Auto-commit 2026-05-02 09:37
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- Code Reviewer completed review and verified all fixes ✅
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- Assigned FRE-588 to Security Reviewer (036d6925-3aac-4939-a0f0-22dc44e618bc)
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- Waiting for Security Reviewer final validation and approval
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---
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## Afternoon Heartbeat
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**Wake time:** 2026-04-30T15:06:27Z
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**Run ID:** 284246dd-dd89-4382-b654-a1743b1793be
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**Wake reason:** continuation from FRE-4492
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### Assignments
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- FRE-4492: Integrate Stripe billing with tier management (completed, in_review)
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- FRE-4497: 5.1 WebRTC Real-Time Call Analysis (completed, in_review)
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- FRE-4501: 5.5 Integration & Testing (in_progress)
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### Actions Taken
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#### FRE-4492: Stripe Billing Integration
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- Fixed 4 code review issues from previous review:
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1. Added fallback defaults for paid tier env vars in `billing.config.ts`
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2. Fixed invoice creation to use proper line_items in `billing.service.ts`
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3. Added `withinLimit` to interface in `billing.middleware.ts`
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4. Added `withSubscription()` middleware and updated `requireSubscription()`
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- Marked as `in_review` and assigned to Code Reviewer
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#### FRE-4497: WebRTC Real-Time Call Analysis
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**Completed implementation:**
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- Created `RealTimeCallAnalysisServer` in `src/lib/call-analysis/real-time-call-server.ts`
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- Integrated WebRTC audio capture (`AudioStreamCapture`)
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- Integrated inference engine (`CallAnalysisEngine`)
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- Implemented WebSocket alert server (port 8089)
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- Features:
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- Real-time sentiment analysis
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- Call quality metrics (MOS score, jitter, latency, packet loss)
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- Event detection (interrupts, overlaps, long pauses, volume spikes, silence)
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- Anomaly detection (background noise, echo, distortion, dropouts, volume inconsistency)
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- Client control commands: start, stop, pause, resume, getMetrics, getEvents, reset
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- Created `src/lib/call-analysis/index.ts` for module exports
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- Created `examples/call-analysis-example.ts` demonstrating usage
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- Marked as `in_review` and assigned to Code Reviewer
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#### FRE-4501: Integration & Testing
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**Started:**
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- Checked out issue for work
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- Documented test coverage plan:
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- Integration tests for RealTimeCallAnalysisServer
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- Performance benchmarks for audio processing pipeline
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- End-to-end tests for WebSocket communication
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- Load testing for concurrent client connections
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- Next: Set up testing infrastructure and create test suites
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### Technical Decisions
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1. **WebSocket Protocol**: Used JSON-based messaging for simplicity and debuggability
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2. **Audio Processing**: 16kHz sample rate, 1024 frame size for real-time analysis
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3. **Analysis Sensitivity**: Configurable anomaly sensitivity (low/medium/high)
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4. **Event Broadcasting**: Only emit significant results to reduce bandwidth
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### Notes
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- Stripe types dependency is a pre-existing issue (not installed)
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- WebRTC signaling server reuses existing WebSocket infrastructure
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- All analysis components are modular and can be used independently
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### Next Action
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- Begin integration test suite for FRE-4501
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- Create test files:
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- `src/lib/call-analysis/real-time-call-server.test.ts`
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- `tests/integration/call-analysis.integration.test.ts`
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- `tests/performance/audio-processing.benchmark.test.ts`
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- `tests/e2e/websocket-call-analysis.e2e.test.ts`
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