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Author SHA1 Message Date
jocelyn
e219702ee2 [ADD]add logic of loop frame 2025-06-10 15:20:17 +08:00
2 changed files with 687 additions and 2 deletions

View File

@ -7,7 +7,6 @@ def play_in_loop_v2(
batch_num,
last_direction,
is_silent,
is_silent_,
first_speak,
last_speak,
):
@ -25,7 +24,6 @@ def play_in_loop_v2(
batch_num: 5
last_direction: 0反向1正向
is_silent: 0说话态1动作态
is_silent_: 目前不明确后面可能废弃
first_speak: 记录是不是第一次讲话
last_speak: 记录是不是讲话结束那一刻
"""
@ -316,3 +314,21 @@ def action2silent(
else:
return 0, cur_pos
if __name__ == "__main__":
startfrom = 0 # 上一个batch的最后一帧
frame_config= []
audio_frame_length = len(mel_chunks) # TODO: 确认是否为 batch_size
startfrom = startfrom if startfrom>= frame_config[0][0] else frame_config[0][0]
first_speak, last_speak = True, False
is_silent= True # 当前batch是否为静默
last_direction = 1 # -1 为反方向
start_idx_list, current_direction = play_in_loop_v2(
frame_config,
startfrom,
audio_frame_length,
last_direction,
is_silent,
first_speak,
last_speak,
)

669
utils/wav2lip_processor.py Normal file
View File

@ -0,0 +1,669 @@
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""
@File : self.py
@Time : 2023/10/11 18:23:49
@Author : LiWei
@Version : 1.0
"""
import os
import threading
import time
import traceback
from concurrent.futures import ThreadPoolExecutor
import cv2
import numpy as np
import torch
import constants
from config import config
from face_parsing.model import BiSeNet
from face_parsing.swap import cal_mask_single_img, image_to_parsing_ori
from high_perf.buffer.infer_buffer import Infer_Cache
from high_perf.buffer.video_buff import Video_Cache
from log import logger
from models.wav2lipv2 import Wav2Lip
from utils.util import (
add_alpha,
datagen,
get_actor_id,
get_model_local_path,
load_face_box,
load_seg_model,
load_w2l_model,
morphing,
play_in_loop_v2,
save_raw_video,
save_raw_wav,
get_trans_idxes,
load_config
)
USING_JIT = False
LOCAL_MODE = False
index = 1
def save_(current_speechid, result_frames, audio_segments):
raw_vid = save_raw_video(current_speechid, result_frames)
raw_wav = save_raw_wav(current_speechid, audio_segments)
command = "ffmpeg -y -i {} -i {} -c:v copy -c:a aac -strict experimental -shortest {}".format(
raw_wav, raw_vid, f"temp/{current_speechid}_result_final.mp4"
)
import platform
import subprocess
subprocess.call(command, shell=platform.system() != "Windows")
os.remove(raw_wav)
class Wav2lip_Processor:
def __init__(
self,
task_queue,
result_queue,
stop_event,
resolution,
channel,
device: str = "cuda:0",
) -> None:
self.task_queue = task_queue
self.result_queue = result_queue
self.stop_event = stop_event
self.write_lock = threading.Lock()
self.device = device
self.resolution = resolution
self.channel = channel
pool_size = config.read_frame_pool_size
self.using_pool = True
if pool_size <= 0:
self.using_pool = False
self.img_reading_pool = None
if self.using_pool:
self.img_reading_pool = ThreadPoolExecutor(max_workers=pool_size)
self.face_det_results = {}
self.w2l_models = {}
self.seg_net = None # 分割模型
mel_block_size = (config.wav2lip_batch_size, 80, 16)
audio_block_size = (np.prod(mel_block_size).tolist(), 1)
# 缓存区最大,缓存两秒的视频,超出的会被覆盖掉
self.video_cache = Video_Cache(
