AI数字人直播系统
你提供的代码已经涵盖了 AI 数字人直播系统的多项核心功能。如果要进一步完善并梳理所有功能,以下是对代码各部分功能及改进方向的详细说明;
- 语音识别功能:
- 代码部分:
python
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 ""
- 功能:使用
SpeechRecognition
库,通过计算机麦克风获取用户语音输入,并利用谷歌语音识别服务将其转换为文本。 - 改进方向:可以添加更多语音识别引擎的支持,如百度语音识别、科大讯飞语音识别等,以提高识别的准确性和适应性。同时,增加对语音识别结果的后处理,例如去除噪声词汇、纠正常见识别错误等。
- 自然语言理解与回复生成功能:
- 代码部分:
python
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
- 功能:借助
transformers
库中的DialoGPT - medium
模型,对用户输入的文本进行自然语言理解,并生成相应的回复。 - 改进方向:可以考虑使用更强大或针对特定领域训练的语言模型,如 GPT - 4(如果有接口)或经过领域微调的模型,以提升回复的质量和专业性。同时,优化对话管理逻辑,实现多轮对话的连贯性和上下文理解。
- 语音合成功能:
- 代码部分:
python
def text_to_speech(text, lang='zh - CN'):
tts = gTTS(text=text, lang=lang)
tts.save("response.mp3")
return "response.mp3"
- 功能:利用
gTTS
库将生成的回复文本转换为语音,并保存为 MP3 文件。 - 改进方向:可以尝试使用其他语音合成引擎,如字节跳动的云雀语音合成(如果有开放接口),以获得更自然、多样的语音效果。还可以添加对语音音色、语速、语调等参数的设置,满足不同场景需求。
- 唇形同步模拟功能:
- 代码部分:
python
def lip_sync_video(video_path, audio_path):
video = VideoFileClip(video_path)
audio = AudioFileClip(audio_path)
video_duration = video.duration
audio_duration = audio.duration
words = audio.reader.get_metadata()['duration'] / 1000
num_words = len(words.split())
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)
def adjust_lips(frame):
height, width, _ = frame.shape
lips_y = int(height * 0.6)
lips_height = int(height * 0.2)
lips_frame = frame[lips_y:lips_y + lips_height, :]
resized_lips = cv2.resize(lips_frame, None, fx=(1 + num_words / 10), fy=1)
frame[lips_y:lips_y + resized_lips.shape[0], :] = resized_lips
return frame
new_video = new_video.fl(adjust_lips)
new_video.write_videofile("lipsynced_video.mp4", codec='libx264')
return "lipsynced_video.mp4"
- 功能:根据语音的时长和简单的音素特征(这里通过单词数量模拟),调整数字人视频的帧率,并对视频帧中的嘴唇部分进行缩放,以实现唇形同步的效果。
- 改进方向:采用更专业的音素分析工具,精确获取语音的音素信息,从而更准确地驱动数字人的唇形变化。可以结合深度学习模型,如基于神经网络的唇形生成模型,实现更自然、逼真的唇形同步效果。
- 表情和动作模拟功能:
- 代码部分:
python
def simulate_digital_human_expression_and_action(video_path, audio_path):
video = VideoFileClip(video_path)
audio = AudioFileClip(audio_path)
audio_duration = audio.duration
audio_data = audio.to_soundarray(fps=audio.fps)
max_volume = np.max(np.abs(audio_data))
if max_volume > 0.5:
def happy_action(frame):
height, width, _ = frame.shape
M = cv2.getRotationMatrix2D((width / 2, height / 2), 10, 1.1)
return cv2.warpAffine(frame, M, (width, height))
new_video = video.fl(happy_action)
else:
new_video = video
new_video.write_videofile("expression_and_action_simulated_video.mp4", codec='libx264')
return "expression_and_action_simulated_video.mp4"
- 功能:通过分析语音的音量来模拟数字人的表情和动作。当音量超过设定阈值(这里设为 0.5)时,认为是积极情感,对视频进行旋转操作以模拟开心动作。
- 改进方向:使用专业的语音情感分析模型,如基于卷积神经网络或循环神经网络的情感分析模型,更准确地判断语音中的情感。同时,结合骨骼动画、面部表情动画等技术,实现更丰富、细腻的数字人表情和动作变化。
- 直播源获取功能:
- 代码部分:
python
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'
- 功能:利用
pytube
库从 YouTube 链接获取直播源视频,并下载保存为本地文件。 - 改进方向:增加对其他直播平台(如抖音、快手等)直播源的获取支持。同时,优化下载过程,如支持断点续传、多线程下载等,提高下载效率和稳定性。
- 直播推流功能:
- 代码部分:
python
async def bili_live_push(video_path):
room = live.LiveDanmaku(room_id=123456)
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")
await room.send_frame(frame)
cap.release()
await room.close()
- 功能:使用
bili - live - danmaku
库将处理后的视频推流至 B 站直播间,同时可以发送弹幕。 - 改进方向:完善推流过程中的错误处理机制,如网络中断、推流失败等情况的自动重试。增加对推流参数(如分辨率、帧率、码率等)的设置,以适应不同的网络环境和直播需求。同时,实现对观众弹幕的实时获取和处理,增强直播的互动性。
如果要将这些改进整合到代码中,示例如下(以改进语音识别和自然语言理解部分为例):
python
import speech_recognition as sr
from transformers import AutoTokenizer, AutoModelForCausalLM
from gtts import gTTS
from moviepy.editor import VideoFileClip, AudioFileClip, CompositeVideoClip, vfx
import os
import numpy as np
import cv2
from pytube import YouTube
from bilibili_api import live, sync
# 新增百度语音识别相关库
from aip import AipSpeech
# 新增对GPT - 4调用的相关库(假设使用OpenAI官方库)
import openai
# 百度语音识别配置
APP_ID = 'your_app_id'
API_KEY = 'your_api_key'
SECRET_KEY = 'your_secret_key'
client = AipSpeech(APP_ID, API_KEY, SECRET_KEY)
# 改进后的语音识别函数
def recognize_speech():
r = sr.Recognizer()
with sr.Microphone() as source:
print("请说话...")
