Loopy方法模块设计分析

Creato il: 4 ottobre 2024

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Domanda

请分析下面的论文,设计主要的模块,并给出每个模块的参数。要详细的指出模块设计和参数对应于论文中的哪些内容。代码要有详细的中文注释。下面是论文:

Loopy

3 Method

In this section, we will introduce our method, Loopy. First, we will provide an overview of the framework, including the input, output, and key components of Loopy. Second, we will focus on the design of the inter-/intra-temporal modules, including the temporal segment module. Third, we will detail the implementation of the audio condition module. Finally, we will describe the implementation details during the training and testing of Loopy.

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Figure 2: The framework of Loopy. it removes the commonly used face locator and speed layer modules found in existing methods. Instead, it achieves flexible and natural motion generation through the proposed inter/intra-clip temporal layer and audio-to-latents module.

3.1 Framework

Our method is built upon Stable Diffusion (SD) and uses its initialization weights. SD is a textto-image diffusion model based on the Latent Diffusion Model (LDM) (Rombach et al., 2022). It employs a pretrained VQ-VAE (Kingma, 2013; Van Den Oord et al., 2017) E to transform images from pixel space to latent space. During training, images are first converted to latents, i.e., z0 = E(I).

Gaussian noise ϵ is then added to the latents in the latent space based on the Denoising Diffusion Probabilistic Model (DDPM) (Ho et al., 2020) for t steps, resulting in a noisy latent zt. The denoising net takes zt as input to predict ϵ. The training objective can be formulated as follows:

L=Ezt,c,ϵN(0,1),t[ϵϵθ(zt,t,c)22],L=\mathbb{E}_{z_{t},c,\epsilon\sim{\mathcal{N}}(0,1),t}\left[\|\epsilon-\epsilon_{\theta}(z_{t},t,c)\|_{2}^{2}\right],
(1)(1)
, (1)
where ϵθ represents the denoising net, including condition-related attention modules, which is the main part Loopy aims to improve. c represents the text condition embedding in SD, and in Loopy, it includes audio, motion frames, and other additional information influencing the final generation.

During testing, the final image is obtained by sampling from Gaussian noise and removing the noise based on DDIM (Song et al., 2020) or DDPM.

As demonstrated in Figure 2, in the Loopy, the inputs to the denoising net include noisy latents, i.e., the VQ-VAE encoded image latents zt. Unlike original SD, where the input is a single image, here it is a sequence of images representing a video clip. The inputs also include reference latents cref (encoded reference image latents via VQ-VAE), audio embedding caudio (audio features of the current clip), motion frames cmf (image latents of the M frames sequence from the preceding clips), and timestep t. During training, additional facial movement-related features are involved:
clmk (facial keypoint sequence of the current clip), cmov (head motion variance of the current clip), and cexp (expression variance of the current clip). The output is the predicted noise ϵ. The denoising network employs a Dual U-Net architecture (Hu, 2024; Zhu et al., 2023). This architecture includes an additional reference net module, which replicates the original SD U-Net structure but utilizes reference latents cref as input. The reference net operates concurrently with the denoising U-Net.

During the spatial attention layer computation in the denoising U-Net, the key and value features from corresponding positions in the reference net are concatenated with the denoising U-Net's features along the token dimension before proceeding with the attention module computation. This design enables the denoising U-Net to effectively incorporate reference image features from the reference net.

Additionally, the reference net also takes motion frames latents cmf as input for feature extraction, allowing these features to be utilized in subsequent temporal attention computations.

3.2 Inter/Intra- Clip Temporal Layers Design

Here, we introduce the design of the proposed inter-/intra-clip temporal modules. Unlike existing methods (Tian et al., 2024; Xu et al., 2024a; Chen et al., 2024; Wang et al., 2024) that process motion frame latents and noisy latents features simultaneously through a single temporal layer, Loopy employs two temporal attention layers: the inter-clip temporal layer and the intra-clip temporal layer.

The inter-clip temporal layer first handles the cross-clip temporal relationships between motion frame latents and noisy latents, while the intra-clip temporal layer focuses on the temporal relationships within the noisy latents of the current clip.

First, we introduce the inter-clip temporal layer, initially ignoring the temporal segment module in Figure 2. We first collect the m image latents from the preceding clip, referred to as motion frames latents cmf . Similar to cref , these latents are processed frame-by-frame through the reference network for feature extraction. Within each residual block, the features xcmf obtained from the reference network are concatenated with the features x from the denoising U-Net along the temporal dimension. To distinguish the types of latents, we add learnable temporal embeddings. Subsequently, self-attention is computed on the concatenated tokens along the temporal dimension, referred to as temporal attention. The intra-clip temporal layer differs in that its input does not include features from motion frames latents, it only processes features from the noisy latents of the current clip. By separating the dual temporal layers, the model can better handle the aggregation of different semantic temporal features in cross-clip temporal relationships.

