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###############################################################################
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# Copyright (C) 2024 LiveTalking@lipku https://github.com/lipku/LiveTalking
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# email: lipku@foxmail.com
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#
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# Licensed under the Apache License, Version 2.0 (the "License");
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# you may not use this file except in compliance with the License.
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# You may obtain a copy of the License at
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#
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# http://www.apache.org/licenses/LICENSE-2.0
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#
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# Unless required by applicable law or agreed to in writing, software
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# distributed under the License is distributed on an "AS IS" BASIS,
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# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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# See the License for the specific language governing permissions and
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# limitations under the License.
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###############################################################################
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# server.py
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from flask import Flask, render_template,send_from_directory,request, jsonify
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from flask_sockets import Sockets
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import base64
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import json
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#import gevent
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#from gevent import pywsgi
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#from geventwebsocket.handler import WebSocketHandler
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import re
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import numpy as np
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from threading import Thread,Event
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#import multiprocessing
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import torch.multiprocessing as mp
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from aiohttp import web
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import aiohttp
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import aiohttp_cors
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from aiortc import RTCPeerConnection, RTCSessionDescription, RTCIceServer, RTCConfiguration
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from aiortc.rtcrtpsender import RTCRtpSender
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from webrtc import HumanPlayer
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from basereal import BaseReal
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from llm import llm_response
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import argparse
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import random
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import shutil
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import asyncio
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import torch
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from typing import Dict
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from logger import logger
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import gc
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import json
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import multiprocessing as mp
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import sqlite3
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import os
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app = Flask(__name__)
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#sockets = Sockets(app)
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nerfreals:Dict[int, BaseReal] = {} #sessionid:BaseReal
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opt = None
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model = None
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avatar = None
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#####webrtc###############################
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pcs = set()
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def randN(N)->int:
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'''生成长度为 N的随机数 '''
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min = pow(10, N - 1)
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max = pow(10, N)
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return random.randint(min, max - 1)
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# 全局数据库连接和配置字典
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conn = None
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config_dict = {}
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def load_db_config(db_path):
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global conn, config_dict
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# 建立可跨线程复用的连接
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conn = sqlite3.connect(db_path, check_same_thread=False)
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conn.row_factory = sqlite3.Row
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# 性能调优：WAL 模式、不阻塞读取、缓存增大
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conn.execute("PRAGMA journal_mode = WAL;")
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conn.execute("PRAGMA synchronous = NORMAL;")
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conn.execute("PRAGMA cache_size = 10000;")
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# 一次性加载所有 key/value
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cursor = conn.execute("SELECT key, value FROM livetalking_config;")
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config_dict = {row["key"]: row["value"] for row in cursor.fetchall()}
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def set_enable_status(db_path):
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# 连接数据库
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conn = sqlite3.connect(db_path, check_same_thread=False)
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try:
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# 把 enable_status 对应的 value 字段改为 '1'
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conn.execute(
