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# import torch
# import numpy as np
# from typing import Callable
# from config.settings import settings
# import os
# import time
# class SileroVAD:
# def __init__(self):
# self.model = None
# self.utils = None
# self.sample_rate = 16000
# self.is_streaming = False
# self.speech_callback = None
# self.audio_buffer = []
# self.speech_start_time = 0
# self.min_speech_duration = 0.5 # Giây
# # ✅ Thêm cấu hình chunk size cho Silero
# self.chunk_size = 512 # Silero yêu cầu 512 samples cho 16000Hz
# self.chunk_duration = self.chunk_size / self.sample_rate # 0.032 giây
# self._initialize_model()
# def _initialize_model(self):
# """Khởi tạo Silero VAD model"""
# try:
# print("🔄 Đang tải Silero VAD model...")
# self.model, self.utils = torch.hub.load(
# repo_or_dir='snakers4/silero-vad',
# model='silero_vad',
# force_reload=False,
# trust_repo=True
# )
# self.model.eval()
# print("✅ Đã tải Silero VAD model thành công")
# except Exception as e:
# print(f"❌ Lỗi tải Silero VAD model: {e}")
# self._initialize_model_fallback()
# def _initialize_model_fallback(self):
# """Fallback nếu torch.hub.load thất bại"""
# try:
# model_dir = torch.hub.get_dir()
# model_path = os.path.join(
# model_dir, 'snakers4_silero-vad_master', 'files', 'silero_vad.jit'
# )
# if os.path.exists(model_path):
# self.model = torch.jit.load(model_path)
# self.model.eval()
# print("✅ Đã tải Silero VAD model thành công (fallback)")
# else:
# print("❌ Không tìm thấy model file (fallback thất bại)")
# self.model = None
# except Exception as e:
# print(f"❌ Lỗi tải Silero VAD model fallback: {e}")
# self.model = None
# def start_stream(self, speech_callback: Callable):
# """Bắt đầu stream với VAD"""
# if self.model is None:
# print("❌ Silero VAD model chưa được khởi tạo")
# return False
# self.is_streaming = True
# self.speech_callback = speech_callback
# self.audio_buffer = []
# self.speech_start_time = 0
# print("🎙️ Bắt đầu Silero VAD streaming...")
# return True
# def stop_stream(self):
# """Dừng stream"""
# self.is_streaming = False
# self.speech_callback = None
# self.audio_buffer = []
# self.speech_start_time = 0
# print("🛑 Đã dừng Silero VAD streaming")
# def process_stream(self, audio_chunk: np.ndarray, sample_rate: int):
# """Xử lý audio chunk với Silero VAD - ĐÃ SỬA LỖI"""
# if not self.is_streaming or self.model is None:
# return
# try:
# # Resample nếu cần
# if sample_rate != self.sample_rate:
# audio_chunk = self._resample_audio(audio_chunk, sample_rate, self.sample_rate)
# # Thêm vào buffer
# self.audio_buffer.extend(audio_chunk)
# # ✅ Xử lý từng chunk 512 samples (Silero requirement)
# while len(self.audio_buffer) >= self.chunk_size:
# chunk = self.audio_buffer[:self.chunk_size]
# self._process_single_chunk(np.array(chunk))
# # Giữ lại phần thừa cho chunk tiếp theo
# self.audio_buffer = self.audio_buffer[self.chunk_size:]
# except Exception as e:
# print(f"❌ Lỗi xử lý Silero VAD: {e}")
# def _process_single_chunk(self, audio_chunk: np.ndarray):
# """Xử lý một chunk 512 samples duy nhất"""
# try:
# # Chuẩn hóa audio
# audio_chunk = self._normalize_audio(audio_chunk)
# # Đảm bảo đúng kích thước
# if len(audio_chunk) != self.chunk_size:
# # Nếu không đủ, pad với zeros
# if len(audio_chunk) < self.chunk_size:
# padding = np.zeros(self.chunk_size - len(audio_chunk), dtype=np.float32)
# audio_chunk = np.concatenate([audio_chunk, padding])
# else:
# audio_chunk = audio_chunk[:self.chunk_size]
# # Dự đoán xác suất speech
# speech_prob = self._get_speech_probability(audio_chunk)
# print(f"🎯 Silero VAD speech probability: {speech_prob:.3f}")
# # Xử lý logic speech detection
# current_time = time.time()
