ai无人自动直播系统

AI 无人自动直播系统整合了数字人技术、直播推流、自动化控制等多方面功能。以下代码示例以 Python 为基础，结合pytube进行直播流获取（假设从 YouTube 获取直播源，实际可替换为其他直播源），利用moviepy处理视频和音频，通过bili - live - danmaku库实现 B 站直播推流（可按需修改为其他直播平台），并集成之前数字人系统中的语音交互与自然语言处理功能。运行代码前，请确保安装以下库：

import speech_recognition as sr​

from transformers import AutoTokenizer, AutoModelForCausalLM​

from gtts import gTTS​

from moviepy.editor import VideoFileClip, AudioFileClip​

import os​

import numpy as np​

import cv2​

from pytube import YouTube​

from bilibili_api import live, sync​

​

​

# 语音识别函数​

def recognize_speech():​

r = sr.Recognizer()​

with sr.Microphone() as source:​

print("请说话...")​

audio = r.listen(source)​

try:​

text = r.recognize_google(audio)​

print(f"识别到的内容: {text}")​

return text​

except sr.UnknownValueError:​

print("无法识别语音")​

return ""​

except sr.RequestError as e:​

print(f"请求错误; {e}")​

return ""​

​

​

# 自然语言理解与回复生成函数​

def generate_response(user_input):​

tokenizer = AutoTokenizer.from_pretrained("microsoft/DialoGPT - medium")​

model = AutoModelForCausalLM.from_pretrained("microsoft/DialoGPT - medium")​

input_ids = tokenizer.encode(user_input + tokenizer.eos_token, return_tensors='pt')​

output = model.generate(input_ids=input_ids, max_length=1000, pad_token_id=tokenizer.eos_token_id)​

response = tokenizer.decode(output[:, input_ids.shape[-1]:][0], skip_special_tokens=True)​

return response​

​

​

# 语音合成并返回音频文件路径​

def text_to_speech(text, lang='zh - CN'):​

tts = gTTS(text=text, lang=lang)​

tts.save("response.mp3")​

return "response.mp3"​

​

​

# 简单的唇形同步模拟，根据语音时长调整视频帧​

def lip_sync_video(video_path, audio_path):​

video = VideoFileClip(video_path)​

audio = AudioFileClip(audio_path)​

video_duration = video.duration​

audio_duration = audio.duration​

if video_duration > audio_duration:​

new_fps = video.fps * (audio_duration / video_duration)​

new_video = video.set_fps(new_fps)​

new_video = new_video.set_duration(audio_duration)​

else:​

new_video = video.set_duration(audio_duration)​

new_video.write_videofile("lipsynced_video.mp4", codec='libx264')​

return "lipsynced_video.mp4"​

​

​

# 模拟数字人动作（简单示例，根据语音时长调整视频播放速度）​

def simulate_digital_human_action(video_path, audio_path):​

video = VideoFileClip(video_path)​

audio = AudioFileClip(audio_path)​

audio_duration = audio.duration​

if video.duration > audio_duration:​

speed_factor = video.duration / audio_duration​

new_video = video.fx(video.fx.speedx, speed_factor)​

else:​

new_video = video​

new_video.write_videofile("action_simulated_video.mp4", codec='libx264')​

return "action_simulated_video.mp4"​

​

​

# 获取直播源视频（以YouTube为例）​

def get_live_source_video(youtube_url):​

yt = YouTube(youtube_url)​

stream = yt.streams.filter(file_extension='mp4', adaptive=True).first()​

stream.download(filename='live_source_video.mp4')​

return 'live_source_video.mp4'​

​

​

# B站直播推流函数（需自行替换房间号和直播密钥等信息）​

async def bili_live_push(video_path):​

room = live.LiveDanmaku(room_id=123456) # 替换为实际房间号​

await room.connect()​

cap = cv2.VideoCapture(video_path)​

while cap.isOpened():​

ret, frame = cap.read()​

if not ret:​

break​

frame = cv2.cvtColor(frame, cv2.COLOR_BGR2RGB)​

await room.send_danmaku("这是直播内容") # 可根据需求发送弹幕​

await room.send_frame(frame)​

cap.release()​

await room.close()​

​

​

# 主函数，整合所有功能​

def main():​

youtube_url = "https://www.youtube.com/watch?v=xxxxxxx" # 替换为实际YouTube直播链接​

live_source_video_path = get_live_source_video(youtube_url)​

user_input = recognize_speech()​

while user_input.lower() != "退出":​

response = generate_response(user_input)​

print(f"数字人回复: {response}")​

audio_path = text_to_speech(response)​

video_path = live_source_video_path​

synced_video_path = lip_sync_video(video_path, audio_path)​

action_simulated_path = simulate_digital_human_action(synced_video_path, audio_path)​

sync(bili_live_push(action_simulated_path))​

os.remove(audio_path)​

os.remove(synced_video_path)​

os.remove(action_simulated_path)​

user_input = recognize_speech()​

​

​

if __name__ == "__main__":​

main()​