Denoising Diffusion Probabilistic Models (DDPMs) are emerging in text-to-speech (TTS) synthesis because of their strong capability of generating high-fidelity samples. However, their iterative refinement process in high-dimensional data space results in slow inference speed, which restricts their application in real-time systems. Previous works have explored speeding up by minimizing the number of inference steps but at the cost of sample quality. In this work, to improve the inference speed for DDPM-based TTS model while achieving high sample quality, we propose ResGrad, a lightweight diffusion model which learns to refine the output spectrogram of an existing TTS model (e.g., FastSpeech 2) by predicting the residual between the model output and the corresponding ground-truth speech. ResGrad has several advantages: 1) Compare with other acceleration methods for DDPM which need to synthesize speech from scratch, ResGrad reduces the complexity of task by changing the generation target from ground-truth mel-spectrogram to the residual, resulting into a more lightweight model and thus a smaller real-time factor. 2) ResGrad is employed in the inference process of the existing TTS model in a plug-and-play way, without re-training this model. We verify ResGrad on the single-speaker dataset LJSpeech and two more challenging datasets with multiple speakers (LibriTTS) and high sampling rate (VCTK). Experimental results show that in comparison with other speed-up methods of DDPMs: 1) ResGrad achieves better sample quality with the same inference speed measured by real-time factor; 2) with similar speech quality, ResGrad synthesizes speech faster than baseline methods by more than 10 times. Audio samples are available at https://resgrad1.github.io/.

Video dubbing aims to translate the original speech in a film or television program into the speech in a target language, which can be achieved with a cascaded system consisting of speech recognition, machine translation and speech synthesis. To ensure the translated speech to be well aligned with the corresponding video, the length/duration of the translated speech should be as close as possible to that of the original speech, which requires strict length control. Previous works usually control the number of words or characters generated by the machine translation model to be similar to the source sentence, without considering the isochronicity of speech as the speech duration of words/characters in different languages varies. In this paper, we propose a machine translation system tailored for the task of video dubbing, which directly considers the speech duration of each token in translation, to match the length of source and target speech. Specifically, we control the speech length of generated sentence by guiding the prediction of each word with the duration information, including the speech duration of itself as well as how much duration is left for the remaining words. We design experiments on four language directions (German -> English, Spanish -> English, Chinese <-> English), and the results show that the proposed method achieves better length control ability on the generated speech than baseline methods. To make up the lack of real-world datasets, we also construct a real-world test set collected from films to provide comprehensive evaluations on the video dubbing task.

Localizing anatomical landmarks are important tasks in medical image analysis. However, the landmarks to be localized often lack prominent visual features. Their locations are elusive and easily confused with the background, and thus precise localization highly depends on the context formed by their surrounding areas. In addition, the required precision is usually higher than segmentation and object detection tasks. Therefore, localization has its unique challenges different from segmentation or detection. In this paper, we propose a zoom-in attentive network (ZIAN) for anatomical landmark localization in ocular images. First, a coarse-to-fine, or "zoom-in" strategy is utilized to learn the contextualized features in different scales. Then, an attentive fusion module is adopted to aggregate multi-scale features, which consists of 1) a co-attention network with a multiple regions-of-interest (ROIs) scheme that learns complementary features from the multiple ROIs, 2) an attention-based fusion module which integrates the multi-ROIs features and non-ROI features. We evaluated ZIAN on two open challenge tasks, i.e., the fovea localization in fundus images and scleral spur localization in AS-OCT images. Experiments show that ZIAN achieves promising performances and outperforms state-of-the-art localization methods. The source code and trained models of ZIAN are available at https://github.com/leixiaofeng-astar/OMIA9-ZIAN.

本文回顾了AIM 2022上压缩图像和视频超级分辨率的挑战。这项挑战包括两条曲目。轨道1的目标是压缩图像的超分辨率，轨迹〜2靶向压缩视频的超分辨率。在轨道1中，我们使用流行的数据集DIV2K作为培训，验证和测试集。在轨道2中，我们提出了LDV 3.0数据集，其中包含365个视频，包括LDV 2.0数据集（335个视频）和30个其他视频。在这一挑战中，有12支球队和2支球队分别提交了赛道1和赛道2的最终结果。所提出的方法和解决方案衡量了压缩图像和视频上超分辨率的最先进。提出的LDV 3.0数据集可在https://github.com/renyang-home/ldv_dataset上找到。此挑战的首页是在https://github.com/renyang-home/aim22_compresssr。

尽管近年来人的重新识别取得了令人印象深刻的改善，但在实际应用程序场景中，由不同的障碍引起的常见闭塞案例仍然是一个不稳定的问题。现有方法主要通过采用额外网络提供的身体线索来区分可见部分，以解决此问题。然而，助理模型和REID数据集之间的不可避免的域间隙极大地增加了获得有效和有效模型的困难。为了摆脱额外的预训练网络并在端到端可训练网络中实现自动对齐，我们根据两个不言而喻的先验知识提出了一种新型的动态原型掩码（DPM）。具体而言，我们首先设计了一个层次蒙版生成器，该层面生成器利用层次的语义选择高质量的整体原型和闭塞输入图像的特征表示之间的可见图案空间。在这种情况下，可以自发地在选定的子空间中很好地对齐。然后，为了丰富高质量整体原型的特征表示并提供更完整的特征空间，我们引入了一个头部丰富模块，以鼓励不同的头部在整个图像中汇总不同的模式表示。对被遮挡和整体人员重新识别基准进行的广泛的实验评估证明了DPM优于最先进的方法。该代码在https://github.com/stone96123/dpm上发布。

