I am making an app where the user browser records the user speaking and sends it to the server which then passes it on to the Google speech to the text interface. I am using mediaRecorder to get 1-second blobs which are sent to a server. On the server-side, I send these blobs over to the Google speech to the text interface. However, I am getting an empty transcriptions.
I know what the issue is. Mediarecorder's default Mime Type id audio/WebM codec=opus, which is not accepted by google's speech to text API. After doing some research, I realize I need to use ffmpeg to convert blobs to LInear16. However, ffmpeg only accepts audio FILES and I want to be able to convert BLOBS. Then I can send the resulting converted blobs over to the API interface.
server.js
wsserver.on('connection', socket => {
console.log("Listening on port 3002")
audio = {
content: null
}
socket.on('message',function(message){
// const buffer = new Int16Array(message, 0, Math.floor(data.byteLength / 2));
// console.log(`received from a client: ${new Uint8Array(message)}`);
// console.log(message);
audio.content = message.toString('base64')
console.log(audio.content);
livetranscriber.createRequest(audio).then(request => {
livetranscriber.recognizeStream(request);
});
});
});
livetranscriber
module.exports = {
createRequest: function(audio){
const encoding = 'LINEAR16';
const sampleRateHertz = 16000;
const languageCode = 'en-US';
return new Promise((resolve, reject, err) =>{
if (err){
reject(err)
}
else{
const request = {
audio: audio,
config: {
encoding: encoding,
sampleRateHertz: sampleRateHertz,
languageCode: languageCode,
},
interimResults: false, // If you want interim results, set this to true
};
resolve(request);
}
});
},
recognizeStream: async function(request){
const [response] = await client.recognize(request)
const transcription = response.results
.map(result => result.alternatives[0].transcript)
.join('\n');
console.log(`Transcription: ${transcription}`);
// console.log(message);
// message.pipe(recognizeStream);
},
}
client
recorder.ondataavailable = function(e) {
console.log('Data', e.data);
var ws = new WebSocket('ws://localhost:3002/websocket');
ws.onopen = function() {
console.log("opening connection");
// const stream = websocketStream(ws)
// const duplex = WebSocket.createWebSocketStream(ws, { encoding: 'utf8' });
var blob = new Blob(e, { 'type' : 'audio/wav; base64' });
ws.send(blob.data);
// e.data).pipe(stream);
// console.log(e.data);
console.log("Sent the message")
};
// chunks.push(e.data);
// socket.emit('data', e.data);
}
