https://github.com/rviscarra/webrtc-speech-to-text

Speech transcription on the browser using WebRTC and Google Speech
https://github.com/rviscarra/webrtc-speech-to-text

Last synced: 3 months ago
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Speech transcription on the browser using WebRTC and Google Speech

• Host: GitHub
• URL: https://github.com/rviscarra/webrtc-speech-to-text
• Owner: rviscarra
• License: mit
• Created: 2019-06-20T07:07:51.000Z (over 5 years ago)
• Default Branch: master
• Last Pushed: 2019-06-24T06:44:52.000Z (over 5 years ago)
• Last Synced: 2024-08-01T19:45:57.978Z (7 months ago)
• Language: Go
• Size: 275 KB
• Stars: 118
• Watchers: 6
• Forks: 21
• Open Issues: 3
• Metadata Files:
  • Readme: README.md
  • License: LICENSE

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• awesome-pion - rviscarra/webrtc-speech-to-text - Speech transcription on the browser using WebRTC and Google Speech (Media API)

README

        
## WebRTC speech to text server

### Dependencies

The speech to text server only depends on [Go 1.12](https://golang.org/doc/install).

### Disclaimer

**This project is only a proof of concept and SHOULDN'T be deployed on a production 

environment as it lacks even the most basic security measures.**

### Build the project

The project includes a Makefile, to build execute:

```bash

make

```

This should generate a `transcribe-server` binary.

### Running the server

The server receives the following flags through the command line:

`--google.cred` (Required)

Path to a Google Cloud credential's file. The project must have Google Speech activated.

`--http.port` (Optional) 

Specifies the port where the HTTP server should listen, by default the port 9000 is used.

`--stun.server` (Optional)

Allows to speficy a different [STUN](https://es.wikipedia.org/wiki/STUN) server, by default a Google STUN server is used.

### Demo page

The demo works on Chrome 75, Firefox 67 and Safari 12.1.1

![Demo screenshot](docs/demo.png)

To run the demo execute the server and navigate to `http://localhost:9000`. 

After pressing the **Start** button a dialog asking for permission to access the microphone should appear. 

After grating access a WebRTC connection is made to the local server, where audio data is decoded and streamed 

to Google Speech (streaming recognition).

Say something and press the **Stop** button, the results (if any) should appear on screen.

### Architecture

![Architecture and data flow](docs/architecture.png)

### Coming soon

- Unit tests.

- Be able to specify the desired language.

- Support for interim results.

### License

MIT - see [LICENSE](LICENSE) for the full text.