# Proposal: Add Voice Communication System

**Change ID:** `add-voice-communication`
**Status:** Proposed
**Type:** Feature
**Priority:** Critical (MVP)
**Target Release:** v0.1.0

## Summary

Implement the core voice communication system for OpenSpeak, enabling real-time voice transmission between clients through the server. This includes audio capture, Opus encoding, packet routing, and playback functionality.

## Problem Statement

OpenSpeak requires a real-time voice communication system where:
- Users can capture audio from microphones and transmit to server
- Server routes voice packets to all members in a channel
- Users receive and play back multiple concurrent voice streams
- Latency is minimized (<100ms round-trip)
- Audio quality is maintained while optimizing bandwidth

## Solution Overview

Implement a voice streaming system using:
- **Opus codec** for encoding/decoding at 64kbps (8-128kbps configurable)
- **gRPC bidirectional streaming** for real-time packet transport
- **Server broadcast model** where server receives packets and broadcasts to channel
- **Client-side audio mixing** for multiple speakers
- **Jitter buffer** for handling packet timing variations

## Impact

### Affected Capabilities
- New: Voice Communication
- New: Audio Streaming
- New: Voice Routing
- Depends on: Channel Management, Authentication, Presence Tracking

### Users/Stakeholders
- End users: Can speak and hear in voice channels
- Developers: Must implement audio subsystem
- DevOps: Must support audio packet forwarding

## Success Criteria

- [ ] Voice packets route correctly from source to channel members
- [ ] Audio latency is <100ms round-trip in typical network conditions
- [ ] Supports 10+ concurrent speakers in single channel
- [ ] Opus encoding/decoding works with <5% CPU per stream
- [ ] Handles packet loss up to 2% without noticeable degradation
- [ ] Unit test coverage >80% for voice subsystem
- [ ] Integration tests pass for client-server voice communication

## Implementation Phases

### Phase 1: Core Voice Routing (Week 1-2)
- [ ] Define VoicePacket protobuf message
- [ ] Implement server voice router component
- [ ] Implement client voice capture and encoding
- [ ] Implement client voice reception and decoding

### Phase 2: Audio Quality (Week 2-3)
- [ ] Implement jitter buffer for timing
- [ ] Add packet loss handling
- [ ] Tune Opus bitrate settings
- [ ] Add volume normalization

### Phase 3: Integration & Testing (Week 3-4)
- [ ] Integration tests for voice communication
- [ ] Performance benchmarks
- [ ] Stress tests with many speakers
- [ ] Documentation and examples

## Risks & Mitigations

| Risk | Probability | Impact | Mitigation |
|------|-------------|--------|-----------|
| Audio library compatibility issues | Medium | High | Test with PortAudio, have fallback plan |
| Network latency exceeds target | Low | Medium | Implement jitter buffer, tune codec settings |
| Memory usage with many streams | Low | Medium | Implement stream pooling, monitor memory |
| CPU usage too high | Low | High | Profile early, optimize hot paths |

## Open Questions

1. Should we use PortAudio or OS-specific audio APIs?
2. What's the minimum jitter buffer size?
3. Should we implement echo cancellation?
4. Should voice activity detection be enabled by default?

## Approval Checklist

- [ ] Technical lead reviews architecture
- [ ] Audio library selection confirmed
- [ ] Performance targets agreed upon
- [ ] Timeline confirmed with team
