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。
fourier transform的原理
fourier transform的原理Fourier Transform的原理Fourier Transform(傅里叶变换)是一种数学工具,用于将一个函数或信号从时间域转换到频率域。
它是由法国数学家Jean-Baptiste Joseph Fourier 在19世纪提出的。
傅里叶变换在信号处理、图像处理、通信等领域都有广泛的应用。
傅里叶级数在介绍傅里叶变换之前,我们首先了解一下傅里叶级数。
傅里叶级数是傅里叶变换的基础,用于将周期性函数表示为一系列正弦和余弦函数的和。
傅里叶级数的公式如下:f(x)=a0+∑[a n cos(2πnxT)+b n sin(2πnxT)]∞n=1其中,a n和b n是函数f(x)的傅里叶系数,T是函数f(x)的周期。
连续傅里叶变换傅里叶级数适用于周期性函数,但对于非周期性函数,我们需要使用连续傅里叶变换。
连续傅里叶变换将一个非周期性函数f(t)转换为一个连续的频谱F(ω),其公式如下:F(ω)=∫f∞−∞(t)e−iωt dt连续傅里叶变换将时域信号转换为频域信号,其中ω表示角频率。
离散傅里叶变换在实际应用中,我们通常处理的是离散的数字信号。
离散傅里叶变换(DFT)是连续傅里叶变换的一种离散形式,将一个离散的信号序列x(n)转换为离散的频谱X(k),其公式如下:X(k)=∑xN−1n=0(n)e−i2πknN其中,k表示频率索引,N表示信号的长度。
快速傅里叶变换离散傅里叶变换的计算复杂度为O(N2),当N较大时,计算时间将会变得非常长。
为了提高计算效率,我们引入了快速傅里叶变换(FFT)。
FFT 是一种高效的算法,能够将离散傅里叶变换的计算复杂度降低到O(NlogN),使得大规模的信号处理成为可能。
傅里叶变换的应用傅里叶变换在信号处理和频谱分析中有着广泛的应用。
它可以用于图像压缩、音频处理、信号滤波、图像恢复等领域。
例如,在音频处理中,我们可以使用傅里叶变换将时域的声音信号转换为频域的频谱,以便对声音进行频谱分析和滤波处理。
fft曲线平滑-概述说明以及解释
fft曲线平滑-概述说明以及解释1.引言1.1 概述概述FFT(Fast Fourier Transform,快速傅里叶变换)是一种广泛应用于信号处理和频谱分析的算法。
通过将信号从时域转换为频域,FFT能够分析信号中的频率成分,从而实现对信号的特征提取、滤波和谱分析等功能。
在信号处理领域,FFT被广泛应用于音频处理、图像处理、通信系统、雷达系统等众多领域。
通过将信号转换为频域表示,FFT能够快速计算信号的频谱,并提取信号中的频率特征。
这为进一步的信号分析和处理提供了基础。
本文的重点是FFT曲线平滑方法。
在实际应用中,我们常常会遇到从FFT得到的频谱曲线存在噪声或震荡的情况。
这些噪声和震荡会对进一步的信号分析和处理带来困扰。
为了去除这些噪声和震荡,研究人员提出了各种FFT曲线平滑的方法。
这些方法包括基于窗函数的平滑、滑动平均平滑、高斯平滑等。
本文将介绍这些方法的原理和应用,并比较它们的优劣。
通过对FFT曲线的平滑处理,我们可以得到更准确和可靠的频谱结果。
这将有助于在音频处理、图像处理和通信系统等领域中更好地分析和理解信号。
同时,FFT曲线平滑方法的研究也是一个不断发展的领域,未来我们可以期待更多更有效的平滑算法的出现。
通过本文的学习,读者将能够深入了解FFT的基本原理、应用,以及FFT曲线平滑方法的原理、效果和应用。
同时,读者也可以对FFT曲线平滑的未来发展进行展望。
本文的目的是为读者提供一个全面的介绍和参考,帮助读者更好地理解和应用FFT曲线平滑技术。
1.2 文章结构文章结构部分的内容可以是以下几点:本文主要分为引言、正文和结论三个部分。
引言部分主要对文章的研究对象进行概述,介绍FFT曲线平滑的背景和意义。
同时,还会对整个文章的结构进行简要说明,为读者提供一个概览。
正文部分是整篇文章的核心部分,包括FFT的基本原理、FFT在信号处理中的应用以及FFT曲线平滑的方法。
