webrtc: Several fixes for performance and user friendliness

[j1elo]

webrtc: Several fixes for performance and user friendliness

Several small changes that I made to make the demo look better and work more reliably.

Each change is individually self-contained on its own commit.

Changes in this PR:

README

• Do not recommend changing internal Chrome flags. This is very unsecure and could open the door to attacks. Instead, web browsers consider "localhost" as a privileged exception to the Secure Context rule, which means that it can be used without SSL to access the mic.

• Add instructions for running with HTTPS, in case someone wants to test the demo in their LAN instead of localhost.

WebRTC server

• Add support for passing an SSL key file (VOSK_KEY_FILE), in addition to the certificate file (VOSK_CERT_FILE).

• Fix asyncio event loop reference.

  The get_event_loop() call was creating a top-level loop which is not what is intended by the code in __run_audio_xfer(), which runs already as a Task with its own event loop, so get_running_loop() should be used instead, inside the Task. Also, get_event_loop() is deprecated.

• Improved handling of AcceptWaveform() results

  Make process_chunk() return None if an error is raised by the Python layer, or if an empty PartialResult is returned. This avoids lots of redundant DataChannel messages with empty payload.

• webrtc/server: Minor optimizations

  • max_frames_len never changes, so no need to update it on every loop iteration.

  • Send an empty object to the HTML UI, so the user knows when DataChannel has been connected and the demo is ready.

Demo HTML

• webrtc/static: Show a prompt (>) on the "partial" div

  Improve visual feedback to the user, by showing a small prompt indicator on the div where "partial" messagess will appear. This gives the feeling that the partial line does indeed contain intermediate results.

• webrtc/static: Improve DataChannel handling code

  Catch errors, rename Text result div to "text", and use more robut checks on object members.

• webrtc/static: Disable favicon to avoid errors on page loading

[nshmyrev]

Amazing. Huge thanks!

[j1elo]

You're welcome! Actually thank you a lot, for maintaining this server. It wouldn't have occurred to me that aiortc provides a non-standard track.recv() method to retrieve already uncompressed audio frames! That's so useful to avoid a decoding step between WebRTC (OPUS audio) and the Vosk recognizer.