Figure 3: The illustration of the temporal segment module

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and the inter/intra- clip temporal layers. The former allows us to expand the motion frame to cover over 100 frames, while the later enables the modeling of long-term motion dependency.

Due to the independent design of the inter-clip temporal layer, Loopy is better equipped to model the motion relationships among clips. To enhance this capability, we introduce the temporal segment module before cmf enters the reference network. This module not only extends the temporal range covered by the inter-clip temporal layer but also accounts for the variations in information due to the varying distances of different clips from the current clip, as illustrated in Figure 3. The temporal segment module divides the original motion frame into multiple segments and extracts representative motion frames from each segment to abstract the segment. Based on these abstracted motion frames, we recombine them to obtain motion frame latents for subsequent computations. For the segmentation process, we define two hyperparameters: stride s and expand ratio r. The stride s represents the number of abstract motion frames in each segment, while the expand ratio r is used to calculate the length of the original motion frames in each segment. The number of frames in the i-th segment, ordered from closest to farthest from the current clip, is given by s × r
(i−1). For example, with a stride s = 4 and an expand ratio r = 2, the first segment would contain 4 frames, the second segment would contain 8 frames, and the third segment would contain 16 frames. For the abstraction process after segmentation, we default to uniform sampling within each segment. In the experimental section, we investigate different segmentation parameters and abstraction methods. Since segmentation and abstraction directly affect the learning of long-term motion dependency, different methods have a significant impact on the results. The output of the temporal segment module, c o mf , can be defined as:
where start_indexi =Pk−1 j=0 r j· s and end_indexi = start_indexi + r k· s, with k =is
. c o mf,i represents the mean value of the i-th element of the output, calculated from the sequence cmf .

The temporal segment module rapidly expands the temporal coverage of the motion frames input to the inter-clip temporal layer while maintaining acceptable computational complexity. For closer frames, a lower expansion rate retains more details, while for distant frames, a higher expansion rate covers a longer duration. This approach helps the model better capture motion style from long-term information and generate temporally natural motion without spatial templates.

3.3 Audio Condition Module

For the audio condition, we first use wav2vec (Baevski et al., 2020; Schneider et al., 2019) for audio feature extraction. Following the approach in EMO, we concatenate the hidden states from each layer of the wav2vec network to obtain multi-scale audio features. For each video frame, we concatenate the audio features of the two preceding and two succeeding frames, resulting in a 5-frame audio feature as audio embeddings caudio for the current frame. Initially, in each residual block, we use cross-attention with noisy latents as the query and audio embedding caudio as the key and value to compute an attended audio feature. This attended audio feature is then added to the noisy latents feature obtained from self-attention, resulting in a new noisy latents feature. This provides a preliminary audio condition.

Additionally, as illustrated in Figure 4, we introduce the Audio-to-Latents module. This module maps conditions that have both strong and weak correlations with portrait motion, such as audio, to a shared motion latents space. These mapped conditions serve as the final condition features, thereby enhancing the modeling of the relationship between audio and portrait movements based on motion latents. Specifically, we maintain a set of learnable embeddings. For each input condition, we map it to a query feature using a fully connected (FC) layer, while the learnable embeddings serve as key and value features for attention computation to obtain a new feature based on the learnable embeddings. These new features, referred to as motion latents, replace the input condition in subsequent computations. These motion latents are then dimensionally transformed via an FC layer and added to the timestep embedding features for subsequent network computation.

During training, we sample an input condition for the Audio-to-Latents module with equal probability from audio embeddings and facial movement-related features such as landmarks, face absolute motion variance, and face expression motion variance. During testing, we only input audio to generate motion latents. By leveraging features strongly correlated with portrait movement, the model utilizes learnable embeddings to control motion. Consequently, transforming audio embeddings into motion latents can also allows for more direct influence on portrait motion.

Figure 4: The audio-to-latents module.

