############################################################################### # Copyright (C) 2024 LiveTalking@lipku https://github.com/lipku/LiveTalking # email: lipku@foxmail.com # # Licensed under the Apache License, Version 2.0 (the "License"); # you may not use this file except in compliance with the License. # You may obtain a copy of the License at # # http://www.apache.org/licenses/LICENSE-2.0 # # Unless required by applicable law or agreed to in writing, software # distributed under the License is distributed on an "AS IS" BASIS, # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. # See the License for the specific language governing permissions and # limitations under the License. ############################################################################### # server.py from flask import Flask, render_template,send_from_directory,request, jsonify from flask_sockets import Sockets import base64 import time import json #import gevent #from gevent import pywsgi #from geventwebsocket.handler import WebSocketHandler import os import re import numpy as np from threading import Thread,Event #import multiprocessing import torch.multiprocessing as mp from aiohttp import web import aiohttp import aiohttp_cors from aiortc import RTCPeerConnection, RTCSessionDescription from aiortc.rtcrtpsender import RTCRtpSender from webrtc import HumanPlayer import argparse import random import shutil import asyncio import torch app = Flask(__name__) #sockets = Sockets(app) nerfreals = {} opt = None model = None avatar = None # def llm_response(message): # from llm.LLM import LLM # # llm = LLM().init_model('Gemini', model_path= 'gemini-pro',api_key='Your API Key', proxy_url=None) # # llm = LLM().init_model('ChatGPT', model_path= 'gpt-3.5-turbo',api_key='Your API Key') # llm = LLM().init_model('VllmGPT', model_path= 'THUDM/chatglm3-6b') # response = llm.chat(message) # print(response) # return response def llm_response(message,nerfreal): start = time.perf_counter() from openai import OpenAI client = OpenAI( # 如果您没有配置环境变量,请在此处用您的API Key进行替换 api_key=os.getenv("DASHSCOPE_API_KEY"), # 填写DashScope SDK的base_url base_url="https://dashscope.aliyuncs.com/compatible-mode/v1", ) end = time.perf_counter() print(f"llm Time init: {end-start}s") completion = client.chat.completions.create( model="qwen-plus", messages=[{'role': 'system', 'content': 'You are a helpful assistant.'}, {'role': 'user', 'content': message}], stream=True, # 通过以下设置,在流式输出的最后一行展示token使用信息 stream_options={"include_usage": True} ) result="" first = True for chunk in completion: if len(chunk.choices)>0: #print(chunk.choices[0].delta.content) if first: end = time.perf_counter() print(f"llm Time to first chunk: {end-start}s") first = False msg = chunk.choices[0].delta.content lastpos=0 #msglist = re.split('[,.!;:,。!?]',msg) for i, char in enumerate(msg): if char in ",.!;:,。!?:;" : result = result+msg[lastpos:i+1] lastpos = i+1 if len(result)>10: print(result) nerfreal.put_msg_txt(result) result="" result = result+msg[lastpos:] end = time.perf_counter() print(f"llm Time to last chunk: {end-start}s") nerfreal.put_msg_txt(result) #####webrtc############################### pcs = set() def randN(N): '''生成长度为 N的随机数 ''' min = pow(10, N - 1) max = pow(10, N) return random.randint(min, max - 1) def build_nerfreal(sessionid): opt.sessionid=sessionid if opt.model == 'wav2lip': from lipreal import LipReal nerfreal = LipReal(opt,model,avatar) elif opt.model == 'musetalk': from musereal import MuseReal nerfreal = MuseReal(opt,model,avatar) elif opt.model == 'ernerf': from nerfreal import NeRFReal nerfreal = NeRFReal(opt,model,avatar) elif opt.model == 'ultralight': from lightreal import LightReal nerfreal = LightReal(opt,model,avatar) return nerfreal #@app.route('/offer', methods=['POST']) async def offer(request): params = await request.json() offer = RTCSessionDescription(sdp=params["sdp"], type=params["type"]) if len(nerfreals) >= opt.max_session: print('reach max session') return -1 sessionid = randN(6) #len(nerfreals) print('sessionid=',sessionid) nerfreals[sessionid] = None nerfreal = await asyncio.get_event_loop().run_in_executor(None, build_nerfreal,sessionid) nerfreals[sessionid] = nerfreal pc = RTCPeerConnection() pcs.add(pc) @pc.on("connectionstatechange") async def on_connectionstatechange(): print("Connection state is %s" % pc.connectionState) if pc.connectionState == "failed": await pc.close() pcs.discard(pc) del nerfreals[sessionid] if pc.connectionState == "closed": pcs.discard(pc) del nerfreals[sessionid] player = HumanPlayer(nerfreals[sessionid]) audio_sender = pc.addTrack(player.audio) video_sender = pc.addTrack(player.video) capabilities = RTCRtpSender.getCapabilities("video") preferences = list(filter(lambda x: x.name == "H264", capabilities.codecs)) preferences += list(filter(lambda x: x.name == "VP8", capabilities.codecs)) preferences += list(filter(lambda x: x.name == "rtx", capabilities.codecs)) transceiver = pc.getTransceivers()[1] transceiver.setCodecPreferences(preferences) await pc.setRemoteDescription(offer) answer = await pc.createAnswer() await pc.setLocalDescription(answer) #return jsonify({"sdp": pc.localDescription.sdp, "type": pc.localDescription.type}) return web.Response( content_type="application/json", text=json.dumps( {"sdp": pc.localDescription.sdp, "type": pc.localDescription.type, "sessionid":sessionid} ), ) async def human(request): params = await request.json() sessionid = params.get('sessionid',0) if params.get('interrupt'): nerfreals[sessionid].flush_talk() if params['type']=='echo': nerfreals[sessionid].put_msg_txt(params['text']) elif params['type']=='chat': res=await asyncio.get_event_loop().run_in_executor(None, llm_response, params['text'],nerfreals[sessionid]) #nerfreals[sessionid].put_msg_txt(res) return web.Response( content_type="application/json", text=json.dumps( {"code": 0, "data":"ok"} ), ) async def humanaudio(request): try: form= await request.post() sessionid = int(form.get('sessionid',0)) fileobj = form["file"] filename=fileobj.filename filebytes=fileobj.file.read() nerfreals[sessionid].put_audio_file(filebytes) return web.Response( content_type="application/json", text=json.dumps( {"code": 0, "msg":"ok"} ), ) except Exception as e: return web.Response( content_type="application/json", text=json.dumps( {"code": -1, "msg":"err","data": ""+e.args[0]+""} ), ) async def set_audiotype(request): params = await request.json() sessionid = params.get('sessionid',0) nerfreals[sessionid].set_curr_state(params['audiotype'],params['reinit']) return web.Response( content_type="application/json", text=json.dumps( {"code": 0, "data":"ok"} ), ) async def record(request): params = await request.json() sessionid = params.get('sessionid',0) if params['type']=='start_record': # nerfreals[sessionid].put_msg_txt(params['text']) nerfreals[sessionid].start_recording() elif params['type']=='end_record': nerfreals[sessionid].stop_recording() return web.Response( content_type="application/json", text=json.dumps( {"code": 0, "data":"ok"} ), ) async def is_speaking(request): params = await request.json() sessionid = params.get('sessionid',0) return web.Response( content_type="application/json", text=json.dumps( {"code": 0, "data": nerfreals[sessionid].is_speaking()} ), ) async def on_shutdown(app): # close peer connections coros = [pc.close() for pc in pcs] await asyncio.gather(*coros) pcs.clear() async def post(url,data): try: async with aiohttp.ClientSession() as session: async with session.post(url,data=data) as response: return await response.text() except aiohttp.ClientError as e: print(f'Error: {e}') async def run(push_url,sessionid): nerfreal = await asyncio.get_event_loop().run_in_executor(None, build_nerfreal,sessionid) nerfreals[sessionid] = nerfreal pc = RTCPeerConnection() pcs.add(pc) @pc.on("connectionstatechange") async def on_connectionstatechange(): print("Connection state is %s" % pc.connectionState) if pc.connectionState == "failed": await pc.close() pcs.discard(pc) player = HumanPlayer(nerfreals[sessionid]) audio_sender = pc.addTrack(player.audio) 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')) ########################################## # os.environ['MKL_SERVICE_FORCE_INTEL'] = '1' # os.environ['MULTIPROCESSING_METHOD'] = 'forkserver' if __name__ == '__main__': mp.set_start_method('spawn') parser = argparse.ArgumentParser() parser.add_argument('--pose', type=str, default="data/data_kf.json", help="transforms.json, pose source") parser.add_argument('--au', type=str, default="data/au.csv", help="eye blink area") parser.add_argument('--torso_imgs', type=str, default="", help="torso images path") parser.add_argument('-O', action='store_true', help="equals --fp16 --cuda_ray --exp_eye") parser.add_argument('--data_range', type=int, nargs='*', default=[0, -1], help="data range to use") parser.add_argument('--workspace', type=str, default='data/video') parser.add_argument('--seed', type=int, default=0) ### training options parser.add_argument('--ckpt', type=str, default='data/pretrained/ngp_kf.pth') parser.add_argument('--num_rays', type=int, default=4096 * 16, help="num rays sampled per image for each training step") parser.add_argument('--cuda_ray', action='store_true', help="use CUDA raymarching instead of pytorch") parser.add_argument('--max_steps', type=int, default=16, help="max num steps sampled per ray (only valid when using --cuda_ray)") parser.add_argument('--num_steps', type=int, default=16, help="num steps sampled per ray (only valid when NOT using --cuda_ray)") parser.add_argument('--upsample_steps', type=int, default=0, help="num steps up-sampled per ray (only valid when NOT using --cuda_ray)") parser.add_argument('--update_extra_interval', type=int, default=16, help="iter interval to update extra status (only valid when using --cuda_ray)") parser.add_argument('--max_ray_batch', type=int, default=4096, help="batch size of rays at inference to avoid OOM (only valid when NOT using --cuda_ray)") ### loss set parser.add_argument('--warmup_step', type=int, default=10000, help="warm up steps") parser.add_argument('--amb_aud_loss', type=int, default=1, help="use ambient aud loss") parser.add_argument('--amb_eye_loss', type=int, default=1, help="use ambient eye loss") parser.add_argument('--unc_loss', type=int, default=1, help="use uncertainty loss") parser.add_argument('--lambda_amb', type=float, default=1e-4, help="lambda for ambient loss") ### network backbone options parser.add_argument('--fp16', action='store_true', help="use amp mixed precision training") parser.add_argument('--bg_img', type=str, default='white', help="background image") parser.add_argument('--fbg', action='store_true', help="frame-wise bg") parser.add_argument('--exp_eye', action='store_true', help="explicitly control the eyes") parser.add_argument('--fix_eye', type=float, default=-1, help="fixed eye area, negative to disable, set to 0-0.3 for a reasonable eye") parser.add_argument('--smooth_eye', action='store_true', help="smooth the eye area sequence") parser.add_argument('--torso_shrink', type=float, default=0.8, help="shrink bg coords to allow more flexibility in deform") ### dataset options parser.add_argument('--color_space', type=str, default='srgb', help="Color space, supports (linear, srgb)") parser.add_argument('--preload', type=int, default=0, help="0 means load data from disk on-the-fly, 1 means preload to CPU, 2 means GPU.") # (the default value is for the fox dataset) parser.add_argument('--bound', type=float, default=1, help="assume the scene is bounded in box[-bound, bound]^3, if > 1, will invoke adaptive ray marching.") parser.add_argument('--scale', type=float, default=4, help="scale camera location into box[-bound, bound]^3") parser.add_argument('--offset', type=float, nargs='*', default=[0, 0, 0], help="offset of camera location") parser.add_argument('--dt_gamma', type=float, default=1/256, help="dt_gamma (>=0) for adaptive ray marching. set to 0 to disable, >0 to accelerate rendering (but usually with worse quality)") parser.add_argument('--min_near', type=float, default=0.05, help="minimum near distance for camera") parser.add_argument('--density_thresh', type=float, default=10, help="threshold for density grid to be occupied (sigma)") parser.add_argument('--density_thresh_torso', type=float, default=0.01, help="threshold for density grid to be occupied (alpha)") parser.add_argument('--patch_size', type=int, default=1, help="[experimental] render patches in training, so as to apply LPIPS loss. 1 means disabled, use [64, 32, 16] to enable") parser.add_argument('--init_lips', action='store_true', help="init lips region") parser.add_argument('--finetune_lips', action='store_true', help="use LPIPS and landmarks to fine tune lips region") parser.add_argument('--smooth_lips', action='store_true', help="smooth the enc_a in a exponential decay way...") parser.add_argument('--torso', action='store_true', help="fix head and train torso") parser.add_argument('--head_ckpt', type=str, default='', help="head model") ### GUI options parser.add_argument('--gui', action='store_true', help="start a GUI") parser.add_argument('--W', type=int, default=450, help="GUI width") parser.add_argument('--H', type=int, default=450, help="GUI height") parser.add_argument('--radius', type=float, default=3.35, help="default GUI camera radius from center") parser.add_argument('--fovy', type=float, default=21.24, help="default GUI camera fovy") parser.add_argument('--max_spp', type=int, default=1, help="GUI rendering max sample per pixel") ### else parser.add_argument('--att', type=int, default=2, help="audio attention mode (0 = turn off, 1 = left-direction, 2 = bi-direction)") parser.add_argument('--aud', type=str, default='', help="audio source (empty will load the default, else should be a path to a npy file)") parser.add_argument('--emb', action='store_true', help="use audio class + embedding instead of logits") parser.add_argument('--ind_dim', type=int, default=4, help="individual code dim, 0 to turn off") parser.add_argument('--ind_num', type=int, default=10000, help="number of individual codes, should be larger than training dataset size") parser.add_argument('--ind_dim_torso', type=int, default=8, help="individual code