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2020 International Conference on Signal Processing and Communications (SPCOM)最新文献

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Onset detection of Tabla Strokes using LP Analysis 用LP分析检测手板卒中发病
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179587
R. Gowriprasad., K. Murty
Onset detection is an important first step in music analysis. We propose a pre-processing scheme for improved onset detection of complex strokes of Indian percussion instruments with resonance characteristics. In this work, we explore the onset detection of Tabla (Indian percussion instrument) strokes. The resonance characteristics of tabla strokes poses challenges to onset detection. In such cases, the energy-based and spectral flux-based onset detectors are often inaccurate on the raw signal. We propose an onset detection algorithm addressing these challenges using Linear Prediction (LP) analysis and Hilbert envelope (HE) in tandem. Tabla signal is modeled using LP, and its residual highlights the onset time instances very well. Unipolar nature of HE on top of LP residual further enhances the onset instances. Onset detection is performed using energy based, spectral flux based and the state of the art Machine Learning based onset detectors on the Hilbert envelope of LP residual (HELP). Experiments were performed on tabla solo played at various tempi and the results show that the HELP based approach provides 12% relative improvement in F-measures compared to the performance on raw tabla signal.
起始音检测是音乐分析中重要的第一步。提出了一种预处理方案,以改进具有共振特征的印度打击乐器复杂笔划的起始检测。在这项工作中,我们探讨了手鼓(印度打击乐器)笔触的起始检测。手鼓敲击的共振特性给起搏检测带来了挑战。在这种情况下,基于能量和基于光谱通量的起始检测器通常对原始信号不准确。我们提出了一种使用线性预测(LP)分析和希尔伯特包络(HE)串联的开始检测算法来解决这些挑战。Tabla信号采用LP建模,其残差很好地突出了起始时间实例。在LP残馀上HE的单极特性进一步增强了发病实例。在LP残差的希尔伯特包络(HELP)上,使用基于能量、基于光谱通量和最先进的基于机器学习的起始检测器进行起始检测。实验结果表明,与原始手鼓信号相比,基于HELP的方法在f测量方面的性能提高了12%。
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引用次数: 4
Performance Analysis of DF Based mixed VLC-FSO-VLC System 基于DF的VLC-FSO-VLC混合系统性能分析
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179534
Rima Deka, Sanya Anees
In this paper the performance of decode-and-forward (DF) based mixed visible light communication-free space optical communication-visible light communication (VLC-FSO-VLC) co-operative system is presented. In the proposed system, the FSO link is used for providing the outdoor connectivity and VLC environment is used for indoor connectivity. The FSO link is characterized by the impact of Double Generalized Gamma (DGG) distributed atmospheric turbulence and Rayleigh distributed pointing errors and VLC environment follows the Lambertian radiation pattern of the light-emitting-diode (LED). For this system, novel mathematical expressions are derived for probability density function and cumulative distribution function of the system’s signal-to-noise-ratio (SNR). Using these SNR statistics, the closed-form analytical expressions of average bit-error rate (BER) for binary modulation techniques considering the possibility of perfect decoding by the DF relays are obtained. The results show that the BER of the considered VLC-FSOVLC system gives a better performance while using coherent binary phase shift keying (CBPSK) modulation technique as compared to other modulation techniques.
介绍了基于译码转发(DF)的可见光通信-自由空间光通信-可见光通信(VLC-FSO-VLC)混合协作系统的性能。在该系统中,FSO链路用于提供室外连接,VLC环境用于提供室内连接。FSO链路的特点是受双广义伽玛(DGG)分布大气湍流和瑞利分布指向误差的影响,VLC环境遵循发光二极管(LED)的朗伯辐射模式。针对该系统,推导了系统信噪比的概率密度函数和累积分布函数的数学表达式。利用这些信噪比统计,得到了考虑DF中继完美译码可能性的二进制调制技术的平均误码率(BER)的封闭解析表达式。结果表明,采用相干二相移键控(CBPSK)调制技术的VLC-FSOVLC系统的误码率优于其他调制技术。
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引用次数: 3
Virtual Full-Duplex Relaying in a Buffer-Aided Multi-Hop Cooperative Network 缓冲辅助多跳协作网络中的虚拟全双工中继
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179530
A. Shukla, B. Manoj, M. Bhatnagar
In this paper, we propose a virtual full-duplex (VFD) multi-hop cooperative relaying network using buffer-aided halfduplex (HD) relays. Firstly, for a given multi-hop network, the links are partitioned into three groups and then information is transmitted over a single time slot using two selected best links, which enhances the network coverage and reliability of the system. The Markov chain approach is used to analyze the state transition matrix which models the evolution of buffer states. An analytical expression of steady state probability is obtained, using which the outage probability of the system is evaluated. Numerical results validate our analytical findings and it shows that the proposed relaying scheme offers a better outage performance as compared to that of the conventional buffer-aided max-link relay selection scheme in a multi-hop communication system.