# if speech_prob > settings.VAD_THRESHOLD:
# if self.speech_start_time == 0:
# self.speech_start_time = current_time
# print("🎯 Bắt đầu phát hiện speech")
# speech_duration = current_time - self.speech_start_time
# # Nếu đủ thời gian speech, gọi callback
# if speech_duration >= self.min_speech_duration:
# if self.speech_callback:
# # Thu thập tất cả audio từ khi bắt đầu speech
# full_audio = self._collect_speech_audio()
# if len(full_audio) > 0:
# self.speech_callback(full_audio, self.sample_rate)
# self.speech_start_time = 0
# else:
# if self.speech_start_time > 0:
# print("🔇 Kết thúc speech segment")
# self.speech_start_time = 0
# except Exception as e:
# print(f"❌ Lỗi xử lý Silero VAD chunk: {e}")
# def _collect_speech_audio(self) -> np.ndarray:
# """Thu thập toàn bộ audio từ khi bắt đầu speech"""
# # Trong implementation thực tế, bạn cần lưu lại audio
# # từ khi bắt đầu phát hiện speech đến hiện tại
# # Đây là simplified version
# min_samples = int(self.sample_rate * self.min_speech_duration)
# return np.random.randn(min_samples).astype(np.float32) # Placeholder
# def _normalize_audio(self, audio: np.ndarray) -> np.ndarray:
# """Chuẩn hóa audio"""
# if audio.dtype != np.float32:
# audio = audio.astype(np.float32)
# if np.max(np.abs(audio)) > 1.0:
# audio = audio / 32768.0
# return np.clip(audio, -1.0, 1.0)
# def _get_speech_probability(self, audio_chunk: np.ndarray) -> float:
# """Trả về xác suất speech - ĐÃ SỬA LỖI"""
# try:
# # ✅ Đảm bảo đúng kích thước 512 samples
# if len(audio_chunk) != self.chunk_size:
# # Resize về đúng 512 samples
# if len(audio_chunk) > self.chunk_size:
# audio_chunk = audio_chunk[:self.chunk_size]
# else:
# padding = np.zeros(self.chunk_size - len(audio_chunk), dtype=np.float32)
# audio_chunk = np.concatenate([audio_chunk, padding])
# audio_tensor = torch.from_numpy(audio_chunk).float().unsqueeze(0)
# with torch.no_grad():
# return self.model(audio_tensor, self.sample_rate).item()
# except Exception as e:
# print(f"❌ Lỗi lấy speech probability: {e}")
# return 0.0
# def _resample_audio(self, audio: np.ndarray, orig_sr: int, target_sr: int) -> np.ndarray:
# """Resample audio"""
# if orig_sr == target_sr:
# return audio
# try:
# from scipy import signal
# # Tính số samples mới
# duration = len(audio) / orig_sr
# new_length = int(duration * target_sr)
# # Resample
# resampled_audio = signal.resample(audio, new_length)
# return resampled_audio.astype(np.float32)
# except ImportError:
# # Fallback simple resampling
# orig_len = len(audio)
# new_len = int(orig_len * target_sr / orig_sr)
# x_old = np.linspace(0, 1, orig_len)
# x_new = np.linspace(0, 1, new_len)
# return np.interp(x_new, x_old, audio).astype(np.float32)
# except Exception as e:
# print(f"⚠️ Lỗi resample: {e}")
# return audio
# def is_speech(self, audio_chunk: np.ndarray, sample_rate: int) -> bool:
# """Kiểm tra chunk có phải speech không - ĐÃ SỬA"""
# if self.model is None:
# return True
# try:
# if sample_rate != self.sample_rate:
# audio_chunk = self._resample_audio(audio_chunk, sample_rate, self.sample_rate)
# audio_chunk = self._normalize_audio(audio_chunk)
# # ✅ Chia thành các chunk 512 samples và kiểm tra trung bình
# chunk_size = 512
# speech_probs = []
# for i in range(0, len(audio_chunk), chunk_size):
# chunk = audio_chunk[i:i+chunk_size]
# if len(chunk) == chunk_size:
# prob = self._get_speech_probability(chunk)
# speech_probs.append(prob)
# if not speech_probs:
# return False
# avg_prob = np.mean(speech_probs)
# return avg_prob > settings.VAD_THRESHOLD
# except Exception as e:
# print(f"❌ Lỗi kiểm tra speech: {e}")
# return True
# def get_speech_probability(self, audio_chunk: np.ndarray, sample_rate: int) -> float:
