312 lines
9.4 KiB
Python
312 lines
9.4 KiB
Python
import cv2
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import pyaudio
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import wave
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import threading
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import numpy as np
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import time
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from queue import Queue
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import webrtcvad
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import os
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import threading
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from transformers import Qwen2VLForConditionalGeneration, AutoTokenizer, AutoProcessor
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from transformers import AutoModelForCausalLM, AutoTokenizer
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from qwen_vl_utils import process_vision_info
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import torch
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from funasr import AutoModel
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import pygame
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import edge_tts
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import asyncio
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from time import sleep
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import langid
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from langdetect import detect
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# --- 配置huggingFace国内镜像 ---
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import os
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os.environ['HF_ENDPOINT'] = 'https://hf-mirror.com'
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# 参数设置
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AUDIO_RATE = 16000 # 音频采样率
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AUDIO_CHANNELS = 1 # 单声道
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CHUNK = 1024 # 音频块大小
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VAD_MODE = 3 # VAD 模式 (0-3, 数字越大越敏感)
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OUTPUT_DIR = "./output" # 输出目录
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NO_SPEECH_THRESHOLD = 1 # 无效语音阈值，单位：秒
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folder_path = "./Test_QWen2_VL/"
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audio_file_count = 0
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# 确保输出目录存在
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os.makedirs(OUTPUT_DIR, exist_ok=True)
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os.makedirs(folder_path, exist_ok=True)
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# 队列用于音频和视频同步缓存
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audio_queue = Queue()
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video_queue = Queue()
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# 全局变量
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last_active_time = time.time()
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recording_active = True
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segments_to_save = []
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saved_intervals = []
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last_vad_end_time = 0 # 上次保存的 VAD 有效段结束时间
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# 初始化 WebRTC VAD
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vad = webrtcvad.Vad()
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vad.set_mode(VAD_MODE)
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# 音频录制线程
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def audio_recorder():
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global audio_queue, recording_active, last_active_time, segments_to_save, last_vad_end_time
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p = pyaudio.PyAudio()
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stream = p.open(format=pyaudio.paInt16,
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channels=AUDIO_CHANNELS,
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rate=AUDIO_RATE,
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input=True,
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frames_per_buffer=CHUNK)
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audio_buffer = []
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print("音频录制已开始")
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while recording_active:
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data = stream.read(CHUNK)
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audio_buffer.append(data)
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# 每 0.5 秒检测一次 VAD
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if len(audio_buffer) * CHUNK / AUDIO_RATE >= 0.5:
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# 拼接音频数据并检测 VAD
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raw_audio = b''.join(audio_buffer)
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vad_result = check_vad_activity(raw_audio)
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if vad_result:
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print("检测到语音活动")
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last_active_time = time.time()
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segments_to_save.append((raw_audio, time.time()))
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else:
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print("静音中...")
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audio_buffer = [] # 清空缓冲区
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# 检查无效语音时间
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if time.time() - last_active_time > NO_SPEECH_THRESHOLD:
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# 检查是否需要保存
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if segments_to_save and segments_to_save[-1][1] > last_vad_end_time:
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save_audio_video()
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last_active_time = time.time()
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else:
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pass
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# print("无新增语音段，跳过保存")
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stream.stop_stream()
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stream.close()
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p.terminate()
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# 视频录制线程
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def video_recorder():
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global video_queue, recording_active
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cap = cv2.VideoCapture(0) # 使用默认摄像头
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print("视频录制已开始")
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while recording_active:
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ret, frame = cap.read()
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if ret:
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video_queue.put((frame, time.time()))
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# 实时显示摄像头画面
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cv2.imshow("Real Camera", frame)
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if cv2.waitKey(1) & 0xFF == ord('q'): # 按 Q 键退出
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break
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else:
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print("无法获取摄像头画面")
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cap.release()
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cv2.destroyAllWindows()
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# 检测 VAD 活动
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def check_vad_activity(audio_data):
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# 将音频数据分块检测
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num, rate = 0, 0.4
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step = int(AUDIO_RATE * 0.02) # 20ms 块大小
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flag_rate = round(rate * len(audio_data) // step)
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for i in range(0, len(audio_data), step):
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chunk = audio_data[i:i + step]
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if len(chunk) == step:
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if vad.is_speech(chunk, sample_rate=AUDIO_RATE):
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num += 1
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if num > flag_rate:
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return True
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return False
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# 保存音频和视频
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def save_audio_video():
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pygame.mixer.init()
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global segments_to_save, video_queue, last_vad_end_time, saved_intervals
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# 全局变量，用于保存音频文件名计数
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global audio_file_count
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audio_file_count += 1
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audio_output_path = f"{OUTPUT_DIR}/audio_{audio_file_count}.wav"
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# audio_output_path = f"{OUTPUT_DIR}/audio_0.wav"
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if not segments_to_save:
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return
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# 停止当前播放的音频
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if pygame.mixer.music.get_busy():
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pygame.mixer.music.stop()
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print("检测到新的有效音，已停止当前音频播放")
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# 获取有效段的时间范围
