数字信号处理的简单介绍文献翻译及英文原文

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数字信号处理 名词解释-概述说明以及解释

数字信号处理 名词解释-概述说明以及解释

数字信号处理名词解释-概述说明以及解释1.引言1.1 概述数字信号处理(Digital Signal Processing,简称DSP)是一种广泛应用于信号处理领域的技术,它利用数字化的方式对连续时间信号进行处理和分析。

数字信号处理可以实现信号的滤波、频谱分析、模拟与数字信号的转换、信息编码解码等功能,是现代通信、音视频处理、生物医学领域等各个领域中不可或缺的技术手段。

通过数字信号处理技术,我们可以更加精确和高效地处理各种类型的信号,包括声音、图像、视频等。

数字信号处理可以使信号的处理过程更加稳定可靠,同时也可以方便地与计算机等数字系统进行集成,实现更多复杂功能。

在本篇文章中,我们将深入探讨数字信号处理的定义、应用领域以及基本原理,以期让读者对这一重要领域有更加全面的认识和理解。

1.2 文章结构本文将分为三个主要部分,分别是引言、正文和结论。

在引言部分,我们将对数字信号处理进行简要的概述,并介绍文章的结构和目的。

正文部分将详细讨论数字信号处理的定义、应用领域和基本原理。

最后,在结论部分,我们将总结数字信号处理的重要性,探讨未来数字信号处理的发展趋势,并做出最终的结论。

通过这样的结构安排,读者能够清晰地了解数字信号处理的基本概念、应用以及未来发展方向。

1.3 目的:本文旨在介绍数字信号处理的概念、应用领域和基本原理,旨在帮助读者更深入了解数字信号处理的重要性和作用。

通过对数字信号处理的定义和应用领域的介绍,读者可以了解数字信号处理在各个领域中的广泛应用和重要性。

同时,通过对数字信号处理的基本原理的讲解,读者可以更好地理解数字信号处理的工作原理和技术特点。

通过本文的阐述,希望读者能够全面了解数字信号处理的基本概念和工作原理,进而认识到数字信号处理在现代科学技术中的重要性和必要性。

同时,本文也将展望未来数字信号处理的发展趋势,希望能够启发读者对数字信号处理领域的进一步研究和探索。

最终,通过本文的阐述,读者可以更加深入地理解数字信号处理这一重要的科学技术领域。

南邮专业英语报告 信号处理导论完整版(包含翻译,原文和单词)

南邮专业英语报告 信号处理导论完整版(包含翻译,原文和单词)

0
1
2
SWEEP
样值处理算法来计算飘动的滤波器系数,再分别计算每次输入抽样的滤波。
Flanging、合唱、调相三种效果都是把一个简单滤波器的系数设计尾随输入
抽样变化而使滤波器成为时变滤波器。自适应信号处理也是随时间改变滤波器的
系数。系数与时间之间的关系是受某些设计条件的限制,即滤波器系数相对于输
入抽样调节并且优化。自适应算法的实施也就是要求滤波器的样值处理算法当中
英文原文
8.2 Digital Audio Effects Audio effects, such as delay, echo, reverberation, comb filtering, flanging, chorusing, pitch shifting, stereo imaging, distortion, compression, expansion, noise gating, and equalization, are indispensable in music production and performance [115 –151]. Some are also available for home and car audio systems.
程序chorus.m演示的是正弦信号经合唱处理后的情形。 调相(Phase Shifting)对吉他手、键盘演奏人员、歌唱家来说是经常采用的一种 效果。调相是把声音信号用一个窄带陷状滤波器过滤,再把过滤信号的一部分与 源信号相加而得到的。
陷点的频率以可控的方式调节,比如说可以用一个低频振荡器,也可以用脚踏板 控制。陷点附近的频率有较强的漂移,与原来的直接声音结合,使得相位在频率 轴上发生抵消或加强,整个相位在频率轴上出现波动。

电子信息专业英语之数字信号处理篇

电子信息专业英语之数字信号处理篇



In this case, an error message can be sent, and the message be retransmitted(转发). On the compact disc(光盘), 1 and 0 are represented by dimples(凹痕). The leading and trailing edges of a dimple represent a 1(凹痕的前端和后端表示1); no change represents a 0(无变化则为0). 在此情况下,可以发送错误信息,要求重传信号。 在光盘上用凹痕来表示0和1,凹痕的前端和后端表 示1,无变化则为0,


