部分傅里叶变换在信号处理中的研究发展中英翻译
基于短时傅里叶变换的音频信号处理技术研究

基于短时傅里叶变换的音频信号处理技术研究音频信号处理技术是一门非常重要的学科,它应用广泛,主要用于改善音质,增强音乐体验,减少噪音干扰等。
而在音频信号处理技术中,短时傅里叶变换是一种常用的技术手段。
本文将介绍基于短时傅里叶变换的音频信号处理技术研究。
一、短时傅里叶变换的基本原理傅里叶变换是将时域信号转换为频域信号的一种数学变换方式。
而在实际应用中,傅里叶变换总是需要考虑到信号的长期性质,这使得其无法精确反映出一段时间信号的频域特征。
为了解决这种问题,人们提出了短时傅里叶变换(Short Time Fourier Transform,简称STFT)。
STFT是将一段时间内的信号按照一定时间间隔分割成几个小段,分别进行傅里叶变换。
通过这种方法,我们可以得到每一段时间内的频域特征,从而更加准确地反映出信号的频域性质。
二、基于STFT的音频信号处理技术基于STFT的音频信号处理技术常常用于音频降噪、语音增强、音乐合成等方面。
下面将分别从这几个方面介绍其应用。
1. 音频降噪音频降噪是一种常见的音频处理技术,它可以减少音频中噪音的干扰,提高音频的清晰度和质量。
而基于STFT的音频降噪技术就是通过识别信号中的噪音成分,并将其从频域中滤除,从而实现降噪效果。
具体来讲,我们可以通过STFT算法将整个信号分成若干个小段,然后在每个小段中分离出噪音和音频成分。
然后,我们可以设计滤波器,将噪音成分从音频中滤除。
最后,将每个小段重新组合成完整的音频信号,即可实现降噪。
2. 语音增强语音增强技术主要用于提高人们在通信、语音合成等方面的体验和效果。
而基于STFT的语音增强技术则是通过处理语音信号的频域特征,去除杂音和其他噪声成分,使得语音更加清晰、自然。
具体来说,我们可以将整个语音信号分为若干个小段,并将每个小段的频域特征进行STFT转化。
然后,根据频域特征的差异性,去除噪音成分,加强语音成分,以达到语音信号增强的目的。
最后将每个小段重新组合成完整的语音信号。
专业英语翻译之数字信号处理

Signal processingSignal processing is an area of electrical engineering and applied mathematics that deals with operations on or analysis of signals, in either discrete or continuous time, to perform useful operations on those signals. Signals of interest can include sound, images, time-varying measurement values and sensor data, for example biological data such as electrocardiograms, control system signals, telecommunication transmission signals such as radio signals, and many others. Signals are analog or digital electrical representations of time-varying or spatial-varying physical quantities. In the context of signal processing, arbitrary binary data streams and on-off signalling are not considered as signals, but only analog and digital signals that are representations of analog physical quantities.HistoryAccording to Alan V. Oppenheim and Ronald W. Schafer, the principles of signal processing can be found in the classical numerical analysis techniques of the 17th century. They further state that the "digitalization" or digital refinement of these techniques can be found in the digital control systems of the 1940s and 1950s.[2]Categories of signal processingAnalog signal processingAnalog signal processing is for signals that have not been digitized, as in classical radio, telephone, radar, and television systems. This involves linear electronic circuits such as passive filters, active filters, additive mixers, integrators and delay lines. It also involves non-linear circuits such ascompandors, multiplicators (frequency mixers and voltage-controlled amplifiers), voltage-controlled filters, voltage-controlled oscillators andphase-locked loops.Discrete time signal processingDiscrete time signal processing is for sampled signals that are considered as defined only at discrete points in time, and as such are quantized in time, but not in magnitude.Analog discrete-time signal processing is a technology based on electronic devices such as sample and hold circuits, analog time-division multiplexers, analog delay lines and analog feedback shift registers. This technology was a predecessor of digital signal processing (see below), and is still used in advanced processing of gigahertz signals.The concept of discrete-time signal processing also refers to a theoretical discipline that establishes a mathematical basis for digital signal processing, without taking quantization error into consideration.Digital signal processingDigital signal processing is for signals that have been digitized. Processing is done by general-purpose computers or by digital circuits such as ASICs, field-programmable gate arrays or specialized digital signal processors (DSP chips). Typical arithmetical operations include fixed-point and floating-point, real-valued and complex-valued, multiplication and addition. Other typical operations supported by the hardware are circular buffers and look-up tables. Examples of algorithms are the Fast Fourier transform (FFT), finite impulseresponse (FIR) filter, Infinite impulse response (IIR) filter, and adaptive filters such as the Wiener and Kalman filters1.Digital signal processingDigital signal processing (DSP) is concerned with the representation of signals by a sequence of numbers or symbols and the processing of these signals. Digital signal processing and analog signal processing are subfields of signal processing. DSP includes subfields like: audio and speech signal processing, sonar and radar signal processing, sensor array processing, spectral estimation, statistical signal processing, digital image processing, signal processing for communications, control of systems, biomedical signal processing, seismic data processing, etc.The goal of DSP is usually to measure, filter and/or compress continuousreal-world analog signals. The first step is usually to convert the signal from an analog to a digital form, by sampling it using an analog-to-digital converter (ADC), which turns the analog signal into a stream of numbers. However, often, the required output signal is another analog output signal, which requires a digital-to-analog converter (DAC). Even if this process is more complex than analog processing and has a discrete value range, the application of computational power to digital signal processing allows for many advantages over analog processing in many applications, such as error detection and correction in transmission as well as data compression.