Twitter机器人检测已成为打击错误信息，促进社交媒体节制并保持在线话语的完整性的越来越重要的任务。最先进的机器人检测方法通常利用Twitter网络的图形结构，在面对传统方法无法检测到的新型Twitter机器人时，它们表现出令人鼓舞的性能。但是，现有的Twitter机器人检测数据集很少是基于图形的，即使这些基于图形的数据集也遭受有限的数据集量表，不完整的图形结构以及低注释质量。实际上，缺乏解决这些问题的大规模基于图的Twitter机器人检测基准，严重阻碍了基于图形的机器人检测方法的开发和评估。在本文中，我们提出了Twibot-22，这是一个综合基于图的Twitter机器人检测基准，它显示了迄今为止最大的数据集，在Twitter网络上提供了多元化的实体和关系，并且与现有数据集相比具有更好的注释质量。此外，我们重新实施35代表性的Twitter机器人检测基线，并在包括Twibot-22在内的9个数据集上进行评估，以促进对模型性能和对研究进度的整体了解的公平比较。为了促进进一步的研究，我们将所有实施的代码和数据集巩固到Twibot-22评估框架中，研究人员可以在其中始终如一地评估新的模型和数据集。 Twibot-22 Twitter机器人检测基准和评估框架可在https://twibot22.github.io/上公开获得。

Binaural audio plays a significant role in constructing immersive augmented and virtual realities. As it is expensive to record binaural audio from the real world, synthesizing them from mono audio has attracted increasing attention. This synthesis process involves not only the basic physical warping of the mono audio, but also room reverberations and head/ear related filtrations, which, however, are difficult to accurately simulate in traditional digital signal processing. In this paper, we formulate the synthesis process from a different perspective by decomposing the binaural audio into a common part that shared by the left and right channels as well as a specific part that differs in each channel. Accordingly, we propose BinauralGrad, a novel two-stage framework equipped with diffusion models to synthesize them respectively. Specifically, in the first stage, the common information of the binaural audio is generated with a single-channel diffusion model conditioned on the mono audio, based on which the binaural audio is generated by a two-channel diffusion model in the second stage. Combining this novel perspective of two-stage synthesis with advanced generative models (i.e., the diffusion models),the proposed BinauralGrad is able to generate accurate and high-fidelity binaural audio samples. Experiment results show that on a benchmark dataset, BinauralGrad outperforms the existing baselines by a large margin in terms of both object and subject evaluation metrics (Wave L2: 0.128 vs. 0.157, MOS: 3.80 vs. 3.61). The generated audio samples (https://speechresearch.github.io/binauralgrad) and code (https://github.com/microsoft/NeuralSpeech/tree/master/BinauralGrad) are available online.

迅速的学习方法通​​过诱导更好的几次表现，在他们仍然遵循基于参数的学习范式的同时，引起了自然语言处理的波动。学习中的遗忘和死记硬背的记忆问题可能会遇到不稳定的概括问题。具体而言，香草及时的学习可能难以利用死记硬背的非典型实例，在完全监督的培训或过度贴身模式的情况下使用低射击数据。为了减轻此类局限性，我们以将知识从记忆中解耦的动机发展为有助于模型在概括和记忆之间取得平衡。与香草及时学习相反，重新启动构造了培训实例中的开放式知识店，并在输入，培训和推理过程中实现检索机制，从而使该模型能够从培训语料库中检索相关环境作为能力为提示增强。广泛的实验表明，Retroppt可以在几次射击和零拍设置中获得更好的性能。此外，我们进一步说明，我们提出的撤退可以通过新数据集获得更好的概括能力。对记忆的详细分析确实显示逆转可以减少语言模型对记忆的依赖；因此，改善下游任务的概括。

在本文中，我们介绍了VCSL（视频复制段本地化），这是一种新的综合段级注释的视频复制数据集。与受视频级注释或小规模限制的现有复制检测数据集相比，VCSL不仅具有两个段级标签的数据级，其中有160k现实的视频副本对，其中包含超过280k的本地化copied seggment对，而且还包含超过280k涵盖各种视频类别和各种视频持续时间。每个收集的视频对中的所有复制段均经过手动提取，并伴随着精确注释的启动和结束时间戳。除了数据集外，我们还提出了一种新颖的评估协议，该协议可以更好地衡量视频对之间复制重叠段的预测准确性，并在不同情况下显示出改善的适应性。通过使用拟议的数据集和评估指标对几个基线和最先进的细分级视频副本检测方法进行基准测试，我们提供了一项全面的分析，可以揭示当前方法的优势和劣势作品。 VCSL数据集，公制和基准代码均在https://github.com/alipay/vcsl上公开获得。

我们展示了一个新的开源和可扩展知识提取工具包，称为Deepke（基于深度学习的知识提取），支持标准完全监督，低资源少拍摄和文档级方案。 Deepke实现了各种信息提取任务，包括命名实体识别，关系提取和属性提取。使用统一的框架，DeePke允许开发人员和研究人员根据其要求，自定义数据集和模型以从非结构化文本中提取信息。具体而言，DeePke不仅为不同的任务和场景提供了各种功能模块和模型实现，而且还通过一致的框架组织所有组件以维持足够的模块化和可扩展性。此外，我们在\ URL {http://deepke.zjukg.cn/}中介绍一个在线平台，用于实时提取各种任务。 Deepke已经配备了Google Colab教程和初学者的综合文件。我们用演示视频发布\ url {https://github.com/zjunlp/deepke}源代码。