audio = r.listen(source)
try:
# 先尝试谷歌语音识别
text = r.recognize_google(audio)
print(f"谷歌识别到的内容: {text}")
return text
except sr.UnknownValueError:
try:
# 谷歌识别失败,尝试百度语音识别
result = client.asr(audio.get_wav_data(), 'wav', 16000, {'dev_pid': 1536, })
if result['err_no'] == 0:
text = result['result'][0]
print(f"百度识别到的内容: {text}")
return text
else:
print("百度语音识别错误")
return ""
except Exception as e:
print(f"请求错误; {e}")
return ""
except sr.RequestError as e:
print(f"请求错误; {e}")
return ""
# 改进后的自然语言理解与回复生成函数
def generate_response(user_input):
# 这里假设已经配置好OpenAI的API Key
openai.api_key = "your_openai_api_key"
try:
# 尝试调用GPT - 4生成回复
response = openai.ChatCompletion.create(
model="gpt - 4",
messages=[
{"role": "user", "content": user_input}
]
)
return response.choices[0].message.content
except Exception as e:
# GPT - 4调用失败,使用原来的DialoGPT - medium模型
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"
# 唇形同步模拟,根据语音时长和音素特征调整视频帧(更复杂版本)
def lip_sync_video(video_path, audio_path):
video = VideoFileClip(video_path)
audio = AudioFileClip(audio_path)
video_duration = video.duration
audio_duration = audio.duration
# 简单的音素分析模拟(实际需要更专业的音素分析库)
# 这里假设每个单词对应一个音素变化,根据单词数量调整视频帧
words = audio.reader.get_metadata()['duration'] / 1000
num_words = len(words.split())
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)
# 模拟根据音素调整嘴唇动作(简单缩放视频帧中嘴唇部分)
def adjust_lips(frame):
height, width, _ = frame.shape
lips_y = int(height * 0.6)
lips_height = int(height * 0.2)
lips_frame = frame[lips_y:lips_y + lips_height, :]
resized_lips = cv2.resize(lips_frame, None, fx=(1 + num_words / 10), fy=1)
frame[lips_y:lips_y + resized_lips.shape[0], :] = resized_lips
return frame
new_video = new_video.fl(adjust_lips)
new_video.write_videofile("lipsynced_video.mp4", codec='libx264')
return "lipsynced_video.mp4"
# 模拟数字人表情和动作,结合语音情感分析(简单模拟)
def simulate_digital_human_expression_and_action(video_path, audio_path):
video = VideoFileClip(video_path)
audio = AudioFileClip(audio_path)
audio_duration = audio.duration
# 简单的语音情感分析模拟(实际需专业情感分析库)
# 这里根据音频音量判断情感,音量高假设为积极情感
audio_data = audio.to_soundarray(fps=audio.fps)
max_volume = np.max(np.abs(audio_data))
if max_volume > 0.5:
def happy_action(frame):
height, width, _ = frame.shape
M = cv2.getRotationMatrix2D((width / 2, height / 2), 10, 1.1)
return cv2.warpAffine(frame, M, (width, height))
new_video = video.fl(happy_action)
else:
new_video = video
new_video.write_videofile("expression_and_action_simulated_video.mp4", codec='libx264')
return "expression_and_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)
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")
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"
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)
expression_and_action_path = simulate_digital_human_expression_and_action(synced_video_path, audio_path)
sync(bili_live_push(expression_and_action_path))
os.remove(audio_path)
os.remove(synced_video_path)
os.remove(expression_and_action_path)
user_input = recognize_speech()
if __name__ == "__main__":
main()
上述代码在原有的基础上,改进了语音识别部分,增加了百度语音识别作为备用;改进了自然语言理解与回复生成部分,优先尝试调用 GPT - 4 生成回复,若失败则使用原来的DialoGPT - medium
模型。