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Update src/streamlit_app.py
Browse files- src/streamlit_app.py +201 -38
src/streamlit_app.py
CHANGED
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@@ -1,40 +1,203 @@
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import altair as alt
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import numpy as np
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import pandas as pd
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import streamlit as st
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""
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If you have any questions, checkout our [documentation](https://docs.streamlit.io) and [community
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forums](https://discuss.streamlit.io).
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In the meantime, below is an example of what you can do with just a few lines of code:
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"""
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num_points = st.slider("Number of points in spiral", 1, 10000, 1100)
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num_turns = st.slider("Number of turns in spiral", 1, 300, 31)
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indices = np.linspace(0, 1, num_points)
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theta = 2 * np.pi * num_turns * indices
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radius = indices
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x = radius * np.cos(theta)
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y = radius * np.sin(theta)
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df = pd.DataFrame({
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"x": x,
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"y": y,
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"idx": indices,
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"rand": np.random.randn(num_points),
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})
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st.altair_chart(alt.Chart(df, height=700, width=700)
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.mark_point(filled=True)
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.encode(
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x=alt.X("x", axis=None),
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y=alt.Y("y", axis=None),
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color=alt.Color("idx", legend=None, scale=alt.Scale()),
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size=alt.Size("rand", legend=None, scale=alt.Scale(range=[1, 150])),
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))
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import streamlit as st
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import torch
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from transformers import (
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pipeline,
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AutoTokenizer,
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AutoModelForCausalLM,
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)
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import numpy as np
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import time
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# --- 0. Streamlit App Configuration ---
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# Set the page configuration for a cleaner look
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st.set_page_config(
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page_title="Nova Voice Chat (Streamlit)",
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layout="centered",
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initial_sidebar_state="collapsed"
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)
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# A custom component is needed for microphone recording in Streamlit
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try:
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from st_audiorec import st_audiorec
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except ImportError:
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st.error("Please install the st-audiorec component: `pip install st-audiorec`")
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st.stop()
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# --- 1. Global Model Loading (Cached) ---
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@st.cache_resource
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def load_models():
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"""Loads all models and pipes, cached globally."""
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with st.spinner("Loading AI models..."):
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print("Loading models...")
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device = "cuda" if torch.cuda.is_available() else "cpu"
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# 1. Speech-to-Text (STT)
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stt_pipe = pipeline("automatic-speech-recognition", model="openai/whisper-tiny.en", device=device)
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# 2. Large Language Model (LLM)
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model_name = "Qwen/Qwen2-0.5B-Instruct"
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tokenizer = AutoTokenizer.from_pretrained(model_name)
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if tokenizer.pad_token is None:
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tokenizer.pad_token = tokenizer.eos_token
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llm_model = AutoModelForCausalLM.from_pretrained(
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model_name,
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torch_dtype=torch.float16 if device == "cuda" else torch.float32,
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device_map="auto"
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)
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# 3. Text-to-Speech (TTS)
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tts_pipe = pipeline("text-to-speech", model="facebook/mms-tts-eng", device=device)
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print("Models loaded.")
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st.success("Models loaded!")
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return stt_pipe, tokenizer, llm_model, tts_pipe
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# Load the models once
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try:
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STT_PIPE, TOKENIZER, LLM_MODEL, TTS_PIPE = load_models()
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except Exception as e:
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st.error(f"Failed to load models. Please check your hardware and dependencies. Error: {e}")
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st.stop()
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# --- 2. State Initialization and Functions ---
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def get_initial_chat_history():
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"""Returns the initial chat history structure."""
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return [
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{
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"role": "system",
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"content": "You are Nova, an AI assistant. You are friendly and helpful. Respond naturally, as if in a conversation."
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}
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]
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# Initialize session state for chat history
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if 'chat_history' not in st.session_state:
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st.session_state.chat_history = get_initial_chat_history()
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# Placeholder for the status text
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if 'status_text' not in st.session_state:
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st.session_state.status_text = "I'm listening..."
