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2016 Signal Processing: Algorithms, Architectures, Arrangements, and Applications (SPA)最新文献

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Asynchronous down-link cooperative communication scheme in Rayleigh fading wireless environment 瑞利衰落无线环境下异步下行协同通信方案
Ahmed Desouky, A. El-Mahdy
In this paper, a down-link cooperative communication scheme is proposed using full-rate Quasi-Orthogonal Space-time Block Codes (QOSTBC), employing asynchronous mobile Decode and Forward (DF) relays in a Rayleigh fading wireless environment. The channel used is a partially-coherent relay channel (PCRC), where the first hop is non-coherent and the second is coherent. Base station broadcasts using four transmitting antennas, towards all relay nodes in range over the first hop. Each intermediate relay uses Minimum Decoding Complexity (MDC) decoder achieving linear complexity. A relay selection technique is applied, electing the best performing intermediate nodes based on the channel response and path loss due to location relative to the destination node. A power allocation technique is adopted, which ranks based on the chosen relays performance. Relays transmit over four pre-defined time slots per frame. The destination node uses a Maximum-Likelihood (ML) decoder.
在瑞利衰落无线环境下,提出了一种基于全速率准正交空时分组码(QOSTBC)的下行链路协同通信方案,该方案采用异步移动译码和转发(DF)中继。所使用的信道是部分相干中继信道(PCRC),其中第一跳是非相干的,第二跳是相干的。基站使用四个发射天线向第一跳范围内的所有中继节点广播。每个中间中继使用最小解码复杂度(MDC)解码器实现线性复杂度。采用中继选择技术,根据信道响应和相对于目标节点的路径损耗来选择性能最好的中间节点。采用功率分配技术,根据所选继电器的性能进行排序。中继每帧传输四个预定义的时隙。目标节点使用最大似然(ML)解码器。
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引用次数: 3
Smile veracity recognition using 3D texture features for image sequence processing 利用三维纹理特征对图像序列进行微笑真实性识别
K. Nurzynska, B. Smolka
Emotions are one of the most important cues of non-verbal communication, therefore are easily recognized by people. However, correct identification of intentions causes many problems, because telling apart the true emotion expression from the fake one is difficult. Therefore, there is a wide search nowadays for automatic techniques, which use computers to analyze behavior of people and learn the differences between spontaneous and posed emotional gestures. In this research we concentrate on the analysis of facial expressions in order to recognize between Duchenne (real) and fake smiles. Our approach treats the input movie as a dynamic texture and exploits LBP-TOP technique for its description. Additionally, the average facial image is removed from consideration as it is assumed, that only tiny changes differ the smile characters. The results obtained on UVA-NEMO dataset proved that it is possible to classify accurately between the spontaneous and posed smiles. Moreover, the influence of facial parts as well as changes due to a phase of smile are investigated.
情绪是非语言交流中最重要的线索之一,因此很容易被人们识别。然而,正确识别意图会导致许多问题,因为很难区分真实的情感表达和虚假的情感表达。因此,现在对自动技术进行了广泛的研究,这种技术使用计算机来分析人们的行为,并学习自发和做作的情感姿态之间的差异。在这项研究中,我们专注于分析面部表情,以识别杜兴(真实的)和假的微笑。我们的方法将输入影片视为动态纹理,并利用LBP-TOP技术对其进行描述。此外,平均面部图像被排除在考虑之外,因为假设只有微小的变化才会改变微笑的特征。在UVA-NEMO数据集上获得的结果证明,可以准确地区分自然微笑和做作微笑。此外,研究了面部部位的影响以及微笑相位的变化。
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引用次数: 1
DIAST variability illuminated thermal and visible ear images datasets DIAST变异性照明热耳图像和可见耳图像数据集
Syed Mohd Zahid Syed Zainal Ariffin, N. Jamil, P. Rahman
As the research in ear biometrics grows, there is a need to have a publicly available ear images database for researchers to test and validate their methods. There are many available ear databases captured in visible light domain. However, images in those databases are mostly acquired in a controlled environment, not in naturally illuminated environments. Up to the point of writing, there is no thermalbased ear image dataset available for research usage. In this paper, we introduce our ear image datasets that were captured in both visible and thermal domain. We present the detailed process of image acquisition, post-processing, and preparation of the ear datasets. The ear dataset consists of 2,200 images captured from 55 subjects in five different illumination conditions, ranging from 2 lux to 10,700 lux, measured using lux meter.
