server run utility
Browse files- app.py +15 -128
- requirements.txt +3 -2
- server.py +131 -0
app.py
CHANGED
@@ -1,133 +1,20 @@
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from fastapi.staticfiles import StaticFiles
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from fastapi.responses import HTMLResponse
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from fastapi.templating import Jinja2Templates
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import os
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import
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import torch
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from transformers.pipelines.audio_utils import ffmpeg_microphone_live
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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classifier = pipeline(
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"audio-classification", model="MIT/ast-finetuned-speech-commands-v2", device=device
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)
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intent_class_pipe = pipeline(
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"audio-classification", model="anton-l/xtreme_s_xlsr_minds14", device=device
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)
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async def launch_fn(
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wake_word="marvin",
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prob_threshold=0.5,
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chunk_length_s=2.0,
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stream_chunk_s=0.25,
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debug=False,
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):
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if wake_word not in classifier.model.config.label2id.keys():
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raise ValueError(
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f"Wake word {wake_word} not in set of valid class labels, pick a wake word in the set {classifier.model.config.label2id.keys()}."
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)
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sampling_rate = classifier.feature_extractor.sampling_rate
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mic = ffmpeg_microphone_live(
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sampling_rate=sampling_rate,
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chunk_length_s=chunk_length_s,
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stream_chunk_s=stream_chunk_s,
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)
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print("Listening for wake word...")
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for prediction in classifier(mic):
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prediction = prediction[0]
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if debug:
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print(prediction)
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if prediction["label"] == wake_word:
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if prediction["score"] > prob_threshold:
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return True
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async def listen(websocket, chunk_length_s=2.0, stream_chunk_s=2.0):
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sampling_rate = intent_class_pipe.feature_extractor.sampling_rate
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mic = ffmpeg_microphone_live(
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sampling_rate=sampling_rate,
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chunk_length_s=chunk_length_s,
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stream_chunk_s=stream_chunk_s,
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)
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audio_buffer = []
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audio_buffer.append(audio_chunk["raw"])
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prediction = intent_class_pipe(audio_chunk["raw"])
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await websocket.send_text(f"chunk: {prediction[0]['label']} | {i+1} / 4")
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if await is_silence(audio_chunk["raw"], threshold=0.7):
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print("Silence detected, processing audio.")
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break
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combined_audio = np.concatenate(audio_buffer)
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prediction = intent_class_pipe(combined_audio)
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top_3_predictions = prediction[:3]
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formatted_predictions = "\n".join([f"{pred['label']}: {pred['score'] * 100:.2f}%" for pred in top_3_predictions])
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await websocket.send_text(f"classes: \n{formatted_predictions}")
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return
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return True
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else:
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app = FastAPI()
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# Set up static file directory
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app.mount("/static", StaticFiles(directory="static"), name="static")
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# Jinja2 Template for HTML rendering
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templates = Jinja2Templates(directory="templates")
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@app.get("/", response_class=HTMLResponse)
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async def get_home(request: Request):
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return templates.TemplateResponse("index.html", {"request": request})
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@app.websocket("/ws")
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async def websocket_endpoint(websocket: WebSocket):
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await websocket.accept()
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try:
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process_active = False # Flag to track the state of the process
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while True:
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message = await websocket.receive_text()
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if message == "start" and not process_active:
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process_active = True
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await websocket.send_text("Listening for wake word...")
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wake_word_detected = await launch_fn(debug=True)
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if wake_word_detected:
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await websocket.send_text("Wake word detected. Listening for your query...")
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await listen(websocket)
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process_active = False # Reset the process flag
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elif message == "stop":
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if process_active:
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# Implement logic to stop the ongoing process
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# This might involve setting a flag that your launch_fn and listen functions check
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process_active = False
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await websocket.send_text("Process stopped. Ready to restart.")
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break # Or keep the loop running if you want to allow restarting without reconnecting
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except WebSocketDisconnect:
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print("Client disconnected.")
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import json
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import os
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import requests
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import socket
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def start_server():
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os.system("uvicorn server:app --port 8080 --host 0.0.0.0 --workers 2")
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def is_port_in_use(port):
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with socket.socket(socket.AF_INET, socket.SOCK_STREAM) as s:
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return s.connect_ex(('0.0.0.0', port)) == 0
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def main():
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if is_port_in_use(8080):
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print("Port 8080 is already in use. Please kill the process and try again.")