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"UPDATE live_config SET value = '1' WHERE key = 'livetlking_enable_status';"
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)
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conn.commit()
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print("livetlking_enable_status 对应的 value 字段改为 '1'")
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finally:
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conn.close()
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def build_nerfreal(sessionid:int)->BaseReal:
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opt.sessionid=sessionid
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if opt.model == 'wav2lip':
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from lipreal import LipReal
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nerfreal = LipReal(opt,model,avatar)
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return nerfreal
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#@app.route('/offer', methods=['POST'])
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async def offer(request):
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logger.info("开始处理offer请求")
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# 接收请求参数
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try:
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params = await request.json()
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logger.debug("接收到的请求参数: %s", params)
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except Exception as e:
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logger.error("解析请求参数失败: %s", str(e))
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return web.Response(
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content_type="application/json",
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text=json.dumps({"code": -2, "msg": "解析参数失败"})
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)
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# 验证必要参数
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if "sdp" not in params or "type" not in params:
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logger.warning("请求参数缺少sdp或type字段: %s", params)
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return web.Response(
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content_type="application/json",
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text=json.dumps({"code": -3, "msg": "缺少必要参数"})
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)
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# 创建RTCSessionDescription
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try:
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offer = RTCSessionDescription(sdp=params["sdp"], type=params["type"])
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logger.debug("成功创建RTCSessionDescription，类型: %s", params["type"])
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except Exception as e:
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logger.error("创建RTCSessionDescription失败: %s", str(e))
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return web.Response(
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content_type="application/json",
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text=json.dumps({"code": -4, "msg": "创建offer失败"})
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)
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# 生成会话ID并记录
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sessionid = randN(6)
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nerfreals[sessionid] = None
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logger.info('生成新会话，sessionid=%d, 当前会话数量=%d', sessionid, len(nerfreals))
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# 构建nerfreal
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try:
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logger.debug("开始构建nerfreal，sessionid=%d", sessionid)
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nerfreal = await asyncio.get_event_loop().run_in_executor(None, build_nerfreal, sessionid)
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nerfreals[sessionid] = nerfreal
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logger.info("成功构建nerfreal，sessionid=%d", sessionid)
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except Exception as e:
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logger.error("构建nerfreal失败，sessionid=%d: %s", sessionid, str(e))
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del nerfreals[sessionid]
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return web.Response(
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content_type="application/json",
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text=json.dumps({"code": -5, "msg": "构建会话失败"})
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)
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# 配置ICE服务器并创建RTCPeerConnection
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try:
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ice_server = RTCIceServer(urls='stun:stun.miwifi.com:3478')
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logger.debug("使用ICE服务器: %s", ice_server.urls)
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pc = RTCPeerConnection(configuration=RTCConfiguration(iceServers=[ice_server]))
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pcs.add(pc)
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logger.info("创建RTCPeerConnection成功，sessionid=%d", sessionid)
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except Exception as e:
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logger.error("创建RTCPeerConnection失败，sessionid=%d: %s", sessionid, str(e))
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del nerfreals[sessionid]
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return web.Response(
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content_type="application/json",
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text=json.dumps({"code": -6, "msg": "创建连接失败"})
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)
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# 连接状态变化处理
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@pc.on("connectionstatechange")
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async def on_connectionstatechange():
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logger.info("sessionid=%d, 连接状态变更为: %s", sessionid, pc.connectionState)
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if pc.connectionState == "failed":
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logger.warning("sessionid=%d, 连接失败，关闭连接", sessionid)
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await pc.close()
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pcs.discard(pc)
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del nerfreals[sessionid]
