""" WebSocket Handler for Real-time STT/TTS with Barge-in Support """ from fastapi import WebSocket, WebSocketDisconnect from typing import Dict, Any, Optional import json import asyncio import base64 from datetime import datetime from collections import deque from enum import Enum import numpy as np import traceback from session import Session, session_store from config_provider import ConfigProvider from chat_handler import handle_new_message, handle_parameter_followup from stt_factory import STTFactory from tts_factory import TTSFactory from logger import log_info, log_error, log_debug, log_warning # ========================= CONSTANTS ========================= # Default values - will be overridden by config DEFAULT_SILENCE_THRESHOLD_MS = 2000 DEFAULT_AUDIO_CHUNK_SIZE = 4096 DEFAULT_ENERGY_THRESHOLD = 0.01 DEFAULT_AUDIO_BUFFER_MAX_SIZE = 1000 # ========================= ENUMS ========================= class ConversationState(Enum): IDLE = "idle" LISTENING = "listening" PROCESSING_STT = "processing_stt" PROCESSING_LLM = "processing_llm" PROCESSING_TTS = "processing_tts" PLAYING_AUDIO = "playing_audio" # ========================= CLASSES ========================= class AudioBuffer: """Thread-safe circular buffer for audio chunks""" def __init__(self, max_size: int = DEFAULT_AUDIO_BUFFER_MAX_SIZE): self.buffer = deque(maxlen=max_size) self.lock = asyncio.Lock() async def add_chunk(self, chunk_data: str): """Add base64 encoded audio chunk""" async with self.lock: decoded = base64.b64decode(chunk_data) self.buffer.append(decoded) async def get_all_audio(self) -> bytes: """Get all audio data concatenated""" async with self.lock: return b''.join(self.buffer) async def clear(self): """Clear buffer""" async with self.lock: self.buffer.clear() def size(self) -> int: """Get current buffer size""" return len(self.buffer) class SilenceDetector: """Detect silence in audio stream""" def __init__(self, threshold_ms: int = DEFAULT_SILENCE_THRESHOLD_MS, energy_threshold: float = DEFAULT_ENERGY_THRESHOLD): self.threshold_ms = threshold_ms self.energy_threshold = energy_threshold self.silence_start = None self.sample_rate = 16000 def update(self, audio_chunk: bytes) -> int: """Update with new audio chunk and return silence duration in ms""" if self.is_silence(audio_chunk): if self.silence_start is None: self.silence_start = datetime.now() silence_duration = (datetime.now() - self.silence_start).total_seconds() * 1000 return int(silence_duration) else: self.silence_start = None return 0 def is_silence(self, audio_chunk: bytes) -> bool: """Check if audio chunk is silence""" try: # Convert bytes to numpy array (assuming 16-bit PCM) audio_data = np.frombuffer(audio_chunk, dtype=np.int16) # Calculate RMS energy if len(audio_data) == 0: return True rms = np.sqrt(np.mean(audio_data.astype(float) ** 2)) normalized_rms = rms / 32768.0 # Normalize for 16-bit audio return normalized_rms < self.energy_threshold except Exception as e: log_warning(f"Silence detection error: {e}") return False def reset(self): """Reset silence detection""" self.silence_start = None class BargeInHandler: """Handle user interruptions during TTS playback""" def __init__(self): self.active_tts_task: Optional[asyncio.Task] = None self.is_interrupting = False self.lock = asyncio.Lock() async def start_tts_task(self, coro): """Start a cancellable TTS task""" async with self.lock: # Cancel any existing task if self.active_tts_task and not self.active_tts_task.done(): self.active_tts_task.cancel() try: await self.active_tts_task except asyncio.CancelledError: pass # Start new task self.active_tts_task = asyncio.create_task(coro) return self.active_tts_task async def handle_interruption(self, current_state: ConversationState): """Handle barge-in interruption""" async with self.lock: self.is_interrupting = True # Cancel TTS if active if self.active_tts_task and not self.active_tts_task.done(): log_info("Barge-in: Cancelling active TTS") self.active_tts_task.cancel() try: await self.active_tts_task except asyncio.CancelledError: pass # Reset flag after short delay await asyncio.sleep(0.5) self.is_interrupting = False class RealtimeSession: """Manage a real-time conversation session""" def __init__(self, session: Session): self.session = session self.state = ConversationState.IDLE # Get settings from config config = ConfigProvider.get().global_config.stt_provider.settings # Initialize with config values or defaults silence_threshold = config.get("speech_timeout_ms", DEFAULT_SILENCE_THRESHOLD_MS) energy_threshold = config.get("energy_threshold", DEFAULT_ENERGY_THRESHOLD) buffer_max_size = config.get("audio_buffer_max_size", DEFAULT_AUDIO_BUFFER_MAX_SIZE) self.audio_buffer = AudioBuffer(max_size=buffer_max_size) self.silence_detector = SilenceDetector( threshold_ms=silence_threshold, energy_threshold=energy_threshold ) self.barge_in_handler = BargeInHandler() self.stt_manager = None self.current_transcription = "" self.is_streaming = False self.lock = asyncio.Lock() # Store config for later use self.audio_chunk_size = config.get("audio_chunk_size", DEFAULT_AUDIO_CHUNK_SIZE) self.silence_threshold_ms = silence_threshold async def initialize_stt(self): """Initialize STT provider""" try: self.stt_manager = STTFactory.create_provider() if self.stt_manager: config = ConfigProvider.get().global_config.stt_provider.settings await self.stt_manager.start_streaming({ "language": config.get("language", "tr-TR"), "interim_results": config.get("interim_results", True), "single_utterance": False, "enable_punctuation": config.get("enable_punctuation", True), "sample_rate": 16000, "encoding": "WEBM_OPUS" }) log_info("STT manager initialized", session_id=self.session.session_id) return True except Exception as e: log_error(f"Failed to initialize STT", error=str(e), session_id=self.session.session_id) return False async def change_state(self, new_state: ConversationState): """Change conversation state""" async with self.lock: old_state = self.state self.state = new_state log_debug( f"State change: {old_state.value} → {new_state.value}", session_id=self.session.session_id ) async def handle_barge_in(self): """Handle user interruption""" await self.barge_in_handler.handle_interruption(self.state) await self.change_state(ConversationState.LISTENING) async def reset_for_new_utterance(self): """Reset for new user utterance""" await self.audio_buffer.clear() self.silence_detector.reset() self.current_transcription = "" async def cleanup(self): """Clean up resources""" try: if self.stt_manager: await self.stt_manager.stop_streaming() log_info(f"Cleaned up realtime session", session_id=self.session.session_id) except Exception as e: log_warning(f"Cleanup error", error=str(e), session_id=self.session.session_id) # ========================= MAIN HANDLER ========================= async def websocket_endpoint(websocket: WebSocket, session_id: str): """Main WebSocket endpoint for real-time conversation""" await websocket.accept() log_info(f"WebSocket connected", session_id=session_id) # Get session session = session_store.get_session(session_id) if not session: await websocket.send_json({ "type": "error", "message": "Session not found" }) await websocket.close() return # Mark as realtime session session.is_realtime = True session_store.update_session(session) # Initialize conversation realtime_session = RealtimeSession(session) # Initialize STT stt_initialized = await realtime_session.initialize_stt() if not stt_initialized: await websocket.send_json({ "type": "error", "message": "STT initialization failed" }) try: while True: # Receive message message = await websocket.receive_json() message_type = message.get("type") if message_type == "audio_chunk": await handle_audio_chunk(websocket, realtime_session, message) elif message_type == "control": await handle_control_message(websocket, realtime_session, message) elif message_type == "ping": # Keep-alive ping await websocket.send_json({"type": "pong"}) except WebSocketDisconnect: log_info(f"WebSocket disconnected", session_id=session_id) except Exception as e: log_error( f"WebSocket error", error=str(e), traceback=traceback.format_exc(), session_id=session_id ) await websocket.send_json({ "type": "error", "message": str(e) }) finally: await realtime_session.cleanup() # ========================= MESSAGE HANDLERS ========================= async def handle_audio_chunk(websocket: WebSocket, session: RealtimeSession, message: Dict[str, Any]): """Handle incoming audio chunk with barge-in support""" try: audio_data = message.get("data") if not audio_data: return # Check for barge-in during TTS/audio playback if session.state in [ConversationState.PLAYING_AUDIO, ConversationState.PROCESSING_TTS]: await session.handle_barge_in() await websocket.send_json({ "type": "control", "action": "stop_playback" }) log_info(f"Barge-in detected", session_id=session.session.session_id, state=session.state.value) # Change state to listening if idle if session.state == ConversationState.IDLE: await session.change_state(ConversationState.LISTENING) await websocket.send_json({ "type": "state_change", "from": "idle", "to": "listening" }) # Add to buffer - don't lose any audio await session.audio_buffer.add_chunk(audio_data) # Decode for processing decoded_audio = base64.b64decode(audio_data) # Check silence silence_duration = session.silence_detector.update(decoded_audio) # Stream to STT if available if session.stt_manager and session.state == ConversationState.LISTENING: async for result in session.stt_manager.stream_audio(decoded_audio): # Send transcription updates await websocket.send_json({ "type": "transcription", "text": result.text, "is_final": result.is_final, "confidence": result.confidence }) if result.is_final: session.current_transcription = result.text # Process if silence detected and we have transcription if silence_duration > session.silence_threshold_ms and session.current_transcription: log_info( f"User stopped speaking", session_id=session.session.session_id, silence_ms=silence_duration, text=session.current_transcription ) await process_user_input(websocket, session) except Exception as e: log_error( f"Audio chunk handling error", error=str(e), traceback=traceback.format_exc(), session_id=session.session.session_id ) await websocket.send_json({ "type": "error", "message": f"Audio processing error: {str(e)}" }) async def handle_control_message(websocket: WebSocket, session: RealtimeSession, message: Dict[str, Any]): """Handle control