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import gradio as gr
import asyncio
import edge_tts
import os
from huggingface_hub import InferenceClient
import requests
import tempfile
import logging
import io
from pydub import AudioSegment
# Set up logging
logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(levelname)s - %(message)s')
# Get the Hugging Face token from environment variable
hf_token = os.getenv("HF_TOKEN")
if not hf_token:
raise ValueError("HF_TOKEN environment variable is not set")
# Initialize the Hugging Face Inference Client for chat completion
chat_client = InferenceClient("mistralai/Mistral-Nemo-Instruct-2407", token=hf_token)
# Whisper API settings
WHISPER_API_URL = "https://api-inference.huggingface.co/models/openai/whisper-large-v3-turbo"
headers = {"Authorization": f"Bearer {hf_token}"}
# Initialize an empty chat history
chat_history = []
async def text_to_speech_stream(text, voice_volume=1.0):
"""Convert text to speech using edge_tts and return the audio file path."""
communicate = edge_tts.Communicate(text, "en-US-BrianMultilingualNeural")
audio_data = b""
async for chunk in communicate.stream():
if chunk["type"] == "audio":
audio_data += chunk["data"]
# Adjust volume
audio = AudioSegment.from_mp3(io.BytesIO(audio_data))
adjusted_audio = audio + (20 * voice_volume - 20) # Adjust volume (0.0 to 2.0)
with tempfile.NamedTemporaryFile(delete=False, suffix=".mp3") as temp_file:
adjusted_audio.export(temp_file.name, format="mp3")
return temp_file.name
def whisper_speech_to_text(audio_path):
"""Convert speech to text using Hugging Face Whisper API."""
if audio_path is None:
logging.error("Error: No audio file provided")
return ""
if not os.path.exists(audio_path):
logging.error(f"Error: Audio file not found at {audio_path}")
return ""
try:
with open(audio_path, "rb") as audio_file:
data = audio_file.read()
response = requests.post(WHISPER_API_URL, headers=headers, data=data)
response.raise_for_status() # Raise an exception for bad status codes
result = response.json()
transcribed_text = result.get("text", "")
logging.info(f"Transcribed text: {transcribed_text}")
return transcribed_text
except requests.exceptions.RequestException as e:
logging.error(f"Error during API request: {e}")
return ""
except Exception as e:
logging.error(f"Unexpected error in whisper_speech_to_text: {e}")
return ""
async def chat_with_ai(message):
global chat_history
chat_history.append({"role": "user", "content": message})
try:
response = chat_client.chat_completion(
messages=[{"role": "system", "content": "You are a helpful voice assistant. Provide concise and clear responses to user queries."}] + chat_history,
max_tokens=800,
temperature=0.7
)
response_text = response.choices[0].message['content']
chat_history.append({"role": "assistant", "content": response_text})
audio_path = await text_to_speech_stream(response_text)
return response_text, audio_path
except Exception as e:
logging.error(f"Error in chat_with_ai: {e}")
return str(e), None
def transcribe_and_chat(audio):
if audio is None:
return "Sorry, no audio was provided. Please try recording again.", None
text = whisper_speech_to_text(audio)
if not text:
return "Sorry, I couldn't understand the audio or there was an error in transcription. Please try again.", None
response, audio_path = asyncio.run(chat_with_ai(text))
return response, audio_path
def create_demo():
with gr.Blocks() as demo:
gr.Markdown(
"""
# π£οΈ AI Voice Assistant
Welcome to your personal voice assistant! Simply record your voice, and I will respond with both text and speech. Powered by advanced AI models.
"""
)
with gr.Row():
with gr.Column(scale=1):
audio_input = gr.Audio(type="filepath", label="π€ Record your voice", elem_id="audio-input")
clear_button = gr.Button("Clear", variant="secondary", elem_id="clear-button")
voice_volume = gr.Slider(minimum=0, maximum=2, value=1, step=0.1, label="Voice Volume", elem_id="voice-volume")
with gr.Column(scale=1):
chat_output = gr.Textbox(label="π¬ AI Response", elem_id="chat-output", lines=5, interactive=False)
audio_output = gr.Audio(label="π AI Voice Response", autoplay=True, elem_id="audio-output")
# Add some spacing and a divider
gr.Markdown("---")
# Processing the audio input
def process_audio(audio, volume):
logging.info(f"Received audio: {audio}")
if audio is None:
return "No audio detected. Please try recording again.", None, None
response, audio_path = transcribe_and_chat(audio)
# Adjust volume for the response audio
adjusted_audio_path = asyncio.run(text_to_speech_stream(response, volume))
logging.info(f"Response: {response}, Audio path: {adjusted_audio_path}")
return response, adjusted_audio_path, None # Return None to clear the audio input
audio_input.change(process_audio, inputs=[audio_input, voice_volume], outputs=[chat_output, audio_output, audio_input])
clear_button.click(lambda: (None, None, None), None, [chat_output, audio_output, audio_input])
# JavaScript to handle autoplay and automatic submission
demo.load(None, js="""
function() {
document.querySelector("audio").addEventListener("stop", function() {
setTimeout(function() {
document.querySelector('button[title="Submit"]').click();
}, 500);
});
function playAssistantAudio() {
var audioElements = document.querySelectorAll('audio');
if (audioElements.length > 1) {
var assistantAudio = audioElements[1];
if (assistantAudio) {
assistantAudio.play();
}
}
}
document.addEventListener('gradioAudioLoaded', function(event) {
playAssistantAudio();
});
document.addEventListener('gradioUpdated', function(event) {
setTimeout(playAssistantAudio, 100);
});
// Prevent audio from stopping when switching tabs
document.addEventListener("visibilitychange", function() {
var audioElements = document.querySelectorAll('audio');
audioElements.forEach(function(audio) {
audio.play();
});
});
}
""")
return demo
# Launch the Gradio app
if __name__ == "__main__":
demo = create_demo()
demo.launch(server_name="0.0.0.0", server_port=7860) |