How to pass audio stream recorded with WebRTC to Google Speech api for realtime transcription?

3
votes

What I'm trying to do is get real time transcription for video recorded in the browser with webRTC. Use case is basically subtitles in real time like google hangouts has.

So I have a WebRTC program running in the browser. It sends webm objects back to the server. They are linear32 audio encodings. Google speech to text only accepts linear16 or Flac files.

Is there a way to convert linear32 to linear16 in real time?

Otherwise has anyone been able to hook up webRTC with Google speech to get real time transcriptions working?

Any advice on where to look to solve this problem would be great

1
webrtcgoogle-speech-apimatroska

1 Answers

1
votes

Check out this repository it might help you - https://github.com/muaz-khan/Translator

Translator.js is a JavaScript library built top on Google Speech-Recognition & Translation API to transcript and translate voice and text. It supports many locales and brings globalization in WebRTC!