"""Full duplex audio pipeline for AI voice conversation.

This module implements the core duplex pipeline that orchestrates:
- VAD (Voice Activity Detection)
- EOU (End of Utterance) Detection
- ASR (Automatic Speech Recognition) - optional
- LLM (Language Model)
- TTS (Text-to-Speech)

Inspired by pipecat's frame-based architecture and active-call's
event-driven design.
"""

import asyncio
import time
from typing import Optional, Callable, Awaitable
from loguru import logger

from core.transports import BaseTransport
from core.conversation import ConversationManager, ConversationState
from core.events import get_event_bus
from processors.vad import VADProcessor, SileroVAD
from processors.eou import EouDetector
from services.base import BaseLLMService, BaseTTSService, BaseASRService
from services.llm import OpenAILLMService, MockLLMService
from services.tts import EdgeTTSService, MockTTSService
from services.asr import BufferedASRService
from services.siliconflow_tts import SiliconFlowTTSService
from services.siliconflow_asr import SiliconFlowASRService
from app.config import settings


class DuplexPipeline:
    """
    Full duplex audio pipeline for AI voice conversation.
    
    Handles bidirectional audio flow with:
    - User speech detection and transcription
    - AI response generation
    - Text-to-speech synthesis
    - Barge-in (interruption) support
    
    Architecture (inspired by pipecat):
    
        User Audio → VAD → EOU → [ASR] → LLM → TTS → Audio Out
                      ↓
                  Barge-in Detection → Interrupt
    """
    
    def __init__(
        self,
        transport: BaseTransport,
        session_id: str,
        llm_service: Optional[BaseLLMService] = None,
        tts_service: Optional[BaseTTSService] = None,
        asr_service: Optional[BaseASRService] = None,
        system_prompt: Optional[str] = None,
        greeting: Optional[str] = None
    ):
        """
        Initialize duplex pipeline.
        
        Args:
            transport: Transport for sending audio/events
            session_id: Session identifier
            llm_service: LLM service (defaults to OpenAI)
            tts_service: TTS service (defaults to EdgeTTS)
            asr_service: ASR service (optional)
            system_prompt: System prompt for LLM
            greeting: Optional greeting to speak on start
        """
        self.transport = transport
        self.session_id = session_id
        self.event_bus = get_event_bus()
        
        # Initialize VAD
        self.vad_model = SileroVAD(
            model_path=settings.vad_model_path,
            sample_rate=settings.sample_rate
        )
        self.vad_processor = VADProcessor(
            vad_model=self.vad_model,
            threshold=settings.vad_threshold
        )
        
        # Initialize EOU detector
        self.eou_detector = EouDetector(
            silence_threshold_ms=settings.vad_eou_threshold_ms,
            min_speech_duration_ms=settings.vad_min_speech_duration_ms
        )
        
        # Initialize services
        self.llm_service = llm_service
        self.tts_service = tts_service
        self.asr_service = asr_service  # Will be initialized in start()
        
        # Track last sent transcript to avoid duplicates
        self._last_sent_transcript = ""
        
        # Conversation manager
        self.conversation = ConversationManager(
            system_prompt=system_prompt,
            greeting=greeting
        )
        
        # State
        self._running = True
        self._is_bot_speaking = False
        self._current_turn_task: Optional[asyncio.Task] = None
        self._audio_buffer: bytes = b""
        max_buffer_seconds = settings.max_audio_buffer_seconds if hasattr(settings, "max_audio_buffer_seconds") else 30
        self._max_audio_buffer_bytes = int(settings.sample_rate * 2 * max_buffer_seconds)
        self._last_vad_status: str = "Silence"
        self._process_lock = asyncio.Lock()
        
        # Interruption handling
        self._interrupt_event = asyncio.Event()
        
        # Latency tracking - TTFB (Time to First Byte)
        self._turn_start_time: Optional[float] = None
        self._first_audio_sent: bool = False
        