在本文中,我们提出了一个虚拟全双工(VFD)多跳合作中继网络,使用缓冲辅助半双工(HD)中继。首先,对于给定的多跳网络,将链路划分为三组,然后选择两个最佳链路在单个时隙中传输信息,从而提高了网络的覆盖率和系统的可靠性。利用马尔可夫链方法对状态转移矩阵进行了分析,该矩阵模拟了缓冲状态的演化。得到了系统稳态概率的解析表达式,利用该表达式计算了系统的中断概率。数值结果验证了我们的分析结果,表明在多跳通信系统中,与传统的缓冲辅助最大链路中继选择方案相比,所提出的中继方案具有更好的中断性能。
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引用次数: 3
Age-of-Information Aware Scheduling under Markovian Energy Arrivals 马尔可夫能量到达下的信息时代感知调度
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179542
Bejjipuram Sombabu, Sharayu Moharir
We consider the task of scheduling updates from multiple sources to a central monitoring station via a shared communication channel. Each source harvests energy from nature to measure a time-varying quantity and report these measurements to the monitoring station. Prior work in this area focuses on the setting where energy arrivals are assumed to be independent across time. Motivated by the time-correlation in energy generated by many renewable energy sources, we use a Markov process to model the energy arrivals. The goal is to minimize the time average of the weighted sum of the ages-of-information of the sources. We use Whittle’s relaxation and propose a modification of the Whittle Index to design a scheduling policy. We show that our policy outperforms other natural policies via simulations.
我们考虑通过共享通信通道调度多个源到中央监测站的更新任务。每个源从自然中获取能量,测量时变量,并将这些测量结果报告给监测站。该领域先前的工作主要集中在假设能量到达是随时间独立的情况下。由于许多可再生能源产生的能量具有时间相关性,我们使用马尔可夫过程对能量到达进行建模。目标是最小化源的信息年龄加权和的时间平均值。我们利用Whittle的松弛,提出了Whittle索引的修改来设计调度策略。我们通过模拟表明,我们的政策优于其他自然政策。
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引用次数: 3
Classification of Speech vs. Speech with Background Music 语音分类与有背景音乐的语音
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179491
Mrinmoy Bhattacharjee, S. Prasanna, P. Guha
Applications that perform enhancement of speech containing background music require a critical preprocessing step that can efficiently detect such segments. This work proposes such a preprocessing method to detect speech with background music that is mixed at different SNR levels. A bag-of-words approach is proposed in this work. Representative dictionaries from speech and music data are first learned. The signals are processed as spectrograms of 1s intervals. Rows of these spectrograms are used to learn separate speech and music dictionaries. This work proposes a weighting scheme to reduce confusion by suppressing codewords of one class that have similarities to the other class. The proposed feature is a weighted histogram of 1s audio intervals obtained from the learned dictionaries. The classification is performed using a deep neural network classifier. The proposed approach is validated against a baseline and benchmarked over two publicly available datasets. The proposed feature shows promising results, both individually and in combination with the baseline.
对包含背景音乐的语音进行增强的应用程序需要一个关键的预处理步骤来有效地检测这些片段。本研究提出了这样一种预处理方法来检测背景音乐在不同信噪比水平下的混合语音。本文提出了一种词袋方法。首先学习语音和音乐数据中的代表性词典。信号以1s间隔的谱图处理。这些谱图被用来学习单独的语音和音乐词典。这项工作提出了一种加权方案,通过抑制一类与另一类具有相似性的码字来减少混淆。所提出的特征是从学习字典中获得的15个音频间隔的加权直方图。使用深度神经网络分类器进行分类。提出的方法在两个公开可用的数据集上对基线和基准进行了验证。所提出的特性显示了有希望的结果,无论是单独还是与基线结合。
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引用次数: 1
An Improved LSF-based Algorithm for Sinusoidal Frequency Estimation that Achieves Maximum Likelihood Performance 一种改进的基于lsf的最大似然性能正弦频率估计算法
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179546
P. Vishnu, C. S. Ramalingam
In this paper we propose a method for sinusoidal frequency estimation that improves upon our previously proposed LSF-based algorithm that used at most 5p candidate points, where p is the number of sinusoids present. In this paper we propose the following improvements: (i) reduced the number of candidate frequencies to at most 2p points, (ii) reduced the method’s threshold to equal that of ML, and (iii) reduced the computational burden by switching to methods like ESPRIT when the SNR is above threshold. Since neither the SNR nor the threshold is known, we estimate them from the data. The proposed reduction-in-threshold step can be applied to EPUMA (proposed Qian et al.), with which we compare our results. For the well-known two-sinusoid example the proposed method has the same threshold as that of ML; ML performance is also achieved when tested on a new, three-sinusoid example.