# """Lấy xác suất speech trung bình"""
# if self.model is None:
# return 0.0
# try:
# if sample_rate != self.sample_rate:
# audio_chunk = self._resample_audio(audio_chunk, sample_rate, self.sample_rate)
# audio_chunk = self._normalize_audio(audio_chunk)
# # Chia thành các chunk 512 samples
# chunk_size = 512
# speech_probs = []
# for i in range(0, len(audio_chunk), chunk_size):
# chunk = audio_chunk[i:i+chunk_size]
# if len(chunk) == chunk_size:
# prob = self._get_speech_probability(chunk)
# speech_probs.append(prob)
# return np.mean(speech_probs) if speech_probs else 0.0
# except Exception as e:
# print(f"❌ Lỗi lấy speech probability: {e}")
# return 0.0
import io
import numpy as np
import soundfile as sf
import time
import traceback
import threading
import queue
import torch
from groq import Groq
from typing import Optional, Dict, Any, Callable
from config.settings import settings
class SileroVAD:
def __init__(self):
self.model = None
self.sample_rate = 16000
self.is_streaming = False
self.speech_callback = None
self.audio_buffer = []
self.speech_buffer = [] # Buffer cho speech đang diễn ra
self.state = "silence" # silence, speech, processing
self.speech_start_time = 0
self.last_voice_time = 0
# Cấu hình tối ưu
self.chunk_size = 512
self.speech_threshold = settings.VAD_THRESHOLD
self.min_speech_duration = settings.VAD_MIN_SPEECH_DURATION
self.min_silence_duration = settings.VAD_MIN_SILENCE_DURATION
self.speech_pad_duration = settings.VAD_SPEECH_PAD_DURATION
self.pre_speech_buffer = settings.VAD_PRE_SPEECH_BUFFER
# Buffer cho pre-speech
self.pre_speech_samples = int(self.pre_speech_buffer * self.sample_rate)
self.pre_speech_buffer = []
self._initialize_model()
def _initialize_model(self):
"""Khởi tạo Silero VAD model"""
try:
print("🔄 Đang tải Silero VAD model...")
self.model, utils = torch.hub.load(
repo_or_dir='snakers4/silero-vad',
model='silero_vad',
force_reload=False,
trust_repo=True
)
self.model.eval()
print("✅ Đã tải Silero VAD model thành công")
except Exception as e:
print(f"❌ Lỗi tải Silero VAD model: {e}")
self.model = None
def start_stream(self, speech_callback: Callable):
"""Bắt đầu stream với VAD"""
if self.model is None:
return False
self.is_streaming = True
self.speech_callback = speech_callback
self.audio_buffer = []
self.speech_buffer = []
self.pre_speech_buffer = []
self.state = "silence"
self.speech_start_time = 0
self.last_voice_time = 0
print("🎙️ Bắt đầu VAD streaming với cấu hình tối ưu...")
return True
def stop_stream(self):
"""Dừng stream"""
self.is_streaming = False
self.speech_callback = None
self.audio_buffer = []
self.speech_buffer = []
self.pre_speech_buffer = []
self.state = "silence"
print("🛑 Đã dừng VAD streaming")
def process_stream(self, audio_chunk: np.ndarray, sample_rate: int):
"""Xử lý audio chunk với VAD tối ưu"""
if not self.is_streaming or self.model is None:
return
try:
# Resample nếu cần
if sample_rate != self.sample_rate:
audio_chunk = self._resample_audio(audio_chunk, sample_rate, self.sample_rate)
# Thêm vào buffer chính
self.audio_buffer.extend(audio_chunk)
# Xử lý từng chunk
while len(self.audio_buffer) >= self.chunk_size:
chunk = self.audio_buffer[:self.chunk_size]
self._process_vad_chunk(np.array(chunk))
self.audio_buffer = self.audio_buffer[self.chunk_size:]
except Exception as e:
print(f"❌ Lỗi xử lý VAD: {e}")
def _process_vad_chunk(self, audio_chunk: np.ndarray):
"""Xử lý VAD cho một chunk - TỐI ƯU HÓA"""
current_time = time.time()
# Chuẩn hóa audio
audio_chunk = self._normalize_audio(audio_chunk)
# Lấy xác suất speech
speech_prob = self._get_speech_probability(audio_chunk)
# Logic state machine cải tiến
if self.state == "silence":
if speech_prob > self.speech_threshold:
print("🎯 Bắt đầu phát hiện speech")
self.state = "speech"
self.speech_start_time = current_time
self.last_voice_time = current_time
# Khởi tạo speech buffer với pre-speech data
self.speech_buffer = self.pre_speech_buffer.copy()
self.speech_buffer.extend(audio_chunk)
else:
# Lưu pre-speech buffer (giới hạn kích thước)
self.pre_speech_buffer.extend(audio_chunk)
if len(self.pre_speech_buffer) > self.pre_speech_samples:
self.pre_speech_buffer = self.pre_speech_buffer[-self.pre_speech_samples:]
elif self.state == "speech":
# Luôn thêm vào speech buffer
self.speech_buffer.extend(audio_chunk)
# Cập nhật thời gian voice cuối cùng
if speech_prob > self.speech_threshold:
self.last_voice_time = current_time
# Kiểm tra kết thúc speech
silence_duration = current_time - self.last_voice_time
speech_duration = current_time - self.speech_start_time
# Điều kiện kết thúc: im lặng đủ lâu VÀ đã nói đủ dài
if (silence_duration >= self.min_silence_duration and
speech_duration >= self.min_speech_duration):
print(f"🔇 Kết thúc speech segment (duration: {speech_duration:.2f}s)")
self._finalize_speech()
# Hoặc speech quá dài (timeout)
elif speech_duration > settings.MAX_AUDIO_DURATION:
print(f"⏰ Speech timeout ({speech_duration:.2f}s)")
self._finalize_speech()
elif self.state == "processing":
# Đang xử lý, không nhận thêm audio
pass
def _finalize_speech(self):
"""Hoàn thành xử lý speech segment"""
if not self.speech_buffer or len(self.speech_buffer) == 0:
self.state = "silence"
return
# Chuyển sang state processing để tránh nhận thêm audio
self.state = "processing"
# Tạo audio array từ buffer
speech_audio = np.array(self.speech_buffer, dtype=np.float32)
# Gọi callback trong thread riêng
if self.speech_callback:
threading.Thread(
target=self.speech_callback,
args=(speech_audio, self.sample_rate),
daemon=True
).start()
# Reset buffers nhưng giữ pre-speech
self.speech_buffer = []
self.audio_buffer = []
# Quay lại state silence sau khi xử lý
self.state = "silence"
def _normalize_audio(self, audio: np.ndarray) -> np.ndarray:
"""Chuẩn hóa audio"""
if audio.dtype != np.float32:
audio = audio.astype(np.float32)
if np.max(np.abs(audio)) > 1.0:
audio = audio / 32768.0
return np.clip(audio, -1.0, 1.0)
def _get_speech_probability(self, audio_chunk: np.ndarray) -> float:
"""Lấy xác suất speech"""
try:
if len(audio_chunk) != self.chunk_size:
return 0.0
audio_tensor = torch.from_numpy(audio_chunk).float().unsqueeze(0)
with torch.no_grad():
return self.model(audio_tensor, self.sample_rate).item()
except Exception as e:
print(f"❌ Lỗi speech probability: {e}")
return 0.0
def _resample_audio(self, audio: np.ndarray, orig_sr: int, target_sr: int) -> np.ndarray:
"""Resample audio"""
if orig_sr == target_sr:
return audio
try:
from scipy import signal
duration = len(audio) / orig_sr
new_length = int(duration * target_sr)
resampled_audio = signal.resample(audio, new_length)
return resampled_audio.astype(np.float32)
except Exception:
return audio
def is_speech(self, audio_chunk: np.ndarray, sample_rate: int) -> bool:
"""Kiểm tra speech (cho compatibility)"""
if self.model is None:
return True
try:
if sample_rate != self.sample_rate:
audio_chunk = self._resample_audio(audio_chunk, sample_rate, self.sample_rate)
audio_chunk = self._normalize_audio(audio_chunk)
# Kiểm tra multiple chunks
chunk_size = 512
speech_probs = []
for i in range(0, len(audio_chunk), chunk_size):
chunk = audio_chunk[i:i+chunk_size]
if len(chunk) == chunk_size:
prob = self._get_speech_probability(chunk)
speech_probs.append(prob)
return np.mean(speech_probs) > self.speech_threshold if speech_probs else False
except Exception as e:
print(f"❌ Lỗi kiểm tra speech: {e}")
return True |