随着耳朵生物识别技术研究的发展,需要有一个公开的耳朵图像数据库,供研究人员测试和验证他们的方法。在可见光域有许多可用的耳数据库。然而,这些数据库中的图像大多是在受控环境中获取的,而不是在自然光照的环境中获取的。到撰写本文为止,还没有基于热成像的耳朵图像数据集可供研究使用。在本文中,我们介绍了我们在可见光和热域捕获的耳朵图像数据集。我们提出了详细的过程图像采集,后处理和耳数据集的准备。耳朵数据集由55名受试者在5种不同光照条件下拍摄的2200幅图像组成,光照条件从2勒克斯到10,700勒克斯,使用勒克斯计测量。
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引用次数: 7
High efficiency video coding 高效视频编码
K. R. Rao
In the family of video coding standards, HEVC has the promise and potential to replace/supplement all the existing standards (MPEG and H.26x series including H.264/AVC). While the complexity of the HEVC encoder is several times that of the H.264/AVC, the decoder complexity is within the range of the latter. Researchers are exploring about reducing the HEVC encoder complexity · Kim et al have shown that motion estimation (ME) occupies 77–81% of HEVC encoder implementation. Hence the focus has been in reducing the ME complexity. Several researchers have implemented performance comparison of HEVC with other standards such as H.264/AVC, MPEG-4 Part 2 visual, H.262/PEG-2 Video, H.263, and VP9 and also with image coding standards such as JPEG2000, JPEG-LS, and JPEG-XR. Several tests have shown that HEVC provides improved compression efficiency up to 50% bit rate reduction for the same subjective video quality compared to H.264/AVC. Besides addressing all current applications, HEVC is designed and developed to focus on two key issues: increased video resolution — up to 8k×4k — and increased use of parallel processing architecture. Brief description of the HEVC is provided. However for details and implementation, the reader is referred to the JCT-VC documents, overview papers, keynote speeches, tutorials, panel discussions, poster sessions, special issues, test models (TM/HM), web/ftp site, open source software, test sequences, anchor bit streams and the latest books on HEVC. Also researchers are exploring transcoding between HEVC and other standards such as MPEG-2 and H.264. Further extensions to HEVC are scalable video coding (SVC), 3D video/multiview video coding and range extensions which include screen content coding (SCC), bit depths larger than 10 bits and color sampling of 4:2:2 and 4:4:4. SCC in general refers to computer generated objects and screen shots from computer applications (both images and videos) and may require lossless coding. Some of these extensions have been finalized by the end of 2014 (time frame for SCC is late 2016). They also provide fertile ground for R & D. Iguchi et al have already developed a hardware encoder for super hi-vision (SHV) i.e., ultra HDTV at 7680×4320 pixel resolution. Also real-time hardware implementation of HEVC encoder for 1080p HD video has been done. NHK is planning SHV experimental broadcasting in 2016. A 249-Mpixel/s HEVC video decoder chip for 4k Ultra-HD applications has already been developed. Bross et al have shown that real time software decoding of 4K (3840×2160) video with HEVC is feasible on current desktop CPUs using four CPU cores. They also state that encoding 4K video in real time on the other hand is a challenge. Multimedia research group (MRC) predicts 2 billion HEVC based devices by end of 2016.