在2.1节中,我们将介绍FFT的基本原理,包括离散傅里叶变换(DFT)和快速傅里叶变换(FFT)的基本概念和理论基础。
小波分析中英文对照外文翻译文献
小波分析中英文对照外文翻译文献(文档含英文原文和中文翻译)译文:一小波研究的意义与背景在实际应用中,针对不同性质的信号和干扰,寻找最佳的处理方法降低噪声,一直是信号处理领域广泛讨论的重要问题。
目前有很多方法可用于信号降噪,如中值滤波,低通滤波,傅立叶变换等,但它们都滤掉了信号细节中的有用部分。
传统的信号去噪方法以信号的平稳性为前提,仅从时域或频域分别给出统计平均结果。
根据有效信号的时域或频域特性去除噪声,而不能同时兼顾信号在时域和频域的局部和全貌。
更多的实践证明,经典的方法基于傅里叶变换的滤波,并不能对非平稳信号进行有效的分析和处理,去噪效果已不能很好地满足工程应用发展的要求。
常用的硬阈值法则和软阈值法则采用设置高频小波系数为零的方法从信号中滤除噪声。
实践证明,这些小波阈值去噪方法具有近似优化特性,在非平稳信号领域中具有良好表现。
小波理论是在傅立叶变换和短时傅立叶变换的基础上发展起来的,它具有多分辨分析的特点,在时域和频域上都具有表征信号局部特征的能力,是信号时频分析的优良工具。
小波变换具有多分辨性、时频局部化特性及计算的快速性等属性,这使得小波变换在地球物理领域有着广泛的应用。
随着技术的发展,小波包分析(Wavelet Packet Analysis)方法产生并发展起来,小波包分析是小波分析的拓展,具有十分广泛的应用价值。
它能够为信号提供一种更加精细的分析方法,它将频带进行多层次划分,对离散小波变换没有细分的高频部分进一步分析,并能够根据被分析信号的特征,自适应选择相应的频带,使之与信号匹配,从而提高了时频分辨率。
小波包分析(wavelet packet analysis)能够为信号提供一种更加精细的分析方法,它将频带进行多层次划分,对小波分析没有细分的高频部分进一步分解,并能够根据被分析信号的特征,自适应地选择相应频带,使之与信号频谱相匹配,因而小波包具有更广泛的应用价值。
利用小波包分析进行信号降噪,一种直观而有效的小波包去噪方法就是直接对小波包分解系数取阈值,选择相关的滤波因子,利用保留下来的系数进行信号的重构,最终达到降噪的目的。
傅里叶变换在信号处理中的应用
傅里叶变换在信号处理中的应用信号处理是一门研究如何提取、改变和分析信号的学科。
在现代科学和工程领域,信号处理的广泛应用使得我们能够从多种传感器中获取、处理和理解大量的数据。
而傅里叶变换作为信号处理中最基本且最重要的数学工具之一,具有广泛的应用。
傅里叶变换是一种线性算法,它将一个函数(在时间域或空域中的信号)转换为另一个函数(在频域中的信号),从而使我们可以在频域中分析信号。
通过将信号分解成一系列正弦和余弦函数的和,傅里叶变换的主要作用是将复杂的波形分解为更简单的组成部分。
这种分解过程是通过傅里叶级数展开或者傅里叶积分得到的。
在信号处理中,傅里叶变换的应用广泛涉及到多个方面。
以下是傅里叶变换在信号处理中的几个重要应用领域:1. 音频和图像处理:傅里叶变换在音频和图像处理中有着广泛的应用。
通过傅里叶变换,我们可以将时域中的音频信号或图像信号转换为频域中的频谱。
这使得我们可以分析音频或图像的频率成分,进行降噪、去除干扰、滤波、压缩等处理操作。
傅里叶变换在音乐中的应用,如音频压缩、音频合成等,以及在图像处理和计算机视觉领域中的应用,如图像增强、图像去噪、特征提取等,都离不开傅里叶变换。
2. 通信系统:傅里叶变换在通信系统中扮演着重要的角色。
通过傅里叶变换,可以将模拟信号转换为数字信号,并进行频域调制和解调。
傅里叶变换可以用于信号编码、信道估计、去除干扰以及实现调制和解调算法。
许多现代数字通信系统都采用傅里叶变换技术用于信号处理和调制。
3. 语音识别与语音合成:在语音识别与合成领域,傅里叶变换被广泛应用。
通过傅里叶变换,可以将语音信号转换为频域中的频谱,并提取其中的特征参数,用于语音识别、声纹识别和语音合成等应用。
傅里叶变换可以帮助我们实现语音信号的分析和处理,从而使得语音识别系统更加稳定和准确。
