| Commit message (Collapse) | Author | Age |
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Create a simple server with 3 endpoints:
* /create_session: Create a session and return its identifier.
* /set_transcript: Update a session's transcript.
* /get_transcript: Fetch a session's transcript.
Right now the session ID provides authentication *and* authorization.
There is no public/private ID so you have to trust whoever you share
your ID with.
IDs are long and generated by the server, so it should be somewhat
secure against low-effort hacking.
Other updates:
* Drop whisper_requirements.txt - no longer needed.
* Vendor curl to make it easier to interact with the server.
TODO:
* Fuzz test the server.
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Circle goes red when speaking, grey when done. Ideally it would be in
the top right portion of the browser source, but this is a good start.
Also, hard-cap transcripts to 4096 chars. This prevents the STT from
lagging during long sessions.
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Should improve legibility.
* Update README
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Transcription output now streams to localhost:8097.
In OBS:
* Create a browser source.
* url: localhost:8097
* width: 2200
* height: 400
TODO:
* Put behind toggle.
* Create input field for port.
Misc cleanup:
* transcribe.py: Drop frames from audio capture thread instead of the
transcription thread. Doing it the other way would result in
occasional data loss.
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This is a much faster, lower-VRAM reimplementation of Whisper in Python.
Early testing is extremely promising: fast transcription speed,
extremely low resource usage (CPU/RAM/VRAM), high accuracy.
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* Implement HTTPMapper classes
* Browser source respects user-configured source port
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It's a crashy mess, but it sort of works.
* Add Transcript class to send transcription segments between layers
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Browser source queries /api/transcript at 10Hz via jquery and renders
the response.
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