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3.4 Training Strategies

Conditions Mask and Dropout. In the Loopy framework, various conditions are involved, including the reference image cref , audio features caudio, preceding frame motion frames cmf , and motion latents cml representing audio and facial motion conditions. Due to the overlapping nature of the information contained within these conditions, to better learn the unique information specific to each condition, we used distinct masking strategies for the conditions during the training process. During training, caudio and motion latents are masked to all-zero features with a 10% probability. For cref and cmf , we design specific dropout and mask strategies due to their conflicting relationship. cmf also provides appearance information and is closer to the current clip compared to cref , leading the model to heavily rely on motion frames rather than the reference image. This can cause color shifts and artifacts in long video sequences during inference. To address this, cref has a 15% probability of being dropped, meaning the denoising U-Net will not concatenate features from the reference network during self-attention computation. When cref is dropped, motion frames are also dropped, meaning the denoising U-Net will not concatenate features from the reference network during temporal attention computation. Additionally, motion frames have an independent 40% probability of being masked to all-zero features.

Multistage Training. Following AnimateAnyone (Hu, 2024) and EMO (Tian et al., 2024), we employ a 2-stage training process. In the first stage, the model is trained without temporal layers and the audio condition module. The inputs to the model are the noisy latents of the target single-frame image and reference image latents, focusing the model on an image-level pose variations task. After completing the first stage, we proceed to the second stage, where the model is initialized with the reference network and denoising U-Net from the first stage. We then add the inter-/intra-clip temporal layers and the audio condition module for full training to obtain the final model. Inference. During Inference, we perform class-free guidance (Ho & Salimans, 2022) using multiple conditions. Specifically, we conduct three inference runs, differing in whether certain conditions are dropped. The final noise ef inal is computed as:
ef inal = audio_ratio × (eaudio − eref ) + ref_ratio × (eref − ebase) + ebase where eaudio includes all conditions cref , caudio, cmf , cml, eref but mask caudio to all-zero features, and ebase masks caudio to all-zero features and removes the concatenation of reference network features in the denoising U-Net's self-attention. This approach allows us to control the model's final output in terms of adherence to the reference image and alignment with the audio. The audio ratio is set to 5 and the reference ratio is set to 3. We use DDIM sampling with 25 denoising steps to complete the inference.

3.5 Experiments

Datasets. For training Data, We collected talking head video data from the internet, excluding videos with low lip sync scores, excessive head movement, extreme rotation, or incomplete head exposure.

This resulted in 160 hours of cleaned training data. Additionally, we supplemented the training data with public datasets such as HDTF (Zhang et al., 2021). For the test set, we randomly sampled 100 videos from CelebV-HQ (Zhu et al., 2022) (a public high-quality celebrity video dataset with mixed scenes) and RAVDESS (Kaggle) (a public high-definition indoor talking scene dataset with rich emotions). To test the generalization ability of diffusion-based models, we also collected 20 portrait test images, including real people, anime, side face, and humanoid crafts of different materials, along with 20 audio clips, including speeches, singing, rap and emotionally rich speech. We refer to this test set as the openset test set. Implementation Details. We trained our model using 24 Nvidia A100 GPUs with a batch size of 24, using the AdamW (Loshchilov & Hutter, 2017) optimizer with a learning rate of 1e-5. The generated video length was set to 12 frames, and the motion frame length was set to 124 frames, representing the preceding 124 frames of the current 12-frame video. After temporal squeezing, this was compressed to 20 motion frame latents. During training, the reference image was randomly selected from a frame within the video clip. For the facial motion information required to train audo-to-motion module, we used DWPose (Yang et al., 2023) to detect facial keypoints for the current 12 frames. The variance of the absolute position of the nose tip across these 12 frames was used as the absolute head motion variance. The variance of the displacement of the upper half of the facial keypoints (37 keypoints)
relative to the nose tip across these 12 frames was used as the expression variance. The training videos were uniformly processed at 25 fps and cropped to 512x512 portrait videos.

Metrics. We assess image quality with the IQA metric (Wu et al., 2023), video motion with VBench's smooth metric (Huang et al., 2024), and audio-visual synchronization using SyncC and SyncD (Prajwal et al., 2020). For the CelebvHQ and RAVDESS test sets, which have corresponding ground truth videos, we will also compute FVD (Unterthiner et al., 2019), E-FID (Tian et al., 2024)
and FID metrics for comparison. Additionally, to compare the global motion (denoted as Glo) and dynamic expressions (denoted as Exp) of the portrait, we calculated the variance values based on key points of the nose and the upper face, specifically excluding the mouth area. We also computed the values for the ground truth video as a reference for comparison. For the openset testset, which lacks ground truth video references, we conducted subjective evaluations. Ten experienced users were invited to assess six key dimensions: identity consistency, video synthesis quality, audio-emotion
下面是论文中相关图片的描述:

