Streaming Stt Realtime

Browser captures audio at 48 kHz via getUserMedia but the WebSocket is opened to wss://api.assemblyai.com/v2/realtime/ws?sample_rate=16000 and frames stream without resampling.

sample_rate query param must match the actual stream. Either resample client-side to 16 kHz (Universal-streaming's preferred rate) before sending, OR pass sample_rate=48000 and accept higher bandwidth. A mismatch degrades accuracy silently — there is no server-side rejection.

Voice-agent application sets end_utterance_silence_threshold=2000 ms. End-users complain the agent feels slow because turn-end is detected too late.

end_utterance_silence_threshold trades latency against premature endpointing. Lower (e.g., 500-800 ms) for snappier voice agents; raise for dictation where long pauses are expected. Tune per workload and verify FinalTranscript boundaries land where expected. Document the tradeoff.

UI renders PartialTranscript events directly into the captions track. User sees text flicker and rewrite repeatedly as the model revises mid-utterance.

PartialTranscript is volatile and may be rewritten until a FinalTranscript with the matching audio span arrives. Render partials in a 'tentative' style (grey/italic), and commit to the captions track only on FinalTranscript. Do not append partials to a transcript log.