在视频编码标准家族中,HEVC具有替代/补充现有所有标准(MPEG和H.26x系列,包括H.264/AVC)的前景和潜力。虽然HEVC编码器的复杂度是H.264/AVC的数倍,但解码器的复杂度在后者的范围内。研究人员正在探索如何降低HEVC编码器的复杂性·Kim等人已经表明,运动估计(ME)占据了HEVC编码器实现的77-81%。因此,重点是降低ME的复杂性。一些研究人员已经将HEVC与其他标准(如H.264/AVC、MPEG-4 Part 2 visual、H.262/PEG-2 Video、H.263和VP9)以及与图像编码标准(如JPEG2000、JPEG-LS和JPEG-XR)进行了性能比较。几项测试表明,与H.264/AVC相比,HEVC在相同的主观视频质量下提供了更高的压缩效率,比特率降低了50%。除了解决当前所有应用之外,HEVC的设计和开发主要关注两个关键问题:提高视频分辨率(高达8k×4k)和增加并行处理架构的使用。简要介绍了HEVC。然而,关于细节和实现,读者可以参考JCT-VC文档、概述论文、主题演讲、教程、小组讨论、海报会议、特刊、测试模型(TM/HM)、web/ftp站点、开源软件、测试序列、锚位流和最新的HEVC书籍。此外,研究人员还在探索HEVC与其他标准(如MPEG-2和H.264)之间的转码。HEVC的进一步扩展是可扩展视频编码(SVC), 3D视频/多视图视频编码和范围扩展,包括屏幕内容编码(SCC),大于10位的位深和4:2:2和4:4:4的颜色采样。SCC一般是指计算机生成的对象和从计算机应用程序(图像和视频)截取的屏幕截图,可能需要无损编码。其中一些扩展已于2014年底完成(SCC的时间框架为2016年底)。它们也为研发提供了肥沃的土壤。井口等人已经开发了一种用于超高清(SHV)的硬件编码器,即7680×4320像素分辨率的超高清电视。并对1080p高清视频的HEVC编码器进行了实时硬件实现。NHK计划在2016年进行SHV实验广播。目前已经开发出一款用于4k超高清应用的249mpixel /s HEVC视频解码器芯片。Bross等人已经证明,使用HEVC对4K (3840×2160)视频进行实时软件解码在当前使用四个CPU内核的桌面CPU上是可行的。他们还表示,另一方面,对4K视频进行实时编码是一个挑战。多媒体研究集团(MRC)预测,到2016年底,基于HEVC的设备将达到20亿台。
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引用次数: 298
Selective interference alignment in 5G networks 5G网络中的选择性干扰对准
Sarah Imam, A. El-Mahdy
The evolving fifth generation (5G) cellular wireless networks provide higher data rates, enhanced end-user quality-of-experience (QoE), reduced end-to-end latency, and lower energy consumption. The 5G networks will consist of macrocells, femtocells and picocells with heterogeneous characteristics and capacities which will result in a multi-tier architecture. Hence, interference is one of the main problems in such systems and it will cause degradation in the system performance. Interference alignment techniques can be used to solve this problem. In this paper, a proposed cross-tier interference management algorithm for coexisting two-tier networks is introduced. First at each receiver, we calculate the sum of the square of the channel gains from each macro user to this receiver and the average of those sum values is taken as a threshold. Macro users whose sum values are greater than this threshold are selected and aligned at the femto receiver. This alignment is done by using precoders at the transmitters. In the proposed algorithm, the subspace in which the interference signals would be aligned is identified then a distributed algorithm is applied to identify the precoders needed at the selected interferers. Numerical results demonstrate the effectiveness of selective interference alignment for uplink interference management. The Cumulative Distribution Function (CDF) is used to measure the system performance.