[1]DSP algorithms have long been run on standard computers, on specialized processors called digital signal processors (DSPs), or on purpose-built hardware such as application-specific integrated circuit (ASICs). Today thereare additional technologies used for digital signal processing including more powerful general purpose microprocessors, field-programmable gate arrays (FPGAs), digital signal controllers (mostly for industrial apps such as motor control), and stream processors, among others.[2]2. DSP domainsIn DSP, engineers usually study digital signals in one of the following domains: time domain (one-dimensional signals), spatial domain (multidimensional signals), frequency domain, autocorrelation domain, and wavelet domains. They choose the domain in which to process a signal by making an informed guess (or by trying different possibilities) as to which domain best represents the essential characteristics of the signal. A sequence of samples from a measuring device produces a time or spatial domain representation, whereas a discrete Fourier transform produces the frequency domain information, that is the frequency spectrum. Autocorrelation is defined as the cross-correlation of the signal with itself over varying intervals of time or space.3. Signal samplingMain article: Sampling (signal processing)With the increasing use of computers the usage of and need for digital signal processing has increased. In order to use an analog signal on a computer it must be digitized with an analog-to-digital converter. Sampling is usually carried out in two stages, discretization and quantization. In the discretization stage, the space of signals is partitioned into equivalence classes and quantization is carried out by replacing the signal with representative signal of the corresponding equivalence class. In the quantization stage the representative signal values are approximated by values from a finite set.The Nyquist–Shannon sampling theorem states that a signal can be exactly reconstructed from its samples if the sampling frequency is greater than twice the highest frequency of the signal; but requires an infinite number of samples . In practice, the sampling frequency is often significantly more than twice that required by the signal's limited bandwidth.A digital-to-analog converter is used to convert the digital signal back to analog. The use of a digital computer is a key ingredient in digital control systems. 4. Time and space domainsMain article: Time domainThe most common processing approach in the time or space domain is enhancement of the input signal through a method called filtering. Digital filtering generally consists of some linear transformation of a number of surrounding samples around the current sample of the input or output signal. There are various ways to characterize filters; for example:∙ A "linear" filter is a linear transformation of input samples; other filters are "non-linear". Linear filters satisfy the superposition condition, i.e. if an input is a weighted linear combination of different signals, the output is an equally weighted linear combination of the corresponding output signals.∙ A "causal" filter uses only previous samples of the input or output signals; while a "non-causal" filter uses future input samples. A non-causal filter can usually be changed into a causal filter by adding a delay to it.∙ A "time-invariant" filter has constant properties over time; other filters such as adaptive filters change in time.∙Some filters are "stable", others are "unstable". A stable filter produces an output that converges to a constant value with time, or remains bounded within a finite interval. An unstable filter can produce an output that grows without bounds, with bounded or even zero input.∙ A "finite impulse response" (FIR) filter uses only the input signals, while an "infinite impulse response" filter (IIR) uses both the input signal and previous samples ofthe output signal. FIR filters are always stable, while IIR filters may be unstable.Filters can be represented by block diagrams which can then be used to derive a sample processing algorithm to implement the filter using hardware instructions. A filter may also be described as a difference equation, a collection of zeroes and poles or, if it is an FIR filter, an impulse response or step response.The output of a digital filter to any given input may be calculated by convolving the input signal with the impulse response.5. Frequency domainMain article: Frequency domainSignals are converted from time or space domain to the frequency domain usually through the Fourier transform. The Fourier transform converts the signal information to a magnitude and phase component of each frequency. Often the Fourier transform is converted to the power spectrum, which is the magnitude of each frequency component squared.The most common purpose for analysis of signals in the frequency domain is analysis of signal properties. The engineer can study the spectrum todetermine which frequencies are present in the input signal and which are missing.In addition to frequency information, phase information is often needed. This can be obtained from the Fourier transform. With some applications, how the phase varies with frequency can be a significant consideration.Filtering, particularly in non-realtime work can also be achieved by converting to the frequency domain, applying the filter and then converting back to the time domain. This is a fast, O(n log n) operation, and can give essentially any filter shape including excellent approximations to brickwall filters.There are some commonly used frequency domain transformations. For example, the cepstrum converts a signal to the frequency domain through Fourier transform, takes the logarithm, then applies another Fourier transform. This emphasizes the frequency components with smaller magnitude while retaining the order of magnitudes of frequency components.Frequency domain analysis is also called spectrum- or spectral analysis. 