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ICASSP 2019 - 2019 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)最新文献

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Group Sparsity Based Target Localization for Distributed Sensor Array Networks 基于组稀疏度的分布式传感器阵列网络目标定位
Qing Shen, Wei Liu, Li Wang, Yin Liu
The target localization problem for distributed sensor array networks where a sub-array is placed at each receiver is studied, and under the compressive sensing (CS) framework, a group sparsity based two-dimensional localization method is proposed. Instead of fusing the separately estimated angles of arrival (AOAs), it processes the information collected by all the receivers simultaneously to form the final target locations. Simulation results show that the proposed localization method provides a significant performance improvement compared with the commonly used maximum likelihood estimator (MLE).
研究了分布式传感器阵列网络的目标定位问题,在压缩感知(CS)框架下,提出了一种基于群稀疏度的二维定位方法。它不是融合单独估计的到达角(AOAs),而是同时处理所有接收器收集的信息以形成最终目标位置。仿真结果表明,与常用的极大似然估计(MLE)相比,所提出的定位方法具有显著的性能提升。
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引用次数: 3
Estimation of Network Processes via Blind Graph Multi-filter Identification 基于盲图多滤波器辨识的网络过程估计
Yu Zhu, F. J. Garcia, A. Marques, Santiago Segarra
We study the problem of jointly estimating several network processes that are driven by the same input, recasting it as one of blind identification of a bank of graph filters. More precisely, we consider the observation of several graph signals – i.e., signals defined on the nodes of a graph – and we model each of these signals as the output of a different network process (represented by a graph filter) defined on a common known graph and driven by a common unknown input. Our goal is to recover the specifications of every network process by only observing the outputs. Since every process shares the same input, the estimation problems are coupled, and a joint inference method is proposed. We study two different scenarios, one where the orders of the filters are known, and one where they are not. For the former case we propose a least-squares approach and provide conditions for recovery. For the latter case, we put forth a sparse recovery algorithm with theoretical guarantees. Finally, we illustrate the methods here proposed via numerical experiments.
我们研究了由相同输入驱动的多个网络过程的联合估计问题,将其转化为一组图滤波器的盲识别问题。更准确地说,我们考虑对几个图信号的观察-即,在图的节点上定义的信号-我们将这些信号建模为不同网络过程(由图过滤器表示)的输出,这些网络过程定义在一个已知的图上,并由一个共同的未知输入驱动。我们的目标是仅通过观察输出来恢复每个网络流程的规范。由于每个过程共享相同的输入,将估计问题耦合起来,提出了一种联合推理方法。我们研究了两种不同的情况,一种是已知滤波器阶数的情况,另一种是未知的情况。对于前一种情况,我们提出了最小二乘方法,并给出了恢复的条件。对于后一种情况,我们提出了一种具有理论保证的稀疏恢复算法。最后,通过数值实验对本文提出的方法进行了验证。
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引用次数: 4
End-to-end Change Detection Using a Symmetric Fully Convolutional Network for Landslide Mapping 基于对称全卷积网络的滑坡映射端到端变化检测
Tao Lei, Qi Zhang, Dinghua Xue, Tao Chen, H. Meng, A. Nandi
In this paper, we propose a novel approach based on a symmetric fully convolutional network within pyramid pooling (FCN-PP) for landslide mapping (LM). The proposed approach has three advantages. Firstly, this approach is automatic and insensitive to noise because multivariate morphological reconstruction (MMR) is used for image preprocessing. Secondly, it is able to take into account features from multiple convolutional layers and explore efficiently the context of images, which leads to a good tradeoff between wider receptive field and the use of context. Finally, the selected pyramid pooling module addresses the drawback of single-scale pooling employed by convolutional neural network (CNN), fully convolutional network (FCN), U-Net, etc. Experimental results show that the proposed FCN-PP is effective for LM, and it outperforms state-of-the-art approaches in terms of four metrics, Precision, Recall, F -score, and Accuracy.
