project_charles / app.py
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refactor - use app.py
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import asyncio
from collections import deque
import os
import threading
import time
import traceback
import av
import numpy as np
import streamlit as st
from streamlit_webrtc import WebRtcMode, webrtc_streamer
import pydub
import torch
# import av
# import cv2
from sample_utils.turn import get_ice_servers
import json
from typing import List
from vosk import SetLogLevel, Model, KaldiRecognizer
SetLogLevel(-1) # mutes vosk verbosity
from dotenv import load_dotenv
load_dotenv()
webrtc_ctx = None
# Initialize Ray
import ray
if not ray.is_initialized():
# Try to connect to a running Ray cluster
ray_address = os.getenv('RAY_ADDRESS')
if ray_address:
ray.init(ray_address, namespace="project_charles")
else:
ray.init(namespace="project_charles")
async def main():
system_one_audio_status = st.empty()
playing = st.checkbox("Playing", value=True)
system_one_audio_status.write("Initializing streaming")
system_one_audio_output = st.empty()
system_one_video_output = st.empty()
system_one_audio_history = []
system_one_audio_history_output = st.empty()
# Initialize resources if not already done
system_one_audio_status.write("Initializing streaming")
if "streamlit_av_queue" not in st.session_state:
from streamlit_av_queue import StreamlitAVQueue
st.session_state.streamlit_av_queue = StreamlitAVQueue()
system_one_audio_status.write("resources referecned")
system_one_audio_status.write("Initializing webrtc_streamer")
webrtc_ctx = webrtc_streamer(
key="charles",
desired_playing_state=playing,
queued_audio_frames_callback=st.session_state.streamlit_av_queue.queued_audio_frames_callback,
queued_video_frames_callback=st.session_state.streamlit_av_queue.queued_video_frames_callback,
mode=WebRtcMode.SENDRECV,
media_stream_constraints={
"video": True,
"audio": {
"sampleRate": 48000,
"sampleSize": 16,
"noiseSuppression": True,
"echoCancellation": True,
"channelCount": 1,
}
},
rtc_configuration={"iceServers": get_ice_servers()},
async_processing=True,
)
if not webrtc_ctx.state.playing:
exit
system_one_audio_status.write("Initializing speech")
from charles_actor import CharlesActor
charles_actor = None
try:
while True:
if not webrtc_ctx.state.playing:
system_one_audio_status.write("Stopped.")
await asyncio.sleep(0.1)
continue
if charles_actor is None:
try:
charles_actor = ray.get_actor("CharlesActor")
system_one_audio_status.write("Charles is here.")
except ValueError as e:
system_one_audio_status.write("Charles is sleeping.")
pass
if charles_actor is not None:
try:
audio_history = await charles_actor.get_system_one_audio_history_output.remote()
system_one_audio_history_output.markdown(audio_history)
except Exception as e:
# assume we disconnected
charles_actor = None
await asyncio.sleep(0.1)
except Exception as e:
print(f"An error occurred: {e}")
traceback.print_exc()
raise e
if __name__ == "__main__":
try:
asyncio.run(main())
except Exception as e:
if webrtc_ctx is not None:
del webrtc_ctx
webrtc_ctx = None
if "streamlit_av_queue" in st.session_state:
del st.session_state.streamlit_av_queue
finally:
pass