diff --git a/app.py b/app.py
index 85130cf..f665c7c 100644
--- a/app.py
+++ b/app.py
@@ -127,6 +127,9 @@ def build_nerfreal(sessionid):
     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'])
@@ -480,6 +483,12 @@ if __name__ == '__main__':
         #     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,model,160)
 
     if opt.transport=='rtmp':
         thread_quit = Event()
@@ -539,4 +548,4 @@ if __name__ == '__main__':
     # server = pywsgi.WSGIServer(('0.0.0.0', 8000), app, handler_class=WebSocketHandler)
     # server.serve_forever()
     
-    
\ No newline at end of file
+    
diff --git a/lightasr.py b/lightasr.py
new file mode 100644
index 0000000..08020f3
--- /dev/null
+++ b/lightasr.py
@@ -0,0 +1,34 @@
+import time
+import torch
+import numpy as np
+from baseasr import BaseASR
+
+
+class LightASR(BaseASR):
+    def __init__(self, opt, parent, audio_processor):
+        super().__init__(opt, parent)
+        self.audio_processor = audio_processor
+        self.stride_left_size = 32
+        self.stride_right_size = 32
+
+
+    def run_step(self):
+        start_time = time.time()
+        
+        for _ in range(self.batch_size * 2):
+            audio_frame, type_ = self.get_audio_frame()
+            self.frames.append(audio_frame)
+            self.output_queue.put((audio_frame, type_))
+        
+        if len(self.frames) <= self.stride_left_size + self.stride_right_size:
+            return
+        
+        inputs = np.concatenate(self.frames)  # [N * chunk]
+
+        mel = self.audio_processor.get_hubert_from_16k_speech(inputs)
+        mel_chunks=self.audio_processor.feature2chunks(feature_array=mel,fps=self.fps/2,batch_size=self.batch_size,start=self.stride_left_size/2)
+
+        self.feat_queue.put(mel_chunks)
+        self.frames = self.frames[-(self.stride_left_size + self.stride_right_size):]
+        print(f"Processing audio costs {(time.time() - start_time) * 1000}ms")
+
diff --git a/lightreal.py b/lightreal.py
new file mode 100644
index 0000000..61e02c4
--- /dev/null
+++ b/lightreal.py
@@ -0,0 +1,350 @@
+###############################################################################
+#  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.
+###############################################################################
+
+import math
+import torch
+import numpy as np
+
+#from .utils import *
+import os
+import time
+import cv2
+import glob
+import pickle
+import copy
+
+import queue
+from queue import Queue
+from threading import Thread, Event
+import torch.multiprocessing as mp
+
+
+from lightasr import LightASR
+import asyncio
+from av import AudioFrame, VideoFrame
+from basereal import BaseReal
+
+#from imgcache import ImgCache
+
+from tqdm import tqdm
+
+#new
+import os
+import cv2
+import torch
+import numpy as np
+import torch.nn as nn
+from torch import optim
+from tqdm import tqdm
+from transformers import Wav2Vec2Processor, HubertModel
+from torch.utils.data import DataLoader
+from ultralight.unet import Model
+from ultralight.audio2feature import Audio2Feature
+
+
+
+device = 'cuda' if torch.cuda.is_available() else 'cpu'
+print('Using {} for inference.'.format(device))
+
+
+
+def load_model(opt):
+    audio_processor = Audio2Feature()
+    model = Model(6, 'hubert').to(device)  # 假设Model是你自定义的类
+    model.load_state_dict(torch.load('./models/ultralight.pth'))
+    model.eval()
+
+    return model,audio_processor
+
+def load_avatar(avatar_id):
+    avatar_path = f"./data/avatars/{avatar_id}"
+    full_imgs_path = f"{avatar_path}/full_body_img" 
+    land_marks_path = f"{avatar_path}/landmarks" 
+    
+    input_img_list = glob.glob(os.path.join(full_imgs_path, '*.[jpJP][pnPN]*[gG]'))
+    input_img_list = sorted(input_img_list, key=lambda x: int(os.path.splitext(os.path.basename(x))[0]))
+    frame_list_cycle = read_imgs(input_img_list)
+    #self.imagecache = ImgCache(len(self.coord_list_cycle),self.full_imgs_path,1000)
+    land_marks_list = glob.glob(os.path.join(land_marks_path, '*.lms'))
+    land_marks_list = sorted(land_marks_list, key=lambda x: int(os.path.splitext(os.path.basename(x))[0]))
+    lms_list_cycle = read_lms(land_marks_list)
+    lms_list_cycle = np.array(lms_list_cycle, dtype=np.int32)
+    return frame_list_cycle,lms_list_cycle
+
+
+@torch.no_grad()
+def warm_up(batch_size,model,modelres):
+    # ?~D?~C??~G??~U?