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# Placeholder for the audio playback element
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if 'audio_to_play' not in st.session_state:
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st.session_state.audio_to_play = None
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def process_audio_file(wav_audio_data):
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"""
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Handles the entire voice interaction flow: STT -> LLM -> TTS.
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This function is called when a recording is finished.
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"""
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if wav_audio_data is None:
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st.session_state.status_text = "Didn't catch that. Try again."
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st.session_state.audio_to_play = None
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st.rerun() # Rerun to update status
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return
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st.session_state.status_text = "Thinking..."
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st.rerun()
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try:
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# Save the audio data to a temporary file for the STT pipe
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# The Gradio version received a file path, st_audiorec gives raw bytes.
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import tempfile
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with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmp_file:
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tmp_file.write(wav_audio_data)
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audio_filepath = tmp_file.name
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# 1. Speech-to-Text (STT)
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result = STT_PIPE(audio_filepath)
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transcript = result["text"].strip() if result and result["text"] else ""
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if not transcript:
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st.session_state.status_text = "I couldn't hear anything clearly."
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st.session_state.audio_to_play = None
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st.rerun()
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return
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# 2. LLM Inference
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st.session_state.chat_history.append({"role": "user", "content": transcript})
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# Manage context length (keep system prompt + last 9 exchanges)
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if len(st.session_state.chat_history) > 10:
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st.session_state.chat_history = [st.session_state.chat_history[0]] + st.session_state.chat_history[-9:]
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text = TOKENIZER.apply_chat_template(st.session_state.chat_history, tokenize=False, add_generation_prompt=True)
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model_inputs = TOKENIZER([text], return_tensors="pt").to(LLM_MODEL.device)
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with torch.no_grad():
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generated_ids = LLM_MODEL.generate(**model_inputs, max_new_tokens=256, pad_token_id=TOKENIZER.eos_token_id)
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response_ids = generated_ids[0][len(model_inputs.input_ids[0]):]
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response = TOKENIZER.decode(response_ids, skip_special_tokens=True).strip()
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st.session_state.chat_history.append({"role": "assistant", "content": response})
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# 3. Text-to-Speech (TTS)
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st.session_state.status_text = "Responding..."
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st.rerun()
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speech = TTS_PIPE(response)
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# Save audio data and trigger playback
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audio_data = (speech["sampling_rate"], speech["audio"].astype(np.float32))
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st.session_state.audio_to_play = audio_data
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st.session_state.status_text = "I'm listening..."
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st.rerun()
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except Exception as e:
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print(f"Error in process_audio: {e}")
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st.session_state.status_text = "An error occurred."
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st.session_state.audio_to_play = None
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st.rerun()
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# --- 3. Streamlit Interface ---
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st.title("Nova Voice Chat 🎤")
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st.markdown("---")
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# The custom microphone recording component
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wav_audio_data = st_audiorec()
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# When a recording is completed, the component returns the audio data,
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# which triggers the processing function.
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if wav_audio_data is not None:
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process_audio_file(wav_audio_data)
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# Status Text Display
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st.markdown(
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f'<h2 style="text-align: center; color: #1A73E8;">{st.session_state.status_text}</h2>',
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unsafe_allow_html=True
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)
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st.markdown(
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'<p style="text-align: center; color: #8C8C8C; font-size: 14px;">Qwen2-0.5B-Instruct</p>',
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unsafe_allow_html=True
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)
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# Audio Playback
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# Display the audio player only when there's new audio to play
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if st.session_state.audio_to_play is not None:
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sampling_rate, audio_array = st.session_state.audio_to_play
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st.audio(audio_array, sample_rate=sampling_rate)
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# Reset the audio state after playback starts (or immediately, as Streamlit reruns)
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st.session_state.audio_to_play = None
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# Optional: Display chat history in the sidebar
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with st.sidebar:
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st.subheader("Chat History")
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# Display the conversation (excluding the system prompt)
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for message in st.session_state.chat_history[1:]:
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with st.chat_message(message["role"]):
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st.write(message["content"])
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if st.button("Reset Chat"):
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st.session_state.chat_history = get_initial_chat_history()
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st.session_state.status_text = "Chat reset. I'm listening..."
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st.rerun()
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