"""
STT Server

FastAPI WebSocket server for real-time speech-to-text.
Combines Silero VAD (CPU) and Faster-Whisper (GPU) for efficient transcription.

Architecture:
- VAD runs continuously on every audio chunk (CPU)
- Whisper transcribes only when VAD detects speech (GPU)
- Supports multiple concurrent users
- Sends partial and final transcripts via WebSocket
"""

from fastapi import FastAPI, WebSocket, WebSocketDisconnect, HTTPException
from fastapi.responses import JSONResponse
import numpy as np
import asyncio
import logging
from typing import Dict, Optional
from datetime import datetime

from vad_processor import VADProcessor
from whisper_transcriber import WhisperTranscriber

# Configure logging
logging.basicConfig(
    level=logging.INFO,
    format='[%(levelname)s] [%(name)s] %(message)s'
)
logger = logging.getLogger('stt_server')

# Initialize FastAPI app
app = FastAPI(title="Miku STT Server", version="1.0.0")

# Global instances (initialized on startup)
vad_processor: Optional[VADProcessor] = None
whisper_transcriber: Optional[WhisperTranscriber] = None

# User session tracking
user_sessions: Dict[str, dict] = {}


class UserSTTSession:
    """Manages STT state for a single user."""
    
    def __init__(self, user_id: str, websocket: WebSocket):
        self.user_id = user_id
        self.websocket = websocket
        self.audio_buffer = []
        self.is_speaking = False
        self.timestamp_ms = 0.0
        self.transcript_buffer = []
        self.last_transcript = ""
        
        logger.info(f"Created STT session for user {user_id}")
    
    async def process_audio_chunk(self, audio_data: bytes):
        """
        Process incoming audio chunk.
        
        Args:
            audio_data: Raw PCM audio (int16, 16kHz mono)
        """
        # Convert bytes to numpy array (int16)
        audio_np = np.frombuffer(audio_data, dtype=np.int16)
        
        # Calculate timestamp (assuming 16kHz, 20ms chunks = 320 samples)
        chunk_duration_ms = (len(audio_np) / 16000) * 1000
        self.timestamp_ms += chunk_duration_ms
        
        # Run VAD on chunk
        vad_event = vad_processor.detect_speech_segment(audio_np, self.timestamp_ms)
        
        if vad_event:
            event_type = vad_event["event"]
            probability = vad_event["probability"]
            
            # Send VAD event to client
            await self.websocket.send_json({
                "type": "vad",
                "event": event_type,
                "speaking": event_type in ["speech_start", "speaking"],
                "probability": probability,
                "timestamp": self.timestamp_ms
            })
            
            # Handle speech events
            if event_type == "speech_start":
                self.is_speaking = True
                self.audio_buffer = [audio_np]
                logger.debug(f"User {self.user_id} started speaking")
            
            elif event_type == "speaking":
                if self.is_speaking:
                    self.audio_buffer.append(audio_np)
                    
                    # Transcribe partial every ~2 seconds for streaming
                    total_samples = sum(len(chunk) for chunk in self.audio_buffer)
                    duration_s = total_samples / 16000
                    
                    if duration_s >= 2.0:
                        await self._transcribe_partial()
            
            elif event_type == "speech_end":
                self.is_speaking = False
                
                # Transcribe final
                await self._transcribe_final()
                
                # Clear buffer
                self.audio_buffer = []
                logger.debug(f"User {self.user_id} stopped speaking")
        
        else:
            # Still accumulate audio if speaking
            if self.is_speaking:
                self.audio_buffer.append(audio_np)
    
    async def _transcribe_partial(self):
        """Transcribe accumulated audio and send partial result."""
        if not self.audio_buffer:
            return
        
        # Concatenate audio
        audio_full = np.concatenate(self.audio_buffer)
        
        # Transcribe asynchronously
        try:
            text = await whisper_transcriber.transcribe_async(
                audio_full,
                sample_rate=16000,
                initial_prompt=self.last_transcript  # Use previous for context
            )
            
            if text and text != self.last_transcript:
                self.last_transcript = text
                
                # Send partial transcript
                await self.websocket.send_json({
                    "type": "partial",
                    "text": text,
                    "user_id": self.user_id,
                    "timestamp": self.timestamp_ms
                })
                
                logger.info(f"Partial [{self.user_id}]: {text}")
        
        except Exception as e:
            logger.error(f"Partial transcription failed: {e}", exc_info=True)
    
    async def _transcribe_final(self):
        """Transcribe final accumulated audio."""
        if not self.audio_buffer:
            return
        
        # Concatenate all audio
        audio_full = np.concatenate(self.audio_buffer)
        
        try:
            text = await whisper_transcriber.transcribe_async(
                audio_full,
                sample_rate=16000
            )
            
            if text:
                self.last_transcript = text
                
                # Send final transcript
                await self.websocket.send_json({
                    "type": "final",
                    "text": text,
                    "user_id": self.user_id,
                    "timestamp": self.timestamp_ms
                })
                
                logger.info(f"Final [{self.user_id}]: {text}")
        
        except Exception as e:
            logger.error(f"Final transcription failed: {e}", exc_info=True)
    
    async def check_interruption(self, audio_data: bytes) -> bool:
        """
        Check if user is interrupting (for use during Miku's speech).
        