The precise voltage of each symbol is not important, but it is critical that its value lies within one of the two allowable ranges. If the value lies outside the two allowable ranges, the telegraph must make a choice of either 1 or 0, and an error may occur. 每个符号所表示的准确电压值并不重要,关键是它 要落在上述任意一个允许范围之内,否则就必须在 0,1之间作出选择而可能发生错误。


In quantization, if the amplitude(大小) of a discrete-time signal does not fall exactly on a quantization level, then the value must be approximated by a quantization level either by truncation or rounding, in either case, errors will occur. Such errors are called quantization errors. 量化时,如果离散时间信号的大小与量化电平不一 致,那么它就必须以截断或舍入的方式用相应的量 化电平来近似。两种情况下都会产生误差。这种误 差叫量化误差。

专业英语翻译之数字信号处理

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。

dsp外文翻译

dsp外文翻译

外文参考文献翻译英文题目 The Breadth and Depth of DSP 中文题目 DSP的广度和深度学院自动化与电气工程学院专业自动化姓名白学文学号 201108536指导教师王思明2015 年 04月 20日DSP的广度和深度数字信号处理是最强大的技术,将塑造二十一世纪的科学与工程之一。

革命性的变化已经在广泛的领域:通信,医疗成像,雷达和声纳,高保真音乐再现,石油勘探,仅举几例。

上述各领域已建立了深厚的DSP技术,用自己的算法,数学,和专门技术。

这种呼吸和深度的结合,使得它不可能为任何一个人掌握所有已开发的DSP技术。

DSP教育包含两个任务:学习一般适用于作为一个整体领域的概念,并学习您感兴趣的特定领域的专门技术。

本章开始描述DSP已在几个不同领域的戏剧性效果的数字信号处理的世界,我们的旅程。

革命已经开始。

1 DSP的根源独特的数据类型,它使用的信号,数字信号处理是区别于其他计算机科学领域。

在大多数情况下,这些信号源于感觉来自现实世界的数据:地震的震动,视觉图像,声波等DSP是数学,算法,并用来操纵这些信号的技术后,他们已被转换成数字形式。

这包括了各种目标,如:加强视觉图像识别和语音生成,存储和传输的数据压缩,等假设我们重视计算机模拟 - 数字转换器,并用它来获得一个现实世界的数据块。

DSP回答了这个问题:下一步怎么办?DSP的根是在20世纪60年代和70年代数字计算机时首次面世。

电脑是昂贵的,在这个时代,DSP是有限的,只有少数关键应用。

努力开拓,在四个关键领域:雷达和声纳,国家安全风险是石油勘探,可以大量资金;太空探索,其中的数据是不可替代的;和医疗成像,可节省生活。

20世纪80年代和90年代的个人电脑革命,引起新的应用DSP的爆炸。

而不是由军方和政府的需求动机,DSP的突然被带动的商业市场。

任何人士如认为他们可以使资金在迅速扩大的领域突然一个DSP供应商。

DSP的市民等产品达到:移动电话机,光盘播放器,电子语音邮件。

数字信号 外文翻译 外文文献 英文文献 数字信号处理器简介

数字信号 外文翻译 外文文献 英文文献 数字信号处理器简介

Introduction to Digital Signal ProcessingSignal and digital signal processing as a branch of information processing, has infiltrated into the scientific research, technological development, industrial production, national defense and national economy in various fields has achieved fruitful results. Of the signal in time domain and transform domain analysis of the characteristics, processing, enable us to the nature of the signal features and a more clear awareness and understanding of the signals need to get our form of information of use to improve the extent, then in a broader and higher level the access to information. DSP chip, also known as digital signal processors, is particularly suitable for digital signal processing microprocessor. The emergence and development of DSP chips, digital signal processing technology for the improvement of many of the new system, the new algorithm came into being, the application field is expanding. Currently, DSP chips have been widely used in communications, automation, aerospace, military, medical and other fields.Digital signal processing is that the signal to digital form and deal with the theory and technology. Digital signal processing and analog signal processing is a subset of signal processing. The purpose of digital signal processing is a continuous real-world analog signals to measure or filter. Therefore needed before carrying out digital signal processing the signal from the analog domain into the digital domain, this is usually achieved through the ADC. The digital signal processing often have to transform the output to the analog domain, which is achieved through the DAC. The core digital signal processing algorithm is the discrete Fourier transform (DFT), is the DFT to the signal in the digital domain and frequency domain are realized discrete, general-purpose computer which can handle discrete signal. Er Shi digital signal processing theory to practical from the fast Fourier transform (FFT), FFT has significantly reduced the DFT computation to make real-time digital signal processing as possible, greatly promote the development of the discipline.The advantages of digital signal processing system as follows: 1. Good flexibility: when the processing method and parameters change, the processing system simply by changing the software design to meet the appropriate changes. 2. Precision: Signal processing systems can be A / D transformation of the median, the processor word length and the appropriate algorithm to meet the accuracy requirements. 3. Reliability: processing system affected by environmental temperature, humidity, noise and electromagnetic interference caused by the less affected. 4. Can be a large-scale integration: With the technological development of semiconductor integrated circuits, digital circuits can be used for very high integration, small size, low power consumption, and good product consistency.1.The development of DSPThe late 70s early 80s, AMI's S2811 chip, Intel's 2902 chip, the birth marks the beginning of DSP chips. With the rapid development of semiconductor integrated circuits, high-speed real-time digital signal processing and digital signal processing applications demands the continuous extension of the field, in the 80 years since the beginning of ten years, DSP made epoch-making development. View from the operation speed, MAC (multiply and accumulate) time of 400 nsfrom 80 down to 40 ns below, data-processing capacity several times. MIPS (million instructions per second) from the early 80's 5MIPS to the current 40 MIPS or more. Key components within the DSP chip multiplier accounts from the early 80s about 40% of mold Area dropped to less than 5%, on-chip RAM for more than an order of magnitude increase. View from the manufacturing process, the early 80s with 4μm of NMOS t echnology and is now using sub-micron CMOS technology, DSP chips, pin numbers from the early 80s up to 64 to more than 200 now, so the increase in the number of chip pins increase the flexibility of application, so that all external memory expansion and more convenient communication between processors. And earlier than DSP chips, DSP chips now have two floating point and fixed-point data format, floating-point DSP chips for floating-point operations, so computing precision greatly enhanced. DSP chip cost, size, voltage, weight and power consumption earlier DSP chip with a large degree of decline. In the DSP development system, software and hardware development tools continue to improve. At present some of the chip with the corresponding integrated development environment, which supports breakpoint settings and program memory, data memory and DMA access and the program running and tracking a single department, and can use high-level language programming, some manufacturers and some software development business application software for the DSP to prepare a common subroutine library and various algorithms and various interface program, which makes application development easier, shorten development time, thus increasing the efficiency of General characteristics of DSP chips(1) In a cycle to complete a multiplication and an accumulation.(2)Harvard architecture, program and data space separated, you can also access instructions and data.