Fix microphone talk eou missing and clean chat log

Reviewed-on: #1

.gitignore

@@ -143,9 +143,6 @@ cython_debug/
 *~
 
 # Project specific
-assets/*.onnx
-*.wav
-*.mp3
-*.pcm
 recordings/
 logs/
+running/

README.md

@@ -5,3 +5,21 @@ Python Active-Call: real-time audio streaming with WebSocket and WebRTC.
 This repo contains a Python 3.11+ codebase for building low-latency voice
 pipelines (capture, stream, and process audio) using WebRTC and WebSockets.
 It is currently in an early, experimental stage.
+
+# Usage
+
+启动
+
+```
+uvicorn app.main:app --reload --host 0.0.0.0 --port 8000
+```
+
+测试
+
+```
+python examples/test_websocket.py
+```
+
+```
+python mic_client.py
+```

app/config.py

@@ -64,8 +64,8 @@ class Settings(BaseSettings):
     
     # Barge-in (interruption) Configuration
     barge_in_min_duration_ms: int = Field(
-        default=50, 
+        default=200, 
-        description="Minimum speech duration (ms) required to trigger barge-in. 50-100ms recommended."
+        description="Minimum speech duration (ms) required to trigger barge-in. Lower=more sensitive."
     )
 
     # Logging
@@ -84,6 +84,10 @@ class Settings(BaseSettings):
         description="ICE servers configuration"
     )
 
+    # WebSocket heartbeat and inactivity
+    inactivity_timeout_sec: int = Field(default=60, description="Close connection after no message from client (seconds)")
+    heartbeat_interval_sec: int = Field(default=50, description="Send heartBeat event to client every N seconds")
+
     @property
     def chunk_size_bytes(self) -> int:
         """Calculate chunk size in bytes based on sample rate and duration."""

app/main.py

@@ -1,11 +1,14 @@
 """FastAPI application with WebSocket and WebRTC endpoints."""
 
-import uuid
+import asyncio
 import json
-from typing import Dict, Any, Optional
+import time
+import uuid
+from pathlib import Path
+from typing import Dict, Any, Optional, List
 from fastapi import FastAPI, WebSocket, WebSocketDisconnect, HTTPException
 from fastapi.middleware.cors import CORSMiddleware
-from fastapi.responses import JSONResponse
+from fastapi.responses import JSONResponse, FileResponse
 from loguru import logger
 
 # Try to import aiortc (optional for WebRTC functionality)
@@ -17,13 +20,52 @@ except ImportError:
     logger.warning("aiortc not available - WebRTC endpoint will be disabled")
 
 from app.config import settings
-from core.transports import SocketTransport, WebRtcTransport
+from core.transports import SocketTransport, WebRtcTransport, BaseTransport
 from core.session import Session
 from processors.tracks import Resampled16kTrack
 from core.events import get_event_bus, reset_event_bus
 
+# Check interval for heartbeat/timeout (seconds)
+_HEARTBEAT_CHECK_INTERVAL_SEC = 5
+
+
+async def heartbeat_and_timeout_task(
+    transport: BaseTransport,
+    session: Session,
+    session_id: str,
+    last_received_at: List[float],
+    last_heartbeat_at: List[float],
+    inactivity_timeout_sec: int,
+    heartbeat_interval_sec: int,
+) -> None:
+    """
+    Background task: send heartBeat every ~heartbeat_interval_sec and close
+    connection if no message from client for inactivity_timeout_sec.
+    """
+    while True:
+        await asyncio.sleep(_HEARTBEAT_CHECK_INTERVAL_SEC)
+        if transport.is_closed:
+            break
+        now = time.monotonic()
+        if now - last_received_at[0] > inactivity_timeout_sec:
+            logger.info(f"Session {session_id}: {inactivity_timeout_sec}s no message, closing")
+            await session.cleanup()
+            break
+        if now - last_heartbeat_at[0] >= heartbeat_interval_sec:
+            try:
+                await transport.send_event({
+                    "event": "heartBeat",
+                    "timestamp": int(time.time() * 1000),
+                })
+                last_heartbeat_at[0] = now
+            except Exception as e:
+                logger.debug(f"Session {session_id}: heartbeat send failed: {e}")
+                break
+
+
 # Initialize FastAPI
 app = FastAPI(title="Python Active-Call", version="0.1.0")
+_WEB_CLIENT_PATH = Path(__file__).resolve().parent.parent / "examples" / "web_client.html"
 
 # Configure CORS
 app.add_middleware(
@@ -40,7 +82,7 @@ active_sessions: Dict[str, Session] = {}
 # Configure logging
 logger.remove()
 logger.add(
-    "../logs/active_call_{time}.log",
+    "./logs/active_call_{time}.log",
     rotation="1 day",
     retention="7 days",
     level=settings.log_level,
@@ -59,6 +101,24 @@ async def health_check():
     return {"status": "healthy", "sessions": len(active_sessions)}
 
 
+@app.get("/")
+async def web_client_root():
+    """Serve the web client."""
+    if not _WEB_CLIENT_PATH.exists():
+        raise HTTPException(status_code=404, detail="Web client not found")
+    return FileResponse(_WEB_CLIENT_PATH)
+
+
+@app.get("/client")
+async def web_client_alias():
+    """Alias for the web client."""
+    if not _WEB_CLIENT_PATH.exists():
+        raise HTTPException(status_code=404, detail="Web client not found")
+    return FileResponse(_WEB_CLIENT_PATH)
+
+
+
+
 @app.get("/iceservers")
 async def get_ice_servers():
     """Get ICE servers configuration for WebRTC."""
@@ -112,10 +172,25 @@ async def websocket_endpoint(websocket: WebSocket):
 
     logger.info(f"WebSocket connection established: {session_id}")
 
+    last_received_at: List[float] = [time.monotonic()]
+    last_heartbeat_at: List[float] = [0.0]
+    hb_task = asyncio.create_task(
+        heartbeat_and_timeout_task(
+            transport,
+            session,
+            session_id,
+            last_received_at,
+            last_heartbeat_at,
+            settings.inactivity_timeout_sec,
+            settings.heartbeat_interval_sec,
+        )
+    )
+
     try:
         # Receive loop
         while True:
             message = await websocket.receive()
+            last_received_at[0] = time.monotonic()
 
             # Handle binary audio data
             if "bytes" in message:
@@ -132,6 +207,11 @@ async def websocket_endpoint(websocket: WebSocket):
         logger.error(f"WebSocket error: {e}", exc_info=True)
 
     finally:
+        hb_task.cancel()
+        try:
+            await hb_task
+        except asyncio.CancelledError:
+            pass
         # Cleanup session
         if session_id in active_sessions:
             await session.cleanup()
@@ -165,6 +245,20 @@ async def webrtc_endpoint(websocket: WebSocket):
 
     logger.info(f"WebRTC connection established: {session_id}")
 
+    last_received_at: List[float] = [time.monotonic()]
+    last_heartbeat_at: List[float] = [0.0]
+    hb_task = asyncio.create_task(
+        heartbeat_and_timeout_task(
+            transport,
+            session,
+            session_id,
+            last_received_at,
+            last_heartbeat_at,
+            settings.inactivity_timeout_sec,
+            settings.heartbeat_interval_sec,
+        )
+    )
+
     # Track handler for incoming audio
     @pc.on("track")
     def on_track(track):
@@ -202,6 +296,7 @@ async def webrtc_endpoint(websocket: WebSocket):
             if "text" not in message:
                 continue
 
+            last_received_at[0] = time.monotonic()
             data = json.loads(message["text"])
 
             # Handle SDP offer/answer
@@ -238,6 +333,11 @@ async def webrtc_endpoint(websocket: WebSocket):
         logger.error(f"WebRTC error: {e}", exc_info=True)
 
     finally:
+        hb_task.cancel()
+        try:
+            await hb_task
+        except asyncio.CancelledError:
+            pass
         # Cleanup
         await pc.close()
         if session_id in active_sessions:

core/__init__.py

@@ -2,7 +2,6 @@
 
 from core.events import EventBus, get_event_bus
 from core.transports import BaseTransport, SocketTransport, WebRtcTransport
-from core.pipeline import AudioPipeline
 from core.session import Session
 from core.conversation import ConversationManager, ConversationState, ConversationTurn
 from core.duplex_pipeline import DuplexPipeline
@@ -13,7 +12,6 @@ __all__ = [
     "BaseTransport",
     "SocketTransport",
     "WebRtcTransport",
-    "AudioPipeline",
     "Session",
     "ConversationManager",
     "ConversationState",

core/duplex_pipeline.py

@@ -85,8 +85,8 @@ class DuplexPipeline:
         
         # Initialize EOU detector
         self.eou_detector = EouDetector(
-            silence_threshold_ms=600,
+            silence_threshold_ms=settings.vad_eou_threshold_ms,
-            min_speech_duration_ms=200
+            min_speech_duration_ms=settings.vad_min_speech_duration_ms
         )
         
         # Initialize services
@@ -108,11 +108,18 @@ class DuplexPipeline:
         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
@@ -202,6 +209,7 @@ class DuplexPipeline:
             return
         
         try:
+            async with self._process_lock:
                 # 1. Process through VAD
                 vad_result = self.vad_processor.process(pcm_bytes, settings.chunk_size_ms)
             
@@ -259,6 +267,9 @@ class DuplexPipeline:
                 # 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
@@ -364,7 +375,8 @@ class DuplexPipeline:
             # Reset for next utterance
             self._audio_buffer = b""
             self._last_sent_transcript = ""
-            await self.conversation.start_user_turn()
+            # 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]}...")
@@ -383,6 +395,8 @@ class DuplexPipeline:
         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))
     
@@ -396,6 +410,10 @@ class DuplexPipeline:
             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 = ""
@@ -406,7 +424,7 @@ class DuplexPipeline:
             
             # Sentence buffer for streaming TTS
             sentence_buffer = ""
-            sentence_ends = {'.', '!', '?', '。', '！', '？', '；', '\n'}
+            sentence_ends = {'，', '。', '！', '？', '\n'}
             first_audio_sent = False
             
             # Stream LLM response and TTS sentence by sentence
@@ -418,6 +436,15 @@ class DuplexPipeline:
                 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
@@ -446,6 +473,16 @@ class DuplexPipeline:
                     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:
@@ -495,10 +532,33 @@ class DuplexPipeline:
         
         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}")
     
@@ -513,6 +573,10 @@ class DuplexPipeline:
             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",
@@ -528,6 +592,20 @@ class DuplexPipeline:
                     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)
                 
@@ -561,8 +639,17 @@ class DuplexPipeline:
         self._barge_in_speech_frames = 0
         self._barge_in_silence_frames = 0
         
-        # Signal interruption
+        # 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:
@@ -572,18 +659,12 @@ class DuplexPipeline:
         if self.llm_service and hasattr(self.llm_service, 'cancel'):
             self.llm_service.cancel()
         
-        # Interrupt conversation
+        # 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()
         
