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.

Recent mainstream weakly-supervised semantic segmentation (WSSS) approaches mainly relies on image-level classification learning, which has limited representation capacity. In this paper, we propose a novel semantic learning based framework, named SLAMs (Semantic Learning based Activation Map), for WSSS.

由于癌症样品收集和注释的难度，宫颈癌数据集通常表现出长尾数据分布。当训练检测器以检测WSI（整个切片图像）中的癌细胞时，从TCT（ThinPrep细胞学测试）样品捕获的样品时，头部类别（例如正常细胞和炎性细胞）通常比尾巴类别数量更大。 （例如癌细胞）。对象检测中的大多数现有最新的长尾学习方法将重点放在类别分布统计上，以解决长尾方案中的问题，而无需考虑每个样本的“硬度”。为了解决这个问题，在这项工作中，我们提出了一个Grad-libra损失，该损失利用梯度动态校准每个样品的硬度程度，以使不同类别的硬度度重新平衡正面和负样品的梯度。因此，我们的损失可以帮助探测器更加重视头部和尾部类别中的这些硬样品。在长尾的TCT WSI图像数据集上进行了广泛的实验表明，主流检测器，例如对使用我们建议的梯度损失训练的训练，重新点，FCO，ATSS，YOLOF等的地图比使用跨透明分类损失训练的地图要高得多（7.8％）。

由于卷积在提取物体的局部上下文中，在过去十年中，对象检测在过去十年中取得了重大进展。但是，对象的尺度是多样的，当前卷积只能处理单尺度输入。因此，传统卷积具有固定接收场在处理这种规模差异问题方面的能力受到限制。多尺度功能表示已被证明是缓解规模差异问题的有效方法。最近的研究主要与某些量表或各个尺度的总体特征采用部分联系，并专注于整个量表的全球信息。但是，跨空间和深度维度的信息被忽略了。受此启发，我们提出了多尺度卷积（MSCONV）来解决此问题。同时考虑到量表，空间和深度信息，MSCONV能够更全面地处理多尺度输入。 MSCONV是有效的，并且在计算上是有效的，只有少量计算成本增加。对于大多数单阶段对象探测器，在检测头中用MSCONV代替传统的卷积可以带来AP的2.5 \％改进（在Coco 2017数据集上），只有3 \％的拖鞋增加了。 MSCONV对于两阶段对象探测器也具有灵活性和有效性。当扩展到主流两阶段对象检测器时，MSCONV的AP可以提高3.0 \％。我们在单尺度测试下的最佳模型在Coco 2017上实现了48.9 \％AP，\ textit {test-dev} Split，它超过了许多最新方法。

锥体网络是多尺度对象检测的标准方法。当前对特征金字塔网络的研究通常采用层连接来从特征层次结构的某些级别收集特征，并且不考虑它们之间的显着差异。我们提出了一个更好的特征金字塔网络的体系结构，称为选择性多尺度学习（SMSL），以解决此问题。SMSL高效且泛滥，可以将其集成到单阶段和两阶段检测器中以提高检测性能，几乎没有额外的推理成本。视网膜与SMSL的结合获得了可可数据集的AP（从39.1 \％到40.9 \％）的1.8 \％改进。与SMSL集成时，两阶段探测器的AP可以提高1.0 \％。

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.

While class activation map (CAM) generated by image classification network has been widely used for weakly supervised object localization (WSOL) and semantic segmentation (WSSS), such classifiers usually focus on discriminative object regions. In this paper, we propose Contrastive learning for Class-agnostic Activation Map (C$^2$AM) generation only using unlabeled image data, without the involvement of image-level supervision. The core idea comes from the observation that i) semantic information of foreground objects usually differs from their backgrounds; ii) foreground objects with similar appearance or background with similar color/texture have similar representations in the feature space. We form the positive and negative pairs based on the above relations and force the network to disentangle foreground and background with a class-agnostic activation map using a novel contrastive loss. As the network is guided to discriminate cross-image foreground-background, the class-agnostic activation maps learned by our approach generate more complete object regions. We successfully extracted from C$^2$AM class-agnostic object bounding boxes for object localization and background cues to refine CAM generated by classification network for semantic segmentation. Extensive experiments on CUB-200-2011, ImageNet-1K, and PASCAL VOC2012 datasets show that both WSOL and WSSS can benefit from the proposed C$^2$AM.

在本文中，我们建议将面向任务导向的对话系统作为纯粹的自然语言生成任务，以便充分利用像GPT-2这样的大规模预训练模型，并简化了复杂的光学化预备。然而，直接应用这种方法严重遭受了通过删除了替代令牌而导致的对话实体不一致，以及在微调期间灾害模型的灾难性遗忘问题，导致表现不令人满意。为了缓解这些问题，我们设计了一种新颖的GPT-Adapter-CopyNet网络，它将轻量级适配器和CopyNet模块包含到GPT-2中，以实现转移学习和对话实体生成的更好性能。在DSTC8轨道1基准和多种数据集上进行的实验结果表明，我们的建议方法显着优于基线模型，在自动和人类评估中具有显着性能。

Diffusion models have achieved state-of-the-art synthesis quality on visual and audio tasks, and recent works adapt them to textual data by diffusing on the embedding space. But the difference between the continuous data space and the embedding space raises challenges to the diffusion model, which have not been carefully explored. In this paper, we conduct systematic studies and analyze the challenges threefold. Firstly, the data distribution is learnable for embeddings, which may lead to the collapse of the loss function. Secondly, as the norm of embedding varies between popular and rare words, adding the same noise scale will lead to sub-optimal results. In addition, we find that noises sampled from a standard Gaussian distribution may distract the diffusion process. To solve the above challenges, we propose Difformer, a denoising diffusion probabilistic model based on Transformer, which consists of three techniques including utilizing an anchor loss function, a layer normalization module for embeddings, and a norm factor to the Gaussian noise. All techniques are complementary to each other and critical to boosting the model performance together. Experiments are conducted on benchmark datasets over two seminal text generation tasks including machine translation and text summarization. The results show that Difformer significantly outperforms the embedding diffusion baselines, while achieving competitive results with strong autoregressive baselines.

我们考虑了自动生成音乐文本描述的新颖任务。与其他完善的文本生成任务（例如图像标题）相比，富裕的音乐和文本数据集的稀缺性使其成为更具挑战性的任务。在本文中，我们利用众包音乐评论来构建一个新的数据集，并提出一个序列到序列模型以生成音乐的文本描述。更具体地说，我们将扩张的卷积层用作编码器的基本组成部分，基于内存的复发性神经网络作为解码器。为了增强生成文本的真实性和主题，我们进一步建议用歧视者和新的主题评估者微调模型。为了衡量生成的文本的质量，我们还提出了两个新的评估指标，它们比人类评估比传统指标（例如BLEU）更加一致。实验结果验证了我们的模型能够在包含原始音乐的主题和内容信息的同时产生流利而有意义的评论。