(3) Chip with fast RAM, typically by an independent data bus simultaneously in two visits.(4) A low cost or no cost recycling and jump hardware support.(5) Fast interrupt handling, and hardware I / O support.(6) With single-cycle operation in a number of hardware address generator.(7) Can perform multiple operations in parallel.(8) To support pipeline operations, fetch, decode, and execution operations may overlap.2.DSP Applications in Communication SystemDSP technology has been widely used in communications. Mainly in the following areas.(1)Software Defined RadioSoftware radio technology and computer technology is constantly touch merged into the 3rd generation mobile communication systems provide a good user interface. DSP hardware technology and its algorithm is the key to realization of software radio. Software radio system flexibility, openness and compatibility features of the signal processor is mainly centered through a common hardware platform and software to achieve. It is mainly to complete the internal data processing station, modem and codec and so on. As the internal data flow and a large radio filtering, frequency and other processing operations more often, to high-speed, real-time, parallel digital signal processor module or ASIC to meet the demand. To complete such an arduous task, to increase the processing speed required hardware, chip capacity expansion and requested algorithmfor optimization and improvement of the processor. The two aspects of the continuous improvement requirements will be the development of digital signal processing technology unremitting power. The only way to achieve high-speed internal software radio operation and multiple functions of the flexible switching and control. Software implementations are generally two types of devices that use DSP to implement and field-programmable gate array (FPGA) to implement.(2)Speech CodingThe purpose of voice data compression is that in the transmission rate as low as possible gain high-quality audio performance, that is, that narrow bandwidth of voice signals Ke Yi in the channel transmission, the voice of quality and Xiajiangdebu more Huo dropped as much as possible . Speech coding system for the early use of the waveform coding method, also called waveform coding. Essentially follows the Nyquist sampling theorem, strong adaptability, good quality synthetic speech, but the high rate coding, coding efficiency is very low. The parameter coding is different from the efficient encoding of waveform coding method, which is based on the mechanism in speech production, mainly on the extraction characteristics of speech signal encoding parameters, you can achieve very low encoding rate. But only to the effect of synthesized speech, voice quality than waveform coding.Over the past decade, speech coding technology has made breakthrough progress. ITU, one after another through a series of low bit rate speech coding standard telephone band. By the parameter coding and waveform coding the hybrid coding method that is of a synthetic coding, can get better sound quality at the same time reduce the coding rate, the most representative is the linear predictive coding (LPG) and Code Excited Linear Prediction Coding (CELP). This encoding can be 4-16kbit / s of the low coding rate on high-quality reconstructed speech, but the complexity of the algorithm, the computing speed of the processor demanding. On speech processing, the higher the compression ratio, the more complex encoding algorithm, real-time compression can not use the logic circuit, it will not use the bulky, slow, high cost of Microcomputer. The DSP is a suitable choice, the web conferencing, voice communications, surveillance systems are important areas of components. The use of DSP voice compression algorithm not only provides a broad application prospects, and the system design simple, reliability is also greatly improved.(3)GPS SystemGPS was developed by the United States to receive navigation satellite signals based on non-autonomous navigation system. Is widely used in various military and economic fields. With GPS technology in various fields to promote and popularize the application of the receiver's small size, intelligence and algorithms to meet user needs studies is necessary.There are two global positioning system components: the satellite constellation, ground control / monitoring network and user receivers. In GPS applications, GPS receivers often require reprocessing the data collected, or use GPS receivers to provide some information for the development of an industry. DSP small size, high speed, low power, high reliability characteristics. Suited to the complexity of real-time GPS signal processing. OEM version of its composition with the GPS information systems, not only satisfy the real-time GPS signal processing and highcomplexity, and because of the powerful DSP processing capability, the system can be further extensions.Typical Application: (1) General signal processing: convolution, correlation, FFT, Hilbert transform, adaptive filtering, spectrum analysis, waveform generation and so on. (2) Communications: High-speed modulator / demodulator, encoder / decoder, adaptive equalizer, simulation, honeycomb network of mobile phones, echo / noise cancellation, fax, telephone conference, spread spectrum communications, data encryption and compression . (3) voice signal processing: speech recognition, speech synthesis, text changed voice, speech coding vector. (4) graphics image signal processing: two, three-dimensional graphics transformation and processing, robot vision, digital map, image enhancement and recognition, image compression and transmission, animation, desktop publishing systems. (5) control: robot control, engine control, automatic driving, voice and so on. (6) Instruments: function occurs, data acquisition, aerospace wind tunnel testing. (7) Consumer Electronics: Digital TV, digital voice synthesis, toys and games, digital answering machine.To DSP chip as the core structure of the digital signal processing system, data acquisition, transmission, storage and high-speed real-time processing as one can fully realize the advantages of digital signal processing system, can satisfy the precision equipment in the field of manned space flight , reliability, channel bandwidth, power consumption, voltage and weight requirements. Currently, DSP chips are the high-performance, high integration and low cost of direction, the various kinds of common and dedicated DSP chips has introduced new, applied technology and development tools continues to improve. Such as real-time digital signal processing applications - especially in the field of manned space applications in a wider space. We have reason to believe, DSP chips will further the development and application of signal processing on the manned space far-reaching impact.数字信号处理器简介数字信号处理作为信号和信息处理的一个分支学科,已渗透到科学研究、技术开发、工业生产、国防和国民经济的各个领域,取得了丰硕的成果。