4. 数字滤波和系统分析:傅里叶变换在数字滤波和系统分析中发挥着重要作用。
通过将信号转换到频域,我们可以对信号进行滤波操作,去除不需要的频率成分,滤波器的设计与分析可以极大地简化。
音频信号处理算法研究与优化
音频信号处理算法研究与优化随着科技的不断发展,音频处理技术也越来越成熟。
音频信号处理算法是音频处理技术中最关键的一部分,它直接关系到音频信号的质量和效果。
本文将就音频信号处理算法进行深入研究与优化。
一、音频信号处理算法的研究发展音频信号处理算法从诞生之初,就没有停止过探究和完善。
最初的音频信号处理算法是数字信号处理(DSP)技术,虽然相对于模拟信号处理(ASP)技术在处理音频信号方面更加便捷,但它不能充分利用现代计算机的计算能力,并且在处理复杂信号时还存在一定的缺陷。
为了解决数字信号处理技术的问题,学者们开始探究更为先进的信号处理算法。
在研究的初期,人们主要关注的是梅尔倒谱分析法、小波分析法、快速傅里叶变换(FFT)等算法。
随着时间的推移,人们对于信号处理算法的研究逐渐深入,出现了自适应滤波(ADAPTIVE FILTER)、人工神经网络(ARTIFICIAL NEURAL NETWORK)、模糊逻辑控制(FUZZY LOGIC CONTROL)等新算法,这些新算法在处理复杂音频信号方面有了比较显著的效果。
二、音频信号处理算法的优化当前,人们面临的主要问题是如何优化音频信号处理算法。
音频信号处理技术已经进入到了一个相对成熟的阶段,但各种复杂信号集成在一起处理时,算法的优化依然十分关键。
在此,我们探讨几种常见的算法优化方法。
1.优化算子的选择算子是指算法中用于求解某一特定问题的操作符号。
优化算子的选择可以直接影响算法的效果。
在实际的优化过程中,人们通常采用与样本数据集契合度较高的算子,以及具有较低时间复杂度和空间复杂度的算子,这样可以在保证优化效果的前提下,尽可能地降低算法的计算时间和内存占用等方面的成本。
2.改进算法的结构改进算法的结构是可以优化算法的效果的一个重要方法。
一些经典的音频信号处理算法中,可能存在着类似于冗余项的问题,这样就增加了算法的空间复杂度和时间复杂度。
通过改进算法结构,减少算法中的冗余项,可以降低算法的计算成本,提高算法的效率。
傅里叶变换
其中Xk是傅里叶幅度。直接使用这个公式计算的计算复杂度为
,而快速傅里叶变换
(FFT)可以将复杂度改进为
。计算复杂度的降低以及数字电路计算能力的发展
使得DFT成为在信号处理领域十分实用且重要的方法。
在阿贝尔群上的统一描述
以上各种傅里叶变换可以被更统一的表述成任意局部紧致的阿贝尔群上的傅里叶变换。这一
问题属于调和分析的范畴。在调和分析中,一个变换从一个群变换到它的对偶群(dual group)。此外,将傅里叶变换与卷积相联系的卷积定理在调和分析中也有类似的结论。傅
变换
注释
10
矩形脉冲和归一化的sinc函数
11
变换10的频域对应。矩形函数是理想的低通滤波器,sinc函 数是这类滤波器对反因果冲击的响应。
12
tri 是三角形函数
13
变换12的频域对应
14
高斯函数exp( − αt2)的傅里叶变换是他本身.只有当Re(α) > 0 时,这是可积的。
15
光学领域应用较多
若函数 及 都在 傅里叶变换存在,且
自的傅里叶逆变换的卷积。 帕塞瓦尔定理
上绝对可积,则卷积函数
的
。卷积性质的逆形式为 ,即两个函数乘积的傅里叶逆变换等于它们各
/wiki/%E5%82%85%E9%87%8C%E5%8F%B6%E5%8F%98%E6%8D%A2
/wiki/%E5%82%85%E9%87%8C%E5%8F%B6%E5%8F%98%E6%8D%A2
2009-11-27
傅里叶变换 - 维基百科,自由的百科全书
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里叶变换的广义理论基础参见庞特里雅金对偶性(Pontryagin duality)中的介绍。
傅里叶分析和变换在信号处理和音频处理中的应用
傅里叶分析和变换在信号处理和音频处理中的应用信号处理和音频处理是现代科技领域中非常重要的研究方向。
