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inference.py
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inference.py
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# pylint: disable=E1101
# scripts/inference.py
"""
This script contains the main inference pipeline for processing audio and image inputs to generate a video output.
The script imports necessary packages and classes, defines a neural network model,
and contains functions for processing audio embeddings and performing inference.
The main inference process is outlined in the following steps:
1. Initialize the configuration.
2. Set up runtime variables.
3. Prepare the input data for inference (source image, face mask, and face embeddings).
4. Process the audio embeddings.
5. Build and freeze the model and scheduler.
6. Run the inference loop and save the result.
Usage:
This script can be run from the command line with the following arguments:
- audio_path: Path to the audio file.
- image_path: Path to the source image.
- face_mask_path: Path to the face mask image.
- face_emb_path: Path to the face embeddings file.
- output_path: Path to save the output video.
Example:
python scripts/inference.py --audio_path audio.wav --image_path image.jpg
--face_mask_path face_mask.png --face_emb_path face_emb.pt --output_path output.mp4
"""
import argparse
import os
import torch
from diffusers import AutoencoderKL, DDIMScheduler
from omegaconf import OmegaConf
from torch import nn
from hallo.animate.face_animate import FaceAnimatePipeline
from hallo.datasets.audio_processor import AudioProcessor
from hallo.datasets.image_processor import ImageProcessor
from hallo.models.audio_proj import AudioProjModel
from hallo.models.face_locator import FaceLocator
from hallo.models.image_proj import ImageProjModel
from hallo.models.unet_2d_condition import UNet2DConditionModel
from hallo.models.unet_3d import UNet3DConditionModel
from hallo.utils.config import filter_non_none
from hallo.utils.util import tensor_to_video
class Net(nn.Module):
"""
The Net class combines all the necessary modules for the inference process.
Args:
reference_unet (UNet2DConditionModel): The UNet2DConditionModel used as a reference for inference.
denoising_unet (UNet3DConditionModel): The UNet3DConditionModel used for denoising the input audio.
face_locator (FaceLocator): The FaceLocator model used to locate the face in the input image.
imageproj (nn.Module): The ImageProjector model used to project the source image onto the face.
audioproj (nn.Module): The AudioProjector model used to project the audio embeddings onto the face.
"""
def __init__(
self,
reference_unet: UNet2DConditionModel,
denoising_unet: UNet3DConditionModel,
face_locator: FaceLocator,
imageproj,
audioproj,
):
super().__init__()
self.reference_unet = reference_unet
self.denoising_unet = denoising_unet
self.face_locator = face_locator
self.imageproj = imageproj
self.audioproj = audioproj
def forward(self,):
"""
empty function to override abstract function of nn Module
"""
def get_modules(self):
"""
Simple method to avoid too-few-public-methods pylint error
"""
return {
"reference_unet": self.reference_unet,
"denoising_unet": self.denoising_unet,
"face_locator": self.face_locator,
"imageproj": self.imageproj,
"audioproj": self.audioproj,
}
def process_audio_emb(audio_emb):
"""
Process the audio embedding to concatenate with other tensors.
Parameters:
audio_emb (torch.Tensor): The audio embedding tensor to process.
Returns:
concatenated_tensors (List[torch.Tensor]): The concatenated tensor list.
"""
concatenated_tensors = []
for i in range(audio_emb.shape[0]):
vectors_to_concat = [
audio_emb[max(min(i + j, audio_emb.shape[0]-1), 0)]for j in range(-2, 3)]
concatenated_tensors.append(torch.stack(vectors_to_concat, dim=0))
audio_emb = torch.stack(concatenated_tensors, dim=0)
return audio_emb
def inference_process(args: argparse.Namespace):
"""
Perform inference processing.
Args:
args (argparse.Namespace): Command-line arguments.
This function initializes the configuration for the inference process. It sets up the necessary
modules and variables to prepare for the upcoming inference steps.