        # Barge-in filtering - require minimum speech duration to interrupt
        self._barge_in_speech_start_time: Optional[float] = None
        self._barge_in_min_duration_ms: int = settings.barge_in_min_duration_ms if hasattr(settings, 'barge_in_min_duration_ms') else 50
        self._barge_in_speech_frames: int = 0  # Count speech frames
        self._barge_in_silence_frames: int = 0  # Count silence frames during potential barge-in
        self._barge_in_silence_tolerance: int = 3  # Allow up to 3 silence frames (60ms at 20ms chunks)
        
        logger.info(f"DuplexPipeline initialized for session {session_id}")
    
    async def start(self) -> None:
        """Start the pipeline and connect services."""
        try:
            # Connect LLM service
            if not self.llm_service:
                if settings.openai_api_key:
                    self.llm_service = OpenAILLMService(
                        api_key=settings.openai_api_key,
                        base_url=settings.openai_api_url,
                        model=settings.llm_model
                    )
                else:
                    logger.warning("No OpenAI API key - using mock LLM")
                    self.llm_service = MockLLMService()
            
            await self.llm_service.connect()
            
            # Connect TTS service
            if not self.tts_service:
                if settings.tts_provider == "siliconflow" and settings.siliconflow_api_key:
                    self.tts_service = SiliconFlowTTSService(
                        api_key=settings.siliconflow_api_key,
                        voice=settings.tts_voice,
                        model=settings.siliconflow_tts_model,
                        sample_rate=settings.sample_rate,
                        speed=settings.tts_speed
                    )
                    logger.info("Using SiliconFlow TTS service")
                else:
                    self.tts_service = EdgeTTSService(
                        voice=settings.tts_voice,
                        sample_rate=settings.sample_rate
                    )
                    logger.info("Using Edge TTS service")
            
            await self.tts_service.connect()
            
            # Connect ASR service
            if not self.asr_service:
                if settings.asr_provider == "siliconflow" and settings.siliconflow_api_key:
                    self.asr_service = SiliconFlowASRService(
                        api_key=settings.siliconflow_api_key,
                        model=settings.siliconflow_asr_model,
                        sample_rate=settings.sample_rate,
                        interim_interval_ms=settings.asr_interim_interval_ms,
                        min_audio_for_interim_ms=settings.asr_min_audio_ms,
                        on_transcript=self._on_transcript_callback
                    )
                    logger.info("Using SiliconFlow ASR service")
                else:
                    self.asr_service = BufferedASRService(
                        sample_rate=settings.sample_rate
                    )
                    logger.info("Using Buffered ASR service (no real transcription)")
            
            await self.asr_service.connect()
            
            logger.info("DuplexPipeline services connected")
            
            # Speak greeting if configured
            if self.conversation.greeting:
                await self._speak(self.conversation.greeting)
            
        except Exception as e:
            logger.error(f"Failed to start pipeline: {e}")
            raise
    
    async def process_audio(self, pcm_bytes: bytes) -> None:
        """
        Process incoming audio chunk.
        
        This is the main entry point for audio from the user.
        
        Args:
            pcm_bytes: PCM audio data (16-bit, mono, 16kHz)
        """
        if not self._running:
            return
        
        try:
            async with self._process_lock:
                # 1. Process through VAD
                vad_result = self.vad_processor.process(pcm_bytes, settings.chunk_size_ms)
            
                vad_status = "Silence"
                if vad_result:
                    event_type, probability = vad_result
                    vad_status = "Speech" if event_type == "speaking" else "Silence"
                    
                    # Emit VAD event
                    await self.event_bus.publish(event_type, {
                        "trackId": self.session_id,
                        "probability": probability
                    })
                else:
                    # No state change - keep previous status
                    vad_status = self._last_vad_status
                
                # Update state based on VAD
                if vad_status == "Speech" and self._last_vad_status != "Speech":
                    await self._on_speech_start()
                
                self._last_vad_status = vad_status
                
                # 2. Check for barge-in (user speaking while bot speaking)
                # Filter false interruptions by requiring minimum speech duration
                if self._is_bot_speaking:
                    if vad_status == "Speech":
                        # User is speaking while bot is speaking
                        self._barge_in_silence_frames = 0  # Reset silence counter
                        
