TranscribeLive takes audio feeds from WebRTC streams, feeding them to Whisper models for translation.
"Features sub-second latencies by processing audio inputs in overlapping 2-second buffer files."
WebRTC Pipeline
Combines React, Node.js, and Python. Audio streams are routed using WebRTC, sliced into raw buffer segments, and processed with faster-whisper pipelines.
System Architecture
[CLIENT AUDIO CAPTURING]
React.js with MediaRecorder API / WebRTC audio capture tracks
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[REAL-TIME STREAM WORKER]
Node.js WebRTC server / PyTorch Faster-Whisper pipeline / 2-second buffers
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[REAL-TIME TRANSCRIPT DELIVERY]
Socket.io Event Emitter (Direct screen text stream injection)
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