不断发展的第五代(5G)蜂窝无线网络提供更高的数据速率、增强的最终用户体验质量(QoE)、更少的端到端延迟和更低的能耗。5G网络将由具有异构特性和容量的宏蜂窝、飞蜂窝和皮蜂窝组成,这将导致多层架构。因此,干扰是这类系统的主要问题之一,它会导致系统性能的下降。干涉对准技术可以用来解决这个问题。本文提出了一种共存两层网络的跨层干扰管理算法。首先,在每个接收器上,我们计算从每个宏用户到该接收器的信道增益的平方和,并将这些和值的平均值作为阈值。选择总和大于此阈值的宏用户并在femto接收器上对齐。这种校准是通过在发射机上使用预编码器来完成的。该算法首先确定干扰信号对齐的子空间,然后采用分布式算法识别所选干扰处所需的预编码器。数值结果表明了选择性干扰对准在上行干扰管理中的有效性。CDF (Cumulative Distribution Function)用于度量系统的性能。
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引用次数: 4
Real-time advanced speech enhancement using low-power, fixed-point hardware 实时高级语音增强使用低功耗,定点硬件
D. Bismor, Krzysztof Bartnicki
This paper describes a project aiming at development of an embedded system speech enhancement solution. It was assumed that a low-cost, low-power, fixed-point processor will be used. It was also assumed that both the stationary and nonstationary noises should be removed. To meet the assumptions, the EZDSP5535 evaluation board, with the TMS320C5535 fixed-point digital signal processor, was selected as a hardware platform. Three speech enhancement algorithms were implemented: the FIR filtering, the adaptive line enhancement, and the spectral subtraction. The last algorithm, being a frequency-domain algorithm, utilizes the FFT co-processor available in the TMS320C5535 chip. All the algorithms can work simultaneously. Examples of good final performance are shown.
本文介绍了一个旨在开发嵌入式系统语音增强解决方案的方案。假定将使用低成本、低功耗、定点处理器。同时假定平稳噪声和非平稳噪声都要去除。为满足上述假设,选择采用TMS320C5535定点数字信号处理器的EZDSP5535评估板作为硬件平台。实现了三种语音增强算法:FIR滤波、自适应线增强和谱减法。最后一种算法是频域算法,利用TMS320C5535芯片中的FFT协处理器。所有算法可以同时工作。最后给出了一些表现良好的例子。
{"title":"Real-time advanced speech enhancement using low-power, fixed-point hardware","authors":"D. Bismor, Krzysztof Bartnicki","doi":"10.1109/SPA.2016.7763588","DOIUrl":"https://doi.org/10.1109/SPA.2016.7763588","url":null,"abstract":"This paper describes a project aiming at development of an embedded system speech enhancement solution. It was assumed that a low-cost, low-power, fixed-point processor will be used. It was also assumed that both the stationary and nonstationary noises should be removed. To meet the assumptions, the EZDSP5535 evaluation board, with the TMS320C5535 fixed-point digital signal processor, was selected as a hardware platform. Three speech enhancement algorithms were implemented: the FIR filtering, the adaptive line enhancement, and the spectral subtraction. The last algorithm, being a frequency-domain algorithm, utilizes the FFT co-processor available in the TMS320C5535 chip. All the algorithms can work simultaneously. Examples of good final performance are shown.","PeriodicalId":413894,"journal":{"name":"2016 Signal Processing: Algorithms, Architectures, Arrangements, and Applications (SPA)","volume":"42 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131841212","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 2
Rough set-based classification of EEG signals related to real and imagery motion 基于粗糙集的真实运动和图像运动脑电信号分类
P. Szczuko
A rough set-based approach to classification of EEG signals registered while subjects were performing real and imagery motions is presented in the paper. The appropriate subset of EEG channels is selected, the recordings are segmented, and features are extracted, based on time-frequency decomposition of the signal. Rough set classifier is trained in several scenarios, comparing accuracy of classification for real and imagery motion. Results are commented and further research is proposed.