6. Z-domain analysisWhereas analog filters are usually analysed on the s-plane; digital filters are analysed on the z-plane or z-domain in terms of z-transforms.Most filters can be described in Z-domain (a complex number superset of the frequency domain) by their transfer functions. A filter may be analysed in the z-domain by its characteristic collection of zeroes and poles.7. ApplicationsThe main applications of DSP are audio signal processing, audio compression, digital image processing, video compression, speech processing, speech recognition, digital communications, RADAR, SONAR, seismology, and biomedicine. Specific examples are speech compression and transmission in digital mobile phones, room matching equalization of sound in Hifi and sound reinforcement applications, weather forecasting, economic forecasting, seismic data processing, analysis and control of industrial processes, computer-generated animations in movies, medical imaging such as CAT scans and MRI, MP3 compression, image manipulation, high fidelity loudspeaker crossovers and equalization, and audio effects for use with electric guitar amplifiers8. ImplementationDigital signal processing is often implemented using specialised microprocessors such as the DSP56000, the TMS320, or the SHARC. These often process data using fixed-point arithmetic, although some versions are available which use floating point arithmetic and are more powerful. For faster applications FPGAs[3] might be used. Beginning in 2007, multicore implementations of DSPs have started to emerge from companies including Freescale and Stream Processors, Inc. For faster applications with vast usage, ASICs might be designed specifically. For slow applications, a traditional slower processor such as a microcontroller may be adequate. Also a growing number of DSP applications are now being implemented on Embedded Systems using powerful PCs with a Multi-core processor.(翻译)信号处理信号处理是电气工程与应用数学领域,在离散的或连续时间域处理和分析信号,以对这些信号进行所需的有用的处理。
离散傅里叶变换时移-概述说明以及解释

离散傅里叶变换时移-概述说明以及解释1.引言1.1 概述离散傅里叶变换(Discrete Fourier Transform,简称DFT)是一种将一个离散信号(或称时域信号)转换为频域表示的数学工具。
在现代信号处理和通信领域中,DFT被广泛应用于信号分析、滤波、频谱估计等领域。
DFT的概念源于傅里叶分析,它是将一个连续时间函数表示为一组基函数乘以一系列复数系数的线性组合。
而离散傅里叶变换则是将这一思想应用于离散信号,将离散时间序列转换为离散频率表示。
通过使用离散傅里叶变换,我们可以将一个时域上的离散信号转换为频域上的频谱表示,从而可以更加直观地观察信号的频率成分和能量分布。
离散傅里叶变换的时移性质是指当输入信号在时域上发生时移时,其在频域上的表示也随之发生相应的时移。
这一性质使得我们可以通过时移操作对信号进行处理和分析。
具体来说,如果我们对一个信号进行时移操作,即将信号中的每个样本向前或向后平移若干个位置,那么该信号在频域上的表示也会相应地发生同样的平移。
在本文中,我们将着重讨论离散傅里叶变换时移的原理和性质。
我们将介绍离散傅里叶变换的基本概念和原理,包括如何进行DFT变换、如何计算DFT系数以及DFT的逆变换等。
然后,我们将详细解释离散傅里叶变换的时移性质,包括时域上的时移操作如何在频域上体现以及时域和频域之间的变换关系等。
通过对离散傅里叶变换时移性质的研究,我们可以更好地理解信号在时域和频域之间的关系,以及对信号进行时移操作的影响。
同时,我们还将探讨离散傅里叶变换时移的应用,包括在信号处理、通信系统和图像处理等领域中的具体应用案例。
通过这些应用案例,我们将展示离散傅里叶变换时移的重要性以及它在实际问题中的实用价值。
1.2 文章结构文章结构部分的内容:本文主要分为三个部分:引言、正文和结论。
在引言部分,首先概述了离散傅里叶变换时移的主题,介绍了离散傅里叶变换的基本概念和原理。
接着,详细说明了本文的结构,即按照离散傅里叶变换时移的相关性质展开论述。
信号处理中的频谱分析方法比较研究

信号处理中的频谱分析方法比较研究概述频谱分析是信号处理领域中常用的一种技术,用于研究信号的频率和幅度特征。
在实际应用中,有多种频谱分析方法可供选择。
本文将比较几种常见的频谱分析方法,包括傅里叶变换(FFT)、短时傅里叶变换(STFT)、Gabor变换和小波变换。
将分析各个方法的原理、优缺点及适用场景,旨在为信号处理研究者和工程师提供选择合适方法的指导。
傅里叶变换(FFT)傅里叶变换是信号处理中最常用的频谱分析方法之一。
它将信号表示为不同频率的正弦和余弦波的叠加,通过在频域提取信号的频率分量。
优点是简单易懂且计算效率高,适用于稳态信号。
但是,傅里叶变换需要处理整个信号,对于非稳态信号和瞬态信号可能无法提供准确的频谱分析结果。
短时傅里叶变换(STFT)为了克服傅里叶变换的不足,短时傅里叶变换(STFT)应运而生。
STFT将信号分成多个短时片段,并对每个片段进行傅里叶变换,从而获得信号在时间和频率上的局部特征。
这使得STFT适用于非稳态信号和时变信号的频谱分析。
然而,STFT的时间和频率分辨率之间存在一个折衷关系,高频率分辨率意味着低时间分辨率,反之亦然。
Gabor变换Gabor变换是一种时间-频率分析方法,它结合了傅里叶变换和瞬态分析。
Gabor变换通过使用窗函数在时间域上局限信号,然后通过傅里叶变换获得频域特性,从而提供了更好的时间和频率分辨率。
Gabor变换适用于非稳态信号和时变信号,具有较好的谱线分离能力,但计算复杂度较高。
小波变换小波变换是一种非平稳信号分析的有效工具。
与傅里叶变换和短时傅里叶变换相比,小波变换可以提供更好的时频局部化特性。
小波变换使用不同的基函数进行多尺度分解,将信号分解为各个频带,并提供不同频率和时间分辨率的频谱信息。
小波变换适用于非稳态信号、时变信号和具有突变特性的信号。
方法比较和适用场景综上所述,不同的频谱分析方法在时间和频率分辨率、计算复杂度、局部化能力等方面有所差异。
ads 傅里叶变换-概述说明以及解释

ads 傅里叶变换-概述说明以及解释1.引言概述是文章中引言部分的第一个小节,它主要用于介绍和概括整个文章的主题和背景。
在本篇长文中,概述部分的目标是为读者提供关于ADS (傅里叶变换)的基本概念和其在实际应用中的重要性的概览。
以下是概述部分的内容:1.1 概述ADS(Advanced Design System)是一种电子设计自动化软件,它在电子电路设计和分析中扮演着关键的角色。
ADS基于傅里叶变换原理,通过将时域信号转换为频域信号,将复杂的电路分析问题转化为更容易解决的频域分析问题。
傅里叶变换是一种数学工具,用于将一个函数表达式分解成一系列正弦和余弦函数的和。
这种变换能够将信号从时间域转换为频域,揭示出信号中包含的不同频率的成分,从而为电子电路的设计和分析提供了重要的参考依据。
本文将详细介绍傅里叶变换的概念和原理,并探讨其在ADS中的具体应用。
首先,我们将对傅里叶变换的基本概念进行解释,包括正向傅里叶变换和逆向傅里叶变换的定义和数学推导。
接着,我们将深入探讨傅里叶变换在电子电路设计和分析中的应用,包括滤波器设计、频率响应分析等方面。
通过这些实际案例,我们将展示ADS作为一种强大的分析工具,如何利用傅里叶变换帮助工程师们更好地设计和优化复杂的电子电路。
总之,本文旨在为读者介绍傅里叶变换在ADS中的应用以及其在电子电路设计和分析中的重要性。
通过深入理解傅里叶变换的原理和应用,我们可以更好地利用ADS这一工具,在电子领域取得更好的设计和分析效果。
接下来,我们将会详细探究傅里叶变换的概念和其在电子电路中的实际应用,以期展望傅里叶变换的未来发展。
1.2文章结构文章结构部分内容如下:1.2 文章结构本文将按照以下结构进行叙述:第一部分为引言,包括概述、文章结构和目的。
在这部分中,将介绍对于ADS(傅里叶变换)这一主题的基本了解,以及文章的整体结构和分析目的。
第二部分是正文,分为傅里叶变换的概念和傅里叶变换的应用两个部分。
