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[电气工程与系统科学] 幅度采样的增量-斜坡编码器及其随时间的解释 编码 [推广有奖]

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摘要翻译:
传统的带限信号采集的理论基础通常依赖于均匀的时间采样,并假设无限精度的幅度值。本文在假定无限精度定时信息的情况下,研究了基于均匀幅度采样的信号表示和恢复。该方法基于增量-斜坡编码器,该编码器包括对输入信号添加适当的锯齿状波形的结果施加一个一级电平交叉检测器。输出样本是这些电平交叉的时间瞬间,因此表示输入信号的时间编码版本。为了理论上的目的,通过斜坡相加将非单调输入信号可逆地转换成单调输入信号,然后均匀地进行幅值采样,可以对该系统进行等效分析。然后,单调函数由信号通过预定义且等间隔的振幅值集合的时间表示。我们把这种技术称为振幅采样。所产生的时间序列可替换地解释为原始源信号的非均匀时间采样。我们得到了变换所涉及的函数的对偶性和频域性质。提出并实现了恢复源信号的迭代算法。仿真结果表明,对于非均匀采样,本文提出的迭代幅度采样算法比基于帧的重构算法具有更快的收敛速度。在保持采样密度不变的情况下,适当地选择参数可以提高系统的性能。
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英文标题:
《Delta-Ramp Encoder for Amplitude Sampling and its Interpretation as Time
  Encoding》
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作者:
Pablo Mart\'inez-Nuevo, Hsin-Yu Lai, Alan V. Oppenheim
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最新提交年份:
2020
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分类信息:

一级分类:Electrical Engineering and Systems Science        电气工程与系统科学
二级分类:Signal Processing        信号处理
分类描述:Theory, algorithms, performance analysis and applications of signal and data analysis, including physical modeling, processing, detection and parameter estimation, learning, mining, retrieval, and information extraction. The term "signal" includes speech, audio, sonar, radar, geophysical, physiological, (bio-) medical, image, video, and multimodal natural and man-made signals, including communication signals and data. Topics of interest include: statistical signal processing, spectral estimation and system identification; filter design, adaptive filtering / stochastic learning; (compressive) sampling, sensing, and transform-domain methods including fast algorithms; signal processing for machine learning and machine learning for signal processing applications; in-network and graph signal processing; convex and nonconvex optimization methods for signal processing applications; radar, sonar, and sensor array beamforming and direction finding; communications signal processing; low power, multi-core and system-on-chip signal processing; sensing, communication, analysis and optimization for cyber-physical systems such as power grids and the Internet of Things.
信号和数据分析的理论、算法、性能分析和应用,包括物理建模、处理、检测和参数估计、学习、挖掘、检索和信息提取。“信号”一词包括语音、音频、声纳、雷达、地球物理、生理、(生物)医学、图像、视频和多模态自然和人为信号,包括通信信号和数据。感兴趣的主题包括:统计信号处理、谱估计和系统辨识;滤波器设计;自适应滤波/随机学习;(压缩)采样、传感和变换域方法,包括快速算法;用于机器学习的信号处理和用于信号处理应用的机器学习;网络与图形信号处理;信号处理中的凸和非凸优化方法;雷达、声纳和传感器阵列波束形成和测向;通信信号处理;低功耗、多核、片上系统信号处理;信息物理系统的传感、通信、分析和优化,如电网和物联网。
--
一级分类:Computer Science        计算机科学
二级分类:Information Theory        信息论
分类描述:Covers theoretical and experimental aspects of information theory and coding. Includes material in ACM Subject Class E.4 and intersects with H.1.1.
涵盖信息论和编码的理论和实验方面。包括ACM学科类E.4中的材料,并与H.1.1有交集。
--
一级分类:Mathematics        数学
二级分类:Complex Variables        复变数
分类描述:Holomorphic functions, automorphic group actions and forms, pseudoconvexity, complex geometry, analytic spaces, analytic sheaves
全纯函数,自守群作用与形式,伪凸性,复几何,解析空间,解析束
--
一级分类:Mathematics        数学
二级分类:Information Theory        信息论
分类描述:math.IT is an alias for cs.IT. Covers theoretical and experimental aspects of information theory and coding.
它是cs.it的别名。涵盖信息论和编码的理论和实验方面。
--

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英文摘要:
  The theoretical basis for conventional acquisition of bandlimited signals typically relies on uniform time sampling and assumes infinite-precision amplitude values. In this paper, we explore signal representation and recovery based on uniform amplitude sampling with assumed infinite precision timing information. The approach is based on the delta-ramp encoder which consists of applying a one-level level-crossing detector to the result of adding an appropriate sawtooth-like waveform to the input signal. The output samples are the time instants of these level crossings, thus representing a time-encoded version of the input signal. For theoretical purposes, this system can be equivalently analyzed by reversibly transforming through ramp addition a nonmonotonic input signal into a monotonic one which is then uniformly sampled in amplitude. The monotonic function is then represented by the times at which the signal crosses a predefined and equally-spaced set of amplitude values. We refer to this technique as amplitude sampling. The time sequence generated can be interpreted alternatively as nonuniform time sampling of the original source signal. We derive duality and frequency-domain properties for the functions involved in the transformation. Iterative algorithms are proposed and implemented for recovery of the original source signal. As indicated in the simulations, the proposed iterative amplitude-sampling algorithm achieves a faster convergence rate than frame-based reconstruction for nonuniform sampling. The performance can also be improved by appropriate choice of the parameters while maintaining the same sampling density.
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PDF链接:
https://arxiv.org/pdf/1802.04672
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关键词:编码器 Applications Optimization Experimental Construction theoretical based 输入 交叉 时间

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