提出了一种基于粗糙集的脑电信号分类方法。基于信号的时频分解,选择合适的脑电信号通道子集,对记录进行分割,提取特征。粗糙集分类器在几种场景下进行训练,比较了真实运动和图像运动的分类精度。对结果进行了评论,并提出了进一步研究的建议。
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引用次数: 5
Computational models for predicting sound quality 预测音质的计算模型
B. Moore
The quality of an audio device, such as a microphone, amplifier, or headphone, depends on how accurately the device transmits the properties of the sound source to the ear(s) of the listener. Two types of “distortion” can occur in this transmission: (1) “Linear” distortion, which may be described as a deviation of the frequency response from the “target” response; (2) Nonlinear distortion, which is characterised by frequency components in the output of the device that were not present in the input. These two forms of distortion have different perceptual effects. Their effects on sound quality can be predicted using a model of auditory processing with the following stages: (1) A filter to take into account the transmission of sound from the device to the ear of the listener; (2) A filter to simulate the effects of transmission through the middle ear; (3) An array of bandpass filters to simulate the auditory filters that exist in the cochlea of the inner ear. For predicting the perceptual effects of linear distortion, a model operating in the frequency domain can be used. For predicting the perceptual effects of nonlinear distortion, a model operating in the time domain is required, since the detailed waveforms at the outputs of the auditory filters need to be considered. The models described have been shown to give accurate predictions for a wide range of “artificial” and “real” linear and nonlinear distortions.
音频设备(如麦克风、放大器或耳机)的质量取决于这些设备将声源的特性传递到听者耳朵的准确度。在这种传输中可能出现两种类型的“失真”:(1)“线性”失真,它可以被描述为频率响应与“目标”响应的偏差;(2)非线性失真,其特征是设备输出中的频率成分在输入中不存在。这两种形式的扭曲具有不同的感知效果。它们对音质的影响可以通过以下几个阶段的听觉处理模型来预测:(1)考虑声音从设备传输到听者耳朵的过滤器;(2)模拟中耳传播效果的滤波器;(3)一组带通滤波器来模拟存在于内耳耳蜗中的听觉滤波器。为了预测线性失真的感知效应,可以使用在频域中工作的模型。为了预测非线性失真的感知效应,需要一个在时域内运行的模型,因为需要考虑听觉滤波器输出的详细波形。所描述的模型已被证明可以对大范围的“人为的”和“真实的”线性和非线性扭曲做出准确的预测。
{"title":"Computational models for predicting sound quality","authors":"B. Moore","doi":"10.1109/SPA.2016.7763577","DOIUrl":"https://doi.org/10.1109/SPA.2016.7763577","url":null,"abstract":"The quality of an audio device, such as a microphone, amplifier, or headphone, depends on how accurately the device transmits the properties of the sound source to the ear(s) of the listener. Two types of “distortion” can occur in this transmission: (1) “Linear” distortion, which may be described as a deviation of the frequency response from the “target” response; (2) Nonlinear distortion, which is characterised by frequency components in the output of the device that were not present in the input. These two forms of distortion have different perceptual effects. Their effects on sound quality can be predicted using a model of auditory processing with the following stages: (1) A filter to take into account the transmission of sound from the device to the ear of the listener; (2) A filter to simulate the effects of transmission through the middle ear; (3) An array of bandpass filters to simulate the auditory filters that exist in the cochlea of the inner ear. For predicting the perceptual effects of linear distortion, a model operating in the frequency domain can be used. For predicting the perceptual effects of nonlinear distortion, a model operating in the time domain is required, since the detailed waveforms at the outputs of the auditory filters need to be considered. The models described have been shown to give accurate predictions for a wide range of “artificial” and “real” linear and nonlinear distortions.","PeriodicalId":413894,"journal":{"name":"2016 Signal Processing: Algorithms, Architectures, Arrangements, and Applications (SPA)","volume":"33 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133772672","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 1
Multithreaded Java approach to speaker recognition 说话人识别的多线程Java方法
Radoslaw Weychan, T. Marciniak, A. Dabrowski
In this paper an analysis of multithread approach to speaker recognition is presented. Two cases have been investigated — the use of multithreading first, to reduce the total time of training and testing (when processing the speaker database) and second, to reduce the time of a single speaker recognition (e.g. in voice-based access systems). We have investigated the processing time in function of a number of threads in order to find the optimal number for 8-core CPU. The obtained results show that the computation time can be reduced even over 5 times for this kind of processor. The prepared implementation is available on Github source code repository.