信号处理中傅里叶变换简介

傅里叶变换一、傅里叶变换的表述在数学上,对任意函数f(x),可按某一点进行展开,常见的有泰勒展开和傅里叶展开.泰勒展开为各阶次幂函数的线性组合形式,本质上自变量未改变,仍为x,而傅里叶展开则为三角函数的线性组合形式,同时将自变量由x变成ω,且由于三角函数处理比较简单,具有良好的性质,故被广泛地应用在信号分析与处理中,可将时域分析变换到频域进行分析。
信号分析与处理中常见的有CFS(连续时间傅里叶级数)、CFT (连续时间傅里叶变换)、DTFT(离散时间傅里叶变换)、DFS(离散傅里叶级数)、DFT(离散傅里叶变换)。
通过对连续非周期信号x c(t)在时域和频域进行各种处理变换,可推导出以上几种变换,同时可得出这些变换之间的关系。
以下将对上述变换进行简述,同时分析它们之间的关系。
1、CFS(连续时间傅里叶级数)在数学中,周期函数f(x)可展开为由此类比,已知连续周期信号x(t),周期为T0,则其傅里叶级数为其中,为了简写,有其中,为了与复数形式联系,先由欧拉公式e j z=cos z+jsin z得故有令则对于D n,有n≤0时同理.故CFS图示如下:Figure 错误!未定义书签。
理论上,CFS对于周期性信号x(t)在任意处展开都可以做到无误差,只要保证n从-∞取到+∞就可以。
在实践中,只要n取值范围足够大,就可以保证在某一点附近对x(t)展开都有很高的精度。
2、CFT(连续时间傅里叶变换)连续非周期信号x(t),可以将其看成一连续周期信号的周期T0→∞。
当然,从时域上也可以反过来看成x(t)的周期延拓。
将x(t)进行CFS展开,有若令则有T0→∞使得Ω0→0,则由此,定义傅里叶变换与其逆变换如下CFT:CFT-1:x(t)是信号的时域表现形式,X(jΩ)是信号的频域表现形式,二者本质上是统一的,相互间可以转换。
CFT即将x(t)分解,并按频率顺序展开,使其成为频率的函数。
上式中,时域自变量t的单位为秒(s),频域自变量Ω的单位为弧度/秒(rad/s).CFS中的D n与CFT中的X(jΩ)之间有如下关系即从频域上分析,D n是对X(jΩ)的采样(可将Figure 1与Figure 2进行对比).CFT图示如下:Figure 错误!未定义书签。
稀疏表示的字典_文献翻译.docx

从统计学的观点来看, 这个过程把数据当作服从低维高斯分布来建模,因此对于 高斯数据最有效。 与傅里叶变换相比,KLT 在表示效率上更优。然而,这个优势是用非结构性 和明显更复杂的转换换来的。 我们将会看到,这种在效率与自适应性之间的折衷 在现代字典设计方法学中仍扮演着重要的角色。 B. 非线性变革与现代字典设计元素 19 世纪 80 年代,统计学的研究领域出现的新的有力方法,即稳健统计。稳 健统计提倡将稀疏作为大范围的复原与分析任务的关键。 这种理念来源于经典物 理学,发展于近年的信息论,在指导现象描述上提升了简易性与简明性。在这种 理念的影响下,80 年代与 90 年代以搜寻更稀疏的表示和更高效的变换为特征。 增强稀疏性需要偏离线性模式,朝更灵活的非线性规划发展。在非线性的实 例中,每个信号都可以使用同一字典中一组不同的原子,以此实现最佳近似。因 此,近似过程变为
其中������������ (������)是分别适用于每个信号的索引集。 非线性观点为设计更新,更有效的变换铺平了道路。在这个过程中,许多指
导现代字典设计的基本概念形成了。我们将沿着历史的时间线,回溯许多最重要 的现代字典设计概念的出现。大部分概念是在 20 世纪的最后 20 年间形成的。
定位:为了实现稀疏性,变换需要更好的定位。受到集中支撑的原子能基于
其中w(∙)是一个定位在 0 处的低通窗口函数, 且α 和β 控制变换的时间和频率分 解。这种变换的很多数学基础都由 Daubechies,Grossman 和 Meyer 在 19 世纪 80 年代提出。他们从框架理论的角度研究该变换。Feichtinger 和 Grochenig 也是 Gabor 变换数学基础的建立者,他们提出了广义的群理论观点。离散形式变 换的研究及其数值实现紧接着在 19 世纪 90 年代早期开始进行。Wexler,Raz, Qian 和 Chen 对该研究做出了重要贡献。 在更高的维度下, 更复杂的 Gabor 结构被研究出来。这些结构通过改变正弦 波的朝向增加了方向性。 这种结构在 Daugman 的工作中得到了大力支持。他在视 觉皮质的简单细胞接受域中发现了方向性的类 Gabor 模式。 这些结果在 Daugman, Porat 和 Zeevi 的工作的引导下促进了图像处理任务中变换的调度。 现在, Gabor 变换的实际应用主要在于分析和探测方面,表现为一些方向滤波器的集合。
Digital-Signal-Processing数字信号处理大学毕业论文英文文献翻译及原文

毕业设计(论文)外文文献翻译文献、资料中文题目:数字信号处理文献、资料英文题目:Digital Signal Processing 文献、资料来源:文献、资料发表(出版)日期:院(部):专业:班级:姓名:学号:指导教师:翻译日期: 2017.02.14数字信号处理一、导论数字信号处理(DSP)是由一系列的数字或符号来表示这些信号的处理的过程的。
数字信号处理与模拟信号处理属于信号处理领域。
DSP包括子域的音频和语音信号处理,雷达和声纳信号处理,传感器阵列处理,谱估计,统计信号处理,数字图像处理,通信信号处理,生物医学信号处理,地震数据处理等。
由于DSP的目标通常是对连续的真实世界的模拟信号进行测量或滤波,第一步通常是通过使用一个模拟到数字的转换器将信号从模拟信号转化到数字信号。
通常,所需的输出信号却是一个模拟输出信号,因此这就需要一个数字到模拟的转换器。
即使这个过程比模拟处理更复杂的和而且具有离散值,由于数字信号处理的错误检测和校正不易受噪声影响,它的稳定性使得它优于许多模拟信号处理的应用(虽然不是全部)。
DSP算法一直是运行在标准的计算机,被称为数字信号处理器(DSP)的专用处理器或在专用硬件如特殊应用集成电路(ASIC)。
目前有用于数字信号处理的附加技术包括更强大的通用微处理器,现场可编程门阵列(FPGA),数字信号控制器(大多为工业应用,如电机控制)和流处理器和其他相关技术。
在数字信号处理过程中,工程师通常研究数字信号的以下领域:时间域(一维信号),空间域(多维信号),频率域,域和小波域的自相关。
他们选择在哪个领域过程中的一个信号,做一个明智的猜测(或通过尝试不同的可能性)作为该域的最佳代表的信号的本质特征。
从测量装置对样品序列产生一个时间或空间域表示,而离散傅立叶变换产生的频谱的频率域信息。
自相关的定义是互相关的信号本身在不同时间间隔的时间或空间的相关情况。
二、信号采样随着计算机的应用越来越多地使用,数字信号处理的需要也增加了。
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毕业设计(论文)外文资料翻译系别:电子信息系专业:通信工程班级:B090310姓名:孙春甫学号:B09031015外文出处:知网附件: 1. 原文; 2. 译文2013年05月Research Progress of the Fractional Fourier Transformin Signal ProcessingABSTRACTThe fractional Fourier transform is a generalization of the classical Fourier transform, which is introduced from the mathematic aspect by Namias at first and has many applications in optics quickly. Whereas its potential appears to have remained largely unknown to the signal processing community until 1990s. The fractional Fourier transform can be viewed as the chirp-basis expansion directly from its definition, but essentially it can be interpreted as a rotation in the time-frequency plane, i.e. the unified time-frequency transform. With the order from 0 increasing to 1, the fractional Fourier transform can show the characteristics of the signal changing from the time domain to the frequency domain. In this research paper, the fractional Fourier transform has been comprehensively and systematically treated from the signal processing point of view. Our aim is to provide a course from the definition to the applications of the fractional Fourier transform, especially as a reference and an introduction for researchers and interested readers.While solving a heat conduction problem in 1807, a French scientist Jean Baptiste Joseph Fourier, suggested the usage of the Fourier theorem. Thereafter, the Fourier transform (FT) has been applied widely in many scientific disciplines, and has played important role in almost all the science and technology domains. However, with the extension of research objects and scope, the FT has been discovered to have shortcomings. Since the FT is a kind of holistic transform, i.e., through which the whole spectrum is obtained, it cannot obtain the local time-frequency character that is essential and pivotal for processing nonstationary signals. So a series of novel signal analysis theories have been put forward to process nonstationary signals, such as: the fractional Fourier transform, the short-time Fourier transform, Wigner-Ville distribution, Gabor transform, wavelet transform, cyclic statistics, AM/FM signal analysis and so on. Hereinto the fractional Fourier transform (FRFT), as a generalization of the classical FT, has caught more and more attention for its inherentpeculiarities. In the last decade, research