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if pc.connectionState == "closed":
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logger.info("sessionid=%d, 连接已关闭", sessionid)
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pcs.discard(pc)
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del nerfreals[sessionid]
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gc.collect()
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logger.debug("sessionid=%d, 已清理资源", sessionid)
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# 添加媒体轨道
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try:
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player = HumanPlayer(nerfreals[sessionid])
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audio_sender = pc.addTrack(player.audio)
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video_sender = pc.addTrack(player.video)
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logger.debug("sessionid=%d, 已添加音频和视频轨道", sessionid)
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except Exception as e:
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logger.error("sessionid=%d, 添加媒体轨道失败: %s", sessionid, str(e))
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return web.Response(
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content_type="application/json",
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text=json.dumps({"code": -7, "msg": "添加媒体轨道失败"})
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)
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# 配置编解码器偏好
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try:
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capabilities = RTCRtpSender.getCapabilities("video")
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preferences = list(filter(lambda x: x.name == "H264", capabilities.codecs))
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preferences += list(filter(lambda x: x.name == "VP8", capabilities.codecs))
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preferences += list(filter(lambda x: x.name == "rtx", capabilities.codecs))
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logger.debug("sessionid=%d, 编解码器偏好: %s", sessionid, [p.name for p in preferences])
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transceiver = pc.getTransceivers()[1]
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transceiver.setCodecPreferences(preferences)
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logger.debug("sessionid=%d, 已设置编解码器偏好", sessionid)
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except Exception as e:
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logger.error("sessionid=%d, 配置编解码器失败: %s", sessionid, str(e))
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# 处理offer并创建answer
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try:
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await pc.setRemoteDescription(offer)
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logger.debug("sessionid=%d, 已设置远程描述", sessionid)
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answer = await pc.createAnswer()
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await pc.setLocalDescription(answer)
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logger.debug("sessionid=%d, 已创建并设置本地answer", sessionid)
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except Exception as e:
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logger.error("sessionid=%d, 处理offer创建answer失败: %s", sessionid, str(e))
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return web.Response(
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content_type="application/json",
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text=json.dumps({"code": -8, "msg": "处理offer失败"})
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)
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# 返回结果
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response_data = {
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"sdp": pc.localDescription.sdp,
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"type": pc.localDescription.type,
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"sessionid": sessionid
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}
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logger.info("sessionid=%d, offer请求处理完成，返回响应", sessionid)
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return web.Response(
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content_type="application/json",
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text=json.dumps(response_data)
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)
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async def human(request):
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try:
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params = await request.json()
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sessionid = params.get('sessionid',0)
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if params.get('interrupt'):
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nerfreals[sessionid].flush_talk()
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if params['type']=='echo':
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nerfreals[sessionid].speaking = True
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nerfreals[sessionid].put_msg_txt(params['text'])
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elif params['type']=='chat':
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asyncio.get_event_loop().run_in_executor(None, llm_response, params['text'],nerfreals[sessionid])
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#nerfreals[sessionid].put_msg_txt(res)
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": 0, "msg":"ok"}
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),
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)
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except Exception as e:
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logger.exception('exception:')
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": -1, "msg": str(e)}
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),
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)
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async def interrupt_talk(request):
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try:
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params = await request.json()
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sessionid = params.get('sessionid',0)
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nerfreals[sessionid].flush_talk()
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": 0, "msg":"ok"}
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),
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)