数字信号处理Z变换中英对照翻译

数字信号处理Z变换中英对照翻译
������
(线性)
(5.1.3)
延迟特性表示 D 采样单元延迟信号的效果是相当于其 z 变换乘以因子 z-D, 即 X(n) → X(z) ⇒ ������(������ − ������) → ������ −������ ������(������)
������ ������
(延迟)
(5.1.4)
(5.1.1)
或者,明确写下一些术语: X(z)= ···+ x(−2)z2 + x(−1)z + x(0)+x(1)z−1 + x(2)z−2 + ··· 存在与非零信号值 x(n)一样多的项。非零项 z-n 可以被认为是值 x(n)的占位 符。如果信号 x(n)是因果关系,则只是负次幂 z-n,n≥出现在扩张中。如果 x(n)是 严格反的, 则为非零在 n≤-1 只有正次幂才会出现在扩张中, 则 z-n =z|n|时, 当 n≤1。如果 x(n)与因果和非因果部分混合,那么 Z 的负和正幂都会出现。 定义(5.1.1)也可应用于脉冲响应序列 H(n)数字滤波器。H(n)的 z 变换称为滤 波器的传递函数。定义如下:
∞ ∞ ∞ n −n
X(z) =
∑(0.5)n u−n (n)z=
n=−∞
∑(0.5) z
n=0
= ∑(0.5z −1 )n
n=0
由于 x(n)的因果关系,在 n≥ 0 上求和。这个无限的总和可以在无穷几何 级数公式的帮助下完成:

1 + x + x + x + ⋯ = ∑ xn =
n=0
2
3
1 1 1−x
������ ������ ������
δ(n − 2) → z−2,

信号处理中英文对照外文翻译文献

信号处理中英文对照外文翻译文献

信号处理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。

数字信号处理中英文对照外文翻译文献

数字信号处理中英文对照外文翻译文献

中英文对照外文翻译(文档含英文原文和中文翻译)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。

为了在计算机上使用一个模拟信号的计算机,它上面必须使用模拟到数字的转换器(ADC)使其数字化。

采样通常分两阶段进行,离散化和量化。

在离散化阶段,信号的空间被划分成等价类和量化是通过一组有限的具有代表性的信号值来代替信号近似值。

数字信号处理英文文献及翻译

数字信号处理英文文献及翻译

英文原文The simulation and the realization of the digital filterWith the information age and the advent of the digital world, digital signal processing has become one of today's most important disciplines and door technology. Digital signal processing in communications, voice, images, automatic control, radar, military, aerospace, medical and household appliances, and many other fields widely applied. In the digital signal processing applications, the digital filter is important and has been widely applied.1、 figures Unit on :Analog and digital filtersIn signal processing, the function of a filter is to remove unwanted parts of the signal, such as random noise, or to extract useful parts of the signal, such as the components lying within a certain frequency range.The following block diagram illustrates the basic idea.There are two main kinds of filter, analog and digital. They are quitedifferent in their physical makeup and in how they work. An analog filter uses analog electronic circuits made up from components such as resistors, capacitors and op amps to produce the required filtering effect. Such filter circuits are widely used in such applications as noise reduction, video signal enhancement, graphic equalisers in hi-fi systems, and many other areas. There are well-established standard techniques for designing an analog filter circuit for a given requirement. At all stages, the signal being filtered is an electrical voltage or current which is the direct analogue of the physical quantity (e.g. a sound or video signal or transducer output) involved. Adigital filter uses a digital processor to perform numerical calculations on sampled values of the signal. The processor may be a general-purpose computersuch as a PC, or a specialised DSP (Digital Signal Processor) chip. The analog input signal must first be sampled and digitised using an ADC (analog todigital converter). The resulting binary numbers, representing successive sampled values of the input signal, are transferred to theprocessor, which carries out numerical calculations on them. These calculations typically involve multiplying the input values by constants and adding the products together. If necessary, the results of these calculations, which now represent sampled values of the filtered signal, are output through a DAC (digital to analog converter) to convert the signal back to analog form.Note that in a digital filter, the signal is represented by a sequence of numbers, rather than a voltage or current.The following diagram shows the basic setup of such a system.Unit refers to the input signals used to filter hardware or software. If the filter input, output signals are separated, they are bound to respond to the impact of the Unit is separated, such as digital filters filter definition. Digital filter function, which was to import sequences X transformation into export operations through a series Y.According to figures filter function 24-hour live response characteristics, digital filters can be divided into two, namely, unlimited long live long live the corresponding IIR filter and the limited response to FIR filters. IIRfilters have the advantage of the digital filter design can use simulation results, and simulation filter design of a large number of tables mayfacilitate simple. It is the shortcomings of the nonlinear phase; Linear phase if required, will use the entire network phase-correction. Image processing and transmission of data collection is required with linear phase filters identity. And FIR linear phase digital filter to achieve, but an arbitrary margincharacteristics. Impact from the digital filter response of the units can be divided into two broad categories : the impact of the limited response (FIR) filters, and unlimited number ofshocks to (IIR) digital filters.FIR filters can be strictly linear phase, but because the system FIR filter function extremity fixed at the original point, it can only use the higher number of bands to achieve their high selectivity for the same filter design indicators FIR filter called band than a few high-IIR 5-10 times, the cost is higher, Signal delay is also larger. But if the same linear phase, IIR filters must be network-wide calibration phase, the same section also increase the number of filters and network complexity. FIR filters can be used to achieve non-Digui way, not in a limited precision of a shock, and into the homes and quantitative factors of uncertainty arising from the impact of errors than IIR filter small number, and FIR filter can be used FFT algorithms, the computational speed. But unlike IIR filter can filter through the simulation results, there is no ready-made formula FIR filter must use computer-aided design software (such as MATLAB) to calculate. So, a broader application of FIR filters, and IIR filters are not very strict requirements on occasions.Unit from sub-functions can be divided into the following four categories :(1) Low-filter (LