Praxis CI
7b1b296430
client/ — React + Vite + TypeScript scaffolded with the Pipecat client SDK (@pipecat-ai/client-js) and SmallWebRTCTransport (@pipecat-ai/small-webrtc-transport). useVoiceSession.ts hook manages mic permission, WebRTC connect, audio playback, live transcript, and a latency readout (captures the e2e_latency_ms metric the server emits). App.tsx is a minimal one-page session UI: disclaimer, Start/End buttons, status badge, latency readout (within/over 600ms budget), live transcript. vite.config.ts proxies /pipecat + /health to the Python server (port 8789). npm run typecheck + npm run build pass. server/latency.py — LatencyObserver (a Pipecat FrameProcessor) timestamps transcript-ready, LLM-first-token, TTS-first-audio, and playback-start per turn, computes ASR→TTS-first-audio (the v0.1 latency target), and logs it to console with a within/over-budget verdict. Wired into the pipeline between STT/LLM/TTS so it observes without altering the frame stream. 5 unit tests pass (LatencyRecord e2e math + observer construction + reset_turn). Full server suite: 18 passed. ---ci--- phase: 1 milestone: v0.1 plan: 02 task: 02-05,02-06 status: execute persona: frontend-engineer,backend-engineer requirements: covered: [REQ-VOICE-01, REQ-VOICE-02, REQ-VOICE-03, REQ-NFR-LAT-01] ---/ci---
14 lines
358 B
HTML
14 lines
358 B
HTML
<!doctype html>
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<html lang="en">
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<head>
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<meta charset="UTF-8" />
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<link rel="icon" type="image/svg+xml" href="/favicon.svg" />
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<meta name="viewport" content="width=device-width, initial-scale=1.0" />
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<title>client</title>
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</head>
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<body>
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<div id="root"></div>
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<script type="module" src="/src/main.tsx"></script>
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</body>
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</html>
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