Phase 4 STT pipeline implemented — Silero VAD + faster-whisper — still not working well at all
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stt/README.md
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stt/README.md
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# Miku STT (Speech-to-Text) Server
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Real-time speech-to-text service for Miku voice chat using Silero VAD (CPU) and Faster-Whisper (GPU).
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## Architecture
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- **Silero VAD** (CPU): Lightweight voice activity detection, runs continuously
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- **Faster-Whisper** (GPU GTX 1660): Efficient speech transcription using CTranslate2
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- **FastAPI WebSocket**: Real-time bidirectional communication
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## Features
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- ✅ Real-time voice activity detection with conservative settings
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- ✅ Streaming partial transcripts during speech
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- ✅ Final transcript on speech completion
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- ✅ Interruption detection (user speaking over Miku)
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- ✅ Multi-user support with isolated sessions
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- ✅ KV cache optimization ready (partial text for LLM precomputation)
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## API Endpoints
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### WebSocket: `/ws/stt/{user_id}`
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Real-time STT session for a specific user.
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**Client sends:** Raw PCM audio (int16, 16kHz mono, 20ms chunks = 320 samples)
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**Server sends:** JSON events:
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```json
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// VAD events
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{"type": "vad", "event": "speech_start", "speaking": true, "probability": 0.85, "timestamp": 1250.5}
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{"type": "vad", "event": "speaking", "speaking": true, "probability": 0.92, "timestamp": 1270.5}
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{"type": "vad", "event": "speech_end", "speaking": false, "probability": 0.35, "timestamp": 3500.0}
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// Transcription events
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{"type": "partial", "text": "Hello how are", "user_id": "123", "timestamp": 2000.0}
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{"type": "final", "text": "Hello how are you?", "user_id": "123", "timestamp": 3500.0}
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// Interruption detection
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{"type": "interruption", "probability": 0.92, "timestamp": 1500.0}
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```
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### HTTP GET: `/health`
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Health check with model status.
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**Response:**
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```json
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{
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"status": "healthy",
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"models": {
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"vad": {"loaded": true, "device": "cpu"},
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"whisper": {"loaded": true, "model": "small", "device": "cuda"}
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},
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"sessions": {
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"active": 2,
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"users": ["user123", "user456"]
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}
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}
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```
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## Configuration
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### VAD Parameters (Conservative)
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- **Threshold**: 0.5 (speech probability)
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- **Min speech duration**: 250ms (avoid false triggers)
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- **Min silence duration**: 500ms (don't cut off mid-sentence)
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- **Speech padding**: 30ms (context around speech)
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### Whisper Parameters
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- **Model**: small (balanced speed/quality, ~500MB VRAM)
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- **Compute**: float16 (GPU optimization)
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- **Language**: en (English)
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- **Beam size**: 5 (quality/speed balance)
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## Usage Example
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```python
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import asyncio
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import websockets
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import numpy as np
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async def stream_audio():
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uri = "ws://localhost:8001/ws/stt/user123"
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async with websockets.connect(uri) as websocket:
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# Wait for ready
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ready = await websocket.recv()
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print(ready)
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# Stream audio chunks (16kHz, 20ms chunks)
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for audio_chunk in audio_stream:
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# Convert to bytes (int16)
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audio_bytes = audio_chunk.astype(np.int16).tobytes()
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await websocket.send(audio_bytes)
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# Receive events
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event = await websocket.recv()
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print(event)
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asyncio.run(stream_audio())
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```
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## Docker Setup
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### Build
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```bash
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docker-compose build miku-stt
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```
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### Run
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```bash
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docker-compose up -d miku-stt
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```
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### Logs
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```bash
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docker-compose logs -f miku-stt
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```
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### Test
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```bash
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curl http://localhost:8001/health
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```
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## GPU Sharing with Soprano
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Both STT (Whisper) and TTS (Soprano) run on GTX 1660 but at different times:
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1. **User speaking** → Whisper active, Soprano idle
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2. **LLM processing** → Both idle
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3. **Miku speaking** → Soprano active, Whisper idle (VAD monitoring only)
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Interruption detection runs VAD continuously but doesn't use GPU.
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## Performance
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- **VAD latency**: 10-20ms per chunk (CPU)
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- **Whisper latency**: ~1-2s for 2s audio (GPU)
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- **Memory usage**:
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- Silero VAD: ~100MB (CPU)
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- Faster-Whisper small: ~500MB (GPU VRAM)
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## Future Improvements
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- [ ] Multi-language support (auto-detect)
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- [ ] Word-level timestamps for better sync
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- [ ] Custom vocabulary/prompt tuning
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- [ ] Speaker diarization (multiple speakers)
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- [ ] Noise suppression preprocessing
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