into the FRFT theory and application was fruitful, resulting in an upsurge in the study of the FRFT.In 1980, Namias introduced the FRFT as a way to solve certain classes of ordinary and partial differential equations arising in quantum mechanics from classical quadratic Hamiltonians. His results were later refined by McBride and Kerr. They developed an operational calculus to define the FRFT which was the base for the optical version of the FRFT. In 1993, Mendlovic and Ozaktas offered the optical realization of the FRFT to process the optical signal, which was easy to be realized with some optical instruments. So the FRFT has many applications in optics. Although the FRFT may be potentially useful, it appears to have remained largely unknown to the signal processing community for the lack of physical illumination and fast digital computation algorithm until the interpretation as a rotation in the time-frequency plane and the efficient digital computation algorithm of the FRFT emerged in 1993 and 1996 respectively. Thereafter, many relevant research papers have been published. The study of the FRFT did not start too late at home, but still stayed at the immature stage in view of the number and content of the relevant papers. In early 1996, some review papers about the FRFT appeared at home, yet the potential of the FRFT was just explored then. What is more, no review paper of the FRFT from the aspect of signal processing has been published overseas so far. So this paper tries to summarize the research progress of the FRFT in signal processing, and expatiate the theoretic system of the FRFT in the foundation, application-foundation and application fields to provide the reference to relevant researchers.The organization of this paper is as follows: we first provide the definition of the FRFT and its meaning. The properties and the relation between the FRFT and the conventional time-frequency distribution are depicted in section 2, as well as the uncertainty principle in the FRFT domain. We consider the FRFT domain to be interpreted as the unified time-frequency transform domain. In section 3, we systematically summarize some signal analysis tools based on the FRFT. We summarize the applications of the FRFT in signal processing in section 4. Finally, this paper is concluded in section 5.1 Definition of the FRFTThe FRFT is defined as:()[]()()(),p p p X u F x u x t K t u dt +∞-∞==⎰, (1)where ()()()()()22cot 2csc cot 1cot ,,(),2,21j t ut u p j e n K t u t u n t u n παααααπδαπδαπ-+⎧-≠⎪⎪=-=⎨⎪+=±⎪⎩(2) where /2p απ= indicates the rotation angle of the transformed signal for FRFT, p is the transform order of the FRFT, and the FRFT operator is designated by p F . It is obvious that the FRFT is periodic with period 4. If and only if 41p n =+ ()2/2n αππ=+, then the FRFT is just the same as the FT. Let /2u u π= and /2t t π= . Then eq. (1) is equivalent to()[]()()()()()()22cot cot csc 221cot ,2,2,21u t j j jut p p j e x t e dt n X u F x u x u n x u n αααααππαπαπ+∞--∞⎧-≠⎪⎪⎪==⎨⎪-=+⎪⎪⎩⎰ (3) eq. (3) shows that the computation of the FRFT corresponds to the following three steps:a. a product by a chirp, ()()()2cot 21cot t j g t j e x t αα=-;b. a FT (with its argument scaled by csc α),()()ˆcsc p X u G u α= with ()()12jut G u g t e dt π+∞--∞=⎰c. another product by a chirp, ()()2cot 2ˆu j p pX u e X u α= It turns up that the FRFT of ()x t exists in the same conditions in which its FT exists; in other words, if ()X ω exits, ()p X u exits too. Using the computation steps above obtained the unified sampling theorem for the FRFT. Based on chirp-periodicity Erseghe et al.[11] generalized the character of the FT (continuous-time, periodic continuous-time, discrete-time, periodic discrete-time) to four corresponding versions of the FRFT, and deduced the unified sampling theorem for the FRFT.The FRFT can be considered as a decomposition of the signal, for the inverse FRFT is defined as()()()(),p p p p x t F X t X u K t u du +∞---∞⎡⎤==⎣⎦⎰ (4)where ()x t is expressed by a class of orthonormal basis function (),p K t u - with weight factors ()p X u . The basis functions are complex exponentials with linear frequency modulation (LFM). For different values of u , they only differ by a time shift and by a phase factor that depends on u :()()2tan 2,sec ,0u j p p K t u e K t u αα-=- (5)2 Properties of the fractional Fourier transform2.1 Basic propertiesThe FRFT is a generalization of the FT, so most of the properties of the FT have their corresponding generalization versions of the FRFT. The basic properties of the FRFT are listed in the appendix. An important property, convolution theorem of the FRFT, has not been listed in the appendix, for it is not obtained simply. Interested readers may refer to refs. Another important property will be introduced that the FRFT can be interpreted as a rotation in the time-frequency plane with angle α. The property establishes the direct relationship between the FRFT and the time-frequency distribution, and founds the theory that the FRFT domain can be interpreted as a uniform time-frequency domain, which offers the FRFT the advantage to be used in signal processing. With the Wigner distribution as the example, let R φ denote the operator to rotate a 2-D function clockwise:[]()(),cos sin sin cos R y t y t t φωφωφφωφ=+-+ (6)Then the relationship is as follows:()[](),,x u W t R W t αωω= (7)where()*,22jw u p p W t X t X t e d τττωτ+∞--∞⎛⎫⎛⎫=+- ⎪ ⎪⎝⎭⎝⎭⎰ ()*,22jw x W t x t x t e d τττωτ+∞--∞⎛⎫⎛⎫=+- ⎪ ⎪⎝⎭⎝⎭⎰ express the Wigner distribution of ()p X u , ()x t respectively. Such relations still remain available for the ambiguity function, the modified short-time Fouriertransform and the spectrogram. Lohmann generalized eq. (7), and obtained the relationship between the FRFT and Radon-Wigner transform:[]()()2x p W u X u αℜ= (8) where αℜ is the operator of the Radon Transform, expressing the integral projection of a 2-D function with angle /2p απ= to axis t. eq. (8) can also be understood as marginal integral after a rotation of the reference frame with angle α, namely:()()2cos sin ,sin cos x p W u v u v dv X u ϕϕϕϕ+∞-∞-+=⎰ (9)Since the FRFT has such relationship with conventional time-frequency distributions, we want to know whether a more general expression exists. Let()()(),,,x xt f t f W d d τθξψτθτθτθ=--⎰⎰ (10) where (),t f ψ is the transform kernel, (),x W τθ and (),t f ξare the Wigner distribution and the Cohen class of time-frequency distribution of ()x t respectively. Only if the transform kernel (),t f ψ is rotationally symmetric around the origin, then (),p X t f ξ the time-frequency distribution of the FRFT of ()x t is a rotatedversion of the time-frequency distribution of ()x t , (),p X t f ξ. Thus, the FRFTcorresponds to rotation of a relatively large class of time-frequency representations.From the relationship between the FRFT and the time-frequency distributions mentioned above, we see that the FRFT offers an integrative description of the signal from the time domain to the frequency domain. The FRFT can provide more space for time-frequency analysis of signals.2.2 Uncertainty principleSince the FRFT domain is a unified time-frequency transform domain, what is the generalization of the conventional uncertain principle in the FRFT domain? Using the conventional uncertain principle and the three decomposition steps of the FRFT mentioned in section 1, we can obtain the uncertain principle between the two FRFT domains with different transform orders.3 Fractional operator and transformBecause the FRFT is a united time-frequency analysis tool, and can be interpreted as a rotation in the time-frequency plane, we can define some useful fractional operators and transforms based on the FRFT.3.1 Fractional operatorsConvolution and correlation are the two kinds of signal processing operators in common use. The fractional convolution and fractional correlation operator are defined in the time domain and transform domain respectively adapted to signal detection and parameter estimation; adapted to filter design, beam forming and pattern recognition.In the time-frequency analysis theory, the unitary operator and hermitian operator are two important operators. Unitarity is one of the factors needed to consider in designing a transform operator. And different transform domains usually can be related by some hermitian operators. Thus, it attracts the people’s strong interest to deduce the unitary and hermitian fractional operator. Based on the concept of time-shift operator and frequency-shift operator, which are two basic unitary operators, Akay defined the fractional-shift operator ,T φτ, namely unitary fractional operator, shown in (11).[]()()2cos sin 2sin ,cos j j t T x t x t e πτφφπτφφττφ-+=- (11)3.2 Fractional transformThe fractional transforms introduced in this section means some signal analysis tools based on the FRFT, which mainly contains two classes: one is some corresponding generalizations of conventional signal analysis tools based on the FT making use of the fact that the FRFT is the generalization of the FT; the other is some new time-frequency analysis tools based on the time-frequency rotation property of the FRFT. Then we make the summary of the main fractional transforms, and elaborate on their characteristics and advantages respectively.Some corresponding generalizations. Hilbert transform is an important signal processing tool that has many applications in communication modulation, image edge detection and so on. We can obtain the fractional Hilbert transform by generalizing the transfer function of the Hilbert transform from the frequency domain into the FRFT domain:[]()()p Hil p p p x t F X H t -⎡⎤Γ=⋅⎣⎦ (12)The essential of the fractional Hilbert transform is still to suppress the negative portion of the ‘spectrum’, similar to the conventional Hilbert transform. Thedifference lies in the ‘spectrum’, which is not the FT but the FRFT of a signal. Based on this definition, obtained a discrete version of the fractional Hilbert transform using eigenvector decomposition-type discrete FRFT, and did some digital image edge detection simulations. The design and application of the fractional Hilbert transformer has been further investigated, and several design methods about the FIR, IIR Hilbert transformer are presented, as well as a secure single-sideband (SSB) communication system with the transform order of the FRFT as a secrete key for demodulation.Sine transform, cosine transform and Hartley transform all belong to the unitary transform, and have already widely been applied in image compression and adaptive filter. Making use