Files
HooHacks-12/Backend/server.py
2025-03-30 07:18:14 -04:00

256 lines
8.6 KiB
Python

import os
import io
import base64
import time
import torch
import torchaudio
import numpy as np
from flask import Flask, render_template, request
from flask_socketio import SocketIO, emit
from transformers import AutoModelForCausalLM, AutoTokenizer
from faster_whisper import WhisperModel
from generator import load_csm_1b, Segment
from collections import deque
app = Flask(__name__)
app.config['SECRET_KEY'] = 'your-secret-key'
socketio = SocketIO(app, cors_allowed_origins="*")
# Select the best available device
if torch.cuda.is_available():
device = "cuda"
whisper_compute_type = "float16"
elif torch.backends.mps.is_available():
device = "mps"
whisper_compute_type = "float32"
else:
device = "cpu"
whisper_compute_type = "int8"
print(f"Using device: {device}")
# Initialize Faster-Whisper for transcription
print("Loading Whisper model...")
whisper_model = WhisperModel("base", device=device, compute_type=whisper_compute_type)
# Initialize CSM model for audio generation
print("Loading CSM model...")
csm_generator = load_csm_1b(device=device)
# Initialize Llama 3.2 model for response generation
print("Loading Llama 3.2 model...")
llm_model_id = "meta-llama/Llama-3.2-1B" # Choose appropriate size based on resources
llm_tokenizer = AutoTokenizer.from_pretrained(llm_model_id)
llm_model = AutoModelForCausalLM.from_pretrained(
llm_model_id,
torch_dtype=torch.bfloat16,
device_map=device
)
# Store conversation context
conversation_context = {} # session_id -> context
@app.route('/')
def index():
return render_template('index.html')
@socketio.on('connect')
def handle_connect():
print(f"Client connected: {request.sid}")
conversation_context[request.sid] = {
'segments': [],
'speakers': [0, 1], # 0 = user, 1 = bot
'audio_buffer': deque(maxlen=10), # Store recent audio chunks
'is_speaking': False,
'silence_start': None
}
emit('ready', {'message': 'Connection established'})
@socketio.on('disconnect')
def handle_disconnect():
print(f"Client disconnected: {request.sid}")
if request.sid in conversation_context:
del conversation_context[request.sid]
@socketio.on('start_speaking')
def handle_start_speaking():
if request.sid in conversation_context:
conversation_context[request.sid]['is_speaking'] = True
conversation_context[request.sid]['audio_buffer'].clear()
print(f"User {request.sid} started speaking")
@socketio.on('audio_chunk')
def handle_audio_chunk(data):
if request.sid not in conversation_context:
return
context = conversation_context[request.sid]
# Decode audio data
audio_data = base64.b64decode(data['audio'])
audio_numpy = np.frombuffer(audio_data, dtype=np.float32)
audio_tensor = torch.tensor(audio_numpy)
# Add to buffer
context['audio_buffer'].append(audio_tensor)
# Check for silence to detect end of speech
if context['is_speaking'] and is_silence(audio_tensor):
if context['silence_start'] is None:
context['silence_start'] = time.time()
elif time.time() - context['silence_start'] > 1.0: # 1 second of silence
# Process the complete utterance
process_user_utterance(request.sid)
else:
context['silence_start'] = None
@socketio.on('stop_speaking')
def handle_stop_speaking():
if request.sid in conversation_context:
conversation_context[request.sid]['is_speaking'] = False
process_user_utterance(request.sid)
print(f"User {request.sid} stopped speaking")
def is_silence(audio_tensor, threshold=0.02):
"""Check if an audio chunk is silence based on amplitude threshold"""
return torch.mean(torch.abs(audio_tensor)) < threshold
def process_user_utterance(session_id):
"""Process completed user utterance, generate response and send audio back"""
context = conversation_context[session_id]
if not context['audio_buffer']:
return
# Combine audio chunks
full_audio = torch.cat(list(context['audio_buffer']), dim=0)
context['audio_buffer'].clear()
context['is_speaking'] = False
context['silence_start'] = None
# Save audio to temporary WAV file for Whisper transcription
temp_audio_path = f"temp_audio_{session_id}.wav"
torchaudio.save(
temp_audio_path,
full_audio.unsqueeze(0),
44100 # Assuming 44.1kHz from client
)
# Transcribe speech using Faster-Whisper
try:
segments, info = whisper_model.transcribe(temp_audio_path, beam_size=5)
# Collect all text from segments
user_text = ""
for segment in segments:
segment_text = segment.text.strip()
print(f"[{segment.start:.2f}s -> {segment.end:.2f}s] {segment_text}")
user_text += segment_text + " "
user_text = user_text.strip()
# Cleanup temp file
if os.path.exists(temp_audio_path):
os.remove(temp_audio_path)
if not user_text:
print("No speech detected.")