-        # Send interrupt event to client
-        await self.transport.send_event({
-            "event": "interrupt",
-            "trackId": self.session_id,
-            "timestamp": self._get_timestamp_ms()
-        })
-        
         # Reset for new user turn
-        self._is_bot_speaking = False
         await self.conversation.start_user_turn()
         self._audio_buffer = b""
         self.eou_detector.reset()
@@ -598,6 +679,12 @@ class DuplexPipeline:
             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()
     

core/pipeline.py

@@ -1,131 +0,0 @@
-"""Audio processing pipeline."""
-
-import asyncio
-from typing import Optional
-from loguru import logger
-
-from core.transports import BaseTransport
-from core.events import EventBus, get_event_bus
-from processors.vad import VADProcessor, SileroVAD
-from app.config import settings
-
-
-class AudioPipeline:
-    """
-    Audio processing pipeline.
-
-    Processes incoming audio through VAD and emits events.
-    """
-
-    def __init__(self, transport: BaseTransport, session_id: str):
-        """
-        Initialize audio pipeline.
-
-        Args:
-            transport: Transport instance for sending events/audio
-            session_id: Session identifier for event tracking
-        """
-        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,
-            silence_threshold_ms=settings.vad_eou_threshold_ms,
-            min_speech_duration_ms=settings.vad_min_speech_duration_ms
-        )
-
-        # State
-        self.is_bot_speaking = False
-        self.interrupt_signal = asyncio.Event()
-        self._running = True
-
-        logger.info(f"Audio pipeline initialized for session {session_id}")
-
-    async def process_input(self, pcm_bytes: bytes) -> None:
-        """
-        Process incoming audio chunk.
-
-        Args:
-            pcm_bytes: PCM audio data (16-bit, mono, 16kHz)
-        """
-        if not self._running:
-            return
-
-        try:
-            # Process through VAD
-            result = self.vad_processor.process(pcm_bytes, settings.chunk_size_ms)
-
-            if result:
-                event_type, probability = result
-
-                # Emit event through event bus
-                await self.event_bus.publish(event_type, {
-                    "trackId": self.session_id,
-                    "probability": probability
-                })
-
-                # Send event to client
-                if event_type == "speaking":
-                    logger.info(f"User speaking started (session {self.session_id})")
-                    await self.transport.send_event({
-                        "event": "speaking",
-                        "trackId": self.session_id,
-                        "timestamp": self._get_timestamp_ms(),
-                        "startTime": self._get_timestamp_ms()
-                    })
-
-                elif event_type == "silence":
-                    logger.info(f"User speaking stopped (session {self.session_id})")
-                    await self.transport.send_event({
-                        "event": "silence",
-                        "trackId": self.session_id,
-                        "timestamp": self._get_timestamp_ms(),
-                        "startTime": self._get_timestamp_ms(),
-                        "duration": 0  # TODO: Calculate actual duration
-                    })
-
-                elif event_type == "eou":
-                    logger.info(f"EOU detected (session {self.session_id})")
-                    await self.transport.send_event({
-                        "event": "eou",
-                        "trackId": self.session_id,
-                        "timestamp": self._get_timestamp_ms()
-                    })
-
-        except Exception as e:
-            logger.error(f"Pipeline processing error: {e}", exc_info=True)
-
-    async def process_text_input(self, text: str) -> None:
-        """
-        Process text input (chat command).
-
-        Args:
-            text: Text input
-        """
-        logger.info(f"Processing text input: {text[:50]}...")
-        # TODO: Implement text processing (LLM integration, etc.)
-        # For now, just log it
-
-    async def interrupt(self) -> None:
-        """Interrupt current audio playback."""
-        if self.is_bot_speaking:
-            self.interrupt_signal.set()
-            logger.info(f"Pipeline interrupted for session {self.session_id}")
-
-    async def cleanup(self) -> None:
-        """Cleanup pipeline resources."""
-        logger.info(f"Cleaning up pipeline for session {self.session_id}")
-        self._running = False
-        self.interrupt_signal.set()
-
-    def _get_timestamp_ms(self) -> int:
-        """Get current timestamp in milliseconds."""
-        import time
-        return int(time.time() * 1000)

core/session.py

@@ -6,7 +6,7 @@ from typing import Optional, Dict, Any
 from loguru import logger
 
 from core.transports import BaseTransport
-from core.pipeline import AudioPipeline
+from core.duplex_pipeline import DuplexPipeline
 from models.commands import parse_command, TTSCommand, ChatCommand, InterruptCommand, HangupCommand
 from app.config import settings
 
@@ -16,7 +16,7 @@ class Session:
     Manages a single call session.
 
     Handles command routing, audio processing, and session lifecycle.
-    Supports both basic audio pipeline and full duplex voice conversation.
+    Uses full duplex voice conversation pipeline.
     """
 
     def __init__(self, session_id: str, transport: BaseTransport, use_duplex: bool = None):
@@ -30,20 +30,14 @@ class Session:
         """
         self.id = session_id
         self.transport = transport
-        
-        # Determine pipeline mode
         self.use_duplex = use_duplex if use_duplex is not None else settings.duplex_enabled
 
-        if self.use_duplex:
-            from core.duplex_pipeline import DuplexPipeline
         self.pipeline = DuplexPipeline(
             transport=transport,
             session_id=session_id,
             system_prompt=settings.duplex_system_prompt,
             greeting=settings.duplex_greeting
         )
-        else:
-            self.pipeline = AudioPipeline(transport, session_id)
 
         # Session state
         self.created_at = None
@@ -129,10 +123,7 @@ class Session:
             audio_bytes: PCM audio data
         """
         try:
-            if self.use_duplex:
             await self.pipeline.process_audio(audio_bytes)
-            else:
-                await self.pipeline.process_input(audio_bytes)
         except Exception as e:
             logger.error(f"Session {self.id} handle_audio error: {e}", exc_info=True)
 
@@ -148,8 +139,8 @@ class Session:
             "timestamp": self._get_timestamp_ms()
         })
 
-        # Start duplex pipeline if enabled
+        # Start duplex pipeline
-        if self.use_duplex and not self._pipeline_started:
+        if not self._pipeline_started:
             try:
                 await self.pipeline.start()
                 self._pipeline_started = True
@@ -228,9 +219,6 @@ class Session:
             logger.info(f"Session {self.id} graceful interrupt")
         else:
             logger.info(f"Session {self.id} immediate interrupt")
-            if self.use_duplex:
-                await self.pipeline.interrupt()
-            else:
             await self.pipeline.interrupt()
 
     async def _handle_pause(self) -> None:
@@ -267,11 +255,7 @@ class Session:
     async def _handle_chat(self, command: ChatCommand) -> None:
         """Handle chat command."""
         logger.info(f"Session {self.id} chat: {command.text[:50]}...")
-        # Process text input through pipeline
-        if self.use_duplex:
         await self.pipeline.process_text(command.text)
-        else:
-            await self.pipeline.process_text_input(command.text)
 
     async def _send_error(self, sender: str, error_message: str) -> None:
         """

docs/duplex_interaction.svg

@@ -0,0 +1,96 @@
+<svg width="1200" height="620" viewBox="0 0 1200 620" xmlns="http://www.w3.org/2000/svg">
+  <defs>
+    <style>
+      .box { fill:#11131a; stroke:#3a3f4b; stroke-width:1.2; rx:10; ry:10; }
+      .title { font: 600 14px 'Arial'; fill:#f2f3f7; }
+      .text { font: 12px 'Arial'; fill:#c8ccd8; }
+      .arrow { stroke:#7aa2ff; stroke-width:1.6; marker-end:url(#arrow); fill:none; }
+      .arrow2 { stroke:#2dd4bf; stroke-width:1.6; marker-end:url(#arrow); fill:none; }
+      .arrow3 { stroke:#ff6b6b; stroke-width:1.6; marker-end:url(#arrow); fill:none; }
+      .label { font: 11px 'Arial'; fill:#9aa3b2; }
+    </style>
+    <marker id="arrow" markerWidth="8" markerHeight="8" refX="7" refY="4" orient="auto">
+      <path d="M0,0 L8,4 L0,8 Z" fill="#7aa2ff"/>
+    </marker>
+  </defs>
+
+  <rect x="40" y="40" width="250" height="120" class="box"/>
+  <text x="60" y="70" class="title">Web Client</text>
+  <text x="60" y="95" class="text">WS JSON commands</text>
+  <text x="60" y="115" class="text">WS binary PCM audio</text>
+
+  <rect x="350" y="40" width="250" height="120" class="box"/>
+  <text x="370" y="70" class="title">FastAPI /ws</text>
+  <text x="370" y="95" class="text">Session + Transport</text>
+
+  <rect x="660" y="40" width="250" height="120" class="box"/>
+  <text x="680" y="70" class="title">DuplexPipeline</text>
+  <text x="680" y="95" class="text">process_audio / process_text</text>
+
+  <rect x="920" y="40" width="240" height="120" class="box"/>
+  <text x="940" y="70" class="title">ConversationManager</text>
+  <text x="940" y="95" class="text">turns + state</text>
+
+  <rect x="660" y="200" width="180" height="100" class="box"/>
+  <text x="680" y="230" class="title">VADProcessor</text>
+  <text x="680" y="255" class="text">speech/silence</text>
+
+  <rect x="860" y="200" width="180" height="100" class="box"/>
+  <text x="880" y="230" class="title">EOU Detector</text>
+  <text x="880" y="255" class="text">end-of-utterance</text>
+
+  <rect x="1060" y="200" width="120" height="100" class="box"/>
+  <text x="1075" y="230" class="title">ASR</text>
+  <text x="1075" y="255" class="text">transcripts</text>
+
+  <rect x="920" y="350" width="240" height="110" class="box"/>
+  <text x="940" y="380" class="title">LLM (stream)</text>
+  <text x="940" y="405" class="text">llmResponse events</text>
+
+  <rect x="660" y="350" width="220" height="110" class="box"/>
+  <text x="680" y="380" class="title">TTS (stream)</text>
+  <text x="680" y="405" class="text">PCM audio</text>
+
+  <rect x="40" y="350" width="250" height="110" class="box"/>
+  <text x="60" y="380" class="title">Web Client</text>
+  <text x="60" y="405" class="text">audio playback + UI</text>
+
+  <path d="M290 80 L350 80" class="arrow"/>
+  <text x="300" y="70" class="label">JSON / PCM</text>
+
+  <path d="M600 80 L660 80" class="arrow"/>
+  <text x="615" y="70" class="label">dispatch</text>
+
+  <path d="M910 80 L920 80" class="arrow"/>
+  <text x="880" y="70" class="label">turn mgmt</text>
+
+  <path d="M750 160 L750 200" class="arrow"/>
+  <text x="705" y="190" class="label">audio chunks</text>
+
+  <path d="M840 250 L860 250" class="arrow"/>
+  <text x="835" y="240" class="label">vad status</text>
+
+  <path d="M1040 250 L1060 250" class="arrow"/>
+  <text x="1010" y="240" class="label">audio buffer</text>
+
+  <path d="M950 300 L950 350" class="arrow2"/>
+  <text x="930" y="340" class="label">EOU -> LLM</text>
+
+  <path d="M880 405 L920 405" class="arrow2"/>
+  <text x="870" y="395" class="label">text stream</text>
+
+  <path d="M660 405 L290 405" class="arrow2"/>
+  <text x="430" y="395" class="label">PCM audio</text>
+
+  <path d="M660 450 L350 450" class="arrow"/>
+  <text x="420" y="440" class="label">events: trackStart/End</text>
+
+  <path d="M350 450 L290 450" class="arrow"/>
+  <text x="315" y="440" class="label">UI updates</text>
+
+  <path d="M750 200 L750 160" class="arrow3"/>
+  <text x="700" y="145" class="label">barge-in detection</text>
+
+  <path d="M760 170 L920 170" class="arrow3"/>
+  <text x="820" y="160" class="label">interrupt event + cancel</text>
+</svg>