of the relationship between them and the FT, we can obtain the fractional sine, cosine, and Hartley transforms. Note: firstly, the fractional sine, cosine, and Hartley transform are all with a period of 2, different from the FRFT with a period of 4; secondly, the fractional sine transform has no even eigenfunctions, and the fractional cosine transform has no odd eigenfunctions. Therefore, it is better to use the fractional cosine transform to process even functions and use the fractional sine transform to process odd functions. Based on the relationship between the FRFT and Radon-Wigner transform shown in (8), it is easy to find that the invert Radon transform of the FRFT may be an available time-frequency analysis tool. According to this clue proposes a new time-frequency analysis method called the tomography time-frequency transform (TTFT), and reduces the cross-terms through the adaptive filter in the FRFT domain.The adaptive signal expansion is a signal analysis method based on the expanding signal on a group of elementary functions that are energy-limited and fit for analyzing the time-frequency structure. This time-frequency distribution related with adaptive signal expansion is of better time-frequency resolution and free from window effect and cross-term interference.proposes a new signal expansion method based on the FRFT of Gaussian functions as the elementary functions for the reason that the Gaussian functions satisfy the boundary condition of the uncertainty principle. With the application of the FRFT, the selection of elementary function becomes more flexible through changing the transform order of the FRFT, which may result in more precise time-frequency representation of a signal.4 Applications in signal processingThe FRFT is a generalization of the classical FT, and processes signals in theunified time-frequency domain. Compared with the FT, the FRFT is more flexible and suitable for processing non stationary signals. What is more, the fast algorithm of the discrete FRFT has been proposed. Thus, the FRFT has found many applications in signal processing.4.1 Signal detection and parameter estimationBecause the FRFT can be considered as a decomposition of the signal in terms of chirps, it is suitable for the processing of chirp-like signals. Based on the property of the concentration of a chirp energy resulting in a peak in a certain FRFT domain, we can carry out detection and parameter estimation of chirps accurately through searching the peak in the 2-D distribution plane vs. the FRFT domain and the transform order. Using this clue presents a new method for the detection and parameter estimation of multi component linear frequency modulation (LFM) signals. In order to increase the search efficiency and reduce the interference between these components, the Quasi-Newton method and peak mask in cascade are introduced. Error analysis and simulations show that this parameter estimation method is asymptotically unbiased and efficient.4.2 Phase retrieval and signal reconstructionA complex signal can be completely reconstructed (except for a constant phase shift) through phase retrieval from the magnitudes of two of its FRFT ()p X u σ+ and ()p X u σ-. The reason for the exception of a constant term is that the fractional power spectrums square of magnitude of the FRFT, of two functions with only the exception of a constant phase are the same. Currently, the iteration-type and Noniteration-type methods are the two main kinds of phase retrieval methods. The Noniteration-type method retrieves the phase through finding the instantaneous frequency in the FRFT domain based on the relationship between the FRFT and time-frequency distributions.4.3 Applications in image processingThe application of the FRFT in image processing includes digital watermark and image encryption. After the image is processed through 2-D FRFT, the watermark is embedded in the selected transform coefficients in terms of certain rules. Compromises are needed to make to determine the detection threshold and the transform coefficients for embedding the watermark, respectively. For the former, a trade-off is needed between holding robust and avoiding image deformation; for thelatter, between watermark imperceptiveness and probability of false detection (false alarm). In brief, applying the FRFT in image encryption is to execute encryption through multiplying the 2-D FRFT of the original image by a phase key. Decryption is the inverse of encryption, namely, first multiplying by the conjugation of the phase key to erase this key and then recover the original image by the corresponding inverse 2-D FRFT. Encryption based on the FRFT takes better effect than based on the FT or cosine transform due to one extra degree of freedom.4.4 Applications in radar, sonar, and communicationIn addition to beamforming and object recognition, the FRFT has many other applications in radar, sonar, and communication.With the development of array antenna technology, the array signal processing based on the FRFT has attracted increasing attention. The proposed approach first separates LFM signals in the FRFT domain by using the energy-concentration property of the LFM signal in a certain FRFT domain, and constructs the correlation matrix of the sensor array signals in the