本文提出了一种基于金字塔池内对称全卷积网络(FCN-PP)的滑坡映射(LM)新方法。所提出的方法有三个优点。首先,该方法采用多变量形态学重构(multivariate morphological reconstruction, MMR)进行图像预处理,具有自动化和对噪声不敏感的特点。其次,它能够考虑来自多个卷积层的特征,并有效地探索图像的上下文,从而在更宽的接受域和使用上下文之间取得了很好的权衡。最后,所选择的金字塔池化模块解决了卷积神经网络(CNN)、全卷积网络(FCN)、U-Net等采用单尺度池化的缺点。实验结果表明,所提出的FCN-PP对于LM是有效的,并且在四个指标(Precision, Recall, F -score和Accuracy)方面优于目前最先进的方法。
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引用次数: 20
Self-attention Based Prosodic Boundary Prediction for Chinese Speech Synthesis 基于自注意的汉语语音合成韵律边界预测
Chunhui Lu, Pengyuan Zhang, Yonghong Yan
Predicting prosodic boundaries from input text plays an important role in Chinese text-to-speech (TTS) system, which directly influences the naturalness and intelligibility of synthesized speech. In this paper, we propose to combine self-attention with multitask learning for prosodic boundary prediction. Self-attention is used to capture the dependency between two arbitrary characters in the input sentence, while multitask learning models the relationships between prosodic boundaries and lexicon words by setting word segmentation as an auxiliary task. The proposed method can generate prosodic boundary labels directly from Chinese characters and achieve the whole process end-to-end. Experimental results show the effectiveness of our proposed model and prove that the performance can be further improved by pretraining the model with extra word segmentation data.
从输入文本中预测韵律边界在汉语文本到语音(TTS)系统中起着重要的作用,它直接影响到合成语音的自然度和可理解度。在本文中,我们提出将自我注意与多任务学习结合起来进行韵律边界预测。自注意用于捕获输入句子中任意两个字符之间的依赖关系,而多任务学习通过将分词作为辅助任务来建模韵律边界与词汇词之间的关系。该方法可以直接从汉字中生成韵律边界标签,实现端到端生成过程。实验结果表明了该模型的有效性,并证明了使用额外的分词数据对模型进行预训练可以进一步提高模型的性能。
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引用次数: 24
Convex Combination of Constraint Vectors for Set-membership Affine Projection Algorithms 集隶属仿射投影算法约束向量的凸组合
T. Ferreira, W. Martins, Markus V. S. Lima, P. Diniz
Set-membership affine projection (SM-AP) adaptive filters have been increasingly employed in the context of online data-selective learning. A key aspect for their good performance in terms of both convergence speed and steady-state mean-squared error is the choice of the so-called constraint vector. Optimal constraint vectors were recently proposed relying on convex optimization tools, which might sometimes lead to prohibitive computational burden. This paper proposes a convex combination of simpler constraint vectors whose performance approaches the optimal solution closely, utilizing much fewer computations. Some illustrative examples confirm that the sub-optimal solution follows the accomplishments of the optimal one.
集隶属度仿射投影(SM-AP)自适应滤波器越来越多地应用于在线数据选择学习。它们在收敛速度和稳态均方误差方面表现良好的一个关键方面是选择所谓的约束向量。最近提出的最优约束向量依赖于凸优化工具,有时可能导致令人望而却步的计算负担。本文提出了一种更简单约束向量的凸组合,其性能接近最优解,使用的计算量少得多。一些说明性的例子证实了次最优解遵循最优解的结果。
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引用次数: 3
Non-local Self-attention Structure for Function Approximation in Deep Reinforcement Learning 深度强化学习中函数逼近的非局部自注意结构
Z. Wang, Xi Xiao, Guangwu Hu, Yao Yao, Dianyan Zhang, Zhendong Peng, Qing Li, Shutao Xia
Reinforcement learning is a framework to make sequential decisions. The combination with deep neural networks further improves the ability of this framework. Convolutional nerual networks make it possible to make sequential decisions based on raw pixels information directly and make reinforcement learning achieve satisfying performances in series of tasks. However, convolutional neural networks still have own limitations in representing geometric patterns and long-term dependencies that occur consistently in state inputs. To tackle with the limitation, we propose the self-attention architecture to augment the original network. It provides a better balance between ability to model long-range dependencies and computational efficiency. Experiments on Atari games illustrate that self-attention structure is significantly effective for function approximation in deep reinforcement learning.
