Note-level automatic singing transcription, involving the extraction of onset, offset, and pitch information from a singing voice, is a crucial process in the field of Music Information Retrieval (MIR), The recent advancements in deep learning models have led to significant progress in this field. However, annotating a training dataset requires professional music expertise, and the entire annotation process is time-consuming and labor-intensive. Therefore, this field suffers from a severe data scarcity problem. To address this issue, we developed a singing transcription model based on wav2vec 2.0, a pretrained speech representation model. The model can learn from unlabeled speech and weakly-labeled singing data and use this knowledge to benefit the transcription task. The experiments showed that our proposed method achieves a significant improvement over previous approaches on various benchmarks. Moreover, additional experiments demonstrate that our method achieves competitive performance even with a small proportion of training data.

Creating a piece of music is difficult for people who have never been trained to compose. We present an automatic music generation system to lower the threshold of creating music. The system takes the user's humming as input and creates full music based on the humming melody. The system consists of five modules: 1) humming transcription, 2) melody generation, 3) broken chord generation, 4) accompaniment generation, and 5) audio synthesis. The first module transcribes the user's humming audio to a score, and then the melody generation module composes a complete melody based on the user's humming melody. After that, the third module will generate a broken chord track to accompany the full melody, and the fourth module will create more accompanying tracks. Finally, the audio synthesis module mixes all the tracks to generate the music. Through the user experiment, our system can generate high-quality music with natural expression based on the user's humming input.

This paper presents a novel system, named LingGe ("伶歌" in Chinese), to generate songs for ancient Chinese poems automatically. LingGe takes the poem as the lyric, composes music conditioned on the lyric, and finally outputs a full song including the singing and the accompaniment. It consists of four modules: rhythm recognition, melody generation, accompaniment generation, and audio synthesis. Firstly, the rhythm recognition module analyzes the song structure and rhythm according to the poem. Secondly, the melody generation module assembles the rhythm into the template and then generates the melody. Thirdly, the accompaniment generation module predicts the accompaniment in harmony with the melody. Finally, the audio synthesis module generates singing and accompaniment audio and then mixes them to obtain songs. The results show that LingGe can generate high-quality and expressive songs for ancient Chinese poems, both in harmony and rhythm.

It is important for task-oriented dialogue systems to discover the dialogue structure (i.e. the general dialogue flow) from dialogue corpora automatically. Previous work models dialogue structure by extracting latent states for each utterance first and then calculating the transition probabilities among states. These two-stage methods ignore the contextual information when calculating the probabilities, which makes the transitions between the states ambiguous. This paper proposes a conversational graph (CG) to represent deterministic dialogue structure where nodes and edges represent the utterance and context information respectively. An unsupervised Edge-Enhanced Graph Auto-Encoder (EGAE) architecture is designed to model local-contextual and global-structural information for conversational graph learning. Furthermore, a self-supervised objective is introduced with the response selection task to guide the unsupervised learning of the dialogue structure. Experimental results on several public datasets demonstrate that the novel model outperforms several alternatives in aggregating utterances with similar semantics. The effectiveness of the learned dialogue structured is also verified by more than 5\% joint accuracy improvement in the downstream task of low resource dialogue state tracking.

Non-Autoregressive Neural Machine Translation (NAT) has achieved significant inference speedup by generating all tokens simultaneously. Despite its high efficiency, NAT usually suffers from two kinds of translation errors: over-translation (e.g. repeated tokens) and under-translation (e.g. missing translations), which eventually limits the translation quality. In this paper, we argue that these issues of NAT can be addressed through coverage modeling, which has been proved to be useful in autoregressive decoding. We propose a novel Coverage-NAT to model the coverage information directly by a token-level coverage iterative refinement mechanism and a sentence-level coverage agreement, which can remind the model if a source token has been translated or not and improve the semantics consistency between the translation and the source, respectively. Experimental results on WMT14 En↔De and WMT16 En↔Ro translation tasks show that our method can alleviate those errors and achieve strong improvements over the baseline system.

Recent studies in dialogue state tracking (DST) leverage historical information to determine states which are generally represented as slot-value pairs. However, most of them have limitations to efficiently exploit relevant context due to the lack of a powerful mechanism for modeling interactions between the slot and the dialogue history. Besides, existing methods usually ignore the slot imbalance problem and treat all slots indiscriminately, which limits the learning of hard slots and eventually hurts overall performance. In this paper, we propose to enhance the DST through employing a contextual hierarchical attention network to not only discern relevant information at both word level and turn level but also learn contextual representations. We further propose an adaptive objective to alleviate the slot imbalance problem by dynamically adjust weights of different slots during training. Experimental results show that our approach reaches 52.68% and 58.55% joint accuracy on MultiWOZ 2.0 and MultiWOZ 2.1 datasets respectively and achieves new state-of-the-art performance with considerable improvements (+1.24% and +5.98%).

The non‐equilibrium of network traffic data brings about the non‐equilibrium of classification. Feature extraction is an effective method to reduce data dimensions, while it can intensify the influence of non‐equilibrium further. A secondary feature extraction algorithm of multidimensional assessment is proposed in this study. The features of network traffic are evaluated in different dimensions to provide the basis for feature extraction. Furthermore, a model dealing with imbalanced data is proposed based on secondary feature extraction and sampling. The model combines the benefits of dimension reduction and redistribution. The experiment results show that the proposed model can not only increase classification accuracy and decrease non‐equilibrium, but also enhance the performance of different classification algorithms.

For object tracking with occlusion in video application, in this paper, an information fusion tracking model based on level set method is proposed. There are two components. One is an object tracking model with information fusion, which utilize multiple information to achieve moving object segmenting task. The other component is occlusion handling with information fusion. During occlusion occurs this component can located moving objects accurately because of employing objects moving features.