本文分析了一种基于多线程的说话人识别方法。我们研究了两种情况——首先使用多线程来减少训练和测试的总时间(在处理说话人数据库时),其次减少单个说话人识别的时间(例如在基于语音的访问系统中)。为了找到8核CPU的最佳数量,我们研究了线程数函数的处理时间。计算结果表明,该处理器的计算时间可缩短5倍以上。准备好的实现可以在Github源代码存储库中获得。
{"title":"Multithreaded Java approach to speaker recognition","authors":"Radoslaw Weychan, T. Marciniak, A. Dabrowski","doi":"10.1109/SPA.2016.7763630","DOIUrl":"https://doi.org/10.1109/SPA.2016.7763630","url":null,"abstract":"In this paper an analysis of multithread approach to speaker recognition is presented. Two cases have been investigated — the use of multithreading first, to reduce the total time of training and testing (when processing the speaker database) and second, to reduce the time of a single speaker recognition (e.g. in voice-based access systems). We have investigated the processing time in function of a number of threads in order to find the optimal number for 8-core CPU. The obtained results show that the computation time can be reduced even over 5 times for this kind of processor. The prepared implementation is available on Github source code repository.","PeriodicalId":413894,"journal":{"name":"2016 Signal Processing: Algorithms, Architectures, Arrangements, and Applications (SPA)","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114306364","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 0
Performance analysis of proportionate-type LMS algorithms 比例型LMS算法的性能分析
Vinay Chakravarthi Gogineni, Subrahmanyam Mula, R. Das, M. Chakraborty
For real-time sparse systems identification applications, Proportionate-type Least Mean Square (Pt-LMS) algorithms are often preferred to their normalized counterparts (Pt-NLMS) due to lower computational complexity of the former algorithms. In this paper, we present the convergence analysis of Pt-LMS algorithms. Without any assumptions on input, both first and second order convergence analysis are carried out and new convergence bounds are obtained. In particular, it establishes the universality of the steady-state mean square deviation. Detailed simulation results are presented to validate the analytical results.
对于实时稀疏系统识别应用,比例型最小均方(Pt-LMS)算法通常比其标准化对应物(Pt-NLMS)更受青睐,因为前者算法的计算复杂度较低。本文给出了Pt-LMS算法的收敛性分析。在对输入不作任何假设的情况下,进行了一阶和二阶收敛分析,得到了新的收敛界。特别地,它建立了稳态均方差的通用性。详细的仿真结果验证了分析结果。
{"title":"Performance analysis of proportionate-type LMS algorithms","authors":"Vinay Chakravarthi Gogineni, Subrahmanyam Mula, R. Das, M. Chakraborty","doi":"10.1109/SPA.2016.7763608","DOIUrl":"https://doi.org/10.1109/SPA.2016.7763608","url":null,"abstract":"For real-time sparse systems identification applications, Proportionate-type Least Mean Square (Pt-LMS) algorithms are often preferred to their normalized counterparts (Pt-NLMS) due to lower computational complexity of the former algorithms. In this paper, we present the convergence analysis of Pt-LMS algorithms. Without any assumptions on input, both first and second order convergence analysis are carried out and new convergence bounds are obtained. In particular, it establishes the universality of the steady-state mean square deviation. Detailed simulation results are presented to validate the analytical results.","PeriodicalId":413894,"journal":{"name":"2016 Signal Processing: Algorithms, Architectures, Arrangements, and Applications (SPA)","volume":"43 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2016-09-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114576034","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 12
期刊
2016 Signal Processing: Algorithms, Architectures, Arrangements, and Applications (SPA)
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