367 lines
14 KiB
Python
367 lines
14 KiB
Python
import os
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import io
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import base64
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import time
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import torch
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import torchaudio
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import numpy as np
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from flask import Flask, render_template, request
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from flask_socketio import SocketIO, emit
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from transformers import AutoModelForCausalLM, AutoTokenizer
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from collections import deque
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import requests
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import huggingface_hub
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from generator import load_csm_1b, Segment
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# Force CPU mode regardless of what's available
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# This bypasses the CUDA/cuDNN library requirements
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os.environ["CUDA_VISIBLE_DEVICES"] = "" # Hide all CUDA devices
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torch.backends.cudnn.enabled = False # Disable cuDNN
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# Configure environment with longer timeouts
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os.environ["HF_HUB_DOWNLOAD_TIMEOUT"] = "600" # 10 minutes timeout for downloads
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requests.adapters.DEFAULT_TIMEOUT = 60 # Increase default requests timeout
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# Create a models directory for caching
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os.makedirs("models", exist_ok=True)
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app = Flask(__name__)
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app.config['SECRET_KEY'] = 'your-secret-key'
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socketio = SocketIO(app, cors_allowed_origins="*")
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# Force CPU regardless of what hardware is available
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device = "cuda" if torch.cuda.is_available() else "cpu"
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whisper_compute_type = "int8"
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print(f"Forcing CPU mode for all models")
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# Initialize models with proper error handling
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whisper_model = None
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csm_generator = None
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llm_model = None
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llm_tokenizer = None
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def load_models():
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global whisper_model, csm_generator, llm_model, llm_tokenizer
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# Initialize Faster-Whisper for transcription
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try:
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print("Loading Whisper model on CPU...")
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# Import here to avoid immediate import errors if package is missing
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from faster_whisper import WhisperModel
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whisper_model = WhisperModel("tiny", device="cpu", compute_type="int8", download_root="./models/whisper")
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print("Whisper model loaded successfully")
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except Exception as e:
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print(f"Error loading Whisper model: {e}")
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print("Will use backup speech recognition method if available")
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# Initialize CSM model for audio generation
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try:
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print("Loading CSM model on CPU...")
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csm_generator = load_csm_1b(device="cpu")
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print("CSM model loaded successfully")
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except Exception as e:
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print(f"Error loading CSM model: {e}")
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print("Audio generation will not be available")
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# Initialize Llama 3.2 model for response generation
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try:
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print("Loading Llama 3.2 model on CPU...")
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llm_model_id = "meta-llama/Llama-3.2-1B" # Choose appropriate size based on resources
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llm_tokenizer = AutoTokenizer.from_pretrained(llm_model_id, cache_dir="./models/llama")
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llm_model = AutoModelForCausalLM.from_pretrained(
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llm_model_id,
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torch_dtype=torch.float32, # Use float32 on CPU
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device_map="cpu",
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cache_dir="./models/llama",
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low_cpu_mem_usage=True
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)
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print("Llama 3.2 model loaded successfully")
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except Exception as e:
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print(f"Error loading Llama 3.2 model: {e}")
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print("Will use a fallback response generation method")
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# Store conversation context
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conversation_context = {} # session_id -> context
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@app.route('/')
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def index():
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return render_template('index.html')
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@socketio.on('connect')
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def handle_connect():
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print(f"Client connected: {request.sid}")
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conversation_context[request.sid] = {
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'segments': [],
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'speakers': [0, 1], # 0 = user, 1 = bot
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'audio_buffer': deque(maxlen=10), # Store recent audio chunks
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'is_speaking': False,
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'silence_start': None
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}
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emit('ready', {'message': 'Connection established'})
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@socketio.on('disconnect')
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def handle_disconnect():
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print(f"Client disconnected: {request.sid}")
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if request.sid in conversation_context:
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del conversation_context[request.sid]
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@socketio.on('start_speaking')
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def handle_start_speaking():
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if request.sid in conversation_context:
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conversation_context[request.sid]['is_speaking'] = True
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conversation_context[request.sid]['audio_buffer'].clear()
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print(f"User {request.sid} started speaking")
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@socketio.on('audio_chunk')
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def handle_audio_chunk(data):
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if request.sid not in conversation_context:
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return
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context = conversation_context[request.sid]
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# Decode audio data