                        if self._barge_in_speech_start_time is None:
                            # Start tracking speech duration
                            self._barge_in_speech_start_time = time.time()
                            self._barge_in_speech_frames = 1
                            logger.debug("Potential barge-in detected, tracking duration...")
                        else:
                            self._barge_in_speech_frames += 1
                            # Check if speech duration exceeds threshold
                            speech_duration_ms = (time.time() - self._barge_in_speech_start_time) * 1000
                            if speech_duration_ms >= self._barge_in_min_duration_ms:
                                logger.info(f"Barge-in confirmed after {speech_duration_ms:.0f}ms of speech ({self._barge_in_speech_frames} frames)")
                                await self._handle_barge_in()
                    else:
                        # Silence frame during potential barge-in
                        if self._barge_in_speech_start_time is not None:
                            self._barge_in_silence_frames += 1
                            # Allow brief silence gaps (VAD flickering)
                            if self._barge_in_silence_frames > self._barge_in_silence_tolerance:
                                # Too much silence - reset barge-in tracking
                                logger.debug(f"Barge-in cancelled after {self._barge_in_silence_frames} silence frames")
                                self._barge_in_speech_start_time = None
                                self._barge_in_speech_frames = 0
                                self._barge_in_silence_frames = 0
                
                # 3. Buffer audio for ASR
                if vad_status == "Speech" or self.conversation.state == ConversationState.LISTENING:
                    self._audio_buffer += pcm_bytes
                    if len(self._audio_buffer) > self._max_audio_buffer_bytes:
                        # Keep only the most recent audio to cap memory usage
                        self._audio_buffer = self._audio_buffer[-self._max_audio_buffer_bytes:]
                    await self.asr_service.send_audio(pcm_bytes)
                    
                    # For SiliconFlow ASR, trigger interim transcription periodically
                    # The service handles timing internally via start_interim_transcription()
                
                # 4. Check for End of Utterance - this triggers LLM response
                if self.eou_detector.process(vad_status):
                    await self._on_end_of_utterance()
            
        except Exception as e:
            logger.error(f"Pipeline audio processing error: {e}", exc_info=True)
    
    async def process_text(self, text: str) -> None:
        """
        Process text input (chat command).
        
        Allows direct text input to bypass ASR.
        
        Args:
            text: User text input
        """
        if not self._running:
            return
        
        logger.info(f"Processing text input: {text[:50]}...")
        
        # Cancel any current speaking
        await self._stop_current_speech()
        
        # Start new turn
        await self.conversation.end_user_turn(text)
        self._current_turn_task = asyncio.create_task(self._handle_turn(text))
    
    async def interrupt(self) -> None:
        """Interrupt current bot speech (manual interrupt command)."""
        await self._handle_barge_in()
    
    async def _on_transcript_callback(self, text: str, is_final: bool) -> None:
        """
        Callback for ASR transcription results.
        
        Streams transcription to client for display.
        
        Args:
            text: Transcribed text
            is_final: Whether this is the final transcription
        """
        # Avoid sending duplicate transcripts
        if text == self._last_sent_transcript and not is_final:
            return
        
        self._last_sent_transcript = text
        
        # Send transcript event to client
        await self.transport.send_event({
            "event": "transcript",
            "trackId": self.session_id,
            "text": text,
            "isFinal": is_final,
            "timestamp": self._get_timestamp_ms()
        })
        
        logger.debug(f"Sent transcript ({'final' if is_final else 'interim'}): {text[:50]}...")
    
    async def _on_speech_start(self) -> None:
        """Handle user starting to speak."""
        if self.conversation.state == ConversationState.IDLE:
            await self.conversation.start_user_turn()
            self._audio_buffer = b""
            self._last_sent_transcript = ""
            self.eou_detector.reset()
            