return
print(f"Transcribed: {user_text}")
# Add to conversation segments
user_segment = Segment(
text=user_text,
speaker=0, # User is speaker 0
audio=full_audio
)
context['segments'].append(user_segment)
# Generate bot response
bot_response = generate_llm_response(user_text, context['segments'])
print(f"Bot response: {bot_response}")
# Convert to audio using CSM
bot_audio = generate_audio_response(bot_response, context['segments'])
# Convert audio to base64 for sending over websocket
audio_bytes = io.BytesIO()
torchaudio.save(audio_bytes, bot_audio.unsqueeze(0).cpu(), csm_generator.sample_rate, format="wav")
audio_bytes.seek(0)
audio_b64 = base64.b64encode(audio_bytes.read()).decode('utf-8')
# Add bot response to conversation history
bot_segment = Segment(
text=bot_response,
speaker=1, # Bot is speaker 1
audio=bot_audio
)
context['segments'].append(bot_segment)
# Send transcribed text to client
emit('transcription', {'text': user_text}, room=session_id)
# Send audio response to client
emit('audio_response', {
'audio': audio_b64,
'text': bot_response
}, room=session_id)
except Exception as e:
print(f"Error processing speech: {e}")
emit('error', {'message': f'Error processing speech: {str(e)}'}, room=session_id)
# Cleanup temp file in case of error
if os.path.exists(temp_audio_path):
os.remove(temp_audio_path)
def generate_llm_response(user_text, conversation_segments):
"""Generate text response using Llama 3.2"""
# Format conversation history for the LLM
conversation_history = ""
for segment in conversation_segments[-5:]: # Use last 5 utterances for context
speaker_name = "User" if segment.speaker == 0 else "Assistant"
conversation_history += f"{speaker_name}: {segment.text}\n"
# Add the current user query
conversation_history += f"User: {user_text}\nAssistant:"
# Generate response
inputs = llm_tokenizer(conversation_history, return_tensors="pt").to(device)
output = llm_model.generate(
inputs.input_ids,
max_new_tokens=150,
temperature=0.7,
top_p=0.9,
do_sample=True
)
response = llm_tokenizer.decode(output[0][inputs.input_ids.shape[1]:], skip_special_tokens=True)
return response.strip()
def generate_audio_response(text, conversation_segments):
"""Generate audio response using CSM"""
# Use the last few conversation segments as context
context_segments = conversation_segments[-4:] if len(conversation_segments) > 4 else conversation_segments
# Generate audio for bot response
audio = csm_generator.generate(
text=text,
speaker=1, # Bot is speaker 1
context=context_segments,
max_audio_length_ms=10000, # 10 seconds max
temperature=0.9,
topk=50
)
return audio
if __name__ == '__main__':
# Ensure the existing index.html file is in the correct location
if not os.path.exists('templates'):
os.makedirs('templates')
if os.path.exists('index.html') and not os.path.exists('templates/index.html'):
os.rename('index.html', 'templates/index.html')
socketio.run(app, host='0.0.0.0', port=5000, debug=False)