docs/proejct_todo.md

@@ -0,0 +1,187 @@
+# OmniSense: 12-Week Sprint Board + Tech Stack (Python Backend) — TODO
+
+## Scope
+- [ ] Build a realtime AI SaaS (OmniSense) focused on web-first audio + video with WebSocket + WebRTC endpoints
+- [ ] Deliver assistant builder, tool execution, observability, evals, optional telephony later
+- [ ] Keep scope aligned to 2-person team, self-hosted services
+
+---
+
+## Sprint Board (12 weeks, 2-week sprints)
+Team assumption: 2 engineers. Scope prioritized to web-first audio + video, with BYO-SFU adapters.
+
+### Sprint 1 (Weeks 1–2) — Realtime Core MVP (WebSocket + WebRTC Audio)
+- Deliverables
+  - [ ] WebSocket transport: audio in/out streaming (1:1)
+  - [ ] WebRTC transport: audio in/out streaming (1:1)
+  - [ ] Adapter contract wired into runtime (transport-agnostic session core)
+  - [ ] ASR → LLM → TTS pipeline, streaming both directions
+  - [ ] Basic session state (start/stop, silence timeout)
+  - [ ] Transcript persistence
+- Acceptance criteria
+  - [ ] < 1.5s median round-trip for short responses
+  - [ ] Stable streaming for 10+ minute session
+
+### Sprint 2 (Weeks 3–4) — Video + Realtime UX
+- Deliverables
+  - [ ] WebRTC video capture + streaming (assistant can “see” frames)
+  - [ ] WebSocket video streaming for local/dev mode
+  - [ ] Low-latency UI: push-to-talk, live captions, speaking indicator
+  - [ ] Recording + transcript storage (web sessions)
+- Acceptance criteria
+  - [ ] Video < 2.5s end-to-end latency for analysis
+  - [ ] Audio quality acceptable (no clipping, jitter handling)
+
+### Sprint 3 (Weeks 5–6) — Assistant Builder v1
+- Deliverables
+  - [ ] Assistant schema + versioning
+  - [ ] UI: Model/Voice/Transcriber/Tools/Video/Transport tabs
+  - [ ] “Test/Chat/Talk to Assistant” (web)
+- Acceptance criteria
+  - [ ] Create/publish assistant and run a live web session
+  - [ ] All config changes tracked by version
+
+### Sprint 4 (Weeks 7–8) — Tooling + Structured Outputs
+- Deliverables
+  - [ ] Tool registry + custom HTTP tools
+  - [ ] Tool auth secrets management
+  - [ ] Structured outputs (JSON extraction)
+- Acceptance criteria
+  - [ ] Tool calls executed with retries/timeouts
+  - [ ] Structured JSON stored per call/session
+
+### Sprint 5 (Weeks 9–10) — Observability + QA + Dev Platform
+- Deliverables
+  - [ ] Session logs + chat logs + media logs
+  - [ ] Evals engine + test suites
+  - [ ] Basic analytics dashboard
+  - [ ] Public WebSocket API spec + message schema
+  - [ ] JS/TS SDK (connect, send audio/video, receive transcripts)
+- Acceptance criteria
+  - [ ] Reproducible test suite runs
+  - [ ] Log filters by assistant/time/status
+  - [ ] SDK demo app runs end-to-end
+
+### Sprint 6 (Weeks 11–12) — SaaS Hardening
+- Deliverables
+  - [ ] Org/RBAC + API keys + rate limits
+  - [ ] Usage metering + credits
+  - [ ] Stripe billing integration
+  - [ ] Self-hosted DB ops (migrations, backup/restore, monitoring)
+- Acceptance criteria
+  - [ ] Metered usage per org
+  - [ ] Credits decrement correctly
+  - [ ] Optional telephony spike documented (defer build)
+  - [ ] Enterprise adapter guide published (BYO-SFU)
+
+---
+
+## Tech Stack by Service (Self-Hosted, Web-First)
+
+### 1) Transport Gateway (Realtime)
+- [ ] WebRTC (browser) + WebSocket (lightweight/dev) protocols
+- [ ] BYO-SFU adapter (enterprise) + LiveKit optional adapter + WS transport server
+- [ ] Python core (FastAPI + asyncio) + Node.js mediasoup adapters when needed
+- [ ] Media: Opus/VP8, jitter buffer, VAD, echo cancellation
+- [ ] Storage: S3-compatible (MinIO) for recordings
+
+### 2) ASR Service
+- [ ] Whisper (self-hosted) baseline
+- [ ] gRPC/WebSocket streaming transport
+- [ ] Python native service
+- [ ] Optional cloud provider fallback (later)
+
+### 3) TTS Service
+- [ ] Piper or Coqui TTS (self-hosted)
+- [ ] gRPC/WebSocket streaming transport
+- [ ] Python native service
+- [ ] Redis cache for common phrases
+
+### 4) LLM Orchestrator
+- [ ] Self-hosted (vLLM + open model)
+- [ ] Python (FastAPI + asyncio)
+- [ ] Streaming, tool calling, JSON mode
+- [ ] Safety filters + prompt templates
+
+### 5) Assistant Config Service
+- [ ] PostgreSQL
+- [ ] Python (SQLAlchemy or SQLModel)
+- [ ] Versioning, publish/rollback
+
+### 6) Session Service
+- [ ] PostgreSQL + Redis
+- [ ] Python
+- [ ] State machine, timeouts, events
+
+### 7) Tool Execution Layer
+- [ ] PostgreSQL
+- [ ] Python
+- [ ] Auth secret vault, retry policies, tool schemas
+
+### 8) Observability + Logs
+- [ ] Postgres (metadata), ClickHouse (logs/metrics)
+- [ ] OpenSearch for search
+- [ ] Prometheus + Grafana metrics
+- [ ] OpenTelemetry tracing
+
+### 9) Billing + Usage Metering
+- [ ] Stripe billing
+- [ ] PostgreSQL
+- [ ] NATS JetStream (events) + Redis counters
+
+### 10) Web App (Dashboard)
+- [ ] React + Next.js
+- [ ] Tailwind or Radix UI
+- [ ] WebRTC client + WS client; adapter-based RTC integration
+- [ ] ECharts/Recharts
+
+### 11) Auth + RBAC
+- [ ] Keycloak (self-hosted) or custom JWT
+- [ ] Org/user/role tables in Postgres
+
+### 12) Public WebSocket API + SDK
+- [ ] WS API: versioned schema, binary audio frames + JSON control messages
+- [ ] SDKs: JS/TS first, optional Python/Go clients
+- [ ] Docs: quickstart, auth flow, session lifecycle, examples
+
+---
+
+## Infrastructure (Self-Hosted)
+- [ ] Docker Compose → k3s (later)
+- [ ] Redis Streams or NATS
+- [ ] MinIO object store
+- [ ] GitHub Actions + Helm or kustomize
+- [ ] Self-hosted Postgres + pgbackrest backups
+- [ ] Vault for secrets
+
+---
+
+## Suggested MVP Sequence
+- [ ] WebRTC demo + ASR/LLM/TTS streaming
+- [ ] Assistant schema + versioning (web-first)
+- [ ] Video capture + multimodal analysis
+- [ ] Tool execution + structured outputs
+- [ ] Logs + evals + public WS API + SDK
+- [ ] Telephony (optional, later)
+
+---
+
+## Public WebSocket API (Minimum Spec)
+- [ ] Auth: API key or JWT in initial `hello` message
+- [ ] Core messages: `session.start`, `session.stop`, `audio.append`, `audio.commit`, `video.append`, `transcript.delta`, `assistant.response`, `tool.call`, `tool.result`, `error`
+- [ ] Binary payloads: PCM/Opus frames with metadata in control channel
+- [ ] Versioning: `v1` schema with backward compatibility rules
+
+---
+
+## Self-Hosted DB Ops Checklist
+- [ ] Postgres in Docker/k3s with persistent volumes
+- [ ] Migrations: `alembic` or `atlas`
+- [ ] Backups: `pgbackrest` nightly + on-demand
+- [ ] Monitoring: postgres_exporter + alerts
+
+---
+
+## RTC Adapter Contract (BYO-SFU First)
+- [ ] Keep RTC pluggable; LiveKit optional, not core dependency
+- [ ] Define adapter interface (TypeScript sketch)

examples/mic_client.py

@@ -4,10 +4,12 @@ Microphone client for testing duplex voice conversation.
 
 This client captures audio from the microphone, sends it to the server,
 and plays back the AI's voice response through the speakers.
+It also displays the LLM's text responses in the console.
 
 Usage:
     python examples/mic_client.py --url ws://localhost:8000/ws
     python examples/mic_client.py --url ws://localhost:8000/ws --chat "Hello!"
+    python examples/mic_client.py --url ws://localhost:8000/ws --verbose
 
 Requirements:
     pip install sounddevice soundfile websockets numpy
@@ -17,6 +19,7 @@ import argparse
 import asyncio
 import json
 import sys
+import time
 import threading
 import queue
 from pathlib import Path
@@ -93,6 +96,17 @@ class MicrophoneClient:
         self.is_recording = True
         self.is_playing = True
         
+        # TTFB tracking (Time to First Byte)
+        self.request_start_time = None
+        self.first_audio_received = False
+        
+        # Interrupt handling - discard audio until next trackStart
+        self._discard_audio = False
+        self._audio_sequence = 0  # Track audio sequence to detect stale chunks
+        
+        # Verbose mode for streaming LLM responses
+        self.verbose = False
+    
     async def connect(self) -> None:
         """Connect to WebSocket server."""
         print(f"Connecting to {self.url}...")
@@ -117,6 +131,10 @@ class MicrophoneClient:
     
     async def send_chat(self, text: str) -> None:
         """Send chat message (text input)."""
+        # Reset TTFB tracking for new request
+        self.request_start_time = time.time()
+        self.first_audio_received = False
+        
         await self.send_command({
             "command": "chat",
             "text": text
@@ -236,9 +254,21 @@ class MicrophoneClient:
                         # Audio data received
                         self.bytes_received += len(message)
                         
+                        # Check if we should discard this audio (after interrupt)
+                        if self._discard_audio:
+                            duration_ms = len(message) / (self.sample_rate * 2) * 1000
+                            print(f"← Audio: {duration_ms:.0f}ms (DISCARDED - waiting for new track)")
+                            continue
+                        
                         if self.is_playing:
                             self._add_audio_to_buffer(message)
                         
+                        # Calculate and display TTFB for first audio packet
+                        if not self.first_audio_received and self.request_start_time:
+                            client_ttfb_ms = (time.time() - self.request_start_time) * 1000
+                            self.first_audio_received = True
+                            print(f"← [TTFB] Client first audio latency: {client_ttfb_ms:.0f}ms")
+                        
                         # Show progress (less verbose)
                         with self.audio_output_lock:
                             buffer_ms = len(self.audio_output_buffer) / (self.sample_rate * 2) * 1000
@@ -285,20 +315,47 @@ class MicrophoneClient:
                 # Interim result - show with indicator (overwrite same line)
                 display_text = text[:60] + "..." if len(text) > 60 else text
                 print(f"  [listening] {display_text}".ljust(80), end="\r")
+        elif event_type == "ttfb":
+            # Server-side TTFB event
+            latency_ms = event.get("latencyMs", 0)
+            print(f"← [TTFB] Server reported latency: {latency_ms}ms")
+        elif event_type == "llmResponse":
+            # LLM text response
+            text = event.get("text", "")
+            is_final = event.get("isFinal", False)
+            if is_final:
+                # Print final LLM response
+                print(f"← AI: {text}")
+            elif self.verbose:
+                # Show streaming chunks only in verbose mode
+                display_text = text[:60] + "..." if len(text) > 60 else text
+                print(f"  [streaming] {display_text}")
         elif event_type == "trackStart":
             print("← Bot started speaking")
+            # IMPORTANT: Accept audio again after trackStart
+            self._discard_audio = False
+            self._audio_sequence += 1
+            # Reset TTFB tracking for voice responses (when no chat was sent)
+            if self.request_start_time is None:
+                self.request_start_time = time.time()
+                self.first_audio_received = False
             # Clear any old audio in buffer
             with self.audio_output_lock:
                 self.audio_output_buffer = b""
         elif event_type == "trackEnd":
             print("← Bot finished speaking")
+            # Reset TTFB tracking after response completes
+            self.request_start_time = None
+            self.first_audio_received = False
         elif event_type == "interrupt":
             print("← Bot interrupted!")
-            # IMPORTANT: Clear audio buffer immediately on interrupt
+            # IMPORTANT: Discard all audio until next trackStart
+            self._discard_audio = True
+            # Clear audio buffer immediately
             with self.audio_output_lock:
                 buffer_ms = len(self.audio_output_buffer) / (self.sample_rate * 2) * 1000
                 self.audio_output_buffer = b""
-                print(f"   (cleared {buffer_ms:.0f}ms of buffered audio)")
+                print(f"   (cleared {buffer_ms:.0f}ms, discarding audio until new track)")
         elif event_type == "error":
             print(f"← Error: {event.get('error')}")
         elif event_type == "hangup":
@@ -511,6 +568,11 @@ async def main():
         action="store_true",
         help="Disable interactive mode"
     )
+    parser.add_argument(
+        "--verbose", "-v",
+        action="store_true",
+        help="Show streaming LLM response chunks"
+    )
     
     args = parser.parse_args()
     
