livetalking/app.py

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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
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
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from threading import Thread,Event
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import multiprocessing
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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
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from aiortc.rtcrtpsender import RTCRtpSender
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from webrtc import HumanPlayer
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import argparse
import shutil
import asyncio
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import string
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app = Flask(__name__)
sockets = Sockets(app)
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nerfreals = []
statreals = []
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@sockets.route('/humanecho')
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def echo_socket(ws):
# 获取WebSocket对象
#ws = request.environ.get('wsgi.websocket')
# 如果没有获取到,返回错误信息
if not ws:
print('未建立连接!')
return 'Please use WebSocket'
# 否则,循环接收和发送消息
else:
print('建立连接!')
while True:
message = ws.receive()
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if not message or len(message)==0:
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return '输入信息为空'
else:
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nerfreal.put_msg_txt(message)
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# 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)
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@sockets.route('/humanchat')
def chat_socket(ws):
# 获取WebSocket对象
#ws = request.environ.get('wsgi.websocket')
# 如果没有获取到,返回错误信息
if not ws:
print('未建立连接!')
return 'Please use WebSocket'
# 否则,循环接收和发送消息
else:
print('建立连接!')
while True:
message = ws.receive()
if len(message)==0:
return '输入信息为空'
else:
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res=llm_response(message)
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nerfreal.put_msg_txt(res)
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#####webrtc###############################
pcs = set()
#@app.route('/offer', methods=['POST'])
async def offer(request):
params = await request.json()
offer = RTCSessionDescription(sdp=params["sdp"], type=params["type"])
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sessionid = len(nerfreals)
for index,value in enumerate(statreals):
if value == 0:
sessionid = index
break
if sessionid>=len(nerfreals):
print('reach max session')
return -1
statreals[sessionid] = 1
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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)
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statreals[sessionid] = 0
if pc.connectionState == "closed":
pcs.discard(pc)
statreals[sessionid] = 0
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player = HumanPlayer(nerfreals[sessionid])
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audio_sender = pc.addTrack(player.audio)
video_sender = pc.addTrack(player.video)
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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)
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await pc.setRemoteDescription(offer)
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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(
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{"sdp": pc.localDescription.sdp, "type": pc.localDescription.type, "sessionid":sessionid}
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),
)
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async def human(request):
params = await request.json()
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sessionid = params.get('sessionid',0)
if params.get('interrupt'):
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nerfreals[sessionid].pause_talk()
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if params['type']=='echo':
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nerfreals[sessionid].put_msg_txt(params['text'])
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elif params['type']=='chat':
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res=await asyncio.get_event_loop().run_in_executor(None, llm_response, params['text'],nerfreals[sessionid])
#nerfreals[sessionid].put_msg_txt(res)
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return web.Response(
content_type="application/json",
text=json.dumps(
{"code": 0, "data":"ok"}
),
)
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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]+""}
),
)
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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"}
),
)
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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'])
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nerfreals[sessionid].start_recording("data/record_lasted.mp4")
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elif params['type']=='end_record':
nerfreals[sessionid].stop_recording()
return web.Response(
content_type="application/json",
text=json.dumps(
{"code": 0, "data":"ok"}
),
)
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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()}
),
)
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async def on_shutdown(app):
# close peer connections
coros = [pc.close() for pc in pcs]
await asyncio.gather(*coros)
pcs.clear()
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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):
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)
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player = HumanPlayer(nerfreals[0])
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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'))
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##########################################
# os.environ['MKL_SERVICE_FORCE_INTEL'] = '1'
# os.environ['MULTIPROCESSING_METHOD'] = 'forkserver'
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if __name__ == '__main__':
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multiprocessing.set_start_method('spawn')
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parser = argparse.ArgumentParser()
parser.add_argument('--pose', type=str, default="data/data_kf.json", help="transforms.json, pose source")
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parser.add_argument('--au', type=str, default="data/au.csv", help="eye blink area")
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parser.add_argument('--torso_imgs', type=str, default="", help="torso images path")
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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')
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parser.add_argument('--asr_model', type=str, default='cpierse/wav2vec2-large-xlsr-53-esperanto') #
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# parser.add_argument('--asr_model', type=str, default='facebook/wav2vec2-large-960h-lv60-self')