"video_cache",
"audio_audio_cache",
"speech_id_cache",
self.resolution,
channel=self.channel,
audio_block_size=audio_block_size[0],
create=True,
)
self.data_cache = Infer_Cache(
"infer_cache",
create=True,
mel_block_shape=mel_block_size,
audio_block_shape=audio_block_size,
)
# TODO: 这个verision 不生效
self.version = "v2"
self.current_speechid = ""
# torch.Size([5, 288, 288, 6]) torch.Size([5, 80, 16, 1])
# torch.Size([5, 288, 288, 6]) torch.Size([5, 80, 16, 1])
self.model_mel_batch = torch.rand((5, 1, 80, 16), dtype=torch.float).to(
self.device
)
self.model_img_batch = torch.rand((5, 6, 288, 288), dtype=torch.float).to(
self.device
)
self.infer_silent_idx = 1
self.post_morphint_idx = 1
self.first_start = True # 判断是否为开播瞬间的静默
self.file_index = 1
def prepare(self, model_speaker_id: str, need_update: bool = True):
"""加载好模型文件"""
if not need_update:
return
self.model_url, self.frame_config, self.padding,self.face_classes,self.trans_method, self.infer_silent_num, self.morphing_num, pkl_version = load_config(model_speaker_id)
models_url_list = [self.model_url]
actor_info = [model_speaker_id]
for actor_url in actor_info:
# 这里不再下载资源,相关工作移到主进程
actor_id = get_actor_id(actor_url)
self.face_det_results[actor_id] = load_face_box(actor_id, pkl_version)
self.w2l_process_state = "READY"
self.__init_seg_model(f"{config.model_folder}/79999_iter.pth")
# 预先加载所有model 模型
# TODO: 采用旧的加载方式看下模型model 是否有错
for model_url in models_url_list:
w2l_name = get_model_local_path(model_url)
self.__init_w2l_model(w2l_name)
logger.debug("Prepare w2l data.")
def __init_w2l_model(self, w2l_name):
""" """
# TODO 这个最好是可以放在启动直播间之前----本模型运行时可能会切换,目前咱们不同的模特可能模型不同
if w2l_name in self.w2l_models:
return
w2l_path = f"{config.model_folder}/{w2l_name}"
if USING_JIT:
net = self.__jit_script(f"{config.model_folder}/w2l.pt", model=Wav2Lip())
else:
net = Wav2Lip()
logger.debug(f"加载模型 {w2l_name}")
self.w2l_models[w2l_name] = load_w2l_model(w2l_path, net, self.device)
def __jit_script(self, jit_module_path: str, model):
"""
convert pytorch module to a runnable scripts.
"""
if not os.path.exists(jit_module_path):
scripted_module = torch.jit.script(model)
torch.jit.save(scripted_module, jit_module_path)
net = torch.jit.load(jit_module_path)
return net
def __init_seg_model(self, seg_path: str):
""" """
# 本模型可以视作永不改变
n_classes = 19
net = BiSeNet(n_classes=n_classes)
if USING_JIT:
net = self.__jit_script(f"{config.model_folder}/swap.pt", model=net)
self.seg_net = load_seg_model(seg_path, net, self.device)
def __using_w2lmodel(self, model_url):
""" """
# 取出对应的模型
model_path = get_model_local_path(model_url)
if model_path in self.w2l_models:
return self.w2l_models[model_path]
else:
self.__init_w2l_model(model_path)
return self.w2l_models[model_path]
def infer(
self,
is_silent,
inference_id,
pkl_name,
startfrom=0,
last_direction=1,
duration=0,
end_with_silent=False,
first_speak=False,
last_speak=False,
before_speak=False,
model_url=None,
debug: bool = False,
test_data: tuple = (),
video_idx: list = [],
):
""" """
start = time.perf_counter()
# TODO: 这是啥逻辑
is_silent_ = 0 if duration < 5 else 1
if not model_url:
model_url = self.model_url
model = self.__using_w2lmodel(model_url)
# 控制加载数据人的个数
self.current_speechid = inference_id
