Phase 3: Voice Features - COMPLETE ✓

Date: 2025-11-23 Status: ✅ Complete Commit: eefee13 Branch: claude/review-codebase-planning-01BPQKdZZnAgjqJ8F3ztUYtV

Overview

Phase 3 successfully implements voice input and audio playback features for the VoiceAssist web application, enabling push-to-talk transcription and text-to-speech for assistant responses.

Completed Features

Backend Implementation

Voice API Endpoints (services/api-gateway/app/api/voice.py)

1. POST /voice/transcribe

   • Audio transcription using OpenAI Whisper API
   • Supports multiple audio formats (webm, mp3, wav, etc.)
   • 25MB file size limit
   • Real-time transcription with error handling
   • Authenticated endpoint with user tracking

2. POST /voice/synthesize

   • Text-to-speech using OpenAI TTS API
   • Multiple voice options (alloy, echo, fable, onyx, nova, shimmer)
   • MP3 audio output format
   • 4096 character text limit
   • Streaming audio response

Integration

• Voice router registered in main application (services/api-gateway/app/main.py)
• CORS middleware configured for audio endpoints
• Rate limiting applied
• Comprehensive logging and error handling

Frontend Implementation

Audio Playback (apps/web-app/src/components/chat/MessageBubble.tsx)

1. Play Audio Button

   • Appears on all assistant messages
   • On-demand audio synthesis
   • Loading state during generation
   • Error handling with user-friendly messages

2. Audio Player Integration

   • Custom AudioPlayer component with controls
   • Play/pause functionality
   • Progress bar with seek capability
   • Duration display
   • Auto-cleanup of audio resources

3. Voice Input (Already implemented in MessageInput)

   • Push-to-talk functionality
   • Real-time transcription display
   • MediaRecorder API integration
   • Visual feedback during recording
   • Automatic transcript insertion

Technical Details

API Client Methods

// Transcribe audio to text
apiClient.transcribeAudio(audioBlob: Blob): Promise<string>

// Synthesize speech from text
apiClient.synthesizeSpeech(text: string, voiceId?: string): Promise<Blob>

Voice Components

• VoiceInput.tsx - Push-to-talk recording interface
• AudioPlayer.tsx - Audio playback with controls
• VoiceSettings.tsx - Voice preferences (speed, volume, auto-play)

User Experience

Voice Input Flow

1. User clicks microphone button in message input
2. Voice input panel appears
3. User presses and holds "Record" button
4. Audio is recorded and sent to backend
5. Transcribed text appears in message input
6. User can edit and send the message

Audio Playback Flow

1. Assistant message appears
2. User clicks "Play Audio" button
3. Audio is synthesized on-demand
4. AudioPlayer component appears with controls
5. User can play/pause and seek through audio
6. Audio is cached for repeated playback

Error Handling

• Microphone Access Denied: Clear error message with instructions
• Transcription Failure: Retry option with error details
• Synthesis Failure: Error message with dismiss button
• Network Errors: Timeout handling and user feedback
• File Size Limits: Validation with clear error messages

Performance Considerations

• On-Demand Synthesis: Audio only generated when requested
• Blob Caching: Audio cached in component state for repeat playback
• Lazy Loading: Audio player only rendered when needed
• Resource Cleanup: Proper cleanup of audio URLs and streams

Security & Privacy

• Authentication Required: All voice endpoints require valid JWT
• Input Validation: File type, size, and content validation
• PHI Protection: Audio not persisted server-side
• HTTPS Only: Encrypted transmission of audio data
• User Consent: Microphone access requires browser permission

Testing Status

• ✅ Backend endpoints created and syntax-validated
• ✅ Frontend components integrated
• ⏳ End-to-end testing pending (requires OpenAI API key)
• ⏳ Audio quality testing
• ⏳ Cross-browser compatibility testing

Dependencies

Backend

• OpenAI Whisper API (audio transcription)
• OpenAI TTS API (speech synthesis)
• httpx for async HTTP requests
• FastAPI for API framework

Frontend

• MediaRecorder API (browser)
• Web Audio API (browser)
• React hooks for state management
• Zustand for auth state

Next Steps

1. ✅ COMPLETED: Basic voice features
2. ⏳ Phase 4: File upload (PDF, images)
3. ⏳ Phase 5: Clinical context forms
4. ⏳ Phase 6: Citation sidebar
5. ⏳ Milestone 2: Advanced voice (WebRTC, VAD, barge-in)

Known Limitations (MVP)

• No continuous mode: Only push-to-talk (hands-free mode deferred to Milestone 2)
• No barge-in: Can't interrupt assistant while speaking
• No Voice Activity Detection: Manual start/stop required
• Single voice: Multiple voice options UI prepared but not yet implemented
• No voice settings persistence: Settings reset on page reload

Deferred Features (Milestone 2)

The following advanced voice features are deferred to Milestone 2 (Weeks 19-20):

• WebRTC audio streaming for lower latency
• Voice Activity Detection (VAD) for hands-free mode
• Echo cancellation and noise suppression
• Barge-in support for natural conversation
• Voice authentication
• OpenAI Realtime API integration

Files Changed

Created

• services/api-gateway/app/api/voice.py (+267 lines)

Modified

• services/api-gateway/app/main.py (+2 lines)
• apps/web-app/src/components/chat/MessageBubble.tsx (+111 lines)

Total: 380 lines added across 3 files

Commit Message

feat(voice): implement voice features - transcription and speech synthesis

Phase 3 - Voice Features Implementation
- Backend: OpenAI Whisper + TTS integration
- Frontend: Audio playback for assistant messages
- Voice input already integrated (MessageInput)
- Push-to-talk, on-demand synthesis, proper error handling

Progress: Milestone 1, Phase 3 Complete
Next: Phase 4 (File Upload)

🎉 Phase 3 Complete! Voice features successfully implemented and pushed to GitHub.