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start_time = segments_to_save[0][1]
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end_time = segments_to_save[-1][1]
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# 检查是否与之前的片段重叠
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if saved_intervals and saved_intervals[-1][1] >= start_time:
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print("当前片段与之前片段重叠，跳过保存")
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segments_to_save.clear()
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return
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# 保存音频
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audio_frames = [seg[0] for seg in segments_to_save]
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wf = wave.open(audio_output_path, 'wb')
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wf.setnchannels(AUDIO_CHANNELS)
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wf.setsampwidth(2) # 16-bit PCM
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wf.setframerate(AUDIO_RATE)
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wf.writeframes(b''.join(audio_frames))
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wf.close()
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print(f"音频保存至 {audio_output_path}")
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# Inference()
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# 使用线程执行推理
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inference_thread = threading.Thread(target=Inference, args=(audio_output_path,))
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inference_thread.start()
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# 记录保存的区间
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saved_intervals.append((start_time, end_time))
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# 清空缓冲区
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segments_to_save.clear()
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# --- 播放音频 -
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def play_audio(file_path):
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try:
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pygame.mixer.init()
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pygame.mixer.music.load(file_path)
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pygame.mixer.music.play()
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while pygame.mixer.music.get_busy():
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time.sleep(1) # 等待音频播放结束
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print("播放完成！")
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except Exception as e:
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print(f"播放失败: {e}")
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finally:
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pygame.mixer.quit()
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async def amain(TEXT, VOICE, OUTPUT_FILE) -> None:
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"""Main function"""
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communicate = edge_tts.Communicate(TEXT, VOICE)
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await communicate.save(OUTPUT_FILE)
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# -------- SenceVoice 语音识别 --模型加载-----
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model_dir = r"D:\AI\download\SenseVoiceSmall"
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model_senceVoice = AutoModel( model=model_dir, trust_remote_code=True, )
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# --- QWen2.5大语言模型 ---
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model_name = r"D:\AI\download\Qwen2.5-1.5B-Instruct"
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# model_name = r'D:\AI\download\Qwen2.5-7B-Instruct-GPTQ-Int4'
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model = AutoModelForCausalLM.from_pretrained(
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model_name,
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torch_dtype="auto",
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device_map="auto"
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)
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tokenizer = AutoTokenizer.from_pretrained(model_name)
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def Inference(TEMP_AUDIO_FILE=f"{OUTPUT_DIR}/audio_0.wav"):
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# -------- SenceVoice 推理 ---------
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input_file = (TEMP_AUDIO_FILE)
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res = model_senceVoice.generate(
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input=input_file,
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cache={},
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language="auto", # "zn", "en", "yue", "ja", "ko", "nospeech"
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use_itn=False,
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)
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# prompt = res[0]['text'].split(">")[-1]
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prompt = res[0]['text'].split(">")[-1] #识别结果
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print("ASR OUT:", prompt)
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# ---------SenceVoice --end----------
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# -------- 模型推理阶段，将语音识别结果作为大模型Prompt ------
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messages = [
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{"role": "system", "content": "你叫千问，是一个18岁的女大学生，性格活泼开朗，说话俏皮"},
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{"role": "user", "content": prompt},
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]
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text = tokenizer.apply_chat_template(
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messages,
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tokenize=False,
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add_generation_prompt=True,
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)
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model_inputs = tokenizer([text], return_tensors="pt").to(model.device)
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generated_ids = model.generate(
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**model_inputs,
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max_new_tokens=512,
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)
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generated_ids = [
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output_ids[len(input_ids):] for input_ids, output_ids in zip(model_inputs.input_ids, generated_ids)
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]
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output_text = tokenizer.batch_decode(generated_ids, skip_special_tokens=True)[0]
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print("answer", output_text)
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# 输入文本
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text = output_text
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# 语种识别 -- langid
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language, confidence = langid.classify(text)
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# 语种识别 -- langdetect
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# language = detect(text).split("-")[0]
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language_speaker = {
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"ja" : "ja-JP-NanamiNeural", # ok
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"fr" : "fr-FR-DeniseNeural", # ok
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"es" : "ca-ES-JoanaNeural", # ok
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"de" : "de-DE-KatjaNeural", # ok
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"zh" : "zh-CN-XiaoyiNeural", # ok
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"en" : "en-US-AnaNeural", # ok
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}
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if language not in language_speaker.keys():
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used_speaker = "zh-CN-XiaoyiNeural"
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else:
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used_speaker = language_speaker[language]
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print("检测到语种：", language, "使用音色：", language_speaker[language])
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global audio_file_count
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asyncio.run(amain(text, used_speaker, os.path.join(folder_path,f"sft_{audio_file_count}.mp3")))
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play_audio(f'{folder_path}/sft_{audio_file_count}.mp3')
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# 主函数
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if __name__ == "__main__":
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try:
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# 启动音视频录制线程
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audio_thread = threading.Thread(target=audio_recorder)
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# video_thread = threading.Thread(target=video_recorder)
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audio_thread.start()
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# video_thread.start()
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print("按 Ctrl+C 停止录制")
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while True:
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time.sleep(1)
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except KeyboardInterrupt:
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print("录制停止中...")
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recording_active = False
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audio_thread.join()
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# video_thread.join()
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print("录制已停止")
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