With the development of depth sensors in recent years, RGBD object tracking has received significant attention. Compared with the traditional RGB object tracking, the addition of the depth modality can effectively solve the target and background interference. However, some existing RGBD trackers use the two modalities separately and thus some particularly useful shared information between them is ignored. On the other hand, some methods attempt to fuse the two modalities by treating them equally, resulting in the missing of modality-specific features. To tackle these limitations, we propose a novel Dual-fused Modality-aware Tracker (termed DMTracker) which aims to learn informative and discriminative representations of the target objects for robust RGBD tracking. The first fusion module focuses on extracting the shared information between modalities based on cross-modal attention. The second aims at integrating the RGB-specific and depth-specific information to enhance the fused features. By fusing both the modality-shared and modality-specific information in a modality-aware scheme, our DMTracker can learn discriminative representations in complex tracking scenes. Experiments show that our proposed tracker achieves very promising results on challenging RGBD benchmarks. Code is available at \url{https://github.com/ShangGaoG/DMTracker}.

由于与传统的基于RGB的跟踪相比，多模式跟踪的能力在复杂的情况下更准确和健壮，因此获得了关注。它的关键在于如何融合多模式数据并减少模式之间的差距。但是，多模式跟踪仍然严重遭受数据缺乏症的影响，从而导致融合模块的学习不足。我们没有在本文中构建这样的融合模块，而是通过将重要性附加到多模式的视觉提示中，为多模式跟踪提供了新的视角。我们设计了一种新型的多模式及时跟踪器（Protrack），可以通过及时范式将多模式输入传递到单个模态。通过最好地利用预先训练的RGB跟踪器在大规模学习的跟踪能力，我们的突起即使没有对多模式数据进行任何额外的培训，我们的突起也可以通过更改输入来实现高性能多模式跟踪。 5个基准数据集的广泛实验证明了所提出的突起的有效性。

视频综合孔径雷达（视频 - 萨尔）图像之间的移动目标阴影总是被低散射背景和混乱的噪音干扰，从而导致移动目标阴影检测跟踪性能不良。为了解决这个问题，这封信提出了一个名为SBN-3D-SD的暗影 - 背景3D空间隔离方法，以提高阴影显着性，以提高视频 - 萨尔移动目标影像阴影检测跟踪性能。

最近，先驱工作发现，演讲预训练模型可以解决全堆栈语音处理任务，因为该模型利用底层学习扬声器相关信息和顶层以编码与内容相关的信息。由于网络容量有限，我们认为如果模型专用于音频内容信息学习，则可以进一步提高语音识别性能。为此，我们向自我监督学习（ILS-SSL）提出中间层监督，这将模型通过在中间层上添加额外的SSL丢失来尽可能地专注于内容信息。 LibrisPeech测试 - 其他集合的实验表明，我们的方法显着优于Hubert，这实现了基数/大型模型的W / O语言模型设置的相对字错误率降低了23.5％/ 11.6％。详细分析显示我们模型的底层与拼音单元具有更好的相关性，这与我们的直觉一致，并解释了我们对ASR的方法的成功。

无监督域适应（UDA）旨在将从标记的源域中学习的知识转移到未标记的目标域。以前的工作主要是在卷积神经网络（CNNS）上建立的，以学习域名不变的表示。随着近期应用视觉变压器（VIT）对视力任务的指数增加，然而，在文献中仍未开发了调整跨领域知识的能力。为了填补这一差距，本文首先全面调查了vit的各种域适应任务的可转移性。令人惊讶的是，VIT通过其具有大边缘的基于CNNS的对应物来证明优异的可转移性，而通过掺入抗体适应可以进一步提高性能。尽管如此，直接使用基于CNNS的适应策略未能利用Vit的内在优点（例如，注意机制和顺序图像表示）在知识转移中起重要作用。为了解决这个问题，我们提出了一个统一的框架，即可转换的视觉变压器（TVT），以充分利用VIT的可转换性来实现域适应。具体而言，我们精致地设计了一种新颖且有效的单位，我们术语可转移适应模块（TAM）。通过将学习的传递注入注意块，TAM压迫重点是可转移和辨别特征。此外，我们利用判别聚类来增强在对抗域对齐期间破坏的特征分集和分离。为了验证其多功能性，我们在四个基准测试中对TVT进行了广泛的研究，实验结果表明，与现有的最先进的UDA方法相比，TVT达到了显着的改进。

