Prior works on improving speech quality with visual input typically study each type of auditory distortion separately (e.g., separation, inpainting, video-to-speech) and present tailored algorithms. This paper proposes to unify these subjects and study Generalized Speech Enhancement, where the goal is not to reconstruct the exact reference clean signal, but to focus on improving certain aspects of speech. In particular, this paper concerns intelligibility, quality, and video synchronization. We cast the problem as audio-visual speech resynthesis, which is composed of two steps: pseudo audio-visual speech recognition (P-AVSR) and pseudo text-to-speech synthesis (P-TTS). P-AVSR and P-TTS are connected by discrete units derived from a self-supervised speech model. Moreover, we utilize self-supervised audio-visual speech model to initialize P-AVSR. The proposed model is coined ReVISE. ReVISE is the first high-quality model for in-the-wild video-to-speech synthesis and achieves superior performance on all LRS3 audio-visual enhancement tasks with a single model. To demonstrates its applicability in the real world, ReVISE is also evaluated on EasyCom, an audio-visual benchmark collected under challenging acoustic conditions with only 1.6 hours of training data. Similarly, ReVISE greatly suppresses noise and improves quality. Project page: https://wnhsu.github.io/ReVISE.

In this paper, we revisit the class of iterative shrinkage-thresholding algorithms (ISTA) for solving the linear inverse problem with sparse representation, which arises in signal and image processing. It is shown in the numerical experiment to deblur an image that the convergence behavior in the logarithmic-scale ordinate tends to be linear instead of logarithmic, approximating to be flat. Making meticulous observations, we find that the previous assumption for the smooth part to be convex weakens the least-square model. Specifically, assuming the smooth part to be strongly convex is more reasonable for the least-square model, even though the image matrix is probably ill-conditioned. Furthermore, we improve the pivotal inequality tighter for composite optimization with the smooth part to be strongly convex instead of general convex, which is first found in [Li et al., 2022]. Based on this pivotal inequality, we generalize the linear convergence to composite optimization in both the objective value and the squared proximal subgradient norm. Meanwhile, we set a simple ill-conditioned matrix which is easy to compute the singular values instead of the original blur matrix. The new numerical experiment shows the proximal generalization of Nesterov's accelerated gradient descent (NAG) for the strongly convex function has a faster linear convergence rate than ISTA. Based on the tighter pivotal inequality, we also generalize the faster linear convergence rate to composite optimization, in both the objective value and the squared proximal subgradient norm, by taking advantage of the well-constructed Lyapunov function with a slight modification and the phase-space representation based on the high-resolution differential equation framework from the implicit-velocity scheme.

The recent trend in multiple object tracking (MOT) is jointly solving detection and tracking, where object detection and appearance feature (or motion) are learned simultaneously. Despite competitive performance, in crowded scenes, joint detection and tracking usually fail to find accurate object associations due to missed or false detections. In this paper, we jointly model counting, detection and re-identification in an end-to-end framework, named CountingMOT, tailored for crowded scenes. By imposing mutual object-count constraints between detection and counting, the CountingMOT tries to find a balance between object detection and crowd density map estimation, which can help it to recover missed detections or reject false detections. Our approach is an attempt to bridge the gap of object detection, counting, and re-Identification. This is in contrast to prior MOT methods that either ignore the crowd density and thus are prone to failure in crowded scenes, or depend on local correlations to build a graphical relationship for matching targets. The proposed MOT tracker can perform online and real-time tracking, and achieves the state-of-the-art results on public benchmarks MOT16 (MOTA of 77.6), MOT17 (MOTA of 78.0%) and MOT20 (MOTA of 70.2%).

Many self-supervised speech models, varying in their pre-training objective, input modality, and pre-training data, have been proposed in the last few years. Despite impressive empirical successes on downstream tasks, we still have a limited understanding of the properties encoded by the models and the differences across models. In this work, we examine the intermediate representations for a variety of recent models. Specifically, we measure acoustic, phonetic, and word-level properties encoded in individual layers, using a lightweight analysis tool based on canonical correlation analysis (CCA). We find that these properties evolve across layers differently depending on the model, and the variations relate to the choice of pre-training objective. We further investigate the utility of our analyses for downstream tasks by comparing the property trends with performance on speech recognition and spoken language understanding tasks. We discover that CCA trends provide reliable guidance to choose layers of interest for downstream tasks and that single-layer performance often matches or improves upon using all layers, suggesting implications for more efficient use of pre-trained models.