if debug:
audio_segments, mel_chunks = test_data
else:
audio_segments, mel_chunks = self.data_cache.get_data()
audio_frame_length = len(mel_chunks)
if video_idx:
mel_chunks, start_idx_list, current_direction = mel_chunks, video_idx, 1
else:
startfrom = startfrom if startfrom>= self.frame_config[0][0] else self.frame_config[0][0]
start_idx_list, current_direction = play_in_loop_v2(
self.frame_config,
startfrom,
audio_frame_length,
last_direction,
is_silent,
is_silent_,
first_speak,
last_speak,
)
logger.info(
f"start_idx_list :{start_idx_list}, speechid: {self.current_speechid}"
)
gan = datagen(
self.img_reading_pool,
mel_chunks,
start_idx_list,
pkl_name,
self.face_det_results[pkl_name],
self.padding,
is_silent,
)
self.file_index += 5
# 3.4.0 增加静默推理过渡融合, 说话后的n帧向静默过渡
need_pre_morphing=False
if self.trans_method == constants.TRANS_METHOD.mophing_infer:
if self.infer_silent_num or self.morphing_num or config.PRE_MORPHING_NUM:
logger.info(f"self.first_start{self.first_start}, {before_speak}, {self.current_speechid}")
if (
is_silent == 0
or (
before_speak)
or (
not self.first_start
and is_silent == 1
and ((self.infer_silent_idx < self.infer_silent_num) or (self.post_morphint_idx < self.morphing_num))
)
):
# 开始说话状态时候,要把上次计数清理
need_infer_silent = (
True
if is_silent == 1
and self.infer_silent_idx < self.infer_silent_num and not before_speak and "&" not in self.current_speechid
else False
)
need_pre_morphing = True if before_speak else False
need_post_morphing = (
True
if is_silent == 1
and self.morphing_num
and self.post_morphint_idx < self.morphing_num
and not need_infer_silent
and not need_pre_morphing
else False
)
if is_silent == 0 or "&" in self.current_speechid:
self.infer_silent_idx = 1
self.post_morphint_idx = 1
self.first_start = False
logger.debug(
f"{start_idx_list} is_silent:{is_silent}, infer_silent_idx: {self.infer_silent_idx} morphint_idx:{self.post_morphint_idx} need_post_morphing: {need_post_morphing}, need_infer_silent:{need_infer_silent}, speech:{self.current_speechid},need_pre_morphing:{need_pre_morphing}"
)
if is_silent == 1 and need_post_morphing and self.post_morphint_idx < self.morphing_num:
self.post_morphint_idx += 5
result_frames = self.prediction(
gan,
model,
pkl_name,
need_post_morphing=need_post_morphing,
need_infer_silent=need_infer_silent,
is_silent=is_silent,
need_pre_morphing=need_pre_morphing,
)
if is_silent == 1 and need_infer_silent and self.infer_silent_idx < self.infer_silent_num:
self.infer_silent_idx += 5
# 第一次静默时候,直接进入这个逻辑
else:
self.no_prediction(gan, model, pkl_name)
else:
if is_silent==0:
self.prediction(
gan,
model,
pkl_name,
is_silent=is_silent,
need_pre_morphing=need_pre_morphing,
)
else:
self.no_prediction(gan, model, pkl_name)
elif self.trans_method == constants.TRANS_METHOD.all_infer:
result_frames = self.prediction(gan, model, pkl_name,is_silent=is_silent)
else:
raise ValueError(f"not supported {self.trans_method}")
if LOCAL_MODE:
saveThread = threading.Thread(
target=save_,
args=(self.current_speechid, result_frames, audio_segments),
)
saveThread.start()
self.video_cache.put_audio(audio_segments, self.current_speechid)
logger.debug(
f"视频合成时间{time.perf_counter() - start},inference_id:{self.current_speechid}"
)
return [