@@ -524,6 +586,7 @@ async def main():
         input_device=args.input_device,
         output_device=args.output_device
     )
+    client.verbose = args.verbose
     
     await client.run(
         chat_message=args.chat,

examples/simple_client.py

@@ -12,6 +12,7 @@ import argparse
 import asyncio
 import json
 import sys
+import time
 import wave
 import io
 
@@ -68,6 +69,13 @@ class SimpleVoiceClient:
         # Stats
         self.bytes_received = 0
         
+        # TTFB tracking (Time to First Byte)
+        self.request_start_time = None
+        self.first_audio_received = False
+        
+        # Interrupt handling - discard audio until next trackStart
+        self._discard_audio = False
+    
     async def connect(self):
         """Connect to server."""
         print(f"Connecting to {self.url}...")
@@ -84,6 +92,10 @@ class SimpleVoiceClient:
     
     async def send_chat(self, text: str):
         """Send chat message."""
+        # Reset TTFB tracking for new request
+        self.request_start_time = time.time()
+        self.first_audio_received = False
+        
         await self.ws.send(json.dumps({"command": "chat", "text": text}))
         print(f"-> chat: {text}")
     
@@ -120,6 +132,18 @@ class SimpleVoiceClient:
                     # Audio data
                     self.bytes_received += len(msg)
                     duration_ms = len(msg) / (self.sample_rate * 2) * 1000
+                    
+                    # Check if we should discard this audio (after interrupt)
+                    if self._discard_audio:
+                        print(f"<- audio: {len(msg)} bytes ({duration_ms:.0f}ms) [DISCARDED]")
+                        continue
+                    
+                    # Calculate and display TTFB for first audio packet
+                    if not self.first_audio_received and self.request_start_time:
+                        client_ttfb_ms = (time.time() - self.request_start_time) * 1000
+                        self.first_audio_received = True
+                        print(f"<- [TTFB] Client first audio latency: {client_ttfb_ms:.0f}ms")
+                    
                     print(f"<- audio: {len(msg)} bytes ({duration_ms:.0f}ms)")
                     
                     # Play immediately in executor to not block
@@ -138,6 +162,18 @@ class SimpleVoiceClient:
                             print(f"<- You said: {text}")
                         else:
                             print(f"<- [listening] {text}", end="\r")
+                    elif etype == "ttfb":
+                        # Server-side TTFB event
+                        latency_ms = event.get("latencyMs", 0)
+                        print(f"<- [TTFB] Server reported latency: {latency_ms}ms")
+                    elif etype == "trackStart":
+                        # New track starting - accept audio again
+                        self._discard_audio = False
+                        print(f"<- {etype}")
+                    elif etype == "interrupt":
+                        # Interrupt - discard audio until next trackStart
+                        self._discard_audio = True
+                        print(f"<- {etype} (discarding audio until new track)")
                     elif etype == "hangup":
                         print(f"<- {etype}")
                         self.running = False