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audio_data = base64.b64decode(data['audio'])
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audio_numpy = np.frombuffer(audio_data, dtype=np.float32)
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audio_tensor = torch.tensor(audio_numpy)
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# Add to buffer
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context['audio_buffer'].append(audio_tensor)
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# Check for silence to detect end of speech
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if context['is_speaking'] and is_silence(audio_tensor):
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if context['silence_start'] is None:
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context['silence_start'] = time.time()
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elif time.time() - context['silence_start'] > 1.0: # 1 second of silence
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# Process the complete utterance
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process_user_utterance(request.sid)
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else:
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context['silence_start'] = None
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@socketio.on('stop_speaking')
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def handle_stop_speaking():
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if request.sid in conversation_context:
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conversation_context[request.sid]['is_speaking'] = False
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process_user_utterance(request.sid)
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print(f"User {request.sid} stopped speaking")
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def is_silence(audio_tensor, threshold=0.02):
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"""Check if an audio chunk is silence based on amplitude threshold"""
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return torch.mean(torch.abs(audio_tensor)) < threshold
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def process_user_utterance(session_id):
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"""Process completed user utterance, generate response and send audio back"""
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context = conversation_context[session_id]
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if not context['audio_buffer']:
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return
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# Combine audio chunks
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full_audio = torch.cat(list(context['audio_buffer']), dim=0)
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context['audio_buffer'].clear()
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context['is_speaking'] = False
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context['silence_start'] = None
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# Save audio to temporary WAV file for transcription
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temp_audio_path = f"temp_audio_{session_id}.wav"
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torchaudio.save(
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temp_audio_path,
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full_audio.unsqueeze(0),
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44100 # Assuming 44.1kHz from client
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)
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try:
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# Try using Whisper first if available
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if whisper_model is not None:
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user_text = transcribe_with_whisper(temp_audio_path)
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else:
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# Fallback to Google's speech recognition
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user_text = transcribe_with_google(temp_audio_path)
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if not user_text:
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print("No speech detected.")
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emit('error', {'message': 'No speech detected. Please try again.'}, room=session_id)
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return
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print(f"Transcribed: {user_text}")
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# Add to conversation segments
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user_segment = Segment(
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text=user_text,
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speaker=0, # User is speaker 0
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audio=full_audio
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)
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context['segments'].append(user_segment)
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# Generate bot response
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bot_response = generate_llm_response(user_text, context['segments'])
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print(f"Bot response: {bot_response}")
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# Send transcribed text to client
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emit('transcription', {'text': user_text}, room=session_id)
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# Generate and send audio response if CSM is available
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if csm_generator is not None:
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# Convert to audio using CSM
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bot_audio = generate_audio_response(bot_response, context['segments'])
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# Convert audio to base64 for sending over websocket
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audio_bytes = io.BytesIO()
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torchaudio.save(audio_bytes, bot_audio.unsqueeze(0).cpu(), csm_generator.sample_rate, format="wav")
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audio_bytes.seek(0)
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audio_b64 = base64.b64encode(audio_bytes.read()).decode('utf-8')
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# Add bot response to conversation history
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bot_segment = Segment(
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text=bot_response,
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speaker=1, # Bot is speaker 1
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audio=bot_audio
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)
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context['segments'].append(bot_segment)
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# Send audio response to client
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emit('audio_response', {
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'audio': audio_b64,
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'text': bot_response
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}, room=session_id)
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else:
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# Send text-only response if audio generation isn't available
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emit('text_response', {'text': bot_response}, room=session_id)
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# Add text-only bot response to conversation history
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bot_segment = Segment(
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text=bot_response,
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speaker=1, # Bot is speaker 1
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audio=torch.zeros(1) # Placeholder empty audio
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)
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context['segments'].append(bot_segment)
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except Exception as e:
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print(f"Error processing speech: {e}")
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emit('error', {'message': f'Error processing speech: {str(e)}'}, room=session_id)
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finally:
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# Cleanup temp file
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if os.path.exists(temp_audio_path):
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os.remove(temp_audio_path)
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def transcribe_with_whisper(audio_path):
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"""Transcribe audio using Faster-Whisper"""