在本文中,我们提出了一种正弦频率估计方法,该方法改进了我们之前提出的基于lsf的算法,该算法最多使用5p个候选点,其中p是存在的正弦波数。在本文中,我们提出了以下改进:(i)将候选频率的数量减少到最多2p个点,(ii)将方法的阈值降低到与ML相等,(iii)当信噪比高于阈值时,通过切换到ESPRIT等方法来减少计算负担。由于信噪比和阈值都不知道,我们从数据中估计它们。所提出的阈值还原步骤可以应用于EPUMA (Qian等人提出的),我们将其与我们的结果进行比较。对于众所周知的双正弦例子,所提出的方法具有与ML相同的阈值;当在一个新的三正弦示例上进行测试时,也实现了ML性能。
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引用次数: 2
Kernel based Matching and a Novel training approach for CNN-based QbE-STD 基于核匹配和基于cnn的QbE-STD训练新方法
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179588
P. Naik, M. Gaonkar, Veena Thenkanidiyoor, A. D. Dileep
Query-by-Example based spoken term detection (QbE-STD) to audio search involves matching an audio query with the reference utterances to find the relevant utterances. QbE-STD involves computing a matching matrix between a query and reference utterance using a suitable metric. In this work we propose to use kernel based matching by considering histogram intersection kernel (HIK) as a matching metric. A CNN-based approach to QbE-STD involves first converting a matching matrix to a corresponding size-normalized image and classifying the image as relevant or not [6]. In this work, we propose to train a CNN-based classifier using size-normalized images instead of splitting them into subimages as in [6]. Training approach proposed in this work is expected to be more effective since there is less chance of a CNN based classifier getting confused. The effectiveness of the proposed kernel based matching and novel training approach is studied using TIMIT dataset.
基于实例查询的语音术语检测(QbE-STD)涉及将音频查询与参考话语进行匹配,以查找相关的话语。QbE-STD涉及使用合适的度量来计算查询和参考话语之间的匹配矩阵。在这项工作中,我们提出使用基于核的匹配,考虑直方图交集核(HIK)作为匹配度量。基于cnn的QbE-STD方法首先将匹配矩阵转换为相应的尺寸归一化图像,并将图像分类为相关或不相关[6]。在这项工作中,我们建议使用尺寸归一化的图像来训练基于cnn的分类器,而不是像[6]那样将它们分成子图像。在这项工作中提出的训练方法预计会更有效,因为基于CNN的分类器混淆的可能性更小。利用TIMIT数据集,研究了基于核匹配和新训练方法的有效性。
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引用次数: 1
Three-Dimensional Underwater Acoustic Source Localization by Sparse Signal Reconstruction Techniques 基于稀疏信号重构技术的三维水声声源定位
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179579
A. Koul, G. V. Anand, Sanjeev Gurugopinath, K. Nathwani
Several superresolution source localization algorithms based on the sparse signal reconstruction framework have been developed in recent years. These methods also offer other advantages such as immunity to noise coherence and robustness to reduction in the number of snapshots. The application of these methods is mostly limited to the problem of one dimensional (1-D) direction-of-arrival estimation. In this paper, we have developed 2-D and 3-D versions of two sparse signal reconstruction methods, viz. $ell_{1}$-SVD and re-weighted $ell_{1}$-SVD, and applied them to the problem of 3-D localization of underwater acoustic sources. A vertical linear array is used for estimation of range and depth and a horizontal cross-shaped array is used for bearing estimation. It is shown that the $ell_{1}$-SVD and re-weighted $ell_{1}$-SVD processors outperform the widely used MUSIC and Bartlett processors.