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# parser.add_argument('--asr_model', type=str, default='facebook/hubert-large-ls960-ft')
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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)
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parser.add_argument('-r', type=int, default=10)
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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)
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#musetalk opt
parser.add_argument('--avatar_id', type=str, default='avator_1')
parser.add_argument('--bbox_shift', type=int, default=5)
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parser.add_argument('--batch_size', type=int, default=16)
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# 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)
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parser.add_argument('--customvideo_config', type=str, default='')
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parser.add_argument('--tts', type=str, default='edgetts') #xtts gpt-sovits cosyvoice
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parser.add_argument('--REF_FILE', type=str, default=None)
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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')
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parser.add_argument('--model', type=str, default='ernerf') #musetalk wav2lip
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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
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parser.add_argument('--max_session', type=int, default=1) #multi session count
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parser.add_argument('--listenport', type=int, default=8010)
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opt = parser.parse_args()
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#app.config.from_object(opt)
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#print(app.config)
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opt.customopt = []
if opt.customvideo_config!='':
with open(opt.customvideo_config,'r') as file:
opt.customopt = json.load(file)
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if opt.model == 'ernerf':
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from ernerf.nerf_triplane.provider import NeRFDataset_Test
from ernerf.nerf_triplane.utils import *
from ernerf.nerf_triplane.network import NeRFNetwork
from nerfreal import NeRFReal
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# assert test mode
opt.test = True
opt.test_train = False
#opt.train_camera =True
# explicit smoothing
opt.smooth_path = True
opt.smooth_lips = True
assert opt.pose != '', 'Must provide a pose source'
# if opt.O:
opt.fp16 = True
opt.cuda_ray = True
opt.exp_eye = True
opt.smooth_eye = True
if opt.torso_imgs=='': #no img,use model output
opt.torso = True
# assert opt.cuda_ray, "Only support CUDA ray mode."
opt.asr = True
if opt.patch_size > 1:
# assert opt.patch_size > 16, "patch_size should > 16 to run LPIPS loss."
assert opt.num_rays % (opt.patch_size ** 2) == 0, "patch_size ** 2 should be dividable by num_rays."
seed_everything(opt.seed)
print(opt)
device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
model = NeRFNetwork(opt)
criterion = torch.nn.MSELoss(reduction='none')
metrics = [] # use no metric in GUI for faster initialization...
print(model)
trainer = Trainer('ngp', opt, model, device=device, workspace=opt.workspace, criterion=criterion, fp16=opt.fp16, metrics=metrics, use_checkpoint=opt.ckpt)
test_loader = NeRFDataset_Test(opt, device=device).dataloader()
model.aud_features = test_loader._data.auds
model.eye_areas = test_loader._data.eye_area
# we still need test_loader to provide audio features for testing.
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for _ in range(opt.max_session):
nerfreal = NeRFReal(opt, trainer, test_loader)
nerfreals.append(nerfreal)
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elif opt.model == 'musetalk':
from musereal import MuseReal
print(opt)
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for _ in range(opt.max_session):
nerfreal = MuseReal(opt)
nerfreals.append(nerfreal)
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elif opt.model == 'wav2lip':
from lipreal import LipReal
print(opt)
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for _ in range(opt.max_session):
nerfreal = LipReal(opt)
nerfreals.append(nerfreal)
for _ in range(opt.max_session):
statreals.append(0)
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if opt.transport=='rtmp':
thread_quit = Event()
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rendthrd = Thread(target=nerfreals[0].render,args=(thread_quit,))
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rendthrd.start()
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#############################################################################
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appasync = web.Application()
appasync.on_shutdown.append(on_shutdown)
appasync.router.add_post("/offer", offer)
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appasync.router.add_post("/human", human)
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appasync.router.add_post("/humanaudio", humanaudio)
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appasync.router.add_post("/set_audiotype", set_audiotype)
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appasync.router.add_post("/record", record)
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appasync.router.add_post("/is_speaking", is_speaking)
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appasync.router.add_static('/',path='web')
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# 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)
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pagename='webrtcapi.html'
if opt.transport=='rtmp':
pagename='echoapi.html'
elif opt.transport=='rtcpush':
pagename='rtcpushapi.html'
print('start http server; http://<serverip>:'+str(opt.listenport)+'/'+pagename)
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def run_server(runner):
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
loop.run_until_complete(runner.setup())
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site = web.TCPSite(runner, '0.0.0.0', opt.listenport)
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loop.run_until_complete(site.start())
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if opt.transport=='rtcpush':
loop.run_until_complete(run(opt.push_url))
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loop.run_forever()
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#Thread(target=run_server, args=(web.AppRunner(appasync),)).start()
run_server(web.AppRunner(appasync))
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#app.on_shutdown.append(on_shutdown)
#app.router.add_post("/offer", offer)
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# print('start websocket server')
# server = pywsgi.WSGIServer(('0.0.0.0', 8000), app, handler_class=WebSocketHandler)
# server.serve_forever()
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