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markdown
This image illustrates the overall framework of the Loopy model for audio-driven portrait video generation. The diagram shows the key components and data flow of the system. Let me break it down in detail: 1. Input: - Ref Image: A reference image of the target person - Motion Frames: A sequence of video frames from preceding clips - Audio: The input audio that will drive the portrait animation 2. Appearance Processing: - The reference image and motion frames are passed through an encoder "E" (likely a VQ-VAE encoder) - The encoded features then go through spatial attention and inter-clip temporal layers 3. Audio Processing: - The audio input goes through a Wav2Vec model for feature extraction - These features, along with Time Embedding, Translation, and Expression information, are fed into an Audio2Latents module 4. Temporal Modeling: - A Temporal Segment Module (TSM) processes the motion frames - The output from TSM and the appearance features are combined 5. Reference Net: - The combined features are processed through a Reference Net, which is likely similar in structure to the main denoising U-Net 6. Denoising Net: - The core of the model is a U-Net based denoising network - It takes in noise as input, along with the processed audio and appearance features - The network consists of multiple layers with spatial attention, audio attention, and intra-clip temporal modeling 7. Output: - The final output is a sequence of Generated Images, representing the animated portrait video 8. Key Features: - The model uses "frozen parameters" in certain components (indicated by snowflake icons) - It employs various types of attention mechanisms: spatial, audio, and temporal - The architecture allows for long-term motion dependency modeling through the inter-clip and intra-clip temporal layers 9. Data Flow: - The diagram uses different arrow styles to indicate various types of data flow (e.g., reference flow, motion frames flow, audio flow) This architecture aims to generate natural and expressive portrait animations driven solely by audio input, without relying on additional spatial motion templates or constraints used in previous methods. The inter-clip and intra-clip temporal modeling, along with the audio-to-latents module, are key innovations of this approach.

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markdown
This image illustrates the key components of the Loopy framework for audio-driven portrait avatar generation. The image is divided into two main sections: 1. Temporal Segment Module: - On the left side, there's a film strip showing a sequence of portrait images of a woman. - The film strip is connected to a "segment" block, which leads to an "abstract" representation. - From the abstract representation, arrows point to multiple output frames labeled m^1, m^2, m^3, etc., representing different motion frames. 2. Inter/Intra-Clip Temporal Layer: - On the right side, there's a diagram showing the processing of motion frame latents and noise latents. - It includes boxes for "Motion Frame Latents" and "Noise Latents". - These are combined with "Learnable Embeddings" and processed through layers labeled "Inter-clip temporal layer" and "Intra-clip temporal layer". Key aspects of Loopy illustrated in this image: 1. The Temporal Segment Module allows the model to consider a longer sequence of preceding frames, expanding the temporal context beyond just the immediately preceding frames. 2. The Inter/Intra-Clip Temporal Layer shows how the model processes both inter-clip relationships (between the current clip and previous clips) and intra-clip relationships (within the current clip). 3. The use of learnable embeddings suggests the model can adapt and learn temporal patterns over time. 4. The overall structure demonstrates how Loopy aims to capture long-term motion dependencies, which is crucial for generating natural and consistent movements in the avatar. This diagram effectively visualizes how Loopy addresses the challenge of generating more natural and expressive motions in audio-driven avatars by leveraging longer-term temporal information and separating inter- and intra-clip processing.

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markdown
Based on the content of the Loopy paper, this image illustrates the framework and key components of the Loopy model for audio-driven portrait avatar generation. Let me describe the main elements of the diagram: 1. Input: The diagram shows multiple inputs: - Translation (likely representing text or language input) - Audio (represented by a waveform) - Expression (shown as colored blocks) 2. Feature Extraction: Each input passes through a "fc" (fully connected) layer, represented by yellow boxes. This extracts features from the inputs. 3. Q (Query) Generation: The features from translation, audio, and expression are combined to form a "Q" (Query) for the attention mechanism. 4. Attention Mechanism: The center of the diagram shows an "Attention" block, which is a key component of the Loopy model. It takes Q as input, along with K (Key) and V (Value) from other parts of the model. 5. Time Embedding: There's a clock icon representing "time embedding", which feeds into the attention mechanism. This allows the model to incorporate temporal information. 6. K and V Generation: On the right side, there are two "fc" layers generating K and V for the attention mechanism. These seem to be derived from learnable tokens (represented by pencil icons). 7. Output: The output of the attention mechanism, combined with the time embedding, feeds into a "Denoising UNet" (likely referring to the denoising U-Net architecture mentioned in the paper). 8. Additional Elements: - There's an "Elementwise add" operation, represented by a "+" symbol in a green circle. - Dotted lines indicate "Train only" connections, suggesting these pathways are only active during training. - Solid arrows represent the main data flow in the model. This diagram illustrates Loopy's novel approach of using inter- and intra-clip temporal modules and the audio-to-latents module to generate more natural and expressive portrait animations from audio input, without relying on spatial motion templates used in previous methods.