examples/wav_client.py

@@ -0,0 +1,504 @@
+#!/usr/bin/env python3
+"""
+WAV file client for testing duplex voice conversation.
+
+This client reads audio from a WAV file, sends it to the server,
+and saves the AI's voice response to an output WAV file.
+
+Usage:
+    python examples/wav_client.py --input input.wav --output response.wav
+    python examples/wav_client.py --input input.wav --output response.wav --url ws://localhost:8000/ws
+    python examples/wav_client.py --input input.wav --output response.wav --wait-time 10
+    python wav_client.py --input ../data/audio_examples/two_utterances.wav -o response.wav
+Requirements:
+    pip install soundfile websockets numpy
+"""
+
+import argparse
+import asyncio
+import json
+import sys
+import time
+import wave
+from pathlib import Path
+
+try:
+    import numpy as np
+except ImportError:
+    print("Please install numpy: pip install numpy")
+    sys.exit(1)
+
+try:
+    import soundfile as sf
+except ImportError:
+    print("Please install soundfile: pip install soundfile")
+    sys.exit(1)
+
+try:
+    import websockets
+except ImportError:
+    print("Please install websockets: pip install websockets")
+    sys.exit(1)
+
+
+class WavFileClient:
+    """
+    WAV file client for voice conversation testing.
+    
+    Features:
+    - Read audio from WAV file
+    - Send audio to WebSocket server
+    - Receive and save response audio
+    - Event logging
+    """
+    
+    def __init__(
+        self,
+        url: str,
+        input_file: str,
+        output_file: str,
+        sample_rate: int = 16000,
+        chunk_duration_ms: int = 20,
+        wait_time: float = 15.0,
+        verbose: bool = False
+    ):
+        """
+        Initialize WAV file client.
+        
+        Args:
+            url: WebSocket server URL
+            input_file: Input WAV file path
+            output_file: Output WAV file path
+            sample_rate: Audio sample rate (Hz)
+            chunk_duration_ms: Audio chunk duration (ms) for sending
+            wait_time: Time to wait for response after sending (seconds)
+            verbose: Enable verbose output
+        """
+        self.url = url
+        self.input_file = Path(input_file)
+        self.output_file = Path(output_file)
+        self.sample_rate = sample_rate
+        self.chunk_duration_ms = chunk_duration_ms
+        self.chunk_samples = int(sample_rate * chunk_duration_ms / 1000)
+        self.wait_time = wait_time
+        self.verbose = verbose
+        
+        # WebSocket connection
+        self.ws = None
+        self.running = False
+        
+        # Audio buffers
+        self.received_audio = bytearray()
+        
+        # Statistics
+        self.bytes_sent = 0
+        self.bytes_received = 0
+        
+        # TTFB tracking (per response)
+        self.send_start_time = None
+        self.response_start_time = None  # set on each trackStart
+        self.waiting_for_first_audio = False
+        self.ttfb_ms = None  # last TTFB for summary
+        self.ttfb_list = []  # TTFB for each response
+        
+        # State tracking
+        self.track_started = False
+        self.track_ended = False
+        self.send_completed = False
+        
+        # Events log
+        self.events_log = []
+    
+    def log_event(self, direction: str, message: str):
+        """Log an event with timestamp."""
+        timestamp = time.time()
+        self.events_log.append({
+            "timestamp": timestamp,
+            "direction": direction,
+            "message": message
+        })
+        # Handle encoding errors on Windows
+        try:
+            print(f"{direction} {message}")
+        except UnicodeEncodeError:
+            # Replace problematic characters for console output
+            safe_message = message.encode('ascii', errors='replace').decode('ascii')
+            print(f"{direction} {safe_message}")
+    
+    async def connect(self) -> None:
+        """Connect to WebSocket server."""
+        self.log_event("→", f"Connecting to {self.url}...")
+        self.ws = await websockets.connect(self.url)
+        self.running = True
+        self.log_event("←", "Connected!")
+        
+        # Send invite command
+        await self.send_command({
+            "command": "invite",
+            "option": {
+                "codec": "pcm",
+                "sampleRate": self.sample_rate
+            }
+        })
+    
+    async def send_command(self, cmd: dict) -> None:
+        """Send JSON command to server."""
+        if self.ws:
+            await self.ws.send(json.dumps(cmd))
+            self.log_event("→", f"Command: {cmd.get('command', 'unknown')}")
+    
+    async def send_hangup(self, reason: str = "Session complete") -> None:
+        """Send hangup command."""
+        await self.send_command({
+            "command": "hangup",
+            "reason": reason
+        })
+    
+    def load_wav_file(self) -> tuple[np.ndarray, int]:
+        """
+        Load and prepare WAV file for sending.
+        
+        Returns:
+            Tuple of (audio_data as int16 numpy array, original sample rate)
+        """
+        if not self.input_file.exists():
+            raise FileNotFoundError(f"Input file not found: {self.input_file}")
+        
+        # Load audio file
+        audio_data, file_sample_rate = sf.read(self.input_file)
+        self.log_event("→", f"Loaded: {self.input_file}")
+        self.log_event("→", f"  Original sample rate: {file_sample_rate} Hz")
+        self.log_event("→", f"  Duration: {len(audio_data) / file_sample_rate:.2f}s")
+        
+        # Convert stereo to mono if needed
+        if len(audio_data.shape) > 1:
+            audio_data = audio_data.mean(axis=1)
+            self.log_event("→", "  Converted stereo to mono")
+        
+        # Resample if needed
+        if file_sample_rate != self.sample_rate:
+            # Simple resampling using numpy
+            duration = len(audio_data) / file_sample_rate
+            num_samples = int(duration * self.sample_rate)
+            indices = np.linspace(0, len(audio_data) - 1, num_samples)
+            audio_data = np.interp(indices, np.arange(len(audio_data)), audio_data)
+            self.log_event("→", f"  Resampled to {self.sample_rate} Hz")
+        
+        # Convert to int16
+        if audio_data.dtype != np.int16:
+            # Normalize to [-1, 1] if needed
+            max_val = np.max(np.abs(audio_data))
+            if max_val > 1.0:
+                audio_data = audio_data / max_val
+            audio_data = (audio_data * 32767).astype(np.int16)
+        
+        self.log_event("→", f"  Prepared: {len(audio_data)} samples ({len(audio_data)/self.sample_rate:.2f}s)")
+        
+        return audio_data, file_sample_rate
+    
+    async def audio_sender(self, audio_data: np.ndarray) -> None:
+        """Send audio data to server in chunks."""
+        total_samples = len(audio_data)
+        chunk_size = self.chunk_samples
+        sent_samples = 0
+        
+        self.send_start_time = time.time()
+        self.log_event("→", f"Starting audio transmission ({total_samples} samples)...")
+        
+        while sent_samples < total_samples and self.running:
+            # Get next chunk
+            end_sample = min(sent_samples + chunk_size, total_samples)
+            chunk = audio_data[sent_samples:end_sample]
+            chunk_bytes = chunk.tobytes()
+            
+            # Send to server
+            if self.ws:
+                await self.ws.send(chunk_bytes)
+                self.bytes_sent += len(chunk_bytes)
+            
+            sent_samples = end_sample
+            
+            # Progress logging (every 500ms worth of audio)
+            if self.verbose and sent_samples % (self.sample_rate // 2) == 0:
+                progress = (sent_samples / total_samples) * 100
+                print(f"  Sending: {progress:.0f}%", end="\r")
+            
+            # Delay to simulate real-time streaming
+            # Server expects audio at real-time pace for VAD/ASR to work properly
+            await asyncio.sleep(self.chunk_duration_ms / 1000)
+        
+        self.send_completed = True
+        elapsed = time.time() - self.send_start_time
+        self.log_event("→", f"Audio transmission complete ({elapsed:.2f}s, {self.bytes_sent/1024:.1f} KB)")
+    
+    async def receiver(self) -> None:
+        """Receive messages from server."""
+        try:
+            while self.running:
+                try:
+                    message = await asyncio.wait_for(self.ws.recv(), timeout=0.1)
+                    
+                    if isinstance(message, bytes):
+                        # Audio data received
+                        self.bytes_received += len(message)
+                        self.received_audio.extend(message)
+                        
+                        # Calculate TTFB on first audio of each response
+                        if self.waiting_for_first_audio and self.response_start_time is not None:
+                            ttfb_ms = (time.time() - self.response_start_time) * 1000
+                            self.ttfb_ms = ttfb_ms
+                            self.ttfb_list.append(ttfb_ms)
+                            self.waiting_for_first_audio = False
+                            self.log_event("←", f"[TTFB] First audio latency: {ttfb_ms:.0f}ms")
+                        
+                        # Log progress
+                        duration_ms = len(message) / (self.sample_rate * 2) * 1000
+                        total_ms = len(self.received_audio) / (self.sample_rate * 2) * 1000
+                        if self.verbose:
+                            print(f"← Audio: +{duration_ms:.0f}ms (total: {total_ms:.0f}ms)", end="\r")
+                        
+                    else:
+                        # JSON event
+                        event = json.loads(message)
+                        await self._handle_event(event)
+                        
+                except asyncio.TimeoutError:
+                    continue
+                except websockets.ConnectionClosed:
+                    self.log_event("←", "Connection closed")
+                    self.running = False
+                    break
+                    
+        except asyncio.CancelledError:
+            pass
+        except Exception as e:
+            self.log_event("!", f"Receiver error: {e}")
+            self.running = False
+    
+    async def _handle_event(self, event: dict) -> None:
+        """Handle incoming event."""
+        event_type = event.get("event", "unknown")
+        
+        if event_type == "answer":
+            self.log_event("←", "Session ready!")
+        elif event_type == "speaking":
+            self.log_event("←", "Speech detected")
+        elif event_type == "silence":
+            self.log_event("←", "Silence detected")
+        elif event_type == "transcript":
+            # ASR transcript (interim = asrDelta-style, final = asrFinal-style)
+            text = event.get("text", "")
+            is_final = event.get("isFinal", False)
+            if is_final:
+                # Clear interim line and print final
+                print(" " * 80, end="\r")
+                self.log_event("←", f"→ You: {text}")
+            else:
+                # Interim result - show with indicator (overwrite same line, as in mic_client)
+                display_text = text[:60] + "..." if len(text) > 60 else text
+                print(f"  [listening] {display_text}".ljust(80), end="\r")
+        elif event_type == "ttfb":
+            latency_ms = event.get("latencyMs", 0)
+            self.log_event("←", f"[TTFB] Server latency: {latency_ms}ms")
+        elif event_type == "llmResponse":
+            text = event.get("text", "")
+            is_final = event.get("isFinal", False)
+            if is_final:
+                self.log_event("←", f"LLM Response (final): {text[:100]}{'...' if len(text) > 100 else ''}")
+            elif self.verbose:
+                # Show streaming chunks only in verbose mode
+                self.log_event("←", f"LLM: {text}")
+        elif event_type == "trackStart":
+            self.track_started = True
+            self.response_start_time = time.time()
+            self.waiting_for_first_audio = True
+            self.log_event("←", "Bot started speaking")
+        elif event_type == "trackEnd":
+            self.track_ended = True
+            self.log_event("←", "Bot finished speaking")
+        elif event_type == "interrupt":
+            self.log_event("←", "Bot interrupted!")
+        elif event_type == "error":
+            self.log_event("!", f"Error: {event.get('error')}")
+        elif event_type == "hangup":
+            self.log_event("←", f"Hangup: {event.get('reason')}")
+            self.running = False
+        else:
+            self.log_event("←", f"Event: {event_type}")
+    
+    def save_output_wav(self) -> None:
+        """Save received audio to output WAV file."""
+        if not self.received_audio:
+            self.log_event("!", "No audio received to save")
+            return
+        
+        # Convert bytes to numpy array
+        audio_data = np.frombuffer(bytes(self.received_audio), dtype=np.int16)
+        
+        # Ensure output directory exists
+        self.output_file.parent.mkdir(parents=True, exist_ok=True)
+        
+        # Save using wave module for compatibility
+        with wave.open(str(self.output_file), 'wb') as wav_file:
+            wav_file.setnchannels(1)
+            wav_file.setsampwidth(2)  # 16-bit
+            wav_file.setframerate(self.sample_rate)
+            wav_file.writeframes(audio_data.tobytes())
+        
+        duration = len(audio_data) / self.sample_rate
+        self.log_event("→", f"Saved output: {self.output_file}")
+        self.log_event("→", f"  Duration: {duration:.2f}s ({len(audio_data)} samples)")
+        self.log_event("→", f"  Size: {len(self.received_audio)/1024:.1f} KB")
+    
+    async def run(self) -> None:
+        """Run the WAV file test."""
+        try:
+            # Load input WAV file
+            audio_data, _ = self.load_wav_file()
+            
+            # Connect to server
+            await self.connect()
+            
+            # Wait for answer
+            await asyncio.sleep(0.5)
+            
+            # Start receiver task
+            receiver_task = asyncio.create_task(self.receiver())
+            
+            # Send audio
+            await self.audio_sender(audio_data)
+            
+            # Wait for response
+            self.log_event("→", f"Waiting {self.wait_time}s for response...")
+            
+            wait_start = time.time()
+            while self.running and (time.time() - wait_start) < self.wait_time:
+                # Check if track has ended (response complete)
+                if self.track_ended and self.send_completed:
+                    # Give a little extra time for any remaining audio
+                    await asyncio.sleep(1.0)
+                    break
+                await asyncio.sleep(0.1)
+            
+            # Cleanup
+            self.running = False
+            receiver_task.cancel()
+            
+            try:
+                await receiver_task
+            except asyncio.CancelledError:
+                pass
+            
+            # Save output
+            self.save_output_wav()
+            
+            # Print summary
+            self._print_summary()
+            
+        except FileNotFoundError as e:
+            print(f"Error: {e}")
+            sys.exit(1)
+        except ConnectionRefusedError:
+            print(f"Error: Could not connect to {self.url}")
+            print("Make sure the server is running.")
+            sys.exit(1)
+        except Exception as e:
+            print(f"Error: {e}")
+            import traceback
+            traceback.print_exc()
+            sys.exit(1)
+        finally:
+            await self.close()
+    
+    def _print_summary(self):
+        """Print session summary."""
+        print("\n" + "=" * 50)
+        print("Session Summary")
+        print("=" * 50)
+        print(f"  Input file:  {self.input_file}")
+        print(f"  Output file: {self.output_file}")
+        print(f"  Bytes sent:     {self.bytes_sent / 1024:.1f} KB")
+        print(f"  Bytes received: {self.bytes_received / 1024:.1f} KB")
+        if self.ttfb_list:
+            if len(self.ttfb_list) == 1:
+                print(f"  TTFB:           {self.ttfb_list[0]:.0f} ms")
+            else:
+                print(f"  TTFB (per response): {', '.join(f'{t:.0f}ms' for t in self.ttfb_list)}")
+        if self.received_audio:
+            duration = len(self.received_audio) / (self.sample_rate * 2)
+            print(f"  Response duration: {duration:.2f}s")
+        print("=" * 50)
+    
+    async def close(self) -> None:
+        """Close the connection."""
+        self.running = False
+        if self.ws:
+            try:
+                await self.ws.close()
+            except:
+                pass
+
+
+async def main():
+    parser = argparse.ArgumentParser(
+        description="WAV file client for testing duplex voice conversation"
+    )
+    parser.add_argument(
+        "--input", "-i",
+        required=True,
+        help="Input WAV file path"
+    )
+    parser.add_argument(
+        "--output", "-o",
+        required=True,
+        help="Output WAV file path for response"
+    )
+    parser.add_argument(
+        "--url",
+        default="ws://localhost:8000/ws",
+        help="WebSocket server URL (default: ws://localhost:8000/ws)"
+    )
+    parser.add_argument(
+        "--sample-rate",
+        type=int,
+        default=16000,
+        help="Target sample rate for audio (default: 16000)"
+    )
+    parser.add_argument(
+        "--chunk-duration",
+        type=int,
+        default=20,
+        help="Chunk duration in ms for sending (default: 20)"
+    )
+    parser.add_argument(
+        "--wait-time", "-w",
+        type=float,
+        default=15.0,
+        help="Time to wait for response after sending (default: 15.0)"
+    )
+    parser.add_argument(
+        "--verbose", "-v",
+        action="store_true",
+        help="Enable verbose output"
+    )
+    
+    args = parser.parse_args()
+    
+    client = WavFileClient(
+        url=args.url,
+        input_file=args.input,
+        output_file=args.output,
+        sample_rate=args.sample_rate,
+        chunk_duration_ms=args.chunk_duration,
+        wait_time=args.wait_time,
+        verbose=args.verbose
+    )
+    
+    await client.run()
+
+
+if __name__ == "__main__":
+    try:
+        asyncio.run(main())
+    except KeyboardInterrupt:
+        print("\nInterrupted by user")