近年来出现了几种基于稀疏信号重构框架的超分辨信号源定位算法。这些方法还具有抗噪声、相干性和减少快照数量的鲁棒性等优点。这些方法的应用大多局限于一维到达方向估计问题。本文提出了两种稀疏信号重建方法$ell_{1}$-SVD和重加权$ell_{1}$-SVD的二维和三维版本,并将其应用于水声声源的三维定位问题。垂直线性阵列用于估计距离和深度,水平十字形阵列用于估计方位。结果表明,$ell_{1}$-SVD和重新加权的$ell_{1}$-SVD处理器优于广泛使用的MUSIC和Bartlett处理器。
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引用次数: 7
Accelerated Weighted ℓ1-Minimization for MRI Reconstruction Under Tight Frames in Complex Domain 复杂域紧框架下MRI重构的加速加权最小化算法
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179611
P. Pokala, C. Seelamantula
We propose an improvement of the projected fast iterative soft-thresholding algorithm (pFISTA) and smoothing FISTA (SFISTA) to achieve faster convergence and improved reconstruction accuracy. The pFISTA addresses the problem of compressed sensing magnetic resonance imaging (CS-MRI) reconstruction under tight frames and considers standard $ell_{1}$ norm minimization. The $ell_{1} -$norm weighs each component in a sparse vector equally. However, this is restrictive. We employ the weighted $ell_{1} -$regularizer, defined over a complex-domain as the sparsity-promoting function in CS-MRI reconstruction. The weighted $ell_{1} -$regularizer assigns different weights to the components in a sparse vector to improve upon reconstruction accuracy. The optimization objective in CS-MRI is a real-valued function defined over a complex-domain and is therefore not holomorphic. We derive an algorithm, namely, projected weighted iterative soft-thresholding algorithm (pWISTA) based on Wirtinger calculus to solve the weighted $ell_{1} -$regularized CS-MRI reconstruction under tight frames. We show that the proximal operator for the weighted $ell_{1}$ regularizer over a complex-domain is the soft-thresholding operator, but with a different threshold for each component. We also incorporate Nesterov’s momentum into the pWISTA update to obtain the projected weighted fast iterative soft-thresholding algorithm (pWFISTA), which result in accelerated optimization as shown by the experimental results.
本文提出了一种改进投影快速迭代软阈值算法(pFISTA)和平滑FISTA算法(SFISTA),以实现更快的收敛速度和更高的重建精度。pFISTA解决了紧框架下压缩感知磁共振成像(CS-MRI)重建问题,并考虑了标准$ell_{1}$范数最小化。$ell_{1} -$范数相等地对稀疏向量中的每个分量进行加权。然而,这是限制性的。我们使用加权$ell_{1} -$正则化器,在复域上定义为CS-MRI重构中的稀疏性促进函数。加权$ell_{1} -$正则化器为稀疏向量中的分量分配不同的权重,以提高重构精度。CS-MRI的优化目标是在复域上定义的实值函数,因此不是全纯的。提出了一种基于Wirtinger演算的投影加权迭代软阈值算法(pWISTA),用于求解紧框架下加权$ell_{1} -$正则化CS-MRI重构。我们证明了复域上加权$ell_{1}$正则化器的近端算子是软阈值算子,但每个分量具有不同的阈值。我们还将Nesterov动量引入到pWISTA更新中,得到了投影加权快速迭代软阈值算法(pWFISTA),实验结果表明该算法具有加速优化的效果。
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引用次数: 2
Index Coding at the WiFi Edge: An Implementation Study for Video Delivery WiFi边缘的索引编码:视频传输的实现研究
Pub Date : 2020-07-01 DOI: 10.1109/SPCOM50965.2020.9179525
Lalhruaizela Chhangte, E. Viterbo, D. Manjunath, N. Karamchandani
HTTP based video streaming has become the de facto standard for video content delivery across different video streaming services. However, video content delivery continues to be challenged at the wireless edge by inadequate and highly variable bandwidth. In this paper, we describe WiCode, a platform that improves HTTP based video content delivery at the WiFi edge. WiCode uses coded delivery at the WiFi AP to reduce data transmissions in order to improve the perceived performance of video streaming at the users. WiCode performs index coding on video segments to reduce the number of bits transmitted. Further, it also performs index coding on UDP packets that are retransmitted to reduce the number of bits transmitted. This paper describes the design and implementation of WiCode, and the practical gains achievable due to employing coded delivery in a real system taking into account the overheads introduced by WiCode. The WiCode module at the client side is a browser plugin that does not require any client side device configuration changes. We also show the effect of variable and fixed length segment size on the perceived performance of WiCode.
基于HTTP的视频流已经成为跨不同视频流服务传输视频内容的事实上的标准。然而,视频内容传输在无线边缘继续受到带宽不足和高度可变的挑战。在本文中,我们描述了WiCode,一个在WiFi边缘改进基于HTTP的视频内容交付的平台。WiCode在WiFi AP上使用编码传输来减少数据传输,以提高用户对视频流的感知性能。WiCode在视频片段上执行索引编码,以减少传输的比特数。此外,它还对重传的UDP数据包执行索引编码,以减少传输的位数。本文描述了WiCode的设计和实现,以及考虑到WiCode引入的开销,在实际系统中采用编码交付所获得的实际收益。客户端的WiCode模块是一个浏览器插件,不需要更改任何客户端设备配置。我们还展示了可变和固定长度段大小对WiCode感知性能的影响。
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引用次数: 0
期刊
2020 International Conference on Signal Processing and Communications (SPCOM)
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