而傅里叶分析和变换作为一种强大的数学工具,被广泛应用于这两个领域。
本文将探讨傅里叶分析和变换在信号处理和音频处理中的应用。
一、傅里叶分析在信号处理中的应用在信号处理中,傅里叶分析被用来将信号从时域转换到频域,从而能够更好地理解信号的频率特性。
通过傅里叶分析,我们可以得到信号的频谱信息,包括信号的频率成分和振幅。
这对于信号的特征提取和分析非常重要。
举个例子来说,假设我们需要分析一段音频信号中的音乐和噪声成分。
首先,我们可以对这段音频信号进行傅里叶变换,得到其频谱。
然后,通过观察频谱图,我们可以清楚地看到音乐和噪声在频率上的分布情况。
这样,我们就可以根据频谱信息,将音乐和噪声成分进行分离,从而实现音频信号的降噪处理。
除了音频信号的降噪处理,傅里叶分析还广泛应用于语音信号的处理和图像信号的处理。
通过傅里叶分析,我们可以提取语音信号的频率特征,从而实现语音识别和语音合成等应用。
对于图像信号的处理,傅里叶分析可以帮助我们提取图像的纹理特征,实现图像的增强和去噪等操作。
二、傅里叶变换在音频处理中的应用与傅里叶分析不同,傅里叶变换是一种将信号从时域转换到频域的数学工具。
在音频处理中,傅里叶变换被广泛应用于音频信号的压缩和编码。
音频信号的压缩是指将原始音频信号的数据量减小,以便更好地存储和传输。
傅里叶变换可以将音频信号转换为频域表示,然后通过对频域数据的处理,实现对音频信号的压缩。
其中,一种常用的压缩算法是基于傅里叶变换的MP3算法。
该算法通过对音频信号的频谱进行分析,然后根据信号的特性进行数据压缩,从而实现对音频信号的高效编码和解码。
除了音频信号的压缩编码,傅里叶变换还可以应用于音频信号的滤波处理。
在音频处理中,滤波是一种常见的操作,用于去除音频信号中的噪声或不需要的频率成分。
通过傅里叶变换,我们可以将音频信号转换为频域表示,然后通过对频域数据的滤波操作,实现对音频信号的去噪和频率调整。
傅里叶变换magnitude 和phase
傅里叶变换magnitude 和phase 傅里叶变换是一种十分重要的数学工具,在信号处理、图像处理、通信系统等领域具有广泛的应用。
傅里叶变换可以将一个连续或离散的时间域信号转换为频域信号,通过分析频域信号的幅度与相位信息,我们可以获得关于信号频谱的重要信息。
在傅里叶变换中,幅度和相位是两个最重要的概念,它们分别描述了频域信号的振幅和相对于时间域信号的延迟或者相位差。
首先,我们来谈谈傅里叶变换的幅度谱,也称为magnitude spectrum。
幅度谱描述了频域信号的振幅特性,它告诉我们频域信号中不同频率成分的强弱。
通过分析幅度谱,我们可以得到信号中频率成分的增益或衰减情况。
对于连续时间域信号,我们可以通过连续傅里叶变换(Continuous Fourier Transform)得到幅度谱。
连续傅里叶变换的公式如下:F(ω) = ∫[f(t) * e^(-jωt)] dt其中,F(ω)表示频域信号的复数形式,f(t)是原始信号,ω为角频率,e^(-jωt)是复指数形式的正弦函数。
对于离散时间域信号,我们可以通过离散傅里叶变换(Discrete Fourier Transform)得到幅度谱。
离散傅里叶变换的公式如下:F(k) = Σ[f(n) * e^(-j2πkn/N)]其中,F(k)表示频域信号的复数形式,f(n)是原始信号,k为频率索引,N为信号的长度。
得到频域信号后,我们可以通过计算每个频率分量的幅度,得到幅度谱。
幅度谱的计算公式如下:M agnitude = |F(ω)|其中,|F(ω)|表示频域信号的振幅。
幅度谱通常以频率为横轴,振幅为纵轴进行绘制。
通过分析幅度谱,我们可以得到信号中不同频率成分的强弱,从而可以判断信号的频谱特性。
接下来,我们来讨论傅里叶变换的相位谱,也称为phase spectrum。
相位谱描述了频域信号相对于时间域信号的延迟或者相位差。
相位谱可以告诉我们信号不同频率成分之间的时间关系,从而可以重构信号或者改变信号的相位。
傅里叶变换及应用
傅里叶变换在MATLZB里的应用摘要:在现代数学中,傅里叶变换是一种非常重要的变换,且在数字信号处理中有着广泛的应用。