Risposta

设计核心模块

我正在为音频驱动的肖像视频生成系统设计核心模块,包括基于Stable Diffusion的框架、临时段模块、音频条件模块等。

Risposta

好的,根据您提供的论文内容,下面我将对Loopy方法进行详细分析,设计其主要模块,并为每个模块提供相应的参数。此外,还将提供每个模块设计和参数对应于论文中的具体内容,并附上详细的中文注释代码示例。

一、论文分析与模块设计

1. 框架概述(对应论文3.1节)

Loopy方法基于Stable Diffusion(SD)框架,主要修改和扩展了以下几个部分:

  • 输入

    • 噪声潜在向量(noisy latents)
    • 参考图像潜在向量(reference latents c_ref
    • 音频嵌入(audio embedding c_audio
    • 运动帧潜在向量(motion frames c_mf
    • 时间步长(timestep t
    • 训练时的额外面部运动特征(如面部关键点序列 c_lmk、头部运动方差 c_mov、表情方差 c_exp
  • 输出

    • 预测的噪声 ε
  • 关键组件

    • 双重U-Net架构(Dual U-Net)
    • 参考网络模块(Reference Net)
    • 交互/内部时间层(Inter/Intra-Clip Temporal Layers)
    • 音频条件模块(Audio Condition Module)
    • 时间段模块(Temporal Segment Module)

2. 主要模块设计

根据论文内容,Loopy方法主要由以下几个模块组成:

  1. 编码器模块(Encoder Module)

    • 使用预训练的VQ-VAE编码器将图像转换为潜在空间。
  2. 参考网络模块(Reference Net Module)

    • 处理参考图像潜在向量 c_ref 和运动帧潜在向量 c_mf
  3. 双重U-Net模块(Dual U-Net Module)

    • 包含主U-Net和参考网络U-Net,用于去噪。
  4. 时间段模块(Temporal Segment Module)

    • 对运动帧进行分段和抽象处理,扩展时间覆盖范围。
  5. 交互/内部时间层(Inter/Intra-Clip Temporal Layers Module)

    • 分别处理跨剪辑和内部剪辑的时间关系。
  6. 音频条件模块(Audio Condition Module)

    • 包括Wav2Vec音频特征提取和Audio-to-Latents模块。
  7. 训练策略模块(Training Strategies Module)

    • 包含条件掩码和多阶段训练策略。

3. 各模块详细设计与参数

3.1 编码器模块(Encoder Module)

  • 功能:将输入图像通过VQ-VAE编码器转换为潜在向量。
  • 对应内容:论文3.1节提到的VQ-VAE编码器E
  • 参数
    • 输入图像尺寸:512x512
    • 潜在空间维度:根据VQ-VAE预训练模型设置

3.2 参考网络模块(Reference Net Module)

  • 功能:处理参考图像潜在向量c_ref和运动帧潜在向量c_mf,提取特征用于后续的注意力计算。
  • 对应内容:论文3.1节中Dual U-Net架构中的参考网络部分。
  • 参数
    • 网络层数:与主U-Net一致
    • 输入维度:与VQ-VAE编码器输出维度一致

3.3 双重U-Net模块(Dual U-Net Module)

  • 功能:主U-Net用于去噪,参考网络U-Net用于处理参考特征。空间注意力层中融合参考特征。
  • 对应内容:论文3.1节中提到的Dual U-Net架构。
  • 参数
    • U-Net层数:根据Stable Diffusion架构
    • 注意力机制类型:空间注意力
    • 输入潜在向量维度:与VQ-VAE输出一致

3.4 时间段模块(Temporal Segment Module)

  • 功能:将运动帧分段,抽取每段的代表帧以扩展时间覆盖范围,便于跨剪辑时间关系建模。
  • 对应内容:论文3.2节中关于Temporal Segment Module的设计。
  • 参数
    • 步幅(stride s):4
    • 扩展比例(expand ratio r):2
    • 分段数量:根据运动帧总数和分段策略确定

3.5 交互/内部时间层(Inter/Intra-Clip Temporal Layers Module)