examples/web_client.html

@@ -0,0 +1,742 @@
+<!doctype html>
+<html lang="en">
+  <head>
+    <meta charset="utf-8" />
+    <meta name="viewport" content="width=device-width, initial-scale=1" />
+    <title>Duplex Voice Web Client</title>
+    <style>
+      @import url("https://fonts.googleapis.com/css2?family=Fraunces:opsz,wght@9..144,300;9..144,500;9..144,700&family=Recursive:wght@300;400;600;700&display=swap");
+
+      :root {
+        --bg: #0b0b0f;
+        --panel: #14141c;
+        --panel-2: #101018;
+        --ink: #f2f3f7;
+        --muted: #a7acba;
+        --accent: #ff6b6b;
+        --accent-2: #ffd166;
+        --good: #2dd4bf;
+        --bad: #f87171;
+        --grid: rgba(255, 255, 255, 0.06);
+        --shadow: 0 20px 60px rgba(0, 0, 0, 0.45);
+      }
+
+      * {
+        box-sizing: border-box;
+      }
+
+      html,
+      body {
+        height: 100%;
+        margin: 0;
+        color: var(--ink);
+        background: radial-gradient(1200px 600px at 20% -10%, #1d1d2a 0%, transparent 60%),
+          radial-gradient(800px 800px at 110% 10%, #20203a 0%, transparent 50%),
+          var(--bg);
+        font-family: "Recursive", ui-sans-serif, system-ui, -apple-system, "Segoe UI", sans-serif;
+      }
+
+      .noise {
+        position: fixed;
+        inset: 0;
+        background-image: url("data:image/svg+xml;utf8,<svg xmlns='http://www.w3.org/2000/svg' width='120' height='120' viewBox='0 0 120 120'><filter id='n'><feTurbulence type='fractalNoise' baseFrequency='0.9' numOctaves='2' stitchTiles='stitch'/></filter><rect width='120' height='120' filter='url(%23n)' opacity='0.06'/></svg>");
+        pointer-events: none;
+        mix-blend-mode: soft-light;
+      }
+
+      header {
+        padding: 32px 28px 18px;
+        border-bottom: 1px solid var(--grid);
+      }
+
+      h1 {
+        font-family: "Fraunces", serif;
+        font-weight: 600;
+        margin: 0 0 6px;
+        letter-spacing: 0.4px;
+      }
+
+      .subtitle {
+        color: var(--muted);
+        font-size: 0.95rem;
+      }
+
+      main {
+        display: grid;
+        grid-template-columns: 1.1fr 1.4fr;
+        gap: 24px;
+        padding: 24px 28px 40px;
+      }
+
+      .panel {
+        background: linear-gradient(180deg, rgba(255, 255, 255, 0.02), transparent),
+          var(--panel);
+        border: 1px solid var(--grid);
+        border-radius: 16px;
+        padding: 20px;
+        box-shadow: var(--shadow);
+      }
+
+      .panel h2 {
+        margin: 0 0 12px;
+        font-size: 1.05rem;
+        font-weight: 600;
+      }
+
+      .stack {
+        display: grid;
+        gap: 12px;
+      }
+
+      label {
+        display: block;
+        font-size: 0.85rem;
+        color: var(--muted);
+        margin-bottom: 6px;
+      }
+
+      input,
+      select,
+      button,
+      textarea {
+        font-family: inherit;
+      }
+
+      input,
+      select,
+      textarea {
+        width: 100%;
+        padding: 10px 12px;
+        border-radius: 10px;
+        border: 1px solid var(--grid);
+        background: var(--panel-2);
+        color: var(--ink);
+        outline: none;
+      }
+
+      textarea {
+        min-height: 80px;
+        resize: vertical;
+      }
+
+      .row {
+        display: grid;
+        grid-template-columns: 1fr 1fr;
+        gap: 12px;
+      }
+
+      .btn-row {
+        display: flex;
+        flex-wrap: wrap;
+        gap: 10px;
+      }
+
+      button {
+        border: none;
+        border-radius: 999px;
+        padding: 10px 16px;
+        font-weight: 600;
+        background: var(--ink);
+        color: #111;
+        cursor: pointer;
+        transition: transform 0.2s ease, box-shadow 0.2s ease;
+      }
+
+      button.secondary {
+        background: transparent;
+        color: var(--ink);
+        border: 1px solid var(--grid);
+      }
+
+      button.accent {
+        background: linear-gradient(120deg, var(--accent), #f97316);
+        color: #0b0b0f;
+      }
+
+      button.good {
+        background: linear-gradient(120deg, var(--good), #22c55e);
+        color: #07261f;
+      }
+
+      button.bad {
+        background: linear-gradient(120deg, var(--bad), #f97316);
+        color: #2a0b0b;
+      }
+
+      button:active {
+        transform: translateY(1px) scale(0.99);
+      }
+
+      .status {
+        display: flex;
+        align-items: center;
+        gap: 12px;
+        padding: 12px;
+        background: rgba(255, 255, 255, 0.03);
+        border-radius: 12px;
+        border: 1px dashed var(--grid);
+        font-size: 0.9rem;
+      }
+
+      .dot {
+        width: 10px;
+        height: 10px;
+        border-radius: 999px;
+        background: var(--bad);
+        box-shadow: 0 0 12px rgba(248, 113, 113, 0.5);
+      }
+
+      .dot.on {
+        background: var(--good);
+        box-shadow: 0 0 12px rgba(45, 212, 191, 0.7);
+      }
+
+      .log {
+        height: 320px;
+        overflow: auto;
+        padding: 12px;
+        background: #0d0d14;
+        border-radius: 12px;
+        border: 1px solid var(--grid);
+        font-size: 0.85rem;
+        line-height: 1.4;
+      }
+
+      .chat {
+        height: 260px;
+        overflow: auto;
+        padding: 12px;
+        background: #0d0d14;
+        border-radius: 12px;
+        border: 1px solid var(--grid);
+        font-size: 0.9rem;
+        line-height: 1.45;
+      }
+
+      .chat-entry {
+        padding: 8px 10px;
+        margin-bottom: 8px;
+        border-radius: 10px;
+        background: rgba(255, 255, 255, 0.04);
+        border: 1px solid rgba(255, 255, 255, 0.06);
+      }
+
+      .chat-entry.user {
+        border-left: 3px solid var(--accent-2);
+      }
+
+      .chat-entry.ai {
+        border-left: 3px solid var(--good);
+      }
+
+      .chat-entry.interim {
+        opacity: 0.7;
+        font-style: italic;
+      }
+
+      .log-entry {
+        padding: 6px 8px;
+        border-bottom: 1px dashed rgba(255, 255, 255, 0.06);
+      }
+
+      .log-entry:last-child {
+        border-bottom: none;
+      }
+
+      .tag {
+        display: inline-flex;
+        align-items: center;
+        gap: 6px;
+        padding: 2px 8px;
+        border-radius: 999px;
+        font-size: 0.7rem;
+        text-transform: uppercase;
+        letter-spacing: 0.6px;
+        background: rgba(255, 255, 255, 0.08);
+        color: var(--muted);
+      }
+
+      .tag.event {
+        background: rgba(255, 107, 107, 0.18);
+        color: #ffc1c1;
+      }
+
+      .tag.audio {
+        background: rgba(45, 212, 191, 0.2);
+        color: #c5f9f0;
+      }
+
+      .tag.sys {
+        background: rgba(255, 209, 102, 0.2);
+        color: #ffefb0;
+      }
+
+      .muted {
+        color: var(--muted);
+      }
+
+      footer {
+        padding: 0 28px 28px;
+        color: var(--muted);
+        font-size: 0.8rem;
+      }
+
+      @media (max-width: 1100px) {
+        main {
+          grid-template-columns: 1fr;
+        }
+        .log {
+          height: 360px;
+        }
+        .chat {
+          height: 260px;
+        }
+      }
+    </style>
+  </head>
+  <body>
+    <div class="noise"></div>
+    <header>
+      <h1>Duplex Voice Client</h1>
+      <div class="subtitle">Browser client for the WebSocket duplex pipeline. Device selection + event logging.</div>
+    </header>
+
+    <main>
+      <section class="panel stack">
+        <h2>Connection</h2>
+        <div>
+          <label for="wsUrl">WebSocket URL</label>
+          <input id="wsUrl" value="ws://localhost:8000/ws" />
+        </div>
+        <div class="btn-row">
+          <button class="accent" id="connectBtn">Connect</button>
+          <button class="secondary" id="disconnectBtn">Disconnect</button>
+        </div>
+        <div class="status">
+          <div id="statusDot" class="dot"></div>
+          <div>
+            <div id="statusText">Disconnected</div>
+            <div class="muted" id="statusSub">Waiting for connection</div>
+          </div>
+        </div>
+
+        <h2>Devices</h2>
+        <div class="row">
+          <div>
+            <label for="inputSelect">Input (Mic)</label>
+            <select id="inputSelect"></select>
+          </div>
+          <div>
+            <label for="outputSelect">Output (Speaker)</label>
+            <select id="outputSelect"></select>
+          </div>
+        </div>
+        <div class="btn-row">
+          <button class="secondary" id="refreshDevicesBtn">Refresh Devices</button>
+          <button class="good" id="startMicBtn">Start Mic</button>
+          <button class="secondary" id="stopMicBtn">Stop Mic</button>
+        </div>
+
+        <h2>Chat</h2>
+        <div class="stack">
+          <textarea id="chatInput" placeholder="Type a message, press Send"></textarea>
+          <div class="btn-row">
+            <button class="accent" id="sendChatBtn">Send Chat</button>
+            <button class="secondary" id="clearLogBtn">Clear Log</button>
+          </div>
+        </div>
+      </section>
+
+      <section class="stack">
+        <div class="panel stack">
+          <h2>Chat History</h2>
+          <div class="chat" id="chatHistory"></div>
+        </div>
+        <div class="panel stack">
+          <h2>Event Log</h2>
+          <div class="log" id="log"></div>
+        </div>
+      </section>
+    </main>
+
+    <footer>
+      Output device selection requires HTTPS + a browser that supports <code>setSinkId</code>.
+      Audio is sent as 16-bit PCM @ 16 kHz, matching <code>examples/mic_client.py</code>.
+    </footer>
+
+    <audio id="audioOut" autoplay></audio>
+
+    <script>
+      const wsUrl = document.getElementById("wsUrl");
+      const connectBtn = document.getElementById("connectBtn");
+      const disconnectBtn = document.getElementById("disconnectBtn");
+      const inputSelect = document.getElementById("inputSelect");
+      const outputSelect = document.getElementById("outputSelect");
+      const startMicBtn = document.getElementById("startMicBtn");
+      const stopMicBtn = document.getElementById("stopMicBtn");
+      const refreshDevicesBtn = document.getElementById("refreshDevicesBtn");
+      const sendChatBtn = document.getElementById("sendChatBtn");
+      const clearLogBtn = document.getElementById("clearLogBtn");
+      const chatInput = document.getElementById("chatInput");
+      const logEl = document.getElementById("log");
+      const chatHistory = document.getElementById("chatHistory");
+      const statusDot = document.getElementById("statusDot");
+      const statusText = document.getElementById("statusText");
+      const statusSub = document.getElementById("statusSub");
+      const audioOut = document.getElementById("audioOut");
+
+      let ws = null;
+      let audioCtx = null;
+      let micStream = null;
+      let processor = null;
+      let micSource = null;
+      let playbackDest = null;
+      let playbackTime = 0;
+      let discardAudio = false;
+      let playbackSources = [];
+      let interimUserEl = null;
+      let interimAiEl = null;
+      let interimUserText = "";
+      let interimAiText = "";
+
+      const targetSampleRate = 16000;
+
+      function logLine(type, text, data) {
+        const time = new Date().toLocaleTimeString();
+        const entry = document.createElement("div");
+        entry.className = "log-entry";
+        const tag = document.createElement("span");
+        tag.className = `tag ${type}`;
+        tag.textContent = type.toUpperCase();
+        const msg = document.createElement("span");
+        msg.style.marginLeft = "10px";
+        msg.textContent = `[${time}] ${text}`;
+        entry.appendChild(tag);
+        entry.appendChild(msg);
+        if (data) {
+          const pre = document.createElement("div");
+          pre.className = "muted";
+          pre.textContent = JSON.stringify(data);
+          pre.style.marginTop = "4px";
+          entry.appendChild(pre);
+        }
+        logEl.appendChild(entry);
+        logEl.scrollTop = logEl.scrollHeight;
+      }
+
+      function addChat(role, text) {
+        const entry = document.createElement("div");
+        entry.className = `chat-entry ${role === "AI" ? "ai" : "user"}`;
+        entry.textContent = `${role}: ${text}`;
+        chatHistory.appendChild(entry);
+        chatHistory.scrollTop = chatHistory.scrollHeight;
+      }
+
+      function setInterim(role, text) {
+        const isAi = role === "AI";
+        let el = isAi ? interimAiEl : interimUserEl;
+        if (!text) {
+          if (el) el.remove();
+          if (isAi) interimAiEl = null;
+          else interimUserEl = null;
+          if (isAi) interimAiText = "";
+          else interimUserText = "";
+          return;
+        }
+        if (!el) {
+          el = document.createElement("div");
+          el.className = `chat-entry ${isAi ? "ai" : "user"} interim`;
+          chatHistory.appendChild(el);
+          if (isAi) interimAiEl = el;
+          else interimUserEl = el;
+        }
+        el.textContent = `${role} (interim): ${text}`;
+        chatHistory.scrollTop = chatHistory.scrollHeight;
+      }
+
+      function stopPlayback() {
+        discardAudio = true;
+        playbackTime = audioCtx ? audioCtx.currentTime : 0;
+        playbackSources.forEach((s) => {
+          try {
+            s.stop();
+          } catch (err) {}
+        });
+        playbackSources = [];
+      }
+
+      function setStatus(connected, detail) {
+        statusDot.classList.toggle("on", connected);
+        statusText.textContent = connected ? "Connected" : "Disconnected";
+        statusSub.textContent = detail || "";
+      }
+
+      async function ensureAudioContext() {
+        if (audioCtx) return;
+        audioCtx = new (window.AudioContext || window.webkitAudioContext)();
+        playbackDest = audioCtx.createMediaStreamDestination();
+        audioOut.srcObject = playbackDest.stream;
+        try {
+          await audioOut.play();
+        } catch (err) {
+          logLine("sys", "Audio playback blocked (user gesture needed)", { err: String(err) });
+        }
+        if (outputSelect.value) {
+          await setOutputDevice(outputSelect.value);
+        }
+      }
+
+      function downsampleBuffer(buffer, inRate, outRate) {
+        if (outRate === inRate) return buffer;
+        const ratio = inRate / outRate;
+        const newLength = Math.round(buffer.length / ratio);
+        const result = new Float32Array(newLength);
+        let offsetResult = 0;
+        let offsetBuffer = 0;
+        while (offsetResult < result.length) {
+          const nextOffsetBuffer = Math.round((offsetResult + 1) * ratio);
+          let accum = 0;
+          let count = 0;
+          for (let i = offsetBuffer; i < nextOffsetBuffer && i < buffer.length; i++) {
+            accum += buffer[i];
+            count++;
+          }
+          result[offsetResult] = accum / count;
+          offsetResult++;
+          offsetBuffer = nextOffsetBuffer;
+        }
+        return result;
+      }
+
+      function floatTo16BitPCM(float32) {
+        const out = new Int16Array(float32.length);
+        for (let i = 0; i < float32.length; i++) {
+          const s = Math.max(-1, Math.min(1, float32[i]));
+          out[i] = s < 0 ? s * 0x8000 : s * 0x7fff;
+        }
+        return out;
+      }
+
+      function schedulePlayback(int16Data) {
+        if (!audioCtx || !playbackDest) return;
+        if (discardAudio) return;
+        const float32 = new Float32Array(int16Data.length);
+        for (let i = 0; i < int16Data.length; i++) {
+          float32[i] = int16Data[i] / 32768;
+        }
+        const buffer = audioCtx.createBuffer(1, float32.length, targetSampleRate);
+        buffer.copyToChannel(float32, 0);
+        const source = audioCtx.createBufferSource();
+        source.buffer = buffer;
+        source.connect(playbackDest);
+        const startTime = Math.max(audioCtx.currentTime + 0.02, playbackTime);
+        source.start(startTime);
+        playbackTime = startTime + buffer.duration;
+        playbackSources.push(source);
+        source.onended = () => {
+          playbackSources = playbackSources.filter((s) => s !== source);
+        };
+      }
+
+      async function connect() {
+        if (ws && ws.readyState === WebSocket.OPEN) return;
+        ws = new WebSocket(wsUrl.value.trim());
+        ws.binaryType = "arraybuffer";
+
+        ws.onopen = () => {
+          setStatus(true, "Session open");
+          logLine("sys", "WebSocket connected");
+          ensureAudioContext();
+          sendCommand({ command: "invite", option: { codec: "pcm", sampleRate: targetSampleRate } });
+        };
+
+        ws.onclose = () => {
+          setStatus(false, "Connection closed");
+          logLine("sys", "WebSocket closed");
+          ws = null;
+        };
+
+        ws.onerror = (err) => {
+          logLine("sys", "WebSocket error", { err: String(err) });
+        };
+
+        ws.onmessage = (msg) => {
+          if (typeof msg.data === "string") {
+            const event = JSON.parse(msg.data);
+            handleEvent(event);
+          } else {
+            const audioBuf = msg.data;
+            const int16 = new Int16Array(audioBuf);
+            schedulePlayback(int16);
+            logLine("audio", `Audio ${Math.round((int16.length / targetSampleRate) * 1000)}ms`);
+          }
+        };
+      }
+
+      function disconnect() {
+        if (ws) ws.close();
+        ws = null;
+        setStatus(false, "Disconnected");
+      }
+
+      function sendCommand(cmd) {
+        if (!ws || ws.readyState !== WebSocket.OPEN) {
+          logLine("sys", "Not connected");
+          return;
+        }
+        ws.send(JSON.stringify(cmd));
+        logLine("sys", `→ ${cmd.command}`, cmd);
+      }
+
+      function handleEvent(event) {
+        const type = event.event || "unknown";
+        logLine("event", type, event);
+        if (type === "transcript") {
+          if (event.isFinal && event.text) {
+            setInterim("You", "");
+            addChat("You", event.text);
+          } else if (event.text) {
+            interimUserText += event.text;
+            setInterim("You", interimUserText);
+          }
+        }
+        if (type === "llmResponse") {
+          if (event.isFinal && event.text) {
+            setInterim("AI", "");
+            addChat("AI", event.text);
+          } else if (event.text) {
+            interimAiText += event.text;
+            setInterim("AI", interimAiText);
+          }
+        }
+        if (type === "trackStart") {
+          // New bot audio: stop any previous playback to avoid overlap
+          stopPlayback();
+          discardAudio = false;
+        }
+        if (type === "speaking") {
+          // User started speaking: clear any in-flight audio to avoid overlap
+          stopPlayback();
+        }
+        if (type === "interrupt") {
+          stopPlayback();
+        }
+      }
+
+      async function startMic() {
+        if (!ws || ws.readyState !== WebSocket.OPEN) {
+          logLine("sys", "Connect before starting mic");
+          return;
+        }
+        await ensureAudioContext();
+        const deviceId = inputSelect.value || undefined;
+        micStream = await navigator.mediaDevices.getUserMedia({
+          audio: deviceId ? { deviceId: { exact: deviceId } } : true,
+        });
+        micSource = audioCtx.createMediaStreamSource(micStream);
+        processor = audioCtx.createScriptProcessor(2048, 1, 1);
+        processor.onaudioprocess = (e) => {
+          if (!ws || ws.readyState !== WebSocket.OPEN) return;
+          const input = e.inputBuffer.getChannelData(0);
+          const downsampled = downsampleBuffer(input, audioCtx.sampleRate, targetSampleRate);
+          const pcm16 = floatTo16BitPCM(downsampled);
+          ws.send(pcm16.buffer);
+        };
+        micSource.connect(processor);
+        processor.connect(audioCtx.destination);
+        logLine("sys", "Microphone started");
+      }
+
+      function stopMic() {
+        if (processor) {
+          processor.disconnect();
+          processor = null;
+        }
+        if (micSource) {
+          micSource.disconnect();
+          micSource = null;
+        }
+        if (micStream) {
+          micStream.getTracks().forEach((t) => t.stop());
+          micStream = null;
+        }
+        logLine("sys", "Microphone stopped");
+      }
+
+      async function refreshDevices() {
+        const devices = await navigator.mediaDevices.enumerateDevices();
+        inputSelect.innerHTML = "";
+        outputSelect.innerHTML = "";
+        devices.forEach((d) => {
+          if (d.kind === "audioinput") {
+            const opt = document.createElement("option");
+            opt.value = d.deviceId;
+            opt.textContent = d.label || `Mic ${inputSelect.length + 1}`;
+            inputSelect.appendChild(opt);
+          }
+          if (d.kind === "audiooutput") {
+            const opt = document.createElement("option");
+            opt.value = d.deviceId;
+            opt.textContent = d.label || `Output ${outputSelect.length + 1}`;
+            outputSelect.appendChild(opt);
+          }
+        });
+      }
+
+      async function requestDeviceAccess() {
+        // Needed to reveal device labels in most browsers
+        try {
+          const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+          stream.getTracks().forEach((t) => t.stop());
+          logLine("sys", "Microphone permission granted");
+        } catch (err) {
+          logLine("sys", "Microphone permission denied", { err: String(err) });
+        }
+      }
+
+      async function setOutputDevice(deviceId) {
+        if (!audioOut.setSinkId) {
+          logLine("sys", "setSinkId not supported in this browser");
+          return;
+        }
+        await audioOut.setSinkId(deviceId);
+        logLine("sys", `Output device set`, { deviceId });
+      }
+
+      connectBtn.addEventListener("click", connect);
+      disconnectBtn.addEventListener("click", disconnect);
+      refreshDevicesBtn.addEventListener("click", async () => {
+        await requestDeviceAccess();
+        await refreshDevices();
+      });
+      startMicBtn.addEventListener("click", startMic);
+      stopMicBtn.addEventListener("click", stopMic);
+      sendChatBtn.addEventListener("click", () => {
+        const text = chatInput.value.trim();
+        if (!text) return;
+        ensureAudioContext();
+        addChat("You", text);
+        sendCommand({ command: "chat", text });
+        chatInput.value = "";
+      });
+      clearLogBtn.addEventListener("click", () => {
+        logEl.innerHTML = "";
+        chatHistory.innerHTML = "";
+        setInterim("You", "");
+        setInterim("AI", "");
+        interimUserText = "";
+        interimAiText = "";
+      });
+      inputSelect.addEventListener("change", () => {
+        if (micStream) {
+          stopMic();
+          startMic();
+        }
+      });
+      outputSelect.addEventListener("change", () => setOutputDevice(outputSelect.value));
+
+      navigator.mediaDevices.addEventListener("devicechange", refreshDevices);
+      refreshDevices().catch(() => {});
+    </script>
+  </body>
+</html>