[0, 1, 2, 3, 4],
start_idx_list[-1],
current_direction,
] # frames_return_list: 视频帧数据 res_index 控制下一帧的开始位置 direction: 播放的顺序 正 反
@torch.no_grad()
def batch_to_tensor(self, img_batch, mel_batch, model, padding, frames, coords):
padding = 10
logger.debug(f"非静默,推理: {self.current_speechid} 准备推理数据 {id(model)}")
img_batch_tensor = torch.as_tensor(img_batch, dtype=torch.float32).to(
self.device, non_blocking=True
)
mel_batch_tensor = torch.as_tensor(mel_batch, dtype=torch.float32).to(
self.device, non_blocking=True
)
img_batch_tensor = img_batch_tensor.permute(0, 3, 1, 2)
mel_batch_tensor = mel_batch_tensor.permute(0, 3, 1, 2)
logger.debug(
f"非静默,推理: {self.current_speechid} 即将嘴型生成 {img_batch_tensor.shape} {mel_batch_tensor.shape}"
) # torch.Size([5, 6, 288, 288]) torch.Size([5, 1, 80, 16])
pred_batch = model(mel_batch_tensor, img_batch_tensor) * 255.0
logger.debug(f"非静默,推理: {self.current_speechid} 嘴型生成完成")
pred_clone_batch = pred_batch.clone()
pred_batch_cpu = pred_batch.to(torch.uint8).to("cpu", non_blocking=True)
del pred_batch
del mel_batch_tensor
logger.debug(f"非静默,推理: {self.current_speechid} 生成面部准备好了")
large_faces = [
frame.copy()[:, :, :3][
y1 - padding : y2 + padding, x1 - padding : x2 + padding
]
for frame, box in zip(frames, coords)
for y1, y2, x1, x2 in [box]
]
logger.debug(f"非静默,推理: {self.current_speechid} 即将对生成的面部进行分割")
seg_out = image_to_parsing_ori(
pred_clone_batch, self.seg_net
) # 假设这个函数在 GPU 上执行并输出 GPU tensor
del img_batch_tensor
del pred_clone_batch
torch.cuda.synchronize() # 等待异步结束
seg_out_cpu = seg_out.to(
"cpu", non_blocking=True
) # 发起 seg_out 的异步数据传输
del seg_out
logger.debug(f"非静默,推理: {self.current_speechid} 对生成的面部进行分割结束")
pred_batch_cpu = pred_batch_cpu.numpy().transpose(
0, 2, 3, 1
) # 按需计算,用到了才计算,这样能不能使得GPU往cpu做数据copy的时间被隐藏
for predict, frame, box in zip(pred_batch_cpu, frames, coords):
y1, y2, x1, x2 = box
width, height = x2 - x1, y2 - y1
frame[:, :, :3][y1:y2, x1:x2] = cv2.resize(
predict, (width, height)
) # 无需再转为unit8gpu上直接转为unit8这样传输规模小一些
torch.cuda.synchronize() # 等待异步结束
return large_faces, seg_out_cpu, frames
def no_prediction(self, gan, model, pkl_name):
logger.debug(
f"无需推理: {self.current_speechid} {self.model_img_batch.shape} {self.model_mel_batch.shape}"
)
with torch.no_grad():
pred_ = model(self.model_mel_batch, self.model_img_batch)
seg_out = image_to_parsing_ori(
pred_, self.seg_net
) # 不确定是不是需要也warmup第二个网络
for _, _, frames, full_masks, _, end_idx in gan:
speech_ids = [
f"{self.current_speechid}_{idx}" for idx in range(len(end_idx))
]
offset_list = self.video_cache.occupy_video_pos(len(end_idx))
file_idxes, _, _ = get_trans_idxes(False, False, 0,0, self.file_index)
logger.info(f"self.file_index:{self.file_index} file_idxes:{file_idxes}")
param_list = [
(
speech_id,
frame,
body_mask,
write_pos,
pkl_name,
self.video_cache,
file_idx
)
for speech_id, frame, body_mask, write_pos, file_idx in zip(
speech_ids,
frames,
full_masks,
offset_list,
file_idxes
)
]
if self.using_pool:
futures = [
self.img_reading_pool.submit(self.silent_paste_back, *param)
for param in param_list
]
_ = [future.result() for future in futures]
self.video_cache._inject_real_pos(len(end_idx))
# if config.debug_mode:
# for param in param_list:
# self.silent_paste_back(*param)
del pred_
