Merge branch 'lipku:main' into main
This commit is contained in:
ShelikeSnow
GitHub
commit
994535fe3e
13
lipasr.py
13
lipasr.py
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@ -1,12 +1,9 @@
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import time
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import torch
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import numpy as np
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import soundfile as sf
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import resampy
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import queue
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from queue import Queue
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from io import BytesIO
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import multiprocessing as mp
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from wav2lip import audio
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@ -26,9 +23,9 @@ class LipASR:
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self.batch_size = opt.batch_size
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self.frames = []
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self.stride_left_size = self.stride_right_size = 10
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self.context_size = 10
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self.audio_feats = []
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self.stride_left_size = opt.l
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self.stride_right_size = opt.r
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#self.context_size = 10
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self.feat_queue = mp.Queue(5)
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self.warm_up()
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@ -38,7 +35,7 @@ class LipASR:
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def __get_audio_frame(self):
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try:
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frame = self.queue.get(block=True,timeout=0.018)
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frame = self.queue.get(block=True,timeout=0.01)
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type = 0
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#print(f'[INFO] get frame {frame.shape}')
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except queue.Empty:
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@ -67,7 +64,7 @@ class LipASR:
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# put to output
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self.output_queue.put((frame,type))
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# context not enough, do not run network.
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if len(self.frames) < self.stride_left_size + self.context_size + self.stride_right_size:
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if len(self.frames) <= self.stride_left_size + self.stride_right_size:
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return
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inputs = np.concatenate(self.frames) # [N * chunk]
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36
museasr.py
36
museasr.py
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@ -1,12 +1,9 @@
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import time
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import torch
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import numpy as np
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import soundfile as sf
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import resampy
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import queue
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from queue import Queue
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from io import BytesIO
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import multiprocessing as mp
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from musetalk.whisper.audio2feature import Audio2Feature
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@ -25,8 +22,9 @@ class MuseASR:
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self.audio_processor = audio_processor
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self.batch_size = opt.batch_size
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self.stride_left_size = self.stride_right_size = 6
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self.audio_feats = []
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self.frames = []
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self.stride_left_size = opt.l
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self.stride_right_size = opt.r
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self.feat_queue = mp.Queue(5)
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self.warm_up()
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@ -36,7 +34,7 @@ class MuseASR:
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def __get_audio_frame(self):
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try:
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frame = self.queue.get(block=True,timeout=0.018)
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frame = self.queue.get(block=True,timeout=0.01)
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type = 0
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#print(f'[INFO] get frame {frame.shape}')
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except queue.Empty:
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@ -49,15 +47,10 @@ class MuseASR:
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return self.output_queue.get()
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def warm_up(self):
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frames = []
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for _ in range(self.stride_left_size + self.stride_right_size):
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audio_frame,type=self.__get_audio_frame()
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frames.append(audio_frame)
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self.frames.append(audio_frame)
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self.output_queue.put((audio_frame,type))
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inputs = np.concatenate(frames) # [N * chunk]
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whisper_feature = self.audio_processor.audio2feat(inputs)
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for feature in whisper_feature:
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self.audio_feats.append(feature)
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for _ in range(self.stride_left_size):
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self.output_queue.get()
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@ -65,20 +58,25 @@ class MuseASR:
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def run_step(self):
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############################################## extract audio feature ##############################################
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start_time = time.time()
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frames = []
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for _ in range(self.batch_size*2):
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audio_frame,type=self.__get_audio_frame()
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frames.append(audio_frame)
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self.frames.append(audio_frame)
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self.output_queue.put((audio_frame,type))
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inputs = np.concatenate(frames) # [N * chunk]
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if len(self.frames) <= self.stride_left_size + self.stride_right_size:
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return
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inputs = np.concatenate(self.frames) # [N * chunk]
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whisper_feature = self.audio_processor.audio2feat(inputs)
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for feature in whisper_feature:
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self.audio_feats.append(feature)
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# for feature in whisper_feature:
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# self.audio_feats.append(feature)
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#print(f"processing audio costs {(time.time() - start_time) * 1000}ms, inputs shape:{inputs.shape} whisper_feature len:{len(whisper_feature)}")
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whisper_chunks = self.audio_processor.feature2chunks(feature_array=self.audio_feats,fps=self.fps/2,batch_size=self.batch_size,start=self.stride_left_size/2 )
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whisper_chunks = self.audio_processor.feature2chunks(feature_array=whisper_feature,fps=self.fps/2,batch_size=self.batch_size,start=self.stride_left_size/2 )