强化学习是一个做出连续决策的框架。与深度神经网络的结合进一步提高了该框架的能力。卷积神经网络使直接基于原始像素信息进行序列决策成为可能,并使强化学习在一系列任务中取得令人满意的性能。然而,卷积神经网络在表示几何模式和长期依赖关系方面仍然有自己的局限性,这些依赖关系在状态输入中始终存在。为了解决这个问题,我们提出了自关注架构来增强原有的网络。它在远程依赖关系建模能力和计算效率之间提供了更好的平衡。在Atari游戏上的实验表明,自注意结构对于深度强化学习中的函数逼近是非常有效的。
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引用次数: 0
A Spiking Neural Network Approach to Auditory Source Lateralisation 听觉源侧化的脉冲神经网络方法
R. Luke, D. McAlpine
A novel approach to multi-microphone acoustic source localisation based on spiking neural networks is presented. We demonstrate that a two microphone system connected to a spiking neural network can be used to localise acoustic sources based purely on inter microphone timing differences, with no need for manually configured delay lines. A two sensor example is provided which includes 1) a front end which converts the acoustic signal to a series of spikes, 2) a hidden layer of spiking neurons, 3) an output layer of spiking neurons which represents the location of the acoustic source. We present details on training the network, and evaluation of its performance in quiet and noisy conditions. The system is trained on two locations, and we show that the lateralisation accuracy is 100% when presented with previously unseen data in quiet conditions. We also demonstrate the network generalises to modulation rates and background noise on which it was not trained.
提出了一种基于尖峰神经网络的多传声器声源定位方法。我们证明了连接到尖峰神经网络的两个麦克风系统可以完全基于麦克风间的时间差异来定位声源,而不需要手动配置延迟线。提供了一个双传感器示例,其包括1)将声信号转换为一系列尖峰的前端,2)尖峰神经元的隐藏层,3)表示声源位置的尖峰神经元的输出层。我们详细介绍了网络的训练,以及在安静和噪声条件下对其性能的评估。该系统在两个位置进行了训练,结果表明,在安静的条件下,当提供以前未见过的数据时,侧向化精度达到100%。我们还证明了网络可以泛化到调制速率和背景噪声,而不是它所训练的。
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引用次数: 3
Information Theoretic Lower Bound of Restricted Isometry Property Constant 受限等距性质常数的信息论下界
Gen Li, Jingkai Yan, Yuantao Gu
Compressed sensing seeks to recover an unknown sparse vector from undersampled rate measurements. Since its introduction, there have been enormous works on compressed sensing that develop efficient algorithms for sparse signal recovery. The restricted isometry property (RIP) has become the dominant tool used for the analysis of exact reconstruction from seemingly undersampled measurements. Although the upper bound of the RIP constant has been studied extensively, as far as we know, the result is missing for the lower bound. In this work, we first present a tight lower bound for the RIP constant, filling the gap there. The lower bound is at the same order as the upper bound for the RIP constant. Moreover, we also show that our lower bound is close to the upper bound by numerical simulations. Our bound on the RIP constant provides an information-theoretic lower bound about the sampling rate for the first time, which is the essential question for practitioners.