models/events.py

@@ -179,6 +179,13 @@ class DTMFEvent(BaseEvent):
     digit: str = Field(..., description="DTMF digit (0-9, *, #, A-D)")
 
 
+class HeartBeatEvent(BaseModel):
+    """Server-to-client heartbeat to keep connection alive."""
+
+    event: str = Field(default="heartBeat", description="Event type")
+    timestamp: int = Field(default_factory=current_timestamp_ms, description="Event timestamp in milliseconds")
+
+
 # Event type mapping
 EVENT_TYPES = {
     "incoming": IncomingEvent,
@@ -198,6 +205,7 @@ EVENT_TYPES = {
     "metrics": MetricsEvent,
     "addHistory": AddHistoryEvent,
     "dtmf": DTMFEvent,
+    "heartBeat": HeartBeatEvent,
 }
 
 

processors/vad.py

@@ -6,7 +6,6 @@ from typing import Tuple, Optional
 import numpy as np
 from loguru import logger
 
-from processors.eou import EouDetector
 
 # Try to import onnxruntime (optional for VAD functionality)
 try:
@@ -64,6 +63,7 @@ class SileroVAD:
         self.min_chunk_size = 512
         self.last_label = "Silence"
         self.last_probability = 0.0
+        self._energy_noise_floor = 1e-4
 
     def _reset_state(self):
         # Silero VAD V4+ expects state shape [2, 1, 128]
@@ -82,8 +82,27 @@ class SileroVAD:
             Tuple of (label, probability) where label is "Speech" or "Silence"
         """
         if self.session is None or not ONNX_AVAILABLE:
-            # If model not loaded or onnxruntime not available, assume speech
+            # Fallback energy-based VAD with adaptive noise floor.
-            return "Speech", 1.0
+            if not pcm_bytes:
+                return "Silence", 0.0
+            audio_int16 = np.frombuffer(pcm_bytes, dtype=np.int16)
+            if audio_int16.size == 0:
+                return "Silence", 0.0
+            audio_float = audio_int16.astype(np.float32) / 32768.0
+            rms = float(np.sqrt(np.mean(audio_float * audio_float)))
+
+            # Update adaptive noise floor (slowly rises, faster to fall)
+            if rms < self._energy_noise_floor:
+                self._energy_noise_floor = 0.95 * self._energy_noise_floor + 0.05 * rms
+            else:
+                self._energy_noise_floor = 0.995 * self._energy_noise_floor + 0.005 * rms
+
+            # Compute SNR-like ratio and map to probability
+            denom = max(self._energy_noise_floor, 1e-6)
+            snr = max(0.0, (rms - denom) / denom)
+            probability = min(1.0, snr / 3.0)  # ~3x above noise => strong speech
+            label = "Speech" if probability >= 0.5 else "Silence"
+            return label, probability
 
         # Convert bytes to numpy array of int16
         audio_int16 = np.frombuffer(pcm_bytes, dtype=np.int16)
@@ -148,25 +167,19 @@ class VADProcessor:
     Tracks speech/silence state and emits events on transitions.
     """
 
-    def __init__(self, vad_model: SileroVAD, threshold: float = 0.5,
+    def __init__(self, vad_model: SileroVAD, threshold: float = 0.5):
-                 silence_threshold_ms: int = 1000, min_speech_duration_ms: int = 250):
         """
         Initialize VAD processor.
 