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#print(f"whisper_chunks len:{len(whisper_chunks)},self.audio_feats len:{len(self.audio_feats)},self.output_queue len:{self.output_queue.qsize()}")
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self.audio_feats = self.audio_feats[-(self.stride_left_size + self.stride_right_size):]
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#self.audio_feats = self.audio_feats[-(self.stride_left_size + self.stride_right_size):]
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self.feat_queue.put(whisper_chunks)
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# discard the old part to save memory
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self.frames = self.frames[-(self.stride_left_size + self.stride_right_size):]
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def get_next_feat(self,block,timeout):
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return self.feat_queue.get(block,timeout)
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@ -2,6 +2,7 @@ from PIL import Image
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import numpy as np
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import cv2
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from face_parsing import FaceParsing
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import copy
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fp = FaceParsing()
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@ -84,17 +85,41 @@ def get_image_prepare_material(image,face_box,upper_boundary_ratio = 0.5,expand=
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mask_array = cv2.GaussianBlur(np.array(modified_mask_image), (blur_kernel_size, blur_kernel_size), 0)
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return mask_array,crop_box
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def get_image_blending(image,face,face_box,mask_array,crop_box):
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body = Image.fromarray(image[:,:,::-1])
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face = Image.fromarray(face[:,:,::-1])
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# def get_image_blending(image,face,face_box,mask_array,crop_box):
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# body = Image.fromarray(image[:,:,::-1])
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# face = Image.fromarray(face[:,:,::-1])
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# x, y, x1, y1 = face_box
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# x_s, y_s, x_e, y_e = crop_box
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# face_large = body.crop(crop_box)
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# mask_image = Image.fromarray(mask_array)
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# mask_image = mask_image.convert("L")
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# face_large.paste(face, (x-x_s, y-y_s, x1-x_s, y1-y_s))
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# body.paste(face_large, crop_box[:2], mask_image)
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# body = np.array(body)
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# return body[:,:,::-1]
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def get_image_blending(image,face,face_box,mask_array,crop_box):
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body = image
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x, y, x1, y1 = face_box
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x_s, y_s, x_e, y_e = crop_box
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face_large = body.crop(crop_box)
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face_large = copy.deepcopy(body[y_s:y_e, x_s:x_e])
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face_large[y-y_s:y1-y_s, x-x_s:x1-x_s]=face
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mask_image = Image.fromarray(mask_array)
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mask_image = mask_image.convert("L")
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face_large.paste(face, (x-x_s, y-y_s, x1-x_s, y1-y_s))
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body.paste(face_large, crop_box[:2], mask_image)
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body = np.array(body)
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return body[:,:,::-1]
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mask_image = cv2.cvtColor(mask_array,cv2.COLOR_BGR2GRAY)
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mask_image = (mask_image/255).astype(np.float32)
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# mask_not = cv2.bitwise_not(mask_array)
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# prospect_tmp = cv2.bitwise_and(face_large, face_large, mask=mask_array)
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# background_img = body[y_s:y_e, x_s:x_e]
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# background_img = cv2.bitwise_and(background_img, background_img, mask=mask_not)
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# body[y_s:y_e, x_s:x_e] = prospect_tmp + background_img
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#print(mask_image.shape)
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#print(cv2.minMaxLoc(mask_image))
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body[y_s:y_e, x_s:x_e] = cv2.blendLinear(face_large,body[y_s:y_e, x_s:x_e],mask_image,1-mask_image)
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#body.paste(face_large, crop_box[:2], mask_image)
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return body
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21
webrtc.py
21
webrtc.py
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@ -55,13 +55,13 @@ class PlayerStreamTrack(MediaStreamTrack):
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if hasattr(self, "_timestamp"):
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#self._timestamp = (time.time()-self._start) * VIDEO_CLOCK_RATE
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self._timestamp += int(VIDEO_PTIME * VIDEO_CLOCK_RATE)
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# wait = self._start + (self._timestamp / VIDEO_CLOCK_RATE) - time.time()
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wait = self.timelist[0] + len(self.timelist)*VIDEO_PTIME - time.time()
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wait = self._start + (self._timestamp / VIDEO_CLOCK_RATE) - time.time()
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# wait = self.timelist[0] + len(self.timelist)*VIDEO_PTIME - time.time()
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if wait>0:
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await asyncio.sleep(wait)
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self.timelist.append(time.time())
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if len(self.timelist)>100:
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self.timelist.pop(0)
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# if len(self.timelist)>=100:
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# self.timelist.pop(0)
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# self.timelist.append(time.time())
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else:
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self._start = time.time()
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self._timestamp = 0
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@ -72,13 +72,14 @@ class PlayerStreamTrack(MediaStreamTrack):
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if hasattr(self, "_timestamp"):
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#self._timestamp = (time.time()-self._start) * SAMPLE_RATE
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self._timestamp += int(AUDIO_PTIME * SAMPLE_RATE)
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# wait = self._start + (self._timestamp / SAMPLE_RATE) - time.time()
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wait = self.timelist[0] + len(self.timelist)*AUDIO_PTIME - time.time()
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wait = self._start + (self._timestamp / SAMPLE_RATE) - time.time()
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# wait = self.timelist[0] + len(self.timelist)*AUDIO_PTIME - time.time()
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if wait>0:
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await asyncio.sleep(wait)
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self.timelist.append(time.time())
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if len(self.timelist)>200:
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self.timelist.pop(0)
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# if len(self.timelist)>=200:
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# self.timelist.pop(0)
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# self.timelist.pop(0)
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# self.timelist.append(time.time())
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else:
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self._start = time.time()
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self._timestamp = 0
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