dim, 0 to turn off") parser.add_argument('--amb_dim', type=int, default=2, help="ambient dimension") parser.add_argument('--part', action='store_true', help="use partial training data (1/10)") parser.add_argument('--part2', action='store_true', help="use partial training data (first 15s)") parser.add_argument('--train_camera', action='store_true', help="optimize camera pose") parser.add_argument('--smooth_path', action='store_true', help="brute-force smooth camera pose trajectory with a window size") parser.add_argument('--smooth_path_window', type=int, default=7, help="smoothing window size") # asr parser.add_argument('--asr', action='store_true', help="load asr for real-time app") parser.add_argument('--asr_wav', type=str, default='', help="load the wav and use as input") parser.add_argument('--asr_play', action='store_true', help="play out the audio") #parser.add_argument('--asr_model', type=str, default='deepspeech') parser.add_argument('--asr_model', type=str, default='cpierse/wav2vec2-large-xlsr-53-esperanto') # # parser.add_argument('--asr_model', type=str, default='facebook/wav2vec2-large-960h-lv60-self') # parser.add_argument('--asr_model', type=str, default='facebook/hubert-large-ls960-ft') parser.add_argument('--asr_save_feats', action='store_true') # audio FPS parser.add_argument('--fps', type=int, default=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('--fullbody', action='store_true', help="fullbody human") parser.add_argument('--fullbody_img', type=str, default='data/fullbody/img') parser.add_argument('--fullbody_width', type=int, default=580) parser.add_argument('--fullbody_height', type=int, default=1080) parser.add_argument('--fullbody_offset_x', type=int, default=0) parser.add_argument('--fullbody_offset_y', type=int, default=0) #musetalk opt parser.add_argument('--avatar_id', type=str, default='avator_1') parser.add_argument('--bbox_shift', type=int, default=5) parser.add_argument('--batch_size', type=int, default=16) # parser.add_argument('--customvideo', action='store_true', help="custom video") # parser.add_argument('--customvideo_img', type=str, default='data/customvideo/img') # parser.add_argument('--customvideo_imgnum', type=int, default=1) parser.add_argument('--customvideo_config', type=str, default='') parser.add_argument('--tts', type=str, default='edgetts') #xtts gpt-sovits cosyvoice parser.add_argument('--REF_FILE', type=str, default=None) parser.add_argument('--REF_TEXT', type=str, default=None) 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='ernerf') #musetalk wav2lip parser.add_argument('--transport', type=str, default='rtcpush') #rtmp webrtc rtcpush 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) opt = parser.parse_args() #app.config.from_object(opt) #print(app.config) 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) # we still need test_loader to provide audio features for testing. # for k in range(opt.max_session): # opt.sessionid=k # nerfreal = NeRFReal(opt, trainer, test_loader,audio_processor,audio_model) # nerfreals.append(nerfreal) elif opt.model == 'musetalk': from musereal import MuseReal,load_model,load_avatar,warm_up print(opt) model = load_model() avatar = load_avatar(opt.avatar_id) warm_up(opt.batch_size,model) # for k in range(opt.max_session): # opt.sessionid=k # nerfreal = MuseReal(opt,audio_processor,vae, unet, pe,timesteps) # nerfreals.append(nerfreal) elif opt.model == 'wav2lip': from lipreal import LipReal,load_model,load_avatar,warm_up print(opt) model = load_model("./models/wav2lip.pth") avatar = load_avatar(opt.avatar_id) warm_up(opt.batch_size,model,256) # for k in range(opt.max_session): # opt.sessionid=k # nerfreal = LipReal(opt,model) # nerfreals.append(nerfreal) elif opt.model == 'ultralight': from lightreal import LightReal,load_model,load_avatar,warm_up print(opt) model = load_model(opt) avatar = load_avatar(opt.avatar_id) warm_up(opt.batch_size,avatar,160) if opt.transport=='rtmp': thread_quit = Event() nerfreals[0] = build_nerfreal(0) rendthrd = Thread(target=nerfreals[0].render,args=(thread_quit,)) rendthrd.start() ############################################################################# appasync = web.Application() 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("/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' print('start http server; http://:'+str(opt.listenport)+'/'+pagename) 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() #Thread(target=run_server, args=(web.AppRunner(appasync),)).start() run_server(web.AppRunner(appasync)) #app.on_shutdown.append(on_shutdown) #app.router.add_post("/offer", offer) # print('start websocket server') # server = pywsgi.WSGIServer(('0.0.0.0', 8000), app, handler_class=WebSocketHandler) # server.serve_forever()