messages""" action = message.get("action") config = message.get("config", {}) log_debug(f"Control message", action=action, session_id=session.session.session_id) if action == "start_session": # Session configuration await websocket.send_json({ "type": "session_started", "session_id": session.session.session_id, "config": { "silence_threshold_ms": session.silence_threshold_ms, "audio_chunk_size": session.audio_chunk_size, "supports_barge_in": True } }) elif action == "end_session": # Clean up and close await session.cleanup() await websocket.close() elif action == "interrupt": # Handle explicit interrupt await session.handle_barge_in() await websocket.send_json({ "type": "control", "action": "interrupt_acknowledged" }) elif action == "reset": # Reset conversation state await session.reset_for_new_utterance() await session.change_state(ConversationState.IDLE) await websocket.send_json({ "type": "state_change", "from": session.state.value, "to": "idle" }) elif action == "audio_ended": # Audio playback ended on client if session.state == ConversationState.PLAYING_AUDIO: await session.change_state(ConversationState.IDLE) await websocket.send_json({ "type": "state_change", "from": "playing_audio", "to": "idle" }) # ========================= PROCESSING FUNCTIONS ========================= async def process_user_input(websocket: WebSocket, session: RealtimeSession): """Process complete user input""" try: user_text = session.current_transcription if not user_text: await session.reset_for_new_utterance() await session.change_state(ConversationState.IDLE) return log_info(f"Processing user input", text=user_text, session_id=session.session.session_id) # State: STT Processing await session.change_state(ConversationState.PROCESSING_STT) await websocket.send_json({ "type": "state_change", "from": "listening", "to": "processing_stt" }) # Send final transcription await websocket.send_json({ "type": "transcription", "text": user_text, "is_final": True, "confidence": 0.95 }) # State: LLM Processing await session.change_state(ConversationState.PROCESSING_LLM) await websocket.send_json({ "type": "state_change", "from": "processing_stt", "to": "processing_llm" }) # Add to chat history session.session.add_message("user", user_text) # Get LLM response based on session state if session.session.state == "collect_params": response_text = await handle_parameter_followup(session.session, user_text) else: response_text = await handle_new_message(session.session, user_text) # Add response to history session.session.add_message("assistant", response_text) # Send text response await websocket.send_json({ "type": "assistant_response", "text": response_text }) # Generate TTS if enabled tts_provider = TTSFactory.create_provider() if tts_provider: await session.change_state(ConversationState.PROCESSING_TTS) await websocket.send_json({ "type": "state_change", "from": "processing_llm", "to": "processing_tts" }) # Generate TTS with barge-in support tts_task = session.barge_in_handler.start_tts_task( generate_and_stream_tts(websocket, session, tts_provider, response_text) ) try: await tts_task except asyncio.CancelledError: log_info("TTS cancelled due to barge-in", session_id=session.session.session_id) else: # No TTS, go back to idle await session.change_state(ConversationState.IDLE) await websocket.send_json({ "type": "state_change", "from": "processing_llm", "to": "idle" }) # Reset for next input await session.reset_for_new_utterance() except Exception as e: log_error( f"Error processing user input", error=str(e), traceback=traceback.format_exc(), session_id=session.session.session_id ) await websocket.send_json({ "type": "error", "message": f"Processing error: {str(e)}" }) await session.reset_for_new_utterance() await session.change_state(ConversationState.IDLE) async def generate_and_stream_tts( websocket: WebSocket, session: RealtimeSession, tts_provider, text: str ): """Generate and stream TTS audio with cancellation support""" try: # Generate audio audio_data = await tts_provider.synthesize(text) # Change state to playing await session.change_state(ConversationState.PLAYING_AUDIO) await websocket.send_json({ "type": "state_change", "from": "processing_tts", "to": "playing_audio" }) # Stream audio in chunks chunk_size = session.audio_chunk_size total_chunks = (len(audio_data) + chunk_size - 1) // chunk_size for i in range(0, len(audio_data), chunk_size): # Check for cancellation if asyncio.current_task().cancelled(): break chunk = audio_data[i:i + chunk_size] chunk_index = i // chunk_size await websocket.send_json({ "type": "tts_audio", "data": base64.b64encode(chunk).decode('utf-8'), "chunk_index": chunk_index, "total_chunks": total_chunks, "is_last": chunk_index == total_chunks - 1 }) # Small delay to prevent overwhelming the client await asyncio.sleep(0.01) log_info( f"TTS streaming completed", session_id=session.session.session_id, text_length=len(text), audio_size=len(audio_data) ) except asyncio.CancelledError: log_info("TTS streaming cancelled", session_id=session.session.session_id) raise except Exception as e: log_error( f"TTS generation error", error=str(e), session_id=session.session.session_id ) await websocket.send_json({ "type": "error", "message": f"TTS error: {str(e)}" })