if (stream && stream.active && recognition) {
recognition.abort();
recorder.stop();
for (let track of stream.getTracks()) {
track.stop();
}
console.log("stream.active:", stream.active);
}
}
})
.catch(err => {
console.error(err)
});
</script>
</body>
</html>
The simplest approach would be to use .pause() and .resume(), .stop() methods of MediaRecorder() to allow user to start, pause, and stop recording audio captured utilizing navigator.mediaDevices.getUserMedia() and convert the resulting Blob to an ArrayBuffer, if that is what the api is expecting to be POSTed to server
<!DOCTYPE html>
<html>
<head>
<title>User Media Recording</title>
</head>
<body>
<input type="button" value="Start/resume recording audio" id="start">
<input type="button" value="Pause recording audio" id="pause">
<input type="button" value="Stop recording audio" id="stop">
<script>
navigator.mediaDevices.getUserMedia({
audio: true
})
.then(stream => {
const recorder = new MediaRecorder(stream);
recorder.ondataavailable = async(e) => {
if (stream.active) {
try {
const blobURL = URL.createObjectURL(e.data);
const request = await fetch(blobURL);
const ab = await request.arrayBuffer();
// do stuff with `ArrayBuffer` of recorded audio
console.log(blobURL, ab);
// we do not need the `Blob URL`, we can revoke the object
// URL.revokeObjectURL(blobURL);
} catch (err) {
throw err
}
}
}
recorder.onpause = e => {
console.log("recorder " + recorder.state);
recorder.requestData();
}
stream.oninactive = () => {
console.log("stream ended");
}
document.getElementById("start")
.onclick = () => {
if (recorder.state === "inactive") {
recorder.start();
} else {
recorder.resume();
}
console.log("recorder.state:", recorder.state);
}
document.getElementById("pause")
.onclick = () => {
if (recorder.state === "recording") {
recorder.pause();
}
console.log("recorder.state:", recorder.state);
}
document.getElementById("stop")
.onclick = () => {
if (recorder.state === "recording" || recorder.state === "paused") {
recorder.stop();
}
for (let track of stream.getTracks()) {
track.stop();
}
document.getElementById("start").onclick = null;
document.getElementById("pause").onclick = null;
console.log("recorder.state:", recorder.state
, "stream.active", stream.active);
}
})
.catch(err => {
console.error(err)
});
</script>
</body>
</html>
来源：https://stackoverflow.com/questions/46543341/how-can-i-extract-the-preceding-audio-from-microphone-as-a-buffer-when-silence