         Args:
             vad_model: Silero VAD model instance
             threshold: Speech detection threshold
-            silence_threshold_ms: EOU silence threshold in ms (longer = one EOU across short pauses)
-            min_speech_duration_ms: EOU min speech duration in ms (ignore very short noises)
         """
         self.vad = vad_model
         self.threshold = threshold
-        self._eou_silence_ms = silence_threshold_ms
-        self._eou_min_speech_ms = min_speech_duration_ms
         self.is_speaking = False
         self.speech_start_time: Optional[float] = None
         self.silence_start_time: Optional[float] = None
-        self.eou_detector = EouDetector(silence_threshold_ms, min_speech_duration_ms)
 
     def process(self, pcm_bytes: bytes, chunk_size_ms: int = 20) -> Optional[Tuple[str, float]]:
         """
@@ -184,10 +197,6 @@ class VADProcessor:
         # Check if this is speech based on threshold
         is_speech = probability >= self.threshold
         
-        # Check EOU
-        if self.eou_detector.process("Speech" if is_speech else "Silence"):
-            return ("eou", probability)
-
         # State transition: Silence -> Speech
         if is_speech and not self.is_speaking:
             self.is_speaking = True
@@ -210,4 +219,3 @@ class VADProcessor:
         self.is_speaking = False
         self.speech_start_time = None
         self.silence_start_time = None
-        self.eou_detector = EouDetector(self._eou_silence_ms, self._eou_min_speech_ms)

scripts/README.md

@@ -0,0 +1 @@
+# Development Script

scripts/generate_test_audio/generate_test_audio.py

@@ -0,0 +1,311 @@
+#!/usr/bin/env python3
+"""
+Generate test audio file with utterances using SiliconFlow TTS API.
+
+Creates a 16kHz mono WAV file with real speech segments separated by
+configurable silence (for VAD/testing).
+
+Usage:
+  python generate_test_audio.py [OPTIONS]
+
+Options:
+  -o, --output PATH       Output WAV path (default: data/audio_examples/two_utterances_16k.wav)
+  -u, --utterance TEXT    Utterance text; repeat for multiple (ignored if -j is set)
+  -j, --json PATH         JSON file: array of strings or {"utterances": [...]}
+  --silence-ms MS         Silence in ms between utterances (default: 500)
+  --lead-silence-ms MS    Silence in ms at start (default: 200)
+  --trail-silence-ms MS   Silence in ms at end (default: 300)
+
+Examples:
+  # Default utterances and output
+  python generate_test_audio.py
+
+  # Custom output path
+  python generate_test_audio.py -o out.wav
+
+  # Utterances from command line
+  python generate_test_audio.py -u "Hello" -u "World" -o test.wav
+
+  # Utterancgenerate_test_audio.py -j utterances.json -o test.wav
+
+  # Custom silence (1s between utterances)
+  python generate_test_audio.py -u "One" -u "Two" --silence-ms 1000 -o test.wav
+
+Requires SILICONFLOW_API_KEY in .env.
+"""
+
+import wave
+import struct
+import argparse
+import asyncio
+import aiohttp
+import json
+import os
+from pathlib import Path
+from dotenv import load_dotenv
+
+
+# Load .env file from project root
+project_root = Path(__file__).parent.parent.parent
+load_dotenv(project_root / ".env")
+
+
+# SiliconFlow TTS Configuration
+SILICONFLOW_API_URL = "https://api.siliconflow.cn/v1/audio/speech"
+SILICONFLOW_MODEL = "FunAudioLLM/CosyVoice2-0.5B"
+
+# Available voices
+VOICES = {
+    "alex": "FunAudioLLM/CosyVoice2-0.5B:alex",
+    "anna": "FunAudioLLM/CosyVoice2-0.5B:anna",
+    "bella": "FunAudioLLM/CosyVoice2-0.5B:bella",
+    "benjamin": "FunAudioLLM/CosyVoice2-0.5B:benjamin",
+    "charles": "FunAudioLLM/CosyVoice2-0.5B:charles",
+    "claire": "FunAudioLLM/CosyVoice2-0.5B:claire",
+    "david": "FunAudioLLM/CosyVoice2-0.5B:david",
+    "diana": "FunAudioLLM/CosyVoice2-0.5B:diana",
+}
+
+
+def generate_silence(duration_ms: int, sample_rate: int = 16000) -> bytes:
+    """Generate silence as PCM bytes."""
+    num_samples = int(sample_rate * (duration_ms / 1000.0))
+    return b'\x00\x00' * num_samples
+
+
+async def synthesize_speech(
+    text: str,
+    api_key: str,
+    voice: str = "anna",
+    sample_rate: int = 16000,
+    speed: float = 1.0
+) -> bytes:
+    """
+    Synthesize speech using SiliconFlow TTS API.
+    
+    Args:
+        text: Text to synthesize
+        api_key: SiliconFlow API key
+        voice: Voice name (alex, anna, bella, benjamin, charles, claire, david, diana)
+        sample_rate: Output sample rate (8000, 16000, 24000, 32000, 44100)
+        speed: Speech speed (0.25 to 4.0)
+        
+    Returns:
+        PCM audio bytes (16-bit signed, little-endian)
+    """
+    # Resolve voice name
+    full_voice = VOICES.get(voice, voice)
+    
+    payload = {
+        "model": SILICONFLOW_MODEL,
+        "input": text,
+        "voice": full_voice,
+        "response_format": "pcm",
+        "sample_rate": sample_rate,
+        "stream": False,
+        "speed": speed
+    }
+    
+    headers = {
+        "Authorization": f"Bearer {api_key}",
+        "Content-Type": "application/json"
+    }
+    
+    async with aiohttp.ClientSession() as session:
+        async with session.post(SILICONFLOW_API_URL, json=payload, headers=headers) as response:
+            if response.status != 200:
+                error_text = await response.text()
+                raise RuntimeError(f"SiliconFlow TTS error: {response.status} - {error_text}")
+            
+            return await response.read()
+
+
+async def generate_test_audio(
+    output_path: str,
+    utterances: list[str],
+    silence_ms: int = 500,
+    lead_silence_ms: int = 200,
+    trail_silence_ms: int = 300,
+    voice: str = "anna",
+    sample_rate: int = 16000,
+    speed: float = 1.0
+):
+    """
+    Generate test audio with multiple utterances separated by silence.
+    
+    Args:
+        output_path: Path to save the WAV file
+        utterances: List of text strings for each utterance
+        silence_ms: Silence duration between utterances (milliseconds)
+        lead_silence_ms: Silence at the beginning (milliseconds)
+        trail_silence_ms: Silence at the end (milliseconds)
+        voice: TTS voice to use
+        sample_rate: Audio sample rate
+        speed: TTS speech speed
+    """
+    api_key = os.getenv("SILICONFLOW_API_KEY")
+    if not api_key:
+        raise ValueError(
+            "SILICONFLOW_API_KEY not found in environment.\n"
+            "Please set it in your .env file:\n"
+            "  SILICONFLOW_API_KEY=your-api-key-here"
+        )
+    
+    print(f"Using SiliconFlow TTS API")
+    print(f"  Voice: {voice}")
+    print(f"  Sample rate: {sample_rate}Hz")
+    print(f"  Speed: {speed}x")
+    print()
+    
+    segments = []
+    
+    # Lead-in silence
+    if lead_silence_ms > 0:
+        segments.append(generate_silence(lead_silence_ms, sample_rate))
+        print(f"  [silence: {lead_silence_ms}ms]")
+    
+    # Generate each utterance with silence between
+    for i, text in enumerate(utterances):
+        print(f"  Synthesizing utterance {i + 1}: \"{text}\"")
+        audio = await synthesize_speech(
+            text=text,
+            api_key=api_key,
+            voice=voice,
+            sample_rate=sample_rate,
+            speed=speed
+        )
+        segments.append(audio)
+        
+        # Add silence between utterances (not after the last one)
+        if i < len(utterances) - 1:
+            segments.append(generate_silence(silence_ms, sample_rate))
+            print(f"  [silence: {silence_ms}ms]")
+    
+    # Trail silence
+    if trail_silence_ms > 0:
+        segments.append(generate_silence(trail_silence_ms, sample_rate))
+        print(f"  [silence: {trail_silence_ms}ms]")
+    
+    # Concatenate all segments
+    audio_data = b''.join(segments)
+    
+    # Write WAV file
+    with wave.open(output_path, 'wb') as wf:
+        wf.setnchannels(1)          # Mono
+        wf.setsampwidth(2)          # 16-bit
+        wf.setframerate(sample_rate)
+        wf.writeframes(audio_data)
+    
+    duration_sec = len(audio_data) / (sample_rate * 2)
+    print()
+    print(f"Generated: {output_path}")
+    print(f"  Duration: {duration_sec:.2f}s")
+    print(f"  Sample rate: {sample_rate}Hz")
+    print(f"  Format: 16-bit mono PCM WAV")
+    print(f"  Size: {len(audio_data):,} bytes")
+
+
+def load_utterances_from_json(path: Path) -> list[str]:
+    """
+    Load utterances from a JSON file.
+    
+    Accepts either:
+    - A JSON array: ["utterance 1", "utterance 2"]
+    - A JSON object with "utterances" key: {"utterances": ["a", "b"]}
+    """
+    with open(path, encoding="utf-8") as f:
+        data = json.load(f)
+    if isinstance(data, list):
+        return [str(s) for s in data]
+    if isinstance(data, dict) and "utterances" in data:
+        return [str(s) for s in data["utterances"]]
+    raise ValueError(
+        f"JSON file must be an array of strings or an object with 'utterances' key. "
+        f"Got: {type(data).__name__}"
+    )
+
+
+def parse_args():
+    """Parse command-line arguments."""
+    script_dir = Path(__file__).parent
+    default_output = script_dir.parent / "data" / "audio_examples" / "two_utterances_16k.wav"
+    
+    parser = argparse.ArgumentParser(description="Generate test audio with SiliconFlow TTS (utterances + silence).")
+    parser.add_argument(
+        "-o", "--output",
+        type=Path,
+        default=default_output,
+        help=f"Output WAV file path (default: {default_output})"
+    )
+    parser.add_argument(
+        "-u", "--utterance",
+        action="append",
+        dest="utterances",
+        metavar="TEXT",
+        help="Utterance text (repeat for multiple). Ignored if --json is set."
+    )
+    parser.add_argument(
+        "-j", "--json",
+        type=Path,
+        metavar="PATH",
+        help="JSON file with utterances: array of strings or object with 'utterances' key"
+    )
+    parser.add_argument(
+        "--silence-ms",
+        type=int,
+        default=500,
+        metavar="MS",
+        help="Silence in ms between utterances (default: 500)"
+    )
+    parser.add_argument(
+        "--lead-silence-ms",
+        type=int,
+        default=200,
+        metavar="MS",
+        help="Silence in ms at start of file (default: 200)"
+    )
+    parser.add_argument(
+        "--trail-silence-ms",
+        type=int,
+        default=300,
+        metavar="MS",
+        help="Silence in ms at end of file (default: 300)"
+    )
+    return parser.parse_args()
+
+
+async def main():
+    """Main entry point."""
+    args = parse_args()
+    output_path = args.output
+    output_path.parent.mkdir(parents=True, exist_ok=True)
+    
+    # Resolve utterances: JSON file > -u args > defaults
+    if args.json is not None:
+        if not args.json.is_file():
+            raise FileNotFoundError(f"Utterances JSON file not found: {args.json}")
+        utterances = load_utterances_from_json(args.json)
+        if not utterances:
+            raise ValueError(f"JSON file has no utterances: {args.json}")
+    elif args.utterances:
+        utterances = args.utterances
+    else:
+        utterances = [
+            "Hello, how are you doing today?",
+            "I'm doing great, thank you for asking!"
+        ]
+    
+    await generate_test_audio(
+        output_path=str(output_path),
+        utterances=utterances,
+        silence_ms=args.silence_ms,
+        lead_silence_ms=args.lead_silence_ms,
+        trail_silence_ms=args.trail_silence_ms,
+        voice="anna",
+        sample_rate=16000,
+        speed=1.0
+    )
+
+
+if __name__ == "__main__":
+    asyncio.run(main())