Portrait-Animation / sonic.py
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import os
import torch
import torch.utils.checkpoint
from PIL import Image
import numpy as np
from omegaconf import OmegaConf
from tqdm import tqdm
import cv2
from diffusers import AutoencoderKLTemporalDecoder
from diffusers.schedulers import EulerDiscreteScheduler
from transformers import WhisperModel, CLIPVisionModelWithProjection, AutoFeatureExtractor
from src.utils.util import save_videos_grid, seed_everything
from src.dataset.test_preprocess import process_bbox, image_audio_to_tensor
from src.models.base.unet_spatio_temporal_condition import UNetSpatioTemporalConditionModel, add_ip_adapters
from src.pipelines.pipeline_sonic import SonicPipeline
from src.models.audio_adapter.audio_proj import AudioProjModel
from src.models.audio_adapter.audio_to_bucket import Audio2bucketModel
from src.utils.RIFE.RIFE_HDv3 import RIFEModel
from src.dataset.face_align.align import AlignImage
BASE_DIR = os.path.dirname(os.path.abspath(__file__))
def test(
pipe,
config,
wav_enc,
audio_pe,
audio2bucket,
image_encoder,
width,
height,
batch
):
# 배치 텐서를 (1,B,C,H,W) 형태로
for k, v in batch.items():
if isinstance(v, torch.Tensor):
batch[k] = v.unsqueeze(0).to(pipe.device).float()
ref_img = batch['ref_img']
clip_img = batch['clip_images']
face_mask = batch['face_mask']
image_embeds = image_encoder(clip_img).image_embeds
audio_feature = batch['audio_feature']
audio_len = batch['audio_len']
step = int(config.step)
# window=3000 -> 16000으로 변경(1초 간격)
window = 16000
audio_prompts = []
last_audio_prompts = []
for i in range(0, audio_feature.shape[-1], window):
audio_clip_chunk = audio_feature[:, :, i:i+window]
# Whisper encoder
audio_prompt = wav_enc.encoder(audio_clip_chunk, output_hidden_states=True).hidden_states
last_audio_prompt = wav_enc.encoder(audio_clip_chunk).last_hidden_state
last_audio_prompt = last_audio_prompt.unsqueeze(-2)
audio_prompt = torch.stack(audio_prompt, dim=2)
audio_prompts.append(audio_prompt)
last_audio_prompts.append(last_audio_prompt)
# ★ 여기서 비었으면 예외
if len(audio_prompts) == 0:
raise ValueError(
"[ERROR] No speech recognized from the audio. "
"Please provide a valid speech audio (with clear voice)."
)
audio_prompts = torch.cat(audio_prompts, dim=1)
audio_prompts = audio_prompts[:, :audio_len*2]
audio_prompts = torch.cat([
torch.zeros_like(audio_prompts[:, :4]),
audio_prompts,
torch.zeros_like(audio_prompts[:, :6])
], dim=1)
last_audio_prompts = torch.cat(last_audio_prompts, dim=1)
last_audio_prompts = last_audio_prompts[:, :audio_len*2]
last_audio_prompts = torch.cat([
torch.zeros_like(last_audio_prompts[:, :24]),
last_audio_prompts,
torch.zeros_like(last_audio_prompts[:, :26])
], dim=1)
ref_tensor_list = []
audio_tensor_list = []
uncond_audio_tensor_list = []
motion_buckets = []
for i in tqdm(range(audio_len // step)):
audio_clip = audio_prompts[:, i*2*step : i*2*step + 10].unsqueeze(0)
audio_clip_for_bucket = last_audio_prompts[:, i*2*step : i*2*step + 50].unsqueeze(0)
motion_bucket = audio2bucket(audio_clip_for_bucket, image_embeds)
motion_bucket = motion_bucket * 16 + 16
motion_buckets.append(motion_bucket[0])
cond_audio_clip = audio_pe(audio_clip).squeeze(0)
uncond_audio_clip = audio_pe(torch.zeros_like(audio_clip)).squeeze(0)
ref_tensor_list.append(ref_img[0])
audio_tensor_list.append(cond_audio_clip[0])