  • 功能
    • Inter-clip Temporal Layer:处理跨剪辑的时间关系。
    • Intra-clip Temporal Layer:处理当前剪辑内部的时间关系。
  • 对应内容:论文3.2节中关于Inter/Intra-Clip Temporal Layers的设计。
  • 参数
    • 注意力头数:与主U-Net一致
    • 时间嵌入维度:与时间步长t嵌入一致
    • 学习的时间嵌入参数:可学习参数数量与时间维度一致

3.6 音频条件模块(Audio Condition Module)

  • 功能
    • 使用Wav2Vec提取音频特征。
    • 通过Audio-to-Latents模块将音频特征映射到运动潜在空间。
  • 对应内容:论文3.3节中关于Audio Condition Module的设计。
  • 参数
    • Wav2Vec层数:预训练模型层数
    • Audio-to-Latents映射层:
      • 全连接层输入维度:Wav2Vec特征维度
      • 全连接层输出维度:与运动潜在空间一致
    • 学习嵌入数量:根据运动潜在空间设定

3.7 训练策略模块(Training Strategies Module)

  • 功能
    • 应用条件掩码和dropout策略以增强模型的泛化能力。
    • 采用多阶段训练策略。
  • 对应内容:论文3.4节中关于训练策略的描述。
  • 参数
    • 条件掩码概率:如c_audio和运动潜在向量的10%掩码概率,c_ref的15%掩码概率等。
    • 学习率:1e-5
    • 批次大小:24
    • 优化器:AdamW

二、代码实现

下面将结合上述模块设计,提供一个简化的代码示例,使用PyTorch框架实现Loopy的主要模块。代码中包含详细的中文注释,以便理解每个部分的功能和参数设置。

1. 导入必要的库

python
import torch import torch.nn as nn import torch.nn.functional as F from transformers import Wav2Vec2Model

2. 编码器模块

python
class VQVAEEncoder(nn.Module): def __init__(self, latent_dim=256): super(VQVAEEncoder, self).__init__() # 假设使用预训练的VQ-VAE编码器,这里只是示例 self.encoder = nn.Sequential( nn.Conv2d(3, 64, kernel_size=4, stride=2, padding=1), # 512x512 -> 256x256 nn.ReLU(), nn.Conv2d(64, 128, kernel_size=4, stride=2, padding=1), # 256x256 -> 128x128 nn.ReLU(), nn.Conv2d(128, latent_dim, kernel_size=4, stride=2, padding=1), # 128x128 -> 64x64 nn.ReLU(), ) def forward(self, x): return self.encoder(x)

3. 参考网络模块

python
class ReferenceNet(nn.Module): def __init__(self, latent_dim=256): super(ReferenceNet, self).__init__() # 参考网络与主U-Net结构一致 self.ref_net = nn.Sequential( nn.Conv2d(latent_dim, 64, kernel_size=3, padding=1), nn.ReLU(), nn.Conv2d(64, 128, kernel_size=3, padding=1), nn.ReLU(), nn.Conv2d(128, latent_dim, kernel_size=3, padding=1), nn.ReLU(), ) def forward(self, cref, cmf): # 处理参考图像潜在向量 cref_features = self.ref_net(cref) # 处理运动帧潜在向量 cmf_features = self.ref_net(cmf) return cref_features, cmf_features

4. 时间段模块

python
class TemporalSegmentModule(nn.Module): def __init__(self, stride=4, expand_ratio=2): super(TemporalSegmentModule, self).__init__() self.stride = stride self.expand_ratio = expand_ratio def forward(self, cmf): # cmf: [batch_size, num_frames, latent_dim, H, W] batch_size, num_frames, latent_dim, H, W = cmf.size() segments = [] i = 0 while i < num_frames: segment_length = self.stride * (self.expand_ratio ** i) segment = cmf[:, i:i+segment_length, :, :, :] # 选择当前段 if segment.size(1) == 0: break # 抽取每段的代表帧,这里使用均值 segment_mean = segment.mean(dim=1) # [batch_size, latent_dim, H, W] segments.append(segment_mean) i += segment_length # 将所有段的均值重新组合 c_omf = torch.stack(segments, dim=1) # [batch_size, num_segments, latent_dim, H, W] return c_omf