压缩感知旨在从欠采样率测量中恢复未知的稀疏向量。自从它被引入以来,在压缩感知方面已经有了大量的工作,开发了高效的稀疏信号恢复算法。限制等距特性(RIP)已成为分析从看似欠采样测量中精确重建的主要工具。虽然人们对RIP常数的上界进行了广泛的研究,但据我们所知,对下界的研究结果还很缺乏。在这项工作中,我们首先提出了RIP常数的严格下界,填补了那里的空白。RIP常数的下界与上界处于同一阶。此外,我们还通过数值模拟证明了我们的下界接近上界。我们对RIP常数的取值范围首次提供了一个关于采样率的信息论的下界,这是实践者需要解决的关键问题。
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引用次数: 1
A Variational Adaptive Population Importance Sampler 变分适应种群重要性采样器
Yousef El-Laham, P. Djurić, M. Bugallo
Adaptive importance sampling (AIS) methods are a family of algorithms which can be used to approximate Bayesian posterior distributions. Many AIS algorithms exist in the literature, where the differences arise in the manner by which the proposal distribution is adapted at each iteration. The adaptive population importance sampler (APIS), for example, deterministically samples from a mixture distribution and uses the local information given by the samples and weights to adapt the location parameter of each proposal. The update rules by nature are heuristic, but effective, especially in the case that the target posterior is multimodal. In this work, we introduce a novel AIS scheme which incorporates modern techniques in stochastic optimization to improve the methodology for higher-dimensional posterior inference. More specifically, we derive update rules for the parameters of each proposal by means of deterministic mixture sampling and show that the method outperforms other state-of-the-art approaches in high-dimensional scenarios.
自适应重要性抽样(AIS)方法是一类用于近似贝叶斯后验分布的算法。文献中存在许多AIS算法,其中差异在于每次迭代时适应提案分布的方式。例如,自适应种群重要性采样器(api)从混合分布中确定样本,并使用样本和权重给出的局部信息来适应每个提案的位置参数。更新规则本质上是启发式的,但它是有效的,特别是在目标后验是多模态的情况下。在这项工作中,我们引入了一种新的AIS方案,该方案结合了随机优化中的现代技术,以改进高维后验推理的方法。更具体地说,我们通过确定性混合抽样推导出每个提案参数的更新规则,并表明该方法在高维场景下优于其他最先进的方法。
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引用次数: 6
Towards End-to-end Speech-to-text Translation with Two-pass Decoding 基于双通道解码的端到端语音到文本翻译
Tzu-Wei Sung, Jun-You Liu, Hung-yi Lee, Lin-Shan Lee
Speech-to-text translation (ST) refers to transforming the audio in source language to the text in target language. Mainstream solutions for such tasks are to cascade automatic speech recognition with machine translation, for which the transcriptions of the source language are needed in training. End-to-end approaches for ST tasks have been investigated because of not only technical interests such as to achieve globally optimized solution, but the need for ST tasks for the many source languages worldwide which do not have written form. In this paper, we propose a new end-to-end ST framework with two decoders to handle the relatively deeper relationships between the source language audio and target language text. The first-pass decoder generates some useful latent representations, and the second-pass decoder then integrates the output of both the encoder and the first-pass decoder to generate the text translation in target language. Only paired source language audio and target language text are used in training. Preliminary experiments on several language pairs showed improved performance, and offered some initial analysis.
语音到文本的翻译是指将源语音频转换成目的语文本。此类任务的主流解决方案是将自动语音识别与机器翻译级联,为此在训练中需要源语言的转录。对ST任务的端到端方法进行了调查,因为不仅技术利益,如实现全局优化的解决方案,而且需要ST任务为世界各地的许多源语言没有书面形式。在本文中,我们提出了一个带有两个解码器的新的端到端翻译框架来处理源语言音频和目标语言文本之间相对更深层次的关系。第一遍解码器生成一些有用的潜在表示,然后第二遍解码器集成编码器和第一遍解码器的输出以生成目标语言的文本翻译。在训练中只使用成对的源语言音频和目标语言文本。对几种语言对的初步实验表明,性能有所提高,并提供了一些初步分析。
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引用次数: 25
期刊
ICASSP 2019 - 2019 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)
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