            # Clear ASR buffer and start interim transcriptions
            if hasattr(self.asr_service, 'clear_buffer'):
                self.asr_service.clear_buffer()
            if hasattr(self.asr_service, 'start_interim_transcription'):
                await self.asr_service.start_interim_transcription()
            
            logger.debug("User speech started")
    
    async def _on_end_of_utterance(self) -> None:
        """Handle end of user utterance."""
        if self.conversation.state != ConversationState.LISTENING:
            return
        
        # Stop interim transcriptions
        if hasattr(self.asr_service, 'stop_interim_transcription'):
            await self.asr_service.stop_interim_transcription()
        
        # Get final transcription from ASR service
        user_text = ""
        
        if hasattr(self.asr_service, 'get_final_transcription'):
            # SiliconFlow ASR - get final transcription
            user_text = await self.asr_service.get_final_transcription()
        elif hasattr(self.asr_service, 'get_and_clear_text'):
            # Buffered ASR - get accumulated text
            user_text = self.asr_service.get_and_clear_text()
        
        # Skip if no meaningful text
        if not user_text or not user_text.strip():
            logger.debug("EOU detected but no transcription - skipping")
            # Reset for next utterance
            self._audio_buffer = b""
            self._last_sent_transcript = ""
            # Return to idle; don't force LISTENING which causes buffering on silence
            await self.conversation.set_state(ConversationState.IDLE)
            return
        
        logger.info(f"EOU detected - user said: {user_text[:100]}...")
        
        # Send final transcription to client
        await self.transport.send_event({
            "event": "transcript",
            "trackId": self.session_id,
            "text": user_text,
            "isFinal": True,
            "timestamp": self._get_timestamp_ms()
        })
        
        # Clear buffers
        self._audio_buffer = b""
        self._last_sent_transcript = ""
        
        # Process the turn - trigger LLM response
        # Cancel any existing turn to avoid overlapping assistant responses
        await self._stop_current_speech()
        await self.conversation.end_user_turn(user_text)
        self._current_turn_task = asyncio.create_task(self._handle_turn(user_text))
    
    async def _handle_turn(self, user_text: str) -> None:
        """
        Handle a complete conversation turn.
        
        Uses sentence-by-sentence streaming TTS for lower latency.
        
        Args:
            user_text: User's transcribed text
        """
        try:
            # Start latency tracking
            self._turn_start_time = time.time()
            self._first_audio_sent = False
            
            # Get AI response (streaming)
            messages = self.conversation.get_messages()
            full_response = ""
            
            await self.conversation.start_assistant_turn()
            self._is_bot_speaking = True
            self._interrupt_event.clear()
            
            # Sentence buffer for streaming TTS
            sentence_buffer = ""
            sentence_ends = {'，', '。', '！', '？', '\n'}
            first_audio_sent = False
            
            # Stream LLM response and TTS sentence by sentence
            async for text_chunk in self.llm_service.generate_stream(messages):
                if self._interrupt_event.is_set():
                    break
                
                full_response += text_chunk
                sentence_buffer += text_chunk
                await self.conversation.update_assistant_text(text_chunk)
                
                # Send LLM response streaming event to client
                await self.transport.send_event({
                    "event": "llmResponse",
                    "trackId": self.session_id,
                    "text": text_chunk,
                    "isFinal": False,
                    "timestamp": self._get_timestamp_ms()
                })
                
                # Check for sentence completion - synthesize immediately for low latency
                while any(end in sentence_buffer for end in sentence_ends):
                    # Find first sentence end
                    min_idx = len(sentence_buffer)
                    for end in sentence_ends:
                        idx = sentence_buffer.find(end)
                        if idx != -1 and idx < min_idx:
                            min_idx = idx
                    
                    if min_idx < len(sentence_buffer):
                        sentence = sentence_buffer[:min_idx + 1].strip()
                        sentence_buffer = sentence_buffer[min_idx + 1:]
                        