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except Exception as e:
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logger.exception('exception:')
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": -1, "msg": str(e)}
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),
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)
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async def humanaudio(request):
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try:
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form= await request.post()
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sessionid = int(form.get('sessionid',0))
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fileobj = form["file"]
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filename=fileobj.filename
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filebytes=fileobj.file.read()
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nerfreals[sessionid].put_audio_file(filebytes)
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": 0, "msg":"ok"}
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),
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)
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except Exception as e:
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logger.exception('exception:')
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": -1, "msg": str(e)}
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),
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)
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async def set_audiotype(request):
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try:
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params = await request.json()
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sessionid = params.get('sessionid',0)
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nerfreals[sessionid].set_custom_state(params['audiotype'],params['reinit'])
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": 0, "msg":"ok"}
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),
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)
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except Exception as e:
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logger.exception('exception:')
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": -1, "msg": str(e)}
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),
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)
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async def record(request):
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try:
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params = await request.json()
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sessionid = params.get('sessionid',0)
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if params['type']=='start_record':
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# nerfreals[sessionid].put_msg_txt(params['text'])
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nerfreals[sessionid].start_recording()
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elif params['type']=='end_record':
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nerfreals[sessionid].stop_recording()
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": 0, "msg":"ok"}
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),
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)
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except Exception as e:
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logger.exception('exception:')
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": -1, "msg": str(e)}
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),
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)
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async def is_speaking(request):
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params = await request.json()
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sessionid = params.get('sessionid',0)
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return web.Response(
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content_type="application/json",
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text=json.dumps(
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{"code": 0, "data": nerfreals[sessionid].is_speaking()}
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),
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)
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async def on_shutdown(app):
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# close peer connections
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coros = [pc.close() for pc in pcs]
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await asyncio.gather(*coros)
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pcs.clear()
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# 关闭数据库连接
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global conn
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if conn:
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conn.close()
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print("[INFO] 已关闭数据库连接")
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async def post(url,data):
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try:
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async with aiohttp.ClientSession() as session:
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async with session.post(url,data=data) as response:
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return await response.text()
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except aiohttp.ClientError as e:
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logger.info(f'Error: {e}')
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async def run(push_url,sessionid):
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nerfreal = await asyncio.get_event_loop().run_in_executor(None, build_nerfreal,sessionid)
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nerfreals[sessionid] = nerfreal
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pc = RTCPeerConnection()
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pcs.add(pc)
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@pc.on("connectionstatechange")
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async def on_connectionstatechange():
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logger.info("Connection state is %s" % pc.connectionState)
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if pc.connectionState == "failed":
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await pc.close()
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pcs.discard(pc)
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player = HumanPlayer(nerfreals[sessionid])
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audio_sender = pc.addTrack(player.audio)