uncond_audio_tensor_list.append(uncond_audio_clip[0])
video = pipe(
ref_img,
clip_img,
face_mask,
audio_tensor_list,
uncond_audio_tensor_list,
motion_buckets,
height=height,
width=width,
num_frames=len(audio_tensor_list),
decode_chunk_size=config.decode_chunk_size,
motion_bucket_scale=config.motion_bucket_scale,
fps=config.fps,
noise_aug_strength=config.noise_aug_strength,
min_guidance_scale1=config.min_appearance_guidance_scale,
max_guidance_scale1=config.max_appearance_guidance_scale,
min_guidance_scale2=config.audio_guidance_scale,
max_guidance_scale2=config.audio_guidance_scale,
overlap=config.overlap,
shift_offset=config.shift_offset,
frames_per_batch=config.n_sample_frames,
num_inference_steps=config.num_inference_steps,
i2i_noise_strength=config.i2i_noise_strength
).frames
video = (video * 0.5 + 0.5).clamp(0, 1)
video = torch.cat([video.to(pipe.device)], dim=0).cpu()
return video
class Sonic():
config_file = os.path.join(BASE_DIR, 'config/inference/sonic.yaml')
config = OmegaConf.load(config_file)
def __init__(self,
device_id=0,
enable_interpolate_frame=True,
):
config = self.config
config.use_interframe = enable_interpolate_frame
device = f'cuda:{device_id}' if device_id > -1 else 'cpu'
config.pretrained_model_name_or_path = os.path.join(BASE_DIR, config.pretrained_model_name_or_path)
# VAE
vae = AutoencoderKLTemporalDecoder.from_pretrained(
config.pretrained_model_name_or_path,
subfolder="vae",
variant="fp16")
# 스케줄러
val_noise_scheduler = EulerDiscreteScheduler.from_pretrained(
config.pretrained_model_name_or_path,
subfolder="scheduler")
# CLIP Vision
image_encoder = CLIPVisionModelWithProjection.from_pretrained(
config.pretrained_model_name_or_path,
subfolder="image_encoder",
variant="fp16")
# UNet
unet = UNetSpatioTemporalConditionModel.from_pretrained(
config.pretrained_model_name_or_path,
subfolder="unet",
variant="fp16")
# Adapter
add_ip_adapters(unet, [32], [config.ip_audio_scale])
audio2token = AudioProjModel(
seq_len=10, blocks=5, channels=384,
intermediate_dim=1024, output_dim=1024, context_tokens=32
).to(device)
audio2bucket = Audio2bucketModel(
seq_len=50, blocks=1, channels=384,
clip_channels=1024, intermediate_dim=1024, output_dim=1,
context_tokens=2
).to(device)
# 로컬 체크포인트 로드
unet_checkpoint_path = os.path.join(BASE_DIR, config.unet_checkpoint_path)
audio2token_checkpoint_path = os.path.join(BASE_DIR, config.audio2token_checkpoint_path)
audio2bucket_checkpoint_path = os.path.join(BASE_DIR, config.audio2bucket_checkpoint_path)
unet.load_state_dict(
torch.load(unet_checkpoint_path, map_location="cpu"),
strict=True,
)
audio2token.load_state_dict(
torch.load(audio2token_checkpoint_path, map_location="cpu"),
strict=True,
)
audio2bucket.load_state_dict(
torch.load(audio2bucket_checkpoint_path, map_location="cpu"),
strict=True,
)
# weight_dtype 설정
if config.weight_dtype == "fp16":
weight_dtype = torch.float16
elif config.weight_dtype == "fp32":
weight_dtype = torch.float32
elif config.weight_dtype == "bf16":
weight_dtype = torch.bfloat16
else:
raise ValueError(f"Do not support weight dtype: {config.weight_dtype}")
# Whisper
whisper = WhisperModel.from_pretrained(
os.path.join(BASE_DIR, 'checkpoints/whisper-tiny/')
).to(device).eval()
whisper.requires_grad_(False)
self.feature_extractor = AutoFeatureExtractor.from_pretrained(
os.path.join(BASE_DIR, 'checkpoints/whisper-tiny/')