本文首先介绍了傅里叶变换的基本概念、性质及发展情况;其次,详细介绍了分离变数法及积分变换法在解数学物理方程中的应用。
傅立叶变换将原来难以处理的时域信号转换成了易于分析的频域信号,再利用傅立叶反变换将这些频域信号转换成时域信号。
应用MATLAB实现信号的谱分析和对信号消噪。
关键词:傅里叶变换;MATLAB软件;信号消噪Abstract: In modern mathematics,Fourier transform is a transform is very important ,And has been widely used in digital signal paper first introduces the basic concepts, properties and development situation of Fourier transform ;Secondly, introduces in detail the method of separation of variables and integral transform method in solving equations in Mathematical transformation makes the original time domain signal whose analysis is difficult easy, by transforming it into frequency domain signal that can be transformed into time domain signal by inverse transformation of Fourier. Using Mat lab realizes signal spectral analysis and signal denoising.Key word: Fourier transformation, software of mat lab ,signal denoising1、傅里叶变换的提出及发展在自然科学和工程技术中为了把较复杂的运算转化为较简单的运算,人们常常采用所谓变换的方法来达到目的"例如在初等数学中,数量的乘积和商可以通过对数变换化为较简单的加法和减法运算。
快速傅里叶变换fft mathmatica
快速傅里叶变换(FFT)是一种非常重要的数学工具,它在信号处理、图像处理、计算机视觉等领域有着广泛的应用。
快速傅里叶变换算法的发明有利于对信号频谱的快速计算,从而加快了信号处理的速度。
在本文中,我们将从多个角度来探讨快速傅里叶变换,并深入理解它的原理和应用。
1. 什么是傅里叶变换?傅里叶变换是一种数学工具,它可以将一个函数从时间或空间域转换到频率域。
通过傅里叶变换,我们可以将一个信号拆分成不同频率的成分,从而更好地理解信号的特性。
在信号处理领域,傅里叶变换被广泛应用于声音、图像等数据的分析和处理中。
2. 快速傅里叶变换的原理快速傅里叶变换是一种高效的傅里叶变换算法,它可以在对数时间内完成信号频谱的计算。
其原理是基于分治法和递归思想的,通过将信号分解成子问题,并利用对称性质和周期性质,从而快速计算出频谱信息。
快速傅里叶变换算法的发明极大地加速了信号处理的速度,使得实时处理成为可能。
3. 快速傅里叶变换的应用快速傅里叶变换在信号处理、图像处理、通信等领域有着广泛的应用。
在音频处理中,通过快速傅里叶变换,我们可以快速计算出音频信号的频谱信息,从而进行音频分析、音频合成等操作。
在图像处理中,快速傅里叶变换可以用于图像的频域滤波、图像压缩等操作。
在通信领域,快速傅里叶变换也被应用于调制解调、信道估计等方面。
4. 我对快速傅里叶变换的个人观点和理解作为一种重要的数学工具,快速傅里叶变换在现代科学技术中扮演着不可或缺的角色。
它的高效性和广泛应用性使得它成为了信号处理领域中的核心算法之一。
虽然快速傅里叶变换算法本身较为复杂,但通过对其原理和应用的深入理解,我们可以更好地利用这一工具,为实际问题提供更好的解决方案。