+    print('warmup model...')
+    model1, audio_processor = model
+    img_batch = torch.ones(batch_size, 6, modelres, modelres).to(device)
+    mel_batch = torch.ones(batch_size, 32, 32, 32).to(device)
+    model1(img_batch, mel_batch)
+
+def read_imgs(img_list):
+    frames = []
+    print('reading images...')
+    for img_path in tqdm(img_list):
+        frame = cv2.imread(img_path)
+        frames.append(frame)
+    return frames
+
+def get_audio_features(features, index):
+    left = index - 8
+    right = index + 8
+    pad_left = 0
+    pad_right = 0
+    if left < 0:
+        pad_left = -left
+        left = 0
+    if right > features.shape[0]:
+        pad_right = right - features.shape[0]
+        right = features.shape[0]
+    auds = torch.from_numpy(features[left:right])
+    if pad_left > 0:
+        auds = torch.cat([torch.zeros_like(auds[:pad_left]), auds], dim=0)
+    if pad_right > 0:
+        auds = torch.cat([auds, torch.zeros_like(auds[:pad_right])], dim=0) # [8, 16]
+    return auds
+
+
+def read_lms(lms_list):
+    land_marks = []
+    print('reading lms...')
+    for lms_path in tqdm(lms_list):
+        file_landmarks = []  # Store landmarks for this file
+        with open(lms_path, "r") as f:
+            lines = f.read().splitlines()
+            for line in lines:
+                arr = list(filter(None, line.split(" ")))
+                if arr:
+                    arr = np.array(arr, dtype=np.float32)
+                    file_landmarks.append(arr)
+        land_marks.append(file_landmarks)  # Add the file's landmarks to the overall list
+    return land_marks
+
+def __mirror_index(size, index):
+    #size = len(self.coord_list_cycle)
+    turn = index // size
+    res = index % size
+    if turn % 2 == 0:
+        return res
+    else:
+        return size - res - 1 
+
+
+def inference(quit_event, batch_size, frame_list_cycle, lms_list_cycle, audio_feat_queue, audio_out_queue, res_frame_queue, model):
+    length = len(lms_list_cycle)
+    index = 0
+    count = 0
+    counttime = 0
+    print('start inference')
+
+    while not quit_event.is_set():
+        starttime=time.perf_counter()
+        try:
+            mel_batch = audio_feat_queue.get(block=True, timeout=1)
+        except queue.Empty:
+            continue
+        is_all_silence=True
+        audio_frames = []
+        for _ in range(batch_size*2):
+            frame,type_ = audio_out_queue.get()
+            audio_frames.append((frame,type_))
+            if type_==0:
+                is_all_silence=False
+        if is_all_silence:
+            for i in range(batch_size):
+                res_frame_queue.put((None,__mirror_index(length,index),audio_frames[i*2:i*2+2]))
+                index = index + 1
+        else:
+            t = time.perf_counter()
+            img_batch = []
+
+            for i in range(batch_size):
+                idx = __mirror_index(length, index + i)
+                face = frame_list_cycle[idx]
+                lms = lms_list_cycle[idx]
+                xmin, ymin = lms[1][0], lms[52][1]
+                xmax = lms[31][0]
+                width = xmax - xmin
+                ymax = ymin + width
+                crop_img = face[ymin:ymax, xmin:xmax]
+#                h, w = crop_img.shape[:2]
+                crop_img = cv2.resize(crop_img, (168, 168), cv2.INTER_AREA)
+                crop_img_ori = crop_img.copy()
+                img_real_ex = crop_img[4:164, 4:164].copy()
+                img_real_ex_ori = img_real_ex.copy()
+                img_masked = cv2.rectangle(img_real_ex_ori,(5,5,150,145),(0,0,0),-1)
+    
+                img_masked = img_masked.transpose(2,0,1).astype(np.float32)
+                img_real_ex = img_real_ex.transpose(2,0,1).astype(np.float32)
+    
+                img_real_ex_T = torch.from_numpy(img_real_ex / 255.0)
+                img_masked_T = torch.from_numpy(img_masked / 255.0)
+                img_concat_T = torch.cat([img_real_ex_T, img_masked_T], axis=0)[None]
+                img_batch.append(img_concat_T)
+
+            reshaped_mel_batch = [arr.reshape(32, 32, 32) for arr in mel_batch]
+            mel_batch = torch.stack([torch.from_numpy(arr) for arr in reshaped_mel_batch])
+            img_batch = torch.stack(img_batch).squeeze(1)
+
+
+            with torch.no_grad():
+                pred = model(img_batch.cuda(),mel_batch.cuda())
+            pred = pred.cpu().numpy().transpose(0, 2, 3, 1) * 255.
+
+            counttime += (time.perf_counter() - t)
+            count += batch_size
+            if count >= 100:
+                print(f"------actual avg infer fps:{count / counttime:.4f}")
+                count = 0
+                counttime = 0
+            for i,res_frame in enumerate(pred):
+                #self.__pushmedia(res_frame,loop,audio_track,video_track)
+                res_frame_queue.put((res_frame,__mirror_index(length,index),audio_frames[i*2:i*2+2]))
+                index = index + 1
+
+#            for i, pred_frame in enumerate(pred):
+#                pred_frame_uint8 = np.array(pred_frame, dtype=np.uint8)
+#                res_frame_queue.put((pred_frame_uint8, __mirror_index(length, index), audio_frames[i * 2:i * 2 + 2]))
+#                index = (index + 1) % length
+
+        #print('total batch time:', time.perf_counter() - starttime)
+
+    print('lightreal inference processor stop')
+
+
+class LightReal(BaseReal):
+    @torch.no_grad()
+    def __init__(self, opt, model, avatar):
+        super().__init__(opt)
+        #self.opt = opt # shared with the trainer's opt to support in-place modification of rendering parameters.
+        self.W = opt.W
+        self.H = opt.H
+
+        self.fps = opt.fps # 20 ms per frame
+        
+        self.batch_size = opt.batch_size
+        self.idx = 0
+        self.res_frame_queue = Queue(self.batch_size*2)  #mp.Queue
+        #self.__loadavatar()
+        self.model,audio_processor = model
+        self.frame_list_cycle,self.lms_list_cycle = avatar
+
+        self.asr = LightASR(opt,self,audio_processor)
+        self.asr.warm_up()
+        #self.__warm_up()
+        
+        self.render_event = mp.Event()
+    
+    def __del__(self):
+        print(f'lightreal({self.sessionid}) delete')
+
+   
+    def process_frames(self,quit_event,loop=None,audio_track=None,video_track=None):
+        
+        while not quit_event.is_set():
+            try:
+                res_frame,idx,audio_frames = self.res_frame_queue.get(block=True, timeout=1)
+            except queue.Empty:
+                continue
+            if audio_frames[0][1]!=0 and audio_frames[1][1]!=0: #全为静音数据，只需要取fullimg
+                self.speaking = False
+                audiotype = audio_frames[0][1]
+                if self.custom_index.get(audiotype) is not None: #有自定义视频
+                    mirindex = self.mirror_index(len(self.custom_img_cycle[audiotype]),self.custom_index[audiotype])
+                    combine_frame = self.custom_img_cycle[audiotype][mirindex]
+                    self.custom_index[audiotype] += 1
+                    # if not self.custom_opt[audiotype].loop and self.custom_index[audiotype]>=len(self.custom_img_cycle[audiotype]):
+                    #     self.curr_state = 1  #当前视频不循环播放，切换到静音状态
+                else:
+                    combine_frame = self.frame_list_cycle[idx]
+                    #combine_frame = self.imagecache.get_img(idx)
+            else:
+                self.speaking = True
+                lms = self.lms_list_cycle[idx]
+                combine_frame = copy.deepcopy(self.frame_list_cycle[idx])
+                xmin = lms[1][0]
+                ymin = lms[52][1]
+
+                xmax = lms[31][0]
+                width = xmax - xmin
+                ymax = ymin + width
+                crop_img = combine_frame[ymin:ymax, xmin:xmax]
+                h, w = crop_img.shape[:2]
+                crop_img_ori = cv2.resize(crop_img, (168, 168), cv2.INTER_AREA).copy()
+                #combine_frame = copy.deepcopy(self.imagecache.get_img(idx))
+                res_frame = np.array(res_frame, dtype=np.uint8)
+                crop_img_ori[4:164, 4:164] = res_frame
+                crop_img_ori = cv2.resize(crop_img_ori, (w, h))
+                combine_frame[ymin:ymax, xmin:xmax] = crop_img_ori
+                #print('blending time:',time.perf_counter()-t)
+
+            new_frame = VideoFrame.from_ndarray(combine_frame, format="bgr24")
+            asyncio.run_coroutine_threadsafe(video_track._queue.put(new_frame), loop)
+            self.record_video_data(combine_frame)
+
+            for audio_frame in audio_frames:
+                frame,type_ = audio_frame
+                frame = (frame * 32767).astype(np.int16)
+                new_frame = AudioFrame(format='s16', layout='mono', samples=frame.shape[0])
+                new_frame.planes[0].update(frame.tobytes())
+                new_frame.sample_rate=16000
+                # if audio_track._queue.qsize()>10:
+                #     time.sleep(0.1)
+                asyncio.run_coroutine_threadsafe(audio_track._queue.put(new_frame), loop)
+                self.record_audio_data(frame)
+        print('lightreal process_frames thread stop') 
+            
+    def render(self,quit_event,loop=None,audio_track=None,video_track=None):
+        #if self.opt.asr:
+        #     self.asr.warm_up()
+
+        self.tts.render(quit_event)
+        self.init_customindex()
+        process_thread = Thread(target=self.process_frames, args=(quit_event,loop,audio_track,video_track))
+        process_thread.start()
+        Thread(target=inference, args=(quit_event,self.batch_size,self.frame_list_cycle,self.lms_list_cycle,self.asr.feat_queue,self.asr.output_queue,self.res_frame_queue,
+                                           self.model,)).start()  #mp.Process
+        
+
+        #self.render_event.set() #start infer process render
+        count=0
+        totaltime=0
+        _starttime=time.perf_counter()
+        #_totalframe=0
+        while not quit_event.is_set(): 
+            # update texture every frame
+            # audio stream thread...
+            t = time.perf_counter()
+            self.asr.run_step()
+
+            # if video_track._queue.qsize()>=2*self.opt.batch_size:
+            #     print('sleep qsize=',video_track._queue.qsize())
+            #     time.sleep(0.04*video_track._queue.qsize()*0.8)
+            if video_track._queue.qsize()>=5:
+                print('sleep qsize=',video_track._queue.qsize())
+                time.sleep(0.04*video_track._queue.qsize()*0.8)
+                
+            # delay = _starttime+_totalframe*0.04-time.perf_counter() #40ms
+            # if delay > 0:
+            #     time.sleep(delay)
+        #self.render_event.clear() #end infer process render
+        print('lightreal thread stop')
+            
+
diff --git a/ultralight/audio2feature.py b/ultralight/audio2feature.py
new file mode 100644
index 0000000..dae7b08
--- /dev/null
+++ b/ultralight/audio2feature.py
@@ -0,0 +1,96 @@
+from transformers import Wav2Vec2Processor, HubertModel
+import torch
+import numpy as np
+
+
+class Audio2Feature():
+    def __init__(self):
+        self.device = 'cuda' if torch.cuda.is_available() else 'cpu'
+        self.processor = Wav2Vec2Processor.from_pretrained('./models/hubert-large-ls960-ft')
+        self.model = HubertModel.from_pretrained('./models/hubert-large-ls960-ft').to(self.device)
+
+
+    @torch.no_grad()
+    def get_hubert_from_16k_speech(self, speech):
+        if speech.ndim == 2:
+            speech = speech[:, 0]  # [T, 2] ==> [T,]
+        input_values_all = self.processor(speech, return_tensors="pt", sampling_rate=16000).input_values  # [1, T]
+        input_values_all = input_values_all.to(self.device)
+	    
+        kernel = 400
+        stride = 320
+        clip_length = stride * 1000
+        num_iter = input_values_all.shape[1] // clip_length
+        expected_T = (input_values_all.shape[1] - (kernel-stride)) // stride
+        res_lst = []
+        for i in range(num_iter):
+            if i == 0:
+                start_idx = 0
+                end_idx = clip_length - stride + kernel
+            else:
+                start_idx = clip_length * i
+                end_idx = start_idx + (clip_length - stride + kernel)
+            input_values = input_values_all[:, start_idx: end_idx]
+            hidden_states = self.model.forward(input_values).last_hidden_state  # [B=1, T=pts//320, hid=1024]
+            res_lst.append(hidden_states[0])
+        if num_iter > 0:
+            input_values = input_values_all[:, clip_length * num_iter:]
+        else:
+            input_values = input_values_all
+        if input_values.shape[1] >= kernel:  # if the last batch is shorter than kernel_size, skip it            
+            hidden_states = self.model(input_values).last_hidden_state  # [B=1, T=pts//320, hid=1024]
+        res_lst.append(hidden_states[0])
+        ret = torch.cat(res_lst, dim=0).cpu()  # [T, 1024]
+        assert abs(ret.shape[0] - expected_T) <= 1
+        if ret.shape[0] < expected_T:
+            ret = torch.nn.functional.pad(ret, (0,0,0,expected_T-ret.shape[0]))
+        else:
+            ret = ret[:expected_T]
+        return ret
+
+    def get_sliced_feature(self,
+                           feature_array,
+                           vid_idx,
+                           audio_feat_length=[8,8],
+                           fps=25):
+        """
+        Get sliced features based on a given index
+        :param feature_array: 
+        :param start_idx: the start index of the feature
+        :param audio_feat_length:
+        :return: 
+        """
+        length = len(feature_array)
+        selected_feature = []
+        selected_idx = []
+
+        center_idx = int(vid_idx*50/fps)
+        left_idx = center_idx-audio_feat_length[0]*2
+        right_idx = center_idx + (audio_feat_length[1])*2
+
+        for idx in range(left_idx,right_idx):
+            idx = max(0, idx)
+            idx = min(length-1, idx)
+            x = feature_array[idx]
+            selected_feature.append(x)
+            selected_idx.append(idx)
+
+        selected_feature = np.concatenate(selected_feature, axis=0)
+        selected_feature = selected_feature.reshape(-1, 1024)
+        return selected_feature,selected_idx
+
+    def feature2chunks(self,feature_array,fps,batch_size,audio_feat_length = [8,8],start=0):
+        whisper_chunks = []
+        whisper_idx_multiplier = 50./fps
+        i = 0
+        #print(f"video in {fps} FPS, audio idx in 50FPS")
+        for _ in range(batch_size):
+            # start_idx = int(i * whisper_idx_multiplier)
+            # if start_idx>=len(feature_array):
+            #     break
+            selected_feature,selected_idx = self.get_sliced_feature(feature_array= feature_array,vid_idx = i+start,audio_feat_length=audio_feat_length,fps=fps)
+            #print(f"i:{i},selected_idx {selected_idx}")
+            whisper_chunks.append(selected_feature)
+            i += 1
+
+        return whisper_chunks
diff --git a/ultralight/unet.py b/ultralight/unet.py
new file mode 100644
index 0000000..d60f51f
--- /dev/null
+++ b/ultralight/unet.py
@@ -0,0 +1,283 @@
+import time
+import math
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+
+class InvertedResidual(nn.Module):
+    def __init__(self, inp, oup, stride, use_res_connect, expand_ratio=6):
+        super(InvertedResidual, self).__init__()
+        self.stride = stride
+        assert stride in [1, 2]
+
+        self.use_res_connect = use_res_connect
+
+        self.conv = nn.Sequential(
+            nn.Conv2d(inp, inp * expand_ratio, 1, 1, 0, bias=False),
+            nn.BatchNorm2d(inp * expand_ratio),
+            nn.ReLU(inplace=True),
+            nn.Conv2d(inp * expand_ratio,
+                      inp * expand_ratio,
+                      3,
+                      stride,
+                      1,
+                      groups=inp * expand_ratio,
+                      bias=False),
+            nn.BatchNorm2d(inp * expand_ratio),
+            nn.ReLU(inplace=True),
+            nn.Conv2d(inp * expand_ratio, oup, 1, 1, 0, bias=False),
+            nn.BatchNorm2d(oup),
+        )
+
+    def forward(self, x):
+        if self.use_res_connect:
+            return x + self.conv(x)
+        else:
+            return self.conv(x)
+
+class DoubleConvDW(nn.Module):
+    
+    def __init__(self, in_channels, out_channels, stride=2):
+
+        super(DoubleConvDW, self).__init__() 
+        self.double_conv = nn.Sequential(
+            InvertedResidual(in_channels, out_channels, stride=stride, use_res_connect=False, expand_ratio=2),
+            InvertedResidual(out_channels, out_channels, stride=1, use_res_connect=True, expand_ratio=2)
+        )
+
+    def forward(self, x):
+        return self.double_conv(x)
+
+class InConvDw(nn.Module):
+    def __init__(self, in_channels, out_channels):
+        super(InConvDw, self).__init__() 
+        self.inconv = nn.Sequential(
+            InvertedResidual(in_channels, out_channels, stride=1, use_res_connect=False, expand_ratio=2)
+        )
+    def forward(self, x):
+        return self.inconv(x)
+
+class Down(nn.Module):
+    
+    def __init__(self, in_channels, out_channels):
+
+        super(Down, self).__init__()
+        self.maxpool_conv = nn.Sequential(
+            DoubleConvDW(in_channels, out_channels, stride=2)
+        )
+
+    def forward(self, x):
+        return self.maxpool_conv(x)
+
+class Up(nn.Module):
+
+    def __init__(self, in_channels, out_channels):
+        super(Up, self).__init__()
+        self.up = nn.Upsample(scale_factor=2, mode='bilinear', align_corners=True)
+        self.conv =  DoubleConvDW(in_channels, out_channels, stride=1)
+
+    def forward(self, x1, x2):
+        
+        x1 = self.up(x1)
+        diffY = x2.shape[2] - x1.shape[2]
+        diffX = x2.shape[3] - x1.shape[3]
+        x1 = F.pad(x1, [diffX // 2, diffX - diffX // 2, diffY // 2, diffY - diffY // 2])
+        x = torch.cat([x1, x2], axis=1)
+        
+        return self.conv(x)
+
+class OutConv(nn.Module):
+    def __init__(self, in_channels, out_channels):
+        super(OutConv, self).__init__()
+        self.conv = nn.Conv2d(in_channels, out_channels, kernel_size=1)
+    def forward(self, x):
+        return self.conv(x)
+
+class AudioConvWenet(nn.Module):
+    def __init__(self):
+        super(AudioConvWenet, self).__init__()
+        # ch = [16, 32, 64, 128, 256]   # if you want to run this model on a mobile device, use this. 
+        ch = [32, 64, 128, 256, 512]
+        self.conv1 = InvertedResidual(ch[3], ch[3], stride=1, use_res_connect=True, expand_ratio=2)
+        self.conv2 = InvertedResidual(ch[3], ch[3], stride=1, use_res_connect=True, expand_ratio=2)
+        
+        self.conv3 = nn.Conv2d(ch[3], ch[3], kernel_size=3, padding=1, stride=(1,2))
+        self.bn3 = nn.BatchNorm2d(ch[3])
+        
+        self.conv4 = InvertedResidual(ch[3], ch[3], stride=1, use_res_connect=True, expand_ratio=2)
+        
+        self.conv5 = nn.Conv2d(ch[3], ch[4], kernel_size=3, padding=3, stride=2)
+        self.bn5 = nn.BatchNorm2d(ch[4])
+        self.relu = nn.ReLU()
+        
+        self.conv6 = InvertedResidual(ch[4], ch[4], stride=1, use_res_connect=True, expand_ratio=2)
+        self.conv7 = InvertedResidual(ch[4], ch[4], stride=1, use_res_connect=True, expand_ratio=2)
+    
+    def forward(self, x):
+        
+        x = self.conv1(x)
+        x = self.conv2(x)
+        
+        x = self.relu(self.bn3(self.conv3(x)))
+        
+        x = self.conv4(x)
+        
+        x = self.relu(self.bn5(self.conv5(x)))
+        
+        x = self.conv6(x)
+        x = self.conv7(x)
+    
+        return x
+    
+class AudioConvHubert(nn.Module):
+    def __init__(self):
+        super(AudioConvHubert, self).__init__()
+        # ch = [16, 32, 64, 128, 256]   # if you want to run this model on a mobile device, use this. 
+        ch = [32, 64, 128, 256, 512]
+        self.conv1 = InvertedResidual(32, ch[1], stride=1, use_res_connect=False, expand_ratio=2)
+        self.conv2 = InvertedResidual(ch[1], ch[2], stride=1, use_res_connect=False, expand_ratio=2)
+        
+        self.conv3 = nn.Conv2d(ch[2], ch[3], kernel_size=3, padding=1, stride=(2,2))
+        self.bn3 = nn.BatchNorm2d(ch[3])
+        
+        self.conv4 = InvertedResidual(ch[3], ch[3], stride=1, use_res_connect=True, expand_ratio=2)
+        
+        self.conv5 = nn.Conv2d(ch[3], ch[4], kernel_size=3, padding=3, stride=2)
+        self.bn5 = nn.BatchNorm2d(ch[4])
+        self.relu = nn.ReLU()
+        
+        self.conv6 = InvertedResidual(ch[4], ch[4], stride=1, use_res_connect=True, expand_ratio=2)
+        self.conv7 = InvertedResidual(ch[4], ch[4], stride=1, use_res_connect=True, expand_ratio=2)
+    
+    def forward(self, x):
+        
+        x = self.conv1(x)
+        x = self.conv2(x)
+        
+        x = self.relu(self.bn3(self.conv3(x)))
+        
+        x = self.conv4(x)
+        
+        x = self.relu(self.bn5(self.conv5(x)))
+        
+        x = self.conv6(x)
+        x = self.conv7(x)
+    
+        return x
+
+class Model(nn.Module):
+    def __init__(self,n_channels=6, mode='hubert'):
+        super(Model, self).__init__()
+        self.n_channels = n_channels   #BGR
+        # ch = [16, 32, 64, 128, 256]  # if you want to run this model on a mobile device, use this. 
+        ch = [32, 64, 128, 256, 512]
+        
+        if mode=='hubert':
+            self.audio_model = AudioConvHubert()
+        if mode=='wenet':
+            self.audio_model = AudioConvWenet()
+            
+        self.fuse_conv = nn.Sequential(
+            DoubleConvDW(ch[4]*2, ch[4], stride=1),
+            DoubleConvDW(ch[4], ch[3], stride=1)
+        )
+
+        self.inc = InConvDw(n_channels, ch[0])
+        self.down1 = Down(ch[0], ch[1])
+        self.down2 = Down(ch[1], ch[2])
+        self.down3 = Down(ch[2], ch[3])
+        self.down4 = Down(ch[3], ch[4])
+
+        self.up1 = Up(ch[4], ch[3]//2)
+        self.up2 = Up(ch[3], ch[2]//2)
+        self.up3 = Up(ch[2], ch[1]//2)
+        self.up4 = Up(ch[1], ch[0])
+
+        self.outc = OutConv(ch[0], 3)
+
+    def forward(self, x, audio_feat):
+
+        x1 = self.inc(x)
+        x2 = self.down1(x1)
+        x3 = self.down2(x2)
+        x4 = self.down3(x3)
+        x5 = self.down4(x4)
+        
+        audio_feat  = self.audio_model(audio_feat)
+        x5 = torch.cat([x5, audio_feat], axis=1)
+        x5 = self.fuse_conv(x5)
+        x = self.up1(x5, x4)
+        x = self.up2(x, x3)
+        x = self.up3(x, x2)
+        x = self.up4(x, x1)
+        out = self.outc(x)
+        out = F.sigmoid(out)
+        return out
+
+if __name__ == '__main__':
+    import time
+    import copy
+    import onnx
+    import numpy as np
+    onnx_path = "./unet.onnx"
+
+    from thop import profile, clever_format
+
+    def reparameterize_model(model: torch.nn.Module) -> torch.nn.Module:
+        """ Method returns a model where a multi-branched structure
+            used in training is re-parameterized into a single branch
+            for inference.
+        :param model: MobileOne model in train mode.
+        :return: MobileOne model in inference mode.
+        """
+        # Avoid editing original graph
+        model = copy.deepcopy(model)
+        for module in model.modules():
+            if hasattr(module, 'reparameterize'):
+                module.reparameterize()
+        return model
+    device = torch.device("cuda")
+    def check_onnx(torch_out, torch_in, audio):
+        onnx_model = onnx.load(onnx_path)
+        onnx.checker.check_model(onnx_model)
+        import onnxruntime
+        providers = ["CUDAExecutionProvider"]
+        ort_session = onnxruntime.InferenceSession(onnx_path, providers=providers)
+        print(ort_session.get_providers())
+        ort_inputs = {ort_session.get_inputs()[0].name: torch_in.cpu().numpy(), ort_session.get_inputs()[1].name: audio.cpu().numpy()}
+        ort_outs = ort_session.run(None, ort_inputs)
+        np.testing.assert_allclose(torch_out[0].cpu().numpy(), ort_outs[0][0], rtol=1e-03, atol=1e-05)
+        print("Exported model has been tested with ONNXRuntime, and the result looks good!")
+        
+    net = Model(6).eval().to(device)
+    img = torch.zeros([1, 6, 160, 160]).to(device)
+    audio = torch.zeros([1, 16, 32, 32]).to(device)
+    # net = reparameterize_model(net)
+    flops, params = profile(net, (img,audio))
+    macs, params = clever_format([flops, params], "%3f")
+    print(macs, params)
+    # dynamic_axes= {'input':[2, 3], 'output':[2, 3]}
+    
+    input_dict = {"input": img, "audio": audio}
+    
+    with torch.no_grad():
+        torch_out = net(img, audio)
+        print(torch_out.shape)
+        torch.onnx.export(net, (img, audio), onnx_path, input_names=['input', "audio"],
+                        output_names=['output'], 
+                        # dynamic_axes=dynamic_axes,
+                        # example_outputs=torch_out,
+                        opset_version=11,
+                        export_params=True)
+    check_onnx(torch_out, img, audio)
+
+    # img = torch.zeros([1, 6, 160, 160]).to(device)
+    # audio = torch.zeros([1, 16, 32, 32]).to(device)
+    # with torch.no_grad():
+    #     for i in range(100000):
+    #         t1 = time.time()
+    #         out = net(img, audio)
+    #         t2 = time.time()
+    #         # print(out.shape)
+    #         print('time cost::', t2-t1)
+    # torch.save(net.state_dict(), '1.pth')
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