del seg_out
torch.cuda.empty_cache()
def silent_paste_back(self, speech_id, frame, body_mask,write_pos,pkl_name, video_cache, file_index):
global index
if frame.shape[-1] == 4:
frame[:, :, :3] = frame[:, :, :3][:, :, ::-1]
elif config.output_alpha:
frame = add_alpha(frame, body_mask, config.alpha)
logger.debug(f"self.file_index:{self.file_index}, file_index{file_index}")
if config.debug_mode:
logger.info(f"不用推理: {file_index} {frame.shape}")
if not cv2.imwrite(f"temp/{pkl_name}/new_img{file_index:05d}.jpg", frame):
logger.error(f"save {file_index} err")
index += 1
logger.info(f"silent frame shape:{frame.shape}")
video_cache._put_raw_frame(frame, write_pos, speech_id)
return frame
def prediction(
self,
gan,
model,
pkl_name: str,
need_post_morphing: bool = False,
need_infer_silent: bool = False,
is_silent: int = 1,
need_pre_morphing: bool = False,
):
""" """
logger.debug(f"非静默,推理: {self.current_speechid}")
for img_batch, mel_batch, frames, full_masks, coords, end_idx in gan:
large_faces, seg_out_cpu, frames = self.batch_to_tensor(
img_batch, mel_batch, model, self.padding, frames, coords
)
offset_list = self.video_cache.occupy_video_pos(len(end_idx))
speech_ids = [
f"{self.current_speechid}_{idx}" for idx in range(len(end_idx))
]
file_idxes, post_morphint_idxes, infer_silent_idxes = get_trans_idxes(need_post_morphing, need_infer_silent,self.post_morphint_idx,self.infer_silent_idx, self.file_index)
logger.info(f"self.file_index:{self.file_index}infer_silent_idxes:{infer_silent_idxes},post_morphint_idxes:{post_morphint_idxes} file_idxes:{file_idxes}")
param_list = [
(
speech_id,
padded_image,
frame,
body_mask,
seg_mask,
boxes,
self.padding,
self.video_cache,
write_pos,
pkl_name,
need_infer_silent,
need_post_morphing,
is_silent,
need_pre_morphing,
pre_morphing_idx,
infer_silent_idx,
post_morphint_idx,
file_idx
)
for speech_id, padded_image, frame, body_mask, seg_mask, boxes, write_pos, pre_morphing_idx,infer_silent_idx, post_morphint_idx,file_idx in zip(
speech_ids,
large_faces,
frames,
full_masks,
seg_out_cpu,
coords,
offset_list,
list(range(config.PRE_MORPHING_NUM)),
infer_silent_idxes,
post_morphint_idxes,
file_idxes
)
]
result_frames = []
if self.using_pool:# and not config.debug_mode:
futures = [
self.img_reading_pool.submit(self.paste_back, *param)
for param in param_list
]
_ = [future.result() for future in futures]
self.video_cache._inject_real_pos(len(end_idx))
# if config.debug_mode:
# for param in param_list:
# frame = self.paste_back(*param)
# if LOCAL_MODE:
# result_frames.append(frame)
return result_frames
def paste_back(
self,
current_speechid,
large_face,
frame,
body_mask,
mask,
boxes,
padding,
video_cache,
cache_start_offset,
pkl_name,
need_infer_silent: bool = False,
need_post_morphing: bool = False,
is_silent: bool = False,
need_pre_morphing: bool = False,
pre_morphing_idx: int = 0,
infer_silent_idx: int = 0,
post_morphint_idx: int = 0,
file_index: int = 1,
):
"""
根据模型预测结果和原始图片进行融合
Args:
"""
global index
padding = 10
y1, y2, x1, x2 = boxes
width, height = x2 - x1, y2 - y1
mask = cal_mask_single_img(
mask, use_old_mode=True, face_classes=self.face_classes
)
mask = np.repeat(mask[..., None], 3, axis=-1).astype("uint8")
