The system streams raw voice data via WebRTC, runs speech recognition on the server, translates the output, and clones the voice back to the user.
Explore basic real-time media streaming via WebRTC combined with speech recognition (ASR) pipelines.
The system streams raw voice data via WebRTC, runs speech recognition on the server, translates the output, and clones the voice back to the user.
Deep dive into real-time WebRTC media ingestion, Whisper ASR chunking, LLM semantic translation, and neural TTS voice cloning with lip-sync synchronization.
Analysis of implementing automated geo-targeting, customized currency routing, and structured JSON-LD schemas to achieve sub-100ms loading speeds.