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video_sender = pc.addTrack(player.video)
|
|
|
|
|
|
await pc.setLocalDescription(await pc.createOffer())
|
|
|
answer = await post(push_url,pc.localDescription.sdp)
|
|
|
await pc.setRemoteDescription(RTCSessionDescription(sdp=answer,type='answer'))
|
|
|
##########################################
|
|
|
|
|
|
if __name__ == '__main__':
|
|
|
try:
|
|
|
db_path = os.path.join(os.path.dirname(os.path.abspath(__file__)), '..', '..', 'live_chat.db')
|
|
|
|
|
|
load_db_config(db_path)
|
|
|
|
|
|
mp.set_start_method('spawn')
|
|
|
parser = argparse.ArgumentParser()
|
|
|
parser.add_argument('--config', type=str, default='config/config.json', help="配置文件路径")
|
|
|
|
|
|
# audio FPS
|
|
|
parser.add_argument('--fps', type=int, default=50, help="audio fps,must be 50")
|
|
|
# sliding window left-middle-right length (unit: 20ms)
|
|
|
parser.add_argument('-l', type=int, default=10)
|
|
|
parser.add_argument('-m', type=int, default=8)
|
|
|
parser.add_argument('-r', type=int, default=10)
|
|
|
|
|
|
parser.add_argument('--W', type=int, default=450, help="GUI width")
|
|
|
parser.add_argument('--H', type=int, default=450, help="GUI height")
|
|
|
|
|
|
#musetalk opt
|
|
|
parser.add_argument('--avatar_id', type=str, default='model_4_1', help="define which avatar in data/avatars")
|
|
|
#parser.add_argument('--bbox_shift', type=int, default=5)
|
|
|
parser.add_argument('--batch_size', type=int, default=16, help="infer batch")
|
|
|
|
|
|
parser.add_argument('--customvideo_config', type=str, default='', help="custom action json")
|
|
|
|
|
|
parser.add_argument('--tts', type=str, default='gpt-sovits', help="tts service type") #xtts gpt-sovits cosyvoice
|
|
|
parser.add_argument('--REF_FILE', type=str, default="input/gentle_girl.wav")
|
|
|
parser.add_argument('--REF_TEXT', type=str, default="刚进直播间的宝子们，左上角先点个关注，点亮咱们家的粉丝灯牌！我是你们的主播陈婉婉，今天给大家准备了超级重磅的福利")
|
|
|
parser.add_argument('--TTS_SERVER', type=str, default='http://127.0.0.1:9880') # http://localhost:9000
|
|
|
# parser.add_argument('--CHARACTER', type=str, default='test')
|
|
|
# parser.add_argument('--EMOTION', type=str, default='default')
|
|
|
|
|
|
parser.add_argument('--model', type=str, default='wav2lip') #musetalk wav2lip ultralight
|
|
|
|
|
|
parser.add_argument('--transport', type=str, default='webrtc') #webrtc rtcpush virtualcam
|
|
|
parser.add_argument('--push_url', type=str, default='http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream') #rtmp://localhost/live/livestream
|
|
|
|
|
|
parser.add_argument('--max_session', type=int, default=1) #multi session count
|
|
|
parser.add_argument('--listenport', type=int, default=8010, help="web listen port")
|
|
|
|
|
|
opt = parser.parse_args()
|
|
|
#app.config.from_object(opt)
|
|
|
#print(app.config)
|
|
|
# —— 新增：数据库字段名到 opt 属性名的映射 ——
|
|
|
name_map = {
|
|
|
'listen_port': 'listenport',
|
|
|
'avatar_id': 'avatar_id',
|
|
|
'ref_file': 'REF_FILE',
|
|
|
'ref_text': 'REF_TEXT',
|
|
|
'tts_server': 'TTS_SERVER',
|
|
|
}
|
|
|
|
|
|
for db_key, db_val in config_dict.items():
|
|
|
if db_key not in name_map:
|
|
|
continue
|
|
|
arg_name = name_map[db_key]
|
|
|
# 仅在用户没有显式传参（仍为默认值）时才覆盖
|
|
|
current = getattr(opt, arg_name)
|
|
|
default = parser.get_default(arg_name)
|
|
|
if current == default:
|
|
|
try:
|
|
|
setattr(opt, arg_name, type(default)(db_val))
|
|
|
except Exception:
|
|
|
setattr(opt, arg_name, db_val)
|
|
|
try:
|
|
|
with open(opt.config, 'r', encoding='utf-8') as f:
|
|
|
cfg = json.load(f)
|
|
|
for key, val in cfg.items():
|
|
|
# 如果当前 opt.key 仍然是 parser 定义的默认值，就用配置文件里的
|
|
|
if getattr(opt, key, None) == parser.get_default(key):
|
|
|
setattr(opt, key, val)
|
|
|
except FileNotFoundError:
|
|
|
logger.warning(f"配置文件未找到：{opt.config}，将全部使用命令行/默认参数")
|
|
|
except Exception as e:
|
|
|
logger.warning(f"加载配置文件时出错：{e}，将全部使用命令行/默认参数")
|
|
|
|
|
|
opt.customopt = []
|
|
|
if opt.customvideo_config!='':
|
|
|
with open(opt.customvideo_config,'r') as file:
|
|
|
opt.customopt = json.load(file)
|
|
|
|
|
|
# if opt.model == 'ernerf':
|
|
|
# from nerfreal import NeRFReal,load_model,load_avatar
|
|
|
# model = load_model(opt)
|
|
|
# avatar = load_avatar(opt)
|
|
|
if opt.model == 'wav2lip':
|
|
|
from lipreal import LipReal,load_model,load_avatar,warm_up
|
|
|
logger.info(opt)
|
|
|
model = load_model("./models/wav2lip384.pth")
|
|
|
avatar = load_avatar(opt.avatar_id)
|
|
|
warm_up(opt.batch_size,model,384)
|
|
|
elif opt.model == 'ultralight':
|
|
|
from lightreal import LightReal,load_model,load_avatar,warm_up
|
|
|
logger.info(opt)
|
|
|
model = load_model(opt)
|
|
|
avatar = load_avatar(opt.avatar_id)
|
|
|
warm_up(opt.batch_size,avatar,160)
|
|
|
|
|
|
if opt.transport=='virtualcam':
|
|
|
thread_quit = Event()
|
|
|
nerfreals[0] = build_nerfreal(0)
|
|
|
rendthrd = Thread(target=nerfreals[0].render,args=(thread_quit,))
|
|
|
rendthrd.start()
|
|
|
|
|
|
#############################################################################
|
|
|
appasync = web.Application(client_max_size=1024**2*100)
|
|
|
appasync.on_shutdown.append(on_shutdown)
|
|
|
appasync.router.add_post("/offer", offer)
|
|
|
appasync.router.add_post("/human", human)
|
|
|
appasync.router.add_post("/humanaudio", humanaudio)
|
|
|
appasync.router.add_post("/set_audiotype", set_audiotype)
|
|
|
appasync.router.add_post("/record", record)
|
|
|
appasync.router.add_post("/interrupt_talk", interrupt_talk)
|
|
|
appasync.router.add_post("/is_speaking", is_speaking)
|
|
|
appasync.router.add_static('/',path='web')
|
|
|
|
|
|
# Configure default CORS settings.
|
|
|
cors = aiohttp_cors.setup(appasync, defaults={
|
|
|
"*": aiohttp_cors.ResourceOptions(
|
|
|
allow_credentials=True,
|
|
|
expose_headers="*",
|
|
|
allow_headers="*",
|
|
|
)
|
|
|
})
|
|
|
# Configure CORS on all routes.
|
|
|
for route in list(appasync.router.routes()):
|
|
|
cors.add(route)
|
|
|
|
|
|
pagename='webrtcapi.html'
|
|
|
if opt.transport=='rtmp':
|
|
|
pagename='echoapi.html'
|
|
|
elif opt.transport=='rtcpush':
|
|
|
pagename='rtcpushapi.html'
|
|
|
logger.info('start http server; http://<serverip>:'+str(opt.listenport)+'/'+pagename)
|
|
|
logger.info('如果使用webrtc，推荐访问webrtc集成前端: http://<serverip>:'+str(opt.listenport)+'/dashboard.html')
|
|
|
# 服务开启，将数据库中enable_status 对应的 value 字段改为 '1'
|
|
|
set_enable_status(db_path)
|
|
|
def run_server(runner):
|
|
|
loop = asyncio.new_event_loop()
|
|
|
asyncio.set_event_loop(loop)
|
|
|
loop.run_until_complete(runner.setup())
|
|
|
site = web.TCPSite(runner, '0.0.0.0', opt.listenport)
|
|
|
loop.run_until_complete(site.start())
|
|
|
if opt.transport=='rtcpush':
|
|
|
for k in range(opt.max_session):
|
|
|
push_url = opt.push_url
|
|
|
if k!=0:
|
|
|
push_url = opt.push_url+str(k)
|
|
|
loop.run_until_complete(run(push_url,k))
|
|
|
loop.run_forever()
|
|
|
run_server(web.AppRunner(appasync))
|
|
|
except Exception:
|
|
|
import traceback
|
|
|
traceback.print_exc() # 打印完整的错误堆栈
|
|
|
input("发生异常，按回车键退出…") # 等待用户按回车再退出 |