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