        Args:
            audio_data: Raw PCM audio chunk
        
        Returns:
            True if interruption detected
        """
        audio_np = np.frombuffer(audio_data, dtype=np.int16)
        speech_prob, is_speaking = vad_processor.process_chunk(audio_np)
        
        # Interruption: high probability sustained for threshold duration
        if speech_prob > 0.7:  # Higher threshold for interruption
            await self.websocket.send_json({
                "type": "interruption",
                "probability": speech_prob,
                "timestamp": self.timestamp_ms
            })
            return True
        
        return False


@app.on_event("startup")
async def startup_event():
    """Initialize models on server startup."""
    global vad_processor, whisper_transcriber
    
    logger.info("=" * 50)
    logger.info("Initializing Miku STT Server")
    logger.info("=" * 50)
    
    # Initialize VAD (CPU)
    logger.info("Loading Silero VAD model (CPU)...")
    vad_processor = VADProcessor(
        sample_rate=16000,
        threshold=0.5,
        min_speech_duration_ms=250,  # Conservative
        min_silence_duration_ms=500   # Conservative
    )
    logger.info("✓ VAD ready")
    
    # Initialize Whisper (GPU with cuDNN)
    logger.info("Loading Faster-Whisper model (GPU)...")
    whisper_transcriber = WhisperTranscriber(
        model_size="small",
        device="cuda",
        compute_type="float16",
        language="en"
    )
    logger.info("✓ Whisper ready")
    
    logger.info("=" * 50)
    logger.info("STT Server ready to accept connections")
    logger.info("=" * 50)


@app.on_event("shutdown")
async def shutdown_event():
    """Cleanup on server shutdown."""
    logger.info("Shutting down STT server...")
    
    if whisper_transcriber:
        whisper_transcriber.cleanup()
    
    logger.info("STT server shutdown complete")


@app.get("/")
async def root():
    """Health check endpoint."""
    return {
        "service": "Miku STT Server",
        "status": "running",
        "vad_ready": vad_processor is not None,
        "whisper_ready": whisper_transcriber is not None,
        "active_sessions": len(user_sessions)
    }


@app.get("/health")
async def health():
    """Detailed health check."""
    return {
        "status": "healthy",
        "models": {
            "vad": {
                "loaded": vad_processor is not None,
                "device": "cpu"
            },
            "whisper": {
                "loaded": whisper_transcriber is not None,
                "model": "small",
                "device": "cuda"
            }
        },
        "sessions": {
            "active": len(user_sessions),
            "users": list(user_sessions.keys())
        }
    }


@app.websocket("/ws/stt/{user_id}")
async def websocket_stt(websocket: WebSocket, user_id: str):
    """
    WebSocket endpoint for real-time STT.
    
    Client sends: Raw PCM audio (int16, 16kHz mono, 20ms chunks)
    Server sends: JSON events:
        - {"type": "vad", "event": "speech_start|speaking|speech_end", ...}
        - {"type": "partial", "text": "...", ...}
        - {"type": "final", "text": "...", ...}
        - {"type": "interruption", "probability": 0.xx}
    """
    await websocket.accept()
    logger.info(f"STT WebSocket connected: user {user_id}")
    
    # Create session
    session = UserSTTSession(user_id, websocket)
    user_sessions[user_id] = session
    
    try:
        # Send ready message
        await websocket.send_json({
            "type": "ready",
            "user_id": user_id,
            "message": "STT session started"
        })
        
        # Main loop: receive audio chunks
        while True:
            # Receive binary audio data
            data = await websocket.receive_bytes()
            
            # Process audio chunk
            await session.process_audio_chunk(data)
    
    except WebSocketDisconnect:
        logger.info(f"User {user_id} disconnected")
    
    except Exception as e:
        logger.error(f"Error in STT WebSocket for user {user_id}: {e}", exc_info=True)
    
    finally:
        # Cleanup session
        if user_id in user_sessions:
            del user_sessions[user_id]
        logger.info(f"STT session ended for user {user_id}")


@app.post("/interrupt/check")
async def check_interruption(user_id: str):
    """
    Check if user is interrupting (for use during Miku's speech).
    
    Query param:
        user_id: Discord user ID
    
    Returns:
        {"interrupting": bool, "probability": float}
    """
    session = user_sessions.get(user_id)
    
    if not session:
        raise HTTPException(status_code=404, detail="User session not found")
    
    # Get current VAD state
    vad_state = vad_processor.get_state()
    
    return {
        "interrupting": vad_state["speaking"],
        "user_id": user_id
    }


if __name__ == "__main__":
    import uvicorn
    uvicorn.run(app, host="0.0.0.0", port=8000, log_level="info")