5. 交互/内部时间层模块

python
class TemporalAttentionLayer(nn.Module): def __init__(self, latent_dim=256, num_heads=8): super(TemporalAttentionLayer, self).__init__() self.self_attn = nn.MultiheadAttention(embed_dim=latent_dim, num_heads=num_heads) self.layer_norm = nn.LayerNorm(latent_dim) def forward(self, x): # x: [sequence_length, batch_size, latent_dim] attn_output, _ = self.self_attn(x, x, x) x = self.layer_norm(x + attn_output) return x class InterIntraClipTemporalLayers(nn.Module): def __init__(self, latent_dim=256, num_heads=8): super(InterIntraClipTemporalLayers, self).__init__() # 跨剪辑时间层 self.inter_clip_layer = TemporalAttentionLayer(latent_dim, num_heads) # 内部剪辑时间层 self.intra_clip_layer = TemporalAttentionLayer(latent_dim, num_heads) def forward(self, noise_latents, cmf_features, c_omf): # noise_latents: [sequence_length, batch_size, latent_dim] # cmf_features: [batch_size, num_segments, latent_dim, H, W] # c_omf: [batch_size, num_segments, latent_dim, H, W] # 处理跨剪辑时间关系 # 这里需要将cmf_features和noise_latents合并,并添加时间嵌入 # 简化处理,假设已展平 inter_input = torch.cat([noise_latents, c_omf.view(c_omf.size(0), -1, c_omf.size(2))], dim=0) inter_output = self.inter_clip_layer(inter_input) # 处理内部剪辑时间关系 intra_output = self.intra_clip_layer(noise_latents) return intra_output

6. 音频条件模块

python
class AudioToLatentsModule(nn.Module): def __init__(self, audio_feature_dim=768, latent_dim=256, num_embeddings=512): super(AudioToLatentsModule, self).__init__() self.fc = nn.Linear(audio_feature_dim, latent_dim) self.learnable_embeddings = nn.Parameter(torch.randn(num_embeddings, latent_dim)) self.attention = nn.MultiheadAttention(embed_dim=latent_dim, num_heads=8) self.fc_out = nn.Linear(latent_dim, latent_dim) def forward(self, audio_features): # audio_features: [batch_size, seq_length, audio_feature_dim] # 映射到查询特征 query = self.fc(audio_features) # [batch_size, seq_length, latent_dim] query = query.permute(1, 0, 2) # [seq_length, batch_size, latent_dim] # 使用可学习的嵌入作为键和值 key = self.learnable_embeddings.unsqueeze(1).repeat(1, audio_features.size(0), 1) # [num_embeddings, batch_size, latent_dim] value = self.learnable_embeddings.unsqueeze(1).repeat(1, audio_features.size(0), 1) # [num_embeddings, batch_size, latent_dim] attn_output, _ = self.attention(query, key, value) # 线性变换 motion_latents = self.fc_out(attn_output) # [seq_length, batch_size, latent_dim] return motion_latents

7. 音频特征提取模块

python
class AudioFeatureExtractor(nn.Module): def __init__(self): super(AudioFeatureExtractor, self).__init__() self.wav2vec = Wav2Vec2Model.from_pretrained("facebook/wav2vec2-base") def forward(self, audio_waveforms): # audio_waveforms: [batch_size, num_samples] features = self.wav2vec(audio_waveforms).last_hidden_state # [batch_size, seq_length, feature_dim] return features

8. 双重U-Net模块

python
class DenoisingUNet(nn.Module): def __init__(self, latent_dim=256, num_heads=8): super(DenoisingUNet, self).__init__() self.latent_dim = latent_dim # 主U-Net部分 self.main_unet = nn.Sequential( nn.Conv2d(latent_dim, 128, kernel_size=3, padding=1), nn.ReLU(), nn.Conv2d(128, 256, kernel_size=3, padding=1), nn.ReLU(), nn.Conv2d(256, latent_dim, kernel_size=3, padding=1), nn.ReLU(), ) # 参考网络U-Net部分 self.reference_net = ReferenceNet(latent_dim) # 空间注意力层 self.spatial_attn = nn.MultiheadAttention(embed_dim=latent_dim, num_heads=num_heads) self.layer_norm = nn.LayerNorm(latent_dim) def forward(self, zt, cref, cmf, c_omf, t, audio_latents): # zt: [batch_size, latent_dim, H, W] # cref, cmf: [batch_size, latent_dim, H, W] # c_omf: [batch_size, num_segments, latent_dim, H, W] # t: [batch_size] # audio_latents: [seq_length, batch_size, latent_dim] # 主U-Net去噪 main_features = self.main_unet(zt) # 参考网络特征提取 cref_features, cmf_features = self.reference_net(cref, cmf) # 将参考特征与主U-Net特征在token维度上拼接 # 假设展平空间维度为序列长度 main_features_flat = main_features.view(main_features.size(0), self.latent_dim, -1).permute(2, 0, 1) # [seq_length, batch_size, latent_dim] cref_features_flat = cref_features.view(cref_features.size(0), self.latent_dim, -1).permute(2, 0, 1) # [seq_length, batch_size, latent_dim] combined = torch.cat([main_features_flat, cref_features_flat], dim=0) # 拼接 # 空间注意力 attn_output, _ = self.spatial_attn(combined, combined, combined) attn_output = self.layer_norm(combined + attn_output) # 添加时间嵌入和音频潜在向量 # 简化处理,假设时间嵌入已添加 denoised = attn_output + audio_latents return denoised