mask_temp = np.zeros_like(large_face)
mask_out = cv2.resize(mask.astype(np.float) * 255.0, (width, height)).astype(
np.uint8
)
mask_temp[padding : height + padding, padding : width + padding] = mask_out
kernel = np.ones((9, 9), np.uint8)
mask_temp = cv2.erode(mask_temp, kernel, iterations=1) # 二值的
# gaosi_kernel = int(0.1 * large_face.shape[0] // 2 * 2) + 1
# mask_temp = cv2.GaussianBlur(
# mask_temp, (gaosi_kernel, gaosi_kernel), 0, 0, cv2.BORDER_DEFAULT
# )
mask_temp = cv2.GaussianBlur(mask_temp, (15, 15), 0, 0, cv2.BORDER_DEFAULT)
mask_temp = cv2.GaussianBlur(mask_temp, (5, 5), 0, 0, cv2.BORDER_DEFAULT)
f_background = large_face.copy()
frame[:, :, :3][y1 - padding : y2 + padding, x1 - padding : x2 + padding] = (
f_background * (1 - mask_temp / 255.0)
+ frame[:, :, :3][y1 - padding : y2 + padding, x1 - padding : x2 + padding]
* (mask_temp / 255.0)
)#.astype("uint8")
if self.trans_method == constants.TRANS_METHOD.mophing_infer:
if is_silent == 1:
if need_pre_morphing:
frame = morphing(
large_face,
frame,
boxes,
mp_ratio=1 - ((pre_morphing_idx + 1) / config.PRE_MORPHING_NUM),
file_index=file_index,
)
logger.debug(
f"file_index{file_index},pre morphing_idx {pre_morphing_idx}, speech_id:{current_speechid}"
)
logger.debug(f"pre morphing_idx {pre_morphing_idx}, speech_id:{current_speechid}")
#TODO: @txueduo 处理前过渡问题
elif need_post_morphing and post_morphint_idx:
mp_ratio = (post_morphint_idx) / self.morphing_num
frame = morphing(
large_face,
frame,
boxes,
mp_ratio=mp_ratio,
file_index=file_index
)
logger.debug(f"post_morphint_idx:{post_morphint_idx}, mp_ratio:{mp_ratio}, file_index:{file_index}, speech_id:{current_speechid}")
if frame.shape[-1]==4:# and not config.output_alpha:
frame[:,:,:3] = frame[:,:,:3][:,:,::-1]
if config.output_alpha and frame.shape[-1]!=4:
frame = add_alpha(frame, body_mask, config.alpha)
if config.debug_mode:
logger.info(f"推理:{file_index}")
if not cv2.imwrite(f"temp/{pkl_name}/new_img{file_index:05d}.jpg", frame):
logger.error(f"save {file_index} err")
video_cache._put_raw_frame(frame, cache_start_offset, current_speechid)
index += 1
return frame
def destroy(self):
""" """
self.data_cache.destroy()
self.video_cache.destroy()
self.data_cache = None
self.video_cache = None
if self.using_pool and self.img_reading_pool is not None:
self.img_reading_pool.shutdown()
self.img_reading_pool = None
del self.model_img_batch
del self.model_mel_batch
def process_wav2lip_predict(task_queue, stop_event, output_queue, resolution, channel):
# 实例化推理类
w2l_processor = Wav2lip_Processor(
task_queue, output_queue, stop_event, resolution, channel
)
need_update = True
while True:
if stop_event.is_set():
print("----------------------stop..")
w2l_processor.destroy() # 这一行代码可能有bug没有同步的控制代码可能在读的地方会有异常
break
try:
params = task_queue.get()
# TODO: 这个操作应该放在更前面, 确认是否更新的判定条件
start = time.perf_counter()
w2l_processor.prepare(params[2], need_update)
need_update = False
result = w2l_processor.infer(*params)
logger.debug(
f"推理结束 :{time.perf_counter() - start},inference_id:{params[1]}"
)
output_queue.put(result)
logger.debug(
f"结果通知到主进程{time.perf_counter() - start},inference_id:{params[1]}"
)
except Exception:
logger.error(f"process_wav2lip_predict :{traceback.format_exc()}")