Server API and Webpage update
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@@ -9,7 +9,9 @@ import io
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import whisperx
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from io import BytesIO
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from typing import List, Dict, Any, Optional
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from fastapi import FastAPI, WebSocket, WebSocketDisconnect
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from fastapi import FastAPI, WebSocket, WebSocketDisconnect, Request
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from fastapi.responses import HTMLResponse, FileResponse
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from fastapi.staticfiles import StaticFiles
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from fastapi.middleware.cors import CORSMiddleware
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from pydantic import BaseModel
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from generator import load_csm_1b, Segment
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@@ -17,6 +19,8 @@ import uvicorn
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import time
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import gc
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from collections import deque
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import socket
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import requests
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# Select device
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if torch.cuda.is_available():
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@@ -45,6 +49,32 @@ app.add_middleware(
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allow_headers=["*"],
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)
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# Define the base directory
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base_dir = os.path.dirname(os.path.abspath(__file__))
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# Mount a static files directory if you have any static assets like CSS or JS
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static_dir = os.path.join(base_dir, "static")
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os.makedirs(static_dir, exist_ok=True) # Create the directory if it doesn't exist
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app.mount("/static", StaticFiles(directory=static_dir), name="static")
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# Define route to serve index.html as the main page
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@app.get("/", response_class=HTMLResponse)
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async def get_index():
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try:
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with open(os.path.join(base_dir, "index.html"), "r") as f:
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return HTMLResponse(content=f.read())
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except FileNotFoundError:
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return HTMLResponse(content="<html><body><h1>Error: index.html not found</h1></body></html>")
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# Add a favicon endpoint (optional, but good to have)
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@app.get("/favicon.ico")
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async def get_favicon():
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favicon_path = os.path.join(static_dir, "favicon.ico")
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if os.path.exists(favicon_path):
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return FileResponse(favicon_path)
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else:
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return HTMLResponse(status_code=204) # No content
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# Connection manager to handle multiple clients
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class ConnectionManager:
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def __init__(self):
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@@ -259,6 +289,7 @@ async def websocket_endpoint(websocket: WebSocket):
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energy_window.clear()
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is_silence = False
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last_active_time = time.time()
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print(f"Streaming started with speaker ID: {speaker_id}")
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await websocket.send_json({
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"type": "streaming_status",
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"status": "started"
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@@ -269,6 +300,13 @@ async def websocket_endpoint(websocket: WebSocket):
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energy_window.append(chunk_energy)
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avg_energy = sum(energy_window) / len(energy_window)
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# Debug audio levels
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if len(energy_window) >= 5: # Only start printing after we have enough samples
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if avg_energy > SILENCE_THRESHOLD:
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print(f"[AUDIO] Active sound detected - Energy: {avg_energy:.6f} (threshold: {SILENCE_THRESHOLD})")
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else:
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print(f"[AUDIO] Silence detected - Energy: {avg_energy:.6f} (threshold: {SILENCE_THRESHOLD})")
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# Check if audio is silent
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current_silence = avg_energy < SILENCE_THRESHOLD
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@@ -277,33 +315,53 @@ async def websocket_endpoint(websocket: WebSocket):
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# Transition to silence
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is_silence = True
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last_active_time = time.time()
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print("[STREAM] Transition to silence detected")
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elif is_silence and not current_silence:
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# User started talking again
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is_silence = False
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print("[STREAM] User resumed speaking")
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# Add chunk to buffer regardless of silence state
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streaming_buffer.append(audio_chunk)
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# Debug buffer size periodically
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if len(streaming_buffer) % 10 == 0:
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print(f"[BUFFER] Current size: {len(streaming_buffer)} chunks, ~{len(streaming_buffer)/5:.1f} seconds")
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# Check if silence has persisted long enough to consider "stopped talking"
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silence_elapsed = time.time() - last_active_time
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if is_silence and silence_elapsed >= SILENCE_DURATION_SEC and len(streaming_buffer) > 0:
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# User has stopped talking - process the collected audio
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print(f"[STREAM] Processing audio after {silence_elapsed:.2f}s of silence")
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print(f"[STREAM] Processing {len(streaming_buffer)} audio chunks (~{len(streaming_buffer)/5:.1f} seconds)")
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full_audio = torch.cat(streaming_buffer, dim=0)
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# Log audio statistics
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audio_duration = len(full_audio) / generator.sample_rate
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audio_min = torch.min(full_audio).item()
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audio_max = torch.max(full_audio).item()
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audio_mean = torch.mean(full_audio).item()
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print(f"[AUDIO] Processed audio - Duration: {audio_duration:.2f}s, Min: {audio_min:.4f}, Max: {audio_max:.4f}, Mean: {audio_mean:.4f}")
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# Process with WhisperX speech-to-text
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print("[ASR] Starting transcription with WhisperX...")