"""
# 1. init config
cli_args = filter_non_none(vars(args))
config = OmegaConf.load(args.config)
config = OmegaConf.merge(config, cli_args)
source_image_path = config.source_image
driving_audio_path = config.driving_audio
save_path = config.save_path
if not os.path.exists(save_path):
os.makedirs(save_path)
motion_scale = [config.pose_weight, config.face_weight, config.lip_weight]
# 2. runtime variables
device = torch.device(
"cuda") if torch.cuda.is_available() else torch.device("cpu")
if config.weight_dtype == "fp16":
weight_dtype = torch.float16
elif config.weight_dtype == "bf16":
weight_dtype = torch.bfloat16
elif config.weight_dtype == "fp32":
weight_dtype = torch.float32
else:
weight_dtype = torch.float32
# 3. prepare inference data
# 3.1 prepare source image, face mask, face embeddings
img_size = (config.data.source_image.width,
config.data.source_image.height)
clip_length = config.data.n_sample_frames
face_analysis_model_path = config.face_analysis.model_path
with ImageProcessor(img_size, face_analysis_model_path) as image_processor:
source_image_pixels, \
source_image_face_region, \
source_image_face_emb, \
source_image_full_mask, \
source_image_face_mask, \
source_image_lip_mask = image_processor.preprocess(
source_image_path, save_path, config.face_expand_ratio)
# 3.2 prepare audio embeddings
sample_rate = config.data.driving_audio.sample_rate
assert sample_rate == 16000, "audio sample rate must be 16000"
fps = config.data.export_video.fps
wav2vec_model_path = config.wav2vec.model_path
wav2vec_only_last_features = config.wav2vec.features == "last"
audio_separator_model_file = config.audio_separator.model_path
with AudioProcessor(
sample_rate,
fps,
wav2vec_model_path,
wav2vec_only_last_features,
os.path.dirname(audio_separator_model_file),
os.path.basename(audio_separator_model_file),
os.path.join(save_path, "audio_preprocess")
) as audio_processor:
audio_emb, audio_length = audio_processor.preprocess(driving_audio_path, clip_length)
# 4. build modules
sched_kwargs = OmegaConf.to_container(config.noise_scheduler_kwargs)
if config.enable_zero_snr:
sched_kwargs.update(
rescale_betas_zero_snr=True,
timestep_spacing="trailing",
prediction_type="v_prediction",
)
val_noise_scheduler = DDIMScheduler(**sched_kwargs)
sched_kwargs.update({"beta_schedule": "scaled_linear"})
vae = AutoencoderKL.from_pretrained(config.vae.model_path)
reference_unet = UNet2DConditionModel.from_pretrained(
config.base_model_path, subfolder="unet")
denoising_unet = UNet3DConditionModel.from_pretrained_2d(
config.base_model_path,
config.motion_module_path,
subfolder="unet",
unet_additional_kwargs=OmegaConf.to_container(
config.unet_additional_kwargs),
use_landmark=False,
)
face_locator = FaceLocator(conditioning_embedding_channels=320)
image_proj = ImageProjModel(
cross_attention_dim=denoising_unet.config.cross_attention_dim,
clip_embeddings_dim=512,
clip_extra_context_tokens=4,
)
audio_proj = AudioProjModel(
seq_len=5,
blocks=12, # use 12 layers' hidden states of wav2vec
channels=768, # audio embedding channel
intermediate_dim=512,
output_dim=768,
context_tokens=32,
).to(device=device, dtype=weight_dtype)
audio_ckpt_dir = config.audio_ckpt_dir
# Freeze
vae.requires_grad_(False)
image_proj.requires_grad_(False)
reference_unet.requires_grad_(False)
denoising_unet.requires_grad_(False)
face_locator.requires_grad_(False)
audio_proj.requires_grad_(False)
reference_unet.enable_gradient_checkpointing()
denoising_unet.enable_gradient_checkpointing()
net = Net(
reference_unet,
denoising_unet,
face_locator,
image_proj,
audio_proj,
)
m,u = net.load_state_dict(
torch.load(
os.path.join(audio_ckpt_dir, "net.pth"),
map_location="cpu",
),
)
assert len(m) == 0 and len(u) == 0, "Fail to load correct checkpoint."