FRFT domain. Through estimating the signal and noise subspaces with the eigendecomposition of the correlation matrix, the MUSIC algorithm is used to estimate the DOAs of LFM signals. Simulation results show that the proposed method can give the precise DOA estimation of wide-band LFM signals, and has great performance even when SNR is very low. Whereas this method is for noncoherent LFM signals, the DOA estimation problem still needs further study to settle for coherent LFM signals.As we all know, the resonance could be excitated when the wavelength of illumination frequency is approximately the same dimensions as the overall length of the object, and can be used to detect and identify the object accurately. Whereas the resonances have a turn-on time, which implies that they evolve only after certain time duration, most previous techniques have used late time signals only.5 ConclusionsThis paper summarizes the research progress of the FRFT in signal processing, and systematically expatiates the theoretic system of the FRFT in the foundation, application-foundation and application fields. The relationship between the FRFT domain and time domain, frequency domain shows clearly that the FRFT is actually a unified time-frequency transform, which reflects the characteristics of a signal in thetime-frequency domain. Unlike usual quadratic time-frequency distributions, it reveals the time-frequency characteristics with a single variable, and does not suffer from cross-terms. Compared with the traditional FT (in fact it is a special condition of the FRFT), the FRFT does better in nonstationary signals processing especially in the chirp-like signals processing. Moreover, one extra degree of freedom (the order p ) may sometimes help to obtain better performance than the usual time-frequency distributions or the FT. And its developed fast algorithms lead to little computation load for good performance. Judging from sections 3 and 5, there are six main applications of the FRFT in signal processing nowadays, which embody the six advantages of the FRFT:(1) The FRFT is a unified time-frequency transform. With the order from 0 increasing to 1, the FRFT can reveal the characteristic of the signal gradually changing from the time domain to the frequency domain. As a result, the FRFT can provide more space for time-frequency analysis of signals. The direct utilizing mode of the FRFT is the generalization of the applications in the time, frequency domain to the FRFT domain looking for improvement to some extent, e.g. filtering in the FRFT domain.(2) The FRFT can be considered as a decomposition of a signal in terms of chirps, thus it is fit for processing chirp-like signals which widely exist in radar, communication, sonar and nature.(3) The pth FRFT can be interpreted as a rotation in the time-frequency plane with angle /2p απ= It is easy to derive the relation between the FRFT and time-frequency transforms, which can be used in instantaneous frequency estimating, phase retrieval or designing new time-frequency transform such as TTFT, signal expansion with the FRFT of Gaussian functions as the elementary functions.(4) Compared with the FT, one extra degree of freedom exits in the FRFT, which helps to obtain better performance in some applications such as digital watermarking and image encryption.(5) The FRFT is a linear transform without cross-term interference, and is ascendant in the multicomponent signal processing with additive noise.(6) The fast algorithms of the FRFT are relatively developed now, which assures that the FRFT is able to be applied in the real-time digital signal processing. And other fractional transform may develop each fast algorithms based on the FRFT, e.g. fractional convolution, fractional correlation, fractional Hartley transform, and so on.So far many research results about the FRFT have been obtained, but there still remain many theoretical problems to be settled. For example, the sampling theory in the FRFT domain depicted is only about uniform sampling, and yet ununiform sampling is sometimes inevitable in the real sampling case. For another example, the optimal order must be determined in many applications of the FRFT, but there is no effective method at present yet, and the method based on the location of minimum second-order moment of a signal’s FRFT in p axis has its limitation. Therefore, several directions need further study as follows: Improvement of the existing methods, such as determining the optimal order, better fast algorithm, analysis of the window of the STFRFT, further exploration of applications of the FRFT, and so on; combining the FRFT with multi-rate digital signal processing to constitute the system of multi-rate theory in the FRFT domain, which can reinforce the advantage of the FRFT. proposes an approach to increase the efficiency of the discrete FRFT computation based on polyphase and equivalent transform in the multi-rate theory; and generalization of the theory of the FRFT into the theory of the linear canonical transform (LCT). Like the relationship between the FRFT and the FT, the LCT is the generalization of the FRFT. The LCT has three degrees of freedom, so it is more flexible compared with the FRFT and the FT.部分傅里叶变换在信号处理中的研究发展摘要部分傅里叶变换是广义的经典的傅立叶变换,这是纳米亚首先从数学的方面有很多的应用涉及光学快速发展。