总结在本文中,我们对快速傅里叶变换进行了全面的探讨,从傅里叶变换的基本概念到快速傅里叶变换算法的原理和应用,希望读者能更全面、深刻和灵活地理解这一重要的数学工具。
通过对快速傅里叶变换的研究,我们可以更好地处理和分析信号数据,为实际问题的解决提供更好的方法和工具。
傅里叶变换在信号处理中的应用
傅里叶变换在信号处理中的应用傅里叶变换简单的说,就是把信号从时域变化的频域分析。
传统的傅里叶变换在数字信号处理中使用的并不多,因为傅里叶变换是一般用于连续信号的分析。
使用最多的是离散傅里叶变换(DFT),而DFT是可以使用快速傅里叶变换(FFT)实现的。
也就是运算复杂度小,可以用DSP等硬件轻易实现。
DFT是现代信号处理的基础,应用非常广泛,比如自适应滤波器啊,阵列信号处理、正交频分复用等等都用的到。
傅里叶变换在信号处理中有着很广泛的应用,首先我们来了解一下什么是傅里叶变换。
f(t)是t的函数,如果t满足狄里赫莱条件:具有有限个间断点;具有有限个极值点;绝对可积。
则有下图①式成立。
称为积分运算f(t)的傅立叶变换,②式的积分运算叫做F(ω)的傅立叶逆变换。
F(ω)叫做f(t)的像函数,f(t)叫做F(ω)的像原函数。
F(ω)是f(t)的像。
f(t)是F(ω)原像。
①傅里叶变换②傅里叶逆变换傅里叶变换在物理学、电子类学科、数论、组合数学、信号处理、概率论、统计学、密码学、声学、光学、海洋学、结构动力学等领域都有着广泛的应用(例如在信号处理中,傅里叶变换的典型用途是将信号分解成幅值谱——显示与频率对应的幅值大小)。
尽管最初傅立叶分析是作为热过程的解析分析的工具,但是其思想方法仍然具有典型的还原论和分析主义的特征。
"任意"的函数通过一定的分解,都能够表示为正弦函数的线性组合的形式,而正弦函数在物理上是被充分研究而相对简单的函数类,这一想法跟化学上的原子论想法何其相似!奇妙的是,现代数学发现傅立叶变换具有非常好的性质,使得它如此的好用和有用,让人不得不感叹造物的神奇:1.傅立叶变换是线性算子,若赋予适当的范数,它还是酉算子;2.傅立叶变换的逆变换容易求出,而且形式与正变换非常类似;3.正弦基函数是微分运算的本征函数,从而使得线性微分方程的求解可以转化为常系数的代数方程的求解.在线性时不变的物理系统内,频率是个不变的性质,从而系统对于复杂激励的响应可以通过组合其对不同频率正弦信号的响应来获取;4.著名的卷积定理指出:傅立叶变换可以化复杂的卷积运算为简单的乘积运算,从而提供了计算卷积的一种简单手段;5.离散形式的傅立叶变换可以利用数字计算机快速的算出(其算法称为快速傅立叶变换算法(FFT)).正是由于上述的良好性质,傅里叶变换在物理学、数论、组合数学、信号处理、概率、统计、密码学、声学、光学等领域都有着广泛的应用。
otfs辛傅里叶变换
otfs辛傅里叶变换
OTFS(Over-the-Air-Frame-Synchronization)是一种通过信道进行帧同步的技术,它可以在复杂的无线信道环境下实现高效的同步和通信。
傅里叶变换(Fourier Transform)是一种重要的数学工具,用于将一个函数表示为一系列正弦和余弦函数的加权和。
它在信号处理、图像处理、通信系统等领域中广泛应用。
OTFS中的傅里叶变换主要用于信号的调制和解调过程。
在OTFS系统中,发送信号经过多径信道传播后,会发生时延扩展和频率选择性衰减等问题。
为了解决这些问题,需要对接收到的信号进行傅里叶变换,将时域的信号转换为频域的信号。
通过傅里叶变换,可以将时域上的信号转换为频域上的信号,并得到信号在不同频率上的能量分布情况。
这样可以更好地理解信号的频谱特性,从而有效地进行信号检测、信号估计和信号解调等操作。
总结来说,OTFS中的傅里叶变换是用于将接收到的信号从时域转换为频域,以便更好地处理和分析信号,实现高效的数据传输。
通过傅里叶变换,可以对信号的频谱进行分析和处理,从而提高系统的性能和可靠性。
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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。