尽管视听模型与仅限音频模型相比可以产生卓越的性能和鲁棒性，但由于缺乏标记和未标记的视听数据以及每种方式部署一个模型的成本，它们的开发和采用受到阻碍。在本文中，我们提出了U-Hubert，这是一个自制的预训练框架，可以通过统一的蒙版群集预测目标来利用多模式和单峰语音。通过在预训练期间利用模态辍学，我们证明了一个微调模型可以在PAR上取得比较的性能或比最先进的模态特异性模型更好。此外，我们仅在音频上进行微调的模型可以通过视听和视觉语音输入来表现良好，从而实现了零击的模态概括，以实现语音识别和扬声器验证。特别是，我们的单个模型在带有音频/视听/视觉输入的LRS3上产生1.2％/1.4％/27.2％的语音识别单词错误率。

本文调查了视听扬声器表示的自我监督的预训练，其中显示了视觉流，显示说话者的口腔区域与语音一起用作输入。我们的研究重点是视听隐藏单元BERT（AV-HUBERT）方法，该方法是最近开发的通用音频语音训练前训练框架。我们进行了广泛的实验，以探测预训练和视觉方式的有效性。实验结果表明，AV-Hubert可以很好地概括与说话者相关的下游任务，从而使标签效率提高了大约10倍的仅10倍，仅音频和视听扬声器验证。我们还表明，结合视觉信息，甚至仅仅是唇部区域，都大大提高了性能和噪声稳健性，在清洁条件下将EER降低了38％，在嘈杂的条件下将EER降低了75％。

基于音频的自动语音识别（ASR）在嘈杂的环境中显着降低，并且特别容易受到干扰语音的影响，因为模型无法确定要转录的扬声器。视听语音识别（AVSR）系统通过将音频流与不变噪声不变的可视信息补充，帮助模型对所需扬声器的视觉信息来提高鲁棒性。但是，以前的AVSR工作仅关注监督学习设置;因此，通过可用的标记数据量阻碍了进度。在这项工作中，我们提出了一个自我监督的AVSR框架，建立在视听休伯特（AV-HUBERT），是最先进的视听语音表示学习模型。在最大可用的AVSR基准数据集LRS3中，我们的方法在存在的情况下使用少于10％的标签数据（433HR与30HR）之前的最先进（28.0％与14.1％）优于〜50％（28.0％vs.14.1％）禁止噪声，平均减少了基于音频模型的WER以上超过75％（25.8％与5.8％）。

语音的视频录制包含相关的音频和视觉信息，为语音表示从扬声器的唇部运动和产生的声音提供了强大的信号。我们介绍了视听隐藏单元BERT（AV-HUBERT），是视听语音的自我监督的代表学习框架，这些屏幕屏蔽了多流视频输入并预测自动发现和迭代地精制多模式隐藏单元。 AV-HUBERT学习强大的视听语音表示，这些语音表示受益于唇读和自动语音识别。在最大的公众唇读基准LRS3（433小时）中，AV-Hubert达到32.5％WER，只有30个小时的标签数据，优于前一种最先进的方法（33.6％）培训，达到了一千次转录的视频数据（31k小时）。当使用来自LRS3的所有433小时的标记数据并结合自培训时，唇读WER进一步降低至26.9％。使用我们在相同的基准测试中使用您的视听表示，用于音频语音识别的相对效率为40％，而最先进的性能（1.3％Vs 2.3％）。我们的代码和模型可在https://github.com/facebookResearch/av_hubert获得

最近的2D-3D人类姿势估计工作倾向于利用人体骨架的拓扑形成的图形结构。但是，我们认为这种骨架拓扑太稀疏，无法反映身体结构并遭受严重的2D-3D模糊问题。为了克服这些弱点，我们提出了一种新颖的图表卷积网络架构，层次图形网络（HGN）。它基于我们的多尺度图结构建筑策略产生的密度图形拓扑，从而提供更精细的几何信息。所提出的架构包含三个并行组织的稀疏微小表示子网，其中通过新颖的特征融合策略处理多尺度图形结构特征，并通过新颖的特征融合策略进行交换信息，导致丰富的分层表示。我们还介绍了3D粗网格约束，以进一步提高与细节相关的特征学习。广泛的实验表明，我们的HGN通过减少的网络参数实现了最先进的性能

在本文中，我们提出了一种新颖的观察者来解决视觉同时定位和映射（SLAM）的问题，仅使用来自单眼摄像机和惯性测量单元（IMU）的信息。系统状态在歧管$ se（3）\ times \ mathbb {r} ^ {3n} $上演变，我们在其中仔细设计动态扩展，以便产生不变的叶片，使得问题重新加入在线\ EMPH{常量参数}识别。然后，遵循最近引入的基于参数估计的观察者（PEBO）和动态回归扩展和混合（DREM）过程，我们提供了一个新的简单解决方案。值得注意的优点是，拟议的观察者保证了几乎全局渐近稳定性，既不需要激发的持久性也不是完全可观察性，然而，在大多数现有的工作中广泛采用了保证稳定性。