A polarization camera has great potential for 3D reconstruction since the angle of polarization (AoP) and the degree of polarization (DoP) of reflected light are related to an object's surface normal. In this paper, we propose a novel 3D reconstruction method called Polarimetric Multi-View Inverse Rendering (Polarimetric MVIR) that effectively exploits geometric, photometric, and polarimetric cues extracted from input multi-view color-polarization images. We first estimate camera poses and an initial 3D model by geometric reconstruction with a standard structure-from-motion and multi-view stereo pipeline. We then refine the initial model by optimizing photometric rendering errors and polarimetric errors using multi-view RGB, AoP, and DoP images, where we propose a novel polarimetric cost function that enables an effective constraint on the estimated surface normal of each vertex, while considering four possible ambiguous azimuth angles revealed from the AoP measurement. The weight for the polarimetric cost is effectively determined based on the DoP measurement, which is regarded as the reliability of polarimetric information. Experimental results using both synthetic and real data demonstrate that our Polarimetric MVIR can reconstruct a detailed 3D shape without assuming a specific surface material and lighting condition.

Neural transducer is now the most popular end-to-end model for speech recognition, due to its naturally streaming ability. However, it is challenging to adapt it with text-only data. Factorized neural transducer (FNT) model was proposed to mitigate this problem. The improved adaptation ability of FNT on text-only adaptation data came at the cost of lowered accuracy compared to the standard neural transducer model. We propose several methods to improve the performance of the FNT model. They are: adding CTC criterion during training, adding KL divergence loss during adaptation, using a pre-trained language model to seed the vocabulary predictor, and an efficient adaptation approach by interpolating the vocabulary predictor with the n-gram language model. A combination of these approaches results in a relative word-error-rate reduction of 9.48\% from the standard FNT model. Furthermore, n-gram interpolation with the vocabulary predictor improves the adaptation speed hugely with satisfactory adaptation performance.

Self-supervised learning (SSL) methods such as WavLM have shown promising speech separation (SS) results in small-scale simulation-based experiments. In this work, we extend the exploration of the SSL-based SS by massively scaling up both the pre-training data (more than 300K hours) and fine-tuning data (10K hours). We also investigate various techniques to efficiently integrate the pre-trained model with the SS network under a limited computation budget, including a low frame rate SSL model training setup and a fine-tuning scheme using only the part of the pre-trained model. Compared with a supervised baseline and the WavLM-based SS model using feature embeddings obtained with the previously released 94K hours trained WavLM, our proposed model obtains 15.9% and 11.2% of relative word error rate (WER) reductions, respectively, for a simulated far-field speech mixture test set. For conversation transcription on real meeting recordings using continuous speech separation, the proposed model achieves 6.8% and 10.6% of relative WER reductions over the purely supervised baseline on AMI and ICSI evaluation sets, respectively, while reducing the computational cost by 38%.

Automatic Speech Recognition (ASR) systems typically yield output in lexical form. However, humans prefer a written form output. To bridge this gap, ASR systems usually employ Inverse Text Normalization (ITN). In previous works, Weighted Finite State Transducers (WFST) have been employed to do ITN. WFSTs are nicely suited to this task but their size and run-time costs can make deployment on embedded applications challenging. In this paper, we describe the development of an on-device ITN system that is streaming, lightweight & accurate. At the core of our system is a streaming transformer tagger, that tags lexical tokens from ASR. The tag informs which ITN category might be applied, if at all. Following that, we apply an ITN-category-specific WFST, only on the tagged text, to reliably perform the ITN conversion. We show that the proposed ITN solution performs equivalent to strong baselines, while being significantly smaller in size and retaining customization capabilities.

End-to-end formulation of automatic speech recognition (ASR) and speech translation (ST) makes it easy to use a single model for both multilingual ASR and many-to-many ST. In this paper, we propose streaming language-agnostic multilingual speech recognition and translation using neural transducers (LAMASSU). To enable multilingual text generation in LAMASSU, we conduct a systematic comparison between specified and unified prediction and joint networks. We leverage a language-agnostic multilingual encoder that substantially outperforms shared encoders. To enhance LAMASSU, we propose to feed target LID to encoders. We also apply connectionist temporal classification regularization to transducer training. Experimental results show that LAMASSU not only drastically reduces the model size but also outperforms monolingual ASR and bilingual ST models.