                        if sentence and not self._interrupt_event.is_set():
                            # Send track start on first audio
                            if not first_audio_sent:
                                await self.transport.send_event({
                                    "event": "trackStart",
                                    "trackId": self.session_id,
                                    "timestamp": self._get_timestamp_ms()
                                })
                                first_audio_sent = True
                            
                            # Synthesize and send this sentence immediately
                            await self._speak_sentence(sentence)
                    else:
                        break
            
            # Send final LLM response event
            if full_response and not self._interrupt_event.is_set():
                await self.transport.send_event({
                    "event": "llmResponse",
                    "trackId": self.session_id,
                    "text": full_response,
                    "isFinal": True,
                    "timestamp": self._get_timestamp_ms()
                })
            
            # Speak any remaining text
            if sentence_buffer.strip() and not self._interrupt_event.is_set():
                if not first_audio_sent:
                    await self.transport.send_event({
                        "event": "trackStart",
                        "trackId": self.session_id,
                        "timestamp": self._get_timestamp_ms()
                    })
                    first_audio_sent = True
                await self._speak_sentence(sentence_buffer.strip())
            
            # Send track end
            if first_audio_sent:
                await self.transport.send_event({
                    "event": "trackEnd",
                    "trackId": self.session_id,
                    "timestamp": self._get_timestamp_ms()
                })
            
            # End assistant turn
            await self.conversation.end_assistant_turn(
                was_interrupted=self._interrupt_event.is_set()
            )
            
        except asyncio.CancelledError:
            logger.info("Turn handling cancelled")
            await self.conversation.end_assistant_turn(was_interrupted=True)
        except Exception as e:
            logger.error(f"Turn handling error: {e}", exc_info=True)
            await self.conversation.end_assistant_turn(was_interrupted=True)
        finally:
            self._is_bot_speaking = False
            # Reset barge-in tracking when bot finishes speaking
            self._barge_in_speech_start_time = None
            self._barge_in_speech_frames = 0
            self._barge_in_silence_frames = 0
    
    async def _speak_sentence(self, text: str) -> None:
        """
        Synthesize and send a single sentence.
        
        Args:
            text: Sentence to speak
        """
        if not text.strip() or self._interrupt_event.is_set():
            return
        
        try:
            async for chunk in self.tts_service.synthesize_stream(text):
                # Check interrupt at the start of each iteration
                if self._interrupt_event.is_set():
                    logger.debug("TTS sentence interrupted")
                    break
                
                # Track and log first audio packet latency (TTFB)
                if not self._first_audio_sent and self._turn_start_time:
                    ttfb_ms = (time.time() - self._turn_start_time) * 1000
                    self._first_audio_sent = True
                    logger.info(f"[TTFB] Server first audio packet latency: {ttfb_ms:.0f}ms (session {self.session_id})")
                    
                    # Send TTFB event to client
                    await self.transport.send_event({
                        "event": "ttfb",
                        "trackId": self.session_id,
                        "timestamp": self._get_timestamp_ms(),
                        "latencyMs": round(ttfb_ms)
                    })
                
                # Double-check interrupt right before sending audio
                if self._interrupt_event.is_set():
                    break
                
                await self.transport.send_audio(chunk.audio)
                await asyncio.sleep(0.005)  # Small delay to prevent flooding
        except asyncio.CancelledError:
            logger.debug("TTS sentence cancelled")
        except Exception as e:
            logger.error(f"TTS sentence error: {e}")
    
    async def _speak(self, text: str) -> None:
        """
        Synthesize and send speech.
        