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segments, info = whisper_model.transcribe(audio_path, beam_size=5)
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# Collect all text from segments
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user_text = ""
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for segment in segments:
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segment_text = segment.text.strip()
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print(f"[{segment.start:.2f}s -> {segment.end:.2f}s] {segment_text}")
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user_text += segment_text + " "
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return user_text.strip()
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def transcribe_with_google(audio_path):
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"""Fallback transcription using Google's speech recognition"""
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import speech_recognition as sr
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recognizer = sr.Recognizer()
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with sr.AudioFile(audio_path) as source:
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audio = recognizer.record(source)
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try:
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text = recognizer.recognize_google(audio)
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return text
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except sr.UnknownValueError:
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return ""
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except sr.RequestError:
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# If Google API fails, try a basic energy-based VAD approach
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# This is a very basic fallback and won't give good results
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return "[Speech detected but transcription failed]"
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def generate_llm_response(user_text, conversation_segments):
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"""Generate text response using available model"""
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if llm_model is not None and llm_tokenizer is not None:
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# Format conversation history for the LLM
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conversation_history = ""
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for segment in conversation_segments[-5:]: # Use last 5 utterances for context
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speaker_name = "User" if segment.speaker == 0 else "Assistant"
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conversation_history += f"{speaker_name}: {segment.text}\n"
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# Add the current user query
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conversation_history += f"User: {user_text}\nAssistant:"
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try:
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# Generate response
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inputs = llm_tokenizer(conversation_history, return_tensors="pt").to(device)
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output = llm_model.generate(
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inputs.input_ids,
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max_new_tokens=150,
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temperature=0.7,
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top_p=0.9,
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do_sample=True
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)
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response = llm_tokenizer.decode(output[0][inputs.input_ids.shape[1]:], skip_special_tokens=True)
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return response.strip()
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except Exception as e:
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print(f"Error generating response with LLM: {e}")
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return fallback_response(user_text)
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else:
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return fallback_response(user_text)
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def fallback_response(user_text):
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"""Generate a simple fallback response when LLM is not available"""
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# Simple rule-based responses
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user_text_lower = user_text.lower()
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if "hello" in user_text_lower or "hi" in user_text_lower:
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return "Hello! I'm a simple fallback assistant. The main language model couldn't be loaded, so I have limited capabilities."
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elif "how are you" in user_text_lower:
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return "I'm functioning within my limited capabilities. How can I assist you today?"
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elif "thank" in user_text_lower:
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return "You're welcome! Let me know if there's anything else I can help with."
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elif "bye" in user_text_lower or "goodbye" in user_text_lower:
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return "Goodbye! Have a great day!"
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elif any(q in user_text_lower for q in ["what", "who", "where", "when", "why", "how"]):
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return "I'm running in fallback mode and can't answer complex questions. Please try again when the main language model is available."
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else:
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return "I understand you said something about that. Unfortunately, I'm running in fallback mode with limited capabilities. Please try again later when the main model is available."
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def generate_audio_response(text, conversation_segments):
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"""Generate audio response using CSM"""
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try:
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# Use the last few conversation segments as context
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context_segments = conversation_segments[-4:] if len(conversation_segments) > 4 else conversation_segments
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# Generate audio for bot response
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audio = csm_generator.generate(
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text=text,
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speaker=1, # Bot is speaker 1
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context=context_segments,
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max_audio_length_ms=10000, # 10 seconds max
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temperature=0.9,
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topk=50
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)
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return audio
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except Exception as e:
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print(f"Error generating audio: {e}")
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# Return silence as fallback
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return torch.zeros(csm_generator.sample_rate * 3) # 3 seconds of silence
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if __name__ == '__main__':
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# Ensure the existing index.html file is in the correct location
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if not os.path.exists('templates'):
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os.makedirs('templates')
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if os.path.exists('index.html') and not os.path.exists('templates/index.html'):
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os.rename('index.html', 'templates/index.html')
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# Load models asynchronously before starting the server
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print("Starting CPU-only model loading...")
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# In a production environment, you could load models in a separate thread
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load_models()
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# Start the server
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print("Starting Flask SocketIO server...")
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socketio.run(app, host='0.0.0.0', port=5000, debug=False) |