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else:
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start_server()
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if __name__ == "__main__":
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main()
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requirements.txt
CHANGED
@@ -3,5 +3,6 @@ transformers
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torchaudio
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numpy
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fastapi
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uvicorn
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gradio
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torchaudio
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numpy
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fastapi
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uvicorn
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gradio
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requests
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server.py
ADDED
@@ -0,0 +1,131 @@
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from fastapi import FastAPI, WebSocket, Request, WebSocketDisconnect
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from fastapi.staticfiles import StaticFiles
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from fastapi.responses import HTMLResponse
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from fastapi.templating import Jinja2Templates
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import numpy as np
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from transformers import pipeline
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import torch
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from transformers.pipelines.audio_utils import ffmpeg_microphone_live
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device = "cuda:0" if torch.cuda.is_available() else "cpu"
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classifier = pipeline(
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"audio-classification", model="MIT/ast-finetuned-speech-commands-v2", device=device
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)
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intent_class_pipe = pipeline(
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"audio-classification", model="anton-l/xtreme_s_xlsr_minds14", device=device
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)
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async def launch_fn(
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wake_word="marvin",
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prob_threshold=0.5,
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chunk_length_s=2.0,
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stream_chunk_s=0.25,
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debug=False,
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):
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if wake_word not in classifier.model.config.label2id.keys():
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raise ValueError(
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f"Wake word {wake_word} not in set of valid class labels, pick a wake word in the set {classifier.model.config.label2id.keys()}."
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)
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sampling_rate = classifier.feature_extractor.sampling_rate
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mic = ffmpeg_microphone_live(
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sampling_rate=sampling_rate,
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chunk_length_s=chunk_length_s,
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stream_chunk_s=stream_chunk_s,
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)
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print("Listening for wake word...")
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for prediction in classifier(mic):
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prediction = prediction[0]
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if debug:
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print(prediction)
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if prediction["label"] == wake_word:
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if prediction["score"] > prob_threshold:
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return True
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async def listen(websocket, chunk_length_s=2.0, stream_chunk_s=2.0):
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sampling_rate = intent_class_pipe.feature_extractor.sampling_rate
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mic = ffmpeg_microphone_live(
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sampling_rate=sampling_rate,
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chunk_length_s=chunk_length_s,
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stream_chunk_s=stream_chunk_s,
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)
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audio_buffer = []
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print("Listening")
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for i in range(4):
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audio_chunk = next(mic)
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audio_buffer.append(audio_chunk["raw"])
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prediction = intent_class_pipe(audio_chunk["raw"])
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await websocket.send_text(f"chunk: {prediction[0]['label']} | {i+1} / 4")
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if await is_silence(audio_chunk["raw"], threshold=0.7):
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print("Silence detected, processing audio.")
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break
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combined_audio = np.concatenate(audio_buffer)
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prediction = intent_class_pipe(combined_audio)
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top_3_predictions = prediction[:3]
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formatted_predictions = "\n".join([f"{pred['label']}: {pred['score'] * 100:.2f}%" for pred in top_3_predictions])
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await websocket.send_text(f"classes: \n{formatted_predictions}")
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return
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async def is_silence(audio_chunk, threshold):
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silence = intent_class_pipe(audio_chunk)
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if silence[0]["label"] == "silence" and silence[0]["score"] > threshold:
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return True
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else:
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return False
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# Initialize FastAPI app
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app = FastAPI()
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# Set up static file directory
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app.mount("/static", StaticFiles(directory="static"), name="static")
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# Jinja2 Template for HTML rendering
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templates = Jinja2Templates(directory="templates")
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@app.get("/", response_class=HTMLResponse)
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async def get_home(request: Request):
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return templates.TemplateResponse("index.html", {"request": request})
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@app.websocket("/ws")
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async def websocket_endpoint(websocket: WebSocket):
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await websocket.accept()
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try:
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process_active = False # Flag to track the state of the process
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while True:
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message = await websocket.receive_text()
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if message == "start" and not process_active:
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process_active = True
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await websocket.send_text("Listening for wake word...")
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wake_word_detected = await launch_fn(debug=True)
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if wake_word_detected:
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await websocket.send_text("Wake word detected. Listening for your query...")
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await listen(websocket)
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process_active = False # Reset the process flag
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elif message == "stop":
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if process_active:
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# Implement logic to stop the ongoing process
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# This might involve setting a flag that your launch_fn and listen functions check
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process_active = False
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await websocket.send_text("Process stopped. Ready to restart.")
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break # Or keep the loop running if you want to allow restarting without reconnecting
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except WebSocketDisconnect:
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print("Client disconnected.")
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