)
# Face detect
det_path = os.path.join(BASE_DIR, 'checkpoints/yoloface_v5m.pt')
self.face_det = AlignImage(device, det_path=det_path)
# RIFE 중간프레임 보간
if config.use_interframe:
rife = RIFEModel(device=device)
rife.load_model(os.path.join(BASE_DIR, 'checkpoints', 'RIFE/'))
self.rife = rife
# dtype 변경
image_encoder.to(weight_dtype)
vae.to(weight_dtype)
unet.to(weight_dtype)
# SonicPipeline 초기화
pipe = SonicPipeline(
unet=unet,
image_encoder=image_encoder,
vae=vae,
scheduler=val_noise_scheduler,
)
pipe = pipe.to(device=device, dtype=weight_dtype)
self.pipe = pipe
self.whisper = whisper
self.audio2token = audio2token
self.audio2bucket = audio2bucket
self.image_encoder = image_encoder
self.device = device
print('Sonic init done')
def preprocess(self, image_path, expand_ratio=1.0):
face_image = cv2.imread(image_path)
h, w = face_image.shape[:2]
_, _, bboxes = self.face_det(face_image, maxface=True)
face_num = len(bboxes)
bbox_s = None
if face_num > 0:
x1, y1, ww, hh = bboxes[0]
x2, y2 = x1 + ww, y1 + hh
bbox = x1, y1, x2, y2
bbox_s = process_bbox(bbox, expand_radio=expand_ratio, height=h, width=w)
return {
'face_num': face_num,
'crop_bbox': bbox_s,
}
def crop_image(self, input_image_path, output_image_path, crop_bbox):
face_image = cv2.imread(input_image_path)
crop_image = face_image[crop_bbox[1]:crop_bbox[3], crop_bbox[0]:crop_bbox[2]]
cv2.imwrite(output_image_path, crop_image)
@torch.no_grad()
def process(self,
image_path,
audio_path,
output_path,
min_resolution=512,
inference_steps=25,
dynamic_scale=1.0,
keep_resolution=False,
seed=None):
config = self.config
device = self.device
pipe = self.pipe
whisper = self.whisper
audio2token = self.audio2token
audio2bucket = self.audio2bucket
image_encoder = self.image_encoder
# 시드 설정
if seed:
config.seed = seed
config.num_inference_steps = inference_steps
config.motion_bucket_scale = dynamic_scale
seed_everything(config.seed)
video_path = output_path.replace('.mp4', '_noaudio.mp4')
audio_video_path = output_path
# 오디오+이미지 -> tensor
test_data = image_audio_to_tensor(
self.face_det,
self.feature_extractor,
image_path,
audio_path,
limit=-1, # 전체 사용
image_size=min_resolution,
area=config.area
)
if test_data is None:
return -1
height, width = test_data['ref_img'].shape[-2:]
if keep_resolution:
imSrc_ = Image.open(image_path).convert('RGB')
raw_w, raw_h = imSrc_.size
resolution = f'{raw_w//2*2}x{raw_h//2*2}'
else:
resolution = f'{width}x{height}'
# 여기서 test(...) 호출
video = test(
pipe,
config,
wav_enc=whisper,
audio_pe=audio2token,
audio2bucket=audio2bucket,
image_encoder=image_encoder,
width=width,
height=height,
batch=test_data,
)
# 중간프레임 보간
if config.use_interframe:
rife = self.rife
out = video.to(device)
results = []
video_len = out.shape[2]
for idx in tqdm(range(video_len - 1), ncols=0):
I1 = out[:, :, idx]
I2 = out[:, :, idx + 1]
middle = rife.inference(I1, I2).clamp(0, 1).detach()
results.append(out[:, :, idx])
results.append(middle)
results.append(out[:, :, video_len - 1])
video = torch.stack(results, 2).cpu()
# 비디오 저장
save_videos_grid(video, video_path, n_rows=video.shape[0], fps=config.fps * (2 if config.use_interframe else 1))
# 오디오 합성 후 최종 mp4
os.system(
f"ffmpeg -i '{video_path}' -i '{audio_path}' -s {resolution} "
f"-vcodec libx264 -acodec aac -crf 18 -shortest '{audio_video_path}' -y; rm '{video_path}'"
)
return 0