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transcribed_text = await transcribe_audio(full_audio)
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# Log the transcription
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print(f"Transcribed text: '{transcribed_text}'")
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print(f"[ASR] Transcribed text: '{transcribed_text}'")
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# Add to conversation context
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if transcribed_text:
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print(f"[DIALOG] Adding user utterance to context: '{transcribed_text}'")
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user_segment = Segment(text=transcribed_text, speaker=speaker_id, audio=full_audio)
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context_segments.append(user_segment)
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# Generate a contextual response
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print("[DIALOG] Generating response...")
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response_text = await generate_response(transcribed_text, context_segments)
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print(f"[DIALOG] Response text: '{response_text}'")
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# Send the transcribed text to client
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await websocket.send_json({
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@@ -312,12 +370,14 @@ async def websocket_endpoint(websocket: WebSocket):
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})
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# Generate audio for the response
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print("[TTS] Generating speech for response...")
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audio_tensor = generator.generate(
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text=response_text,
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speaker=1 if speaker_id == 0 else 0, # Use opposite speaker
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context=context_segments,
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max_audio_length_ms=10_000,
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)
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print(f"[TTS] Generated audio length: {len(audio_tensor)/generator.sample_rate:.2f}s")
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# Add response to context
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ai_segment = Segment(
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@@ -326,15 +386,18 @@ async def websocket_endpoint(websocket: WebSocket):
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audio=audio_tensor
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)
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context_segments.append(ai_segment)
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print(f"[DIALOG] Context now has {len(context_segments)} segments")
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# Convert audio to base64 and send back to client
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audio_base64 = await encode_audio_data(audio_tensor)
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print("[STREAM] Sending audio response to client")
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await websocket.send_json({
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"type": "audio_response",
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"text": response_text,
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"audio": audio_base64
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})
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else:
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print("[ASR] Transcription failed or returned empty text")
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# If transcription failed, send a generic response
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await websocket.send_json({
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"type": "error",
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@@ -346,17 +409,20 @@ async def websocket_endpoint(websocket: WebSocket):
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energy_window.clear()
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is_silence = False
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last_active_time = time.time()
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print("[STREAM] Buffer cleared, ready for next utterance")
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# If buffer gets too large without silence, process it anyway
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# This prevents memory issues with very long streams
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elif len(streaming_buffer) >= 30: # ~6 seconds of audio at 5 chunks/sec
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print("Buffer limit reached, processing audio")
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print("[BUFFER] Maximum buffer size reached, processing audio")
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full_audio = torch.cat(streaming_buffer, dim=0)
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# Process with WhisperX speech-to-text
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print("[ASR] Starting forced transcription of long audio...")
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transcribed_text = await transcribe_audio(full_audio)
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if transcribed_text:
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print(f"[ASR] Transcribed long audio: '{transcribed_text}'")
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context_segments.append(Segment(text=transcribed_text, speaker=speaker_id, audio=full_audio))
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# Send the transcribed text to client
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@@ -364,11 +430,17 @@ async def websocket_endpoint(websocket: WebSocket):
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"type": "transcription",
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"text": transcribed_text + " (processing continued speech...)"
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})
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else:
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print("[ASR] No transcription from long audio")
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streaming_buffer = []
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print("[BUFFER] Buffer cleared due to size limit")
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except Exception as e:
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print(f"Error processing streaming audio: {str(e)}")
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print(f"[ERROR] Processing streaming audio: {str(e)}")
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# Print traceback for more detailed error information
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import traceback
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traceback.print_exc()
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await websocket.send_json({
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"type": "error",
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"message": f"Error processing streaming audio: {str(e)}"
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@@ -412,6 +484,53 @@ async def websocket_endpoint(websocket: WebSocket):
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pass
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manager.disconnect(websocket)
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# Add this function to get the public IP address
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def get_public_ip():
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"""Get the server's public IP address using an external service"""
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try:
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# Try multiple services in case one is down
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services = [
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"https://api.ipify.org",
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"https://ifconfig.me/ip",
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"https://checkip.amazonaws.com",
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]
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for service in services:
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try:
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response = requests.get(service, timeout=3)
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if response.status_code == 200:
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return response.text.strip()
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except:
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continue
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# Fallback to socket if external services fail
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s = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
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try:
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# Doesn't need to be reachable, just used to determine interface
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s.connect(('8.8.8.8', 1))
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local_ip = s.getsockname()[0]
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return local_ip
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except:
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return "localhost"
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finally:
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s.close()
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except:
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return "Could not determine IP address"
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# Update the __main__ block
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if __name__ == "__main__":
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public_ip = get_public_ip()
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print(f"\n{'='*50}")
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print(f"💬 Sesame AI Voice Chat Server")
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print(f"{'='*50}")
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print(f"📡 Server Information:")
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print(f" - Public IP: http://{public_ip}:8000")
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print(f" - Local URL: http://localhost:8000")
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print(f" - WebSocket: ws://{public_ip}:8000/ws")
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print(f"{'='*50}")
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print(f"🌐 Connect from web browsers using: http://{public_ip}:8000")
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print(f"🔧 Serving index.html from: {os.path.join(base_dir, 'index.html')}")
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print(f"{'='*50}\n")
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# Start the server
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uvicorn.run(app, host="0.0.0.0", port=8000)
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