print("loaded weight from ", os.path.join(audio_ckpt_dir, "net.pth"))
# 5. inference
pipeline = FaceAnimatePipeline(
vae=vae,
reference_unet=net.reference_unet,
denoising_unet=net.denoising_unet,
face_locator=net.face_locator,
scheduler=val_noise_scheduler,
image_proj=net.imageproj,
)
pipeline.to(device=device, dtype=weight_dtype)
audio_emb = process_audio_emb(audio_emb)
source_image_pixels = source_image_pixels.unsqueeze(0)
source_image_face_region = source_image_face_region.unsqueeze(0)
source_image_face_emb = source_image_face_emb.reshape(1, -1)
source_image_face_emb = torch.tensor(source_image_face_emb)
source_image_full_mask = [
(mask.repeat(clip_length, 1))
for mask in source_image_full_mask
]
source_image_face_mask = [
(mask.repeat(clip_length, 1))
for mask in source_image_face_mask
]
source_image_lip_mask = [
(mask.repeat(clip_length, 1))
for mask in source_image_lip_mask
]
times = audio_emb.shape[0] // clip_length
tensor_result = []
generator = torch.manual_seed(42)
for t in range(times):
print(f"[{t+1}/{times}]")
if len(tensor_result) == 0:
# The first iteration
motion_zeros = source_image_pixels.repeat(
config.data.n_motion_frames, 1, 1, 1)
motion_zeros = motion_zeros.to(
dtype=source_image_pixels.dtype, device=source_image_pixels.device)
pixel_values_ref_img = torch.cat(
[source_image_pixels, motion_zeros], dim=0) # concat the ref image and the first motion frames
else:
motion_frames = tensor_result[-1][0]
motion_frames = motion_frames.permute(1, 0, 2, 3)
motion_frames = motion_frames[0-config.data.n_motion_frames:]
motion_frames = motion_frames * 2.0 - 1.0
motion_frames = motion_frames.to(
dtype=source_image_pixels.dtype, device=source_image_pixels.device)
pixel_values_ref_img = torch.cat(
[source_image_pixels, motion_frames], dim=0) # concat the ref image and the motion frames
pixel_values_ref_img = pixel_values_ref_img.unsqueeze(0)
audio_tensor = audio_emb[
t * clip_length: min((t + 1) * clip_length, audio_emb.shape[0])
]
audio_tensor = audio_tensor.unsqueeze(0)
audio_tensor = audio_tensor.to(
device=net.audioproj.device, dtype=net.audioproj.dtype)
audio_tensor = net.audioproj(audio_tensor)
pipeline_output = pipeline(
ref_image=pixel_values_ref_img,
audio_tensor=audio_tensor,
face_emb=source_image_face_emb,
face_mask=source_image_face_region,
pixel_values_full_mask=source_image_full_mask,
pixel_values_face_mask=source_image_face_mask,
pixel_values_lip_mask=source_image_lip_mask,
width=img_size[0],
height=img_size[1],
video_length=clip_length,
num_inference_steps=config.inference_steps,
guidance_scale=config.cfg_scale,
generator=generator,
motion_scale=motion_scale,
)
tensor_result.append(pipeline_output.videos)
tensor_result = torch.cat(tensor_result, dim=2)
tensor_result = tensor_result.squeeze(0)
tensor_result = tensor_result[:, :audio_length]
output_file = config.output
# save the result after all iteration
tensor_to_video(tensor_result, output_file, driving_audio_path)
return output_file
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument(
"-c", "--config", default="configs/inference/default.yaml")
parser.add_argument("--source_image", type=str, required=False,
help="source image")
parser.add_argument("--driving_audio", type=str, required=False,
help="driving audio")
parser.add_argument(
"--output", type=str, help="output video file name", default=".cache/output.mp4")
parser.add_argument(
"--pose_weight", type=float, help="weight of pose", required=False)
parser.add_argument(
"--face_weight", type=float, help="weight of face", required=False)
parser.add_argument(
"--lip_weight", type=float, help="weight of lip", required=False)
parser.add_argument(
"--face_expand_ratio", type=float, help="face region", required=False)
parser.add_argument(
"--audio_ckpt_dir", "--checkpoint", type=str, help="specific checkpoint dir", required=False)
command_line_args = parser.parse_args()
inference_process(command_line_args)