9. Loopy模型总架构

python
class LoopyModel(nn.Module): def __init__(self, latent_dim=256, num_heads=8, stride=4, expand_ratio=2, audio_feature_dim=768, num_embeddings=512): super(LoopyModel, self).__init__() self.encoder = VQVAEEncoder(latent_dim) self.audio_extractor = AudioFeatureExtractor() self.audio_to_latents = AudioToLatentsModule(audio_feature_dim, latent_dim, num_embeddings) self.temporal_segment = TemporalSegmentModule(stride, expand_ratio) self.temporal_layers = InterIntraClipTemporalLayers(latent_dim, num_heads) self.denoising_unet = DenoisingUNet(latent_dim, num_heads) def forward(self, I, cref, motion_frames, audio_waveforms, t, facial_features=None): """ 参数: I: 输入图像 [batch_size, 3, 512, 512] cref: 参考图像潜在向量 [batch_size, latent_dim, H, W] motion_frames: 运动帧潜在向量 [batch_size, num_frames, latent_dim, H, W] audio_waveforms: 音频波形 [batch_size, num_samples] t: 时间步长 [batch_size] facial_features: 面部运动特征,可选 """ # 编码输入图像 z0 = self.encoder(I) # [batch_size, latent_dim, H, W] # 提取音频特征 audio_features = self.audio_extractor(audio_waveforms) # [batch_size, seq_length, feature_dim] # 映射音频特征到运动潜在空间 audio_latents = self.audio_to_latents(audio_features) # [seq_length, batch_size, latent_dim] # 处理运动帧 c_omf = self.temporal_segment(motion_frames) # [batch_size, num_segments, latent_dim, H, W] # 交互/内部时间层处理 intra_output = self.temporal_layers(z0, cref, c_omf) # [sequence_length, batch_size, latent_dim] # 去噪 denoised = self.denoising_unet(z0, cref, motion_frames, c_omf, t, audio_latents) return denoised

10. 示例训练步骤

python
def train_step(model, optimizer, I, cref, motion_frames, audio_waveforms, t, facial_features=None): """ 单步训练过程 """ model.train() optimizer.zero_grad() # 前向传播 predicted_noise = model(I, cref, motion_frames, audio_waveforms, t, facial_features) # 计算损失(假设目标噪声为真实噪声) loss = F.mse_loss(predicted_noise, true_noise) # 反向传播 loss.backward() optimizer.step() return loss.item()

三、模块与论文内容对应关系

  1. 编码器模块(Encoder Module)

    • 对应论文3.1节中Stable Diffusion基于VQ-VAE的编码器E
  2. 参考网络模块(Reference Net Module)

    • 对应论文3.1节中Dual U-Net架构中的参考网络部分。
  3. 时间段模块(Temporal Segment Module)

    • 对应论文3.2节中Temporal Segment Module的设计,负责分段和抽象处理运动帧。
  4. 交互/内部时间层模块(Inter/Intra-Clip Temporal Layers Module)

    • 对应论文3.2节中Inter-clip和Intra-clip Temporal Layers的设计,分别处理跨剪辑和内部剪辑的时间关系。
  5. 音频条件模块(Audio Condition Module)

    • 对应论文3.3节中Audio Condition Module的设计,包括Wav2Vec音频特征提取和Audio-to-Latents模块。
  6. 双重U-Net模块(Dual U-Net Module)

    • 对应论文3.1节中Dual U-Net架构的整体设计,包含主U-Net和参考网络U-Net。
  7. 训练策略模块(Training Strategies Module)

    • 对应论文3.4节中关于条件掩码、dropout策略和多阶段训练的描述。

四、总结

通过上述分析和代码示例,我们详细设计了Loopy方法的主要模块,并明确了每个模块的参数和其在论文中的对应内容。代码部分提供了一个简化的实现框架,实际应用中可能需要根据具体的预训练模型和数据进行调整和优化。

若需进一步扩展或具体化某些模块(如Dual U-Net的详细实现、具体的注意力机制等),建议参考相关的Stable Diffusion和VQ-VAE实现代码,并根据论文中的描述进行定制化修改。

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