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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---
24 lines
558 B
JSON
24 lines
558 B
JSON
{
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"compilerOptions": {
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"tsBuildInfoFile": "./node_modules/.tmp/tsconfig.node.tsbuildinfo",
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"target": "es2023",
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"lib": ["ES2023"],
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"types": ["node"],
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"skipLibCheck": true,
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/* Bundler mode */
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"module": "nodenext",
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"allowImportingTsExtensions": true,
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"verbatimModuleSyntax": true,
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"moduleDetection": "force",
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"noEmit": true,
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/* Linting */
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"noUnusedLocals": true,
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"noUnusedParameters": true,
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"erasableSyntaxOnly": true,
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"noFallthroughCasesInSwitch": true
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},
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"include": ["vite.config.ts"]
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}
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