        Args:
            text: Text to speak
        """
        if not text.strip():
            return
        
        try:
            # Start latency tracking for greeting
            speak_start_time = time.time()
            first_audio_sent = False
            
            # Send track start event
            await self.transport.send_event({
                "event": "trackStart",
                "trackId": self.session_id,
                "timestamp": self._get_timestamp_ms()
            })
            
            self._is_bot_speaking = True
            
            # Stream TTS audio
            async for chunk in self.tts_service.synthesize_stream(text):
                if self._interrupt_event.is_set():
                    logger.info("TTS interrupted by barge-in")
                    break
                
                # Track and log first audio packet latency (TTFB)
                if not first_audio_sent:
                    ttfb_ms = (time.time() - speak_start_time) * 1000
                    first_audio_sent = True
                    logger.info(f"[TTFB] Greeting first audio packet latency: {ttfb_ms:.0f}ms (session {self.session_id})")
                    
                    # Send TTFB event to client
                    await self.transport.send_event({
                        "event": "ttfb",
                        "trackId": self.session_id,
                        "timestamp": self._get_timestamp_ms(),
                        "latencyMs": round(ttfb_ms)
                    })
                
                # Send audio to client
                await self.transport.send_audio(chunk.audio)
                
                # Small delay to prevent flooding
                await asyncio.sleep(0.01)
            
            # Send track end event
            await self.transport.send_event({
                "event": "trackEnd",
                "trackId": self.session_id,
                "timestamp": self._get_timestamp_ms()
            })
            
        except asyncio.CancelledError:
            logger.info("TTS cancelled")
            raise
        except Exception as e:
            logger.error(f"TTS error: {e}")
        finally:
            self._is_bot_speaking = False
    
    async def _handle_barge_in(self) -> None:
        """Handle user barge-in (interruption)."""
        if not self._is_bot_speaking:
            return
        
        logger.info("Barge-in detected - interrupting bot speech")
        
        # Reset barge-in tracking
        self._barge_in_speech_start_time = None
        self._barge_in_speech_frames = 0
        self._barge_in_silence_frames = 0
        
        # IMPORTANT: Signal interruption FIRST to stop audio sending
        self._interrupt_event.set()
        self._is_bot_speaking = False
        
        # Send interrupt event to client IMMEDIATELY
        # This must happen BEFORE canceling services, so client knows to discard in-flight audio
        await self.transport.send_event({
            "event": "interrupt",
            "trackId": self.session_id,
            "timestamp": self._get_timestamp_ms()
        })
        
        # Cancel TTS
        if self.tts_service:
            await self.tts_service.cancel()
        
        # Cancel LLM
        if self.llm_service and hasattr(self.llm_service, 'cancel'):
            self.llm_service.cancel()
        
        # Interrupt conversation only if there is no active turn task.
        # When a turn task exists, it will handle end_assistant_turn() to avoid double callbacks.
        if not (self._current_turn_task and not self._current_turn_task.done()):
            await self.conversation.interrupt()
        
        # Reset for new user turn
        await self.conversation.start_user_turn()
        self._audio_buffer = b""
        self.eou_detector.reset()
    
    async def _stop_current_speech(self) -> None:
        """Stop any current speech task."""
        if self._current_turn_task and not self._current_turn_task.done():
            self._interrupt_event.set()
            self._current_turn_task.cancel()
            try:
                await self._current_turn_task
            except asyncio.CancelledError:
                pass

        # Ensure underlying services are cancelled to avoid leaking work/audio
        if self.tts_service:
            await self.tts_service.cancel()
        if self.llm_service and hasattr(self.llm_service, 'cancel'):
            self.llm_service.cancel()
        
        self._is_bot_speaking = False
        self._interrupt_event.clear()
    
    async def cleanup(self) -> None:
        """Cleanup pipeline resources."""
        logger.info(f"Cleaning up DuplexPipeline for session {self.session_id}")
        
        self._running = False
        await self._stop_current_speech()
        
        # Disconnect services
        if self.llm_service:
            await self.llm_service.disconnect()
        if self.tts_service:
            await self.tts_service.disconnect()
        if self.asr_service:
            await self.asr_service.disconnect()
    
    def _get_timestamp_ms(self) -> int:
        """Get current timestamp in milliseconds."""
        import time
        return int(time.time() * 1000)
    
    @property
    def is_speaking(self) -> bool:
        """Check if bot is currently speaking."""
        return self._is_bot_speaking
    
    @property
    def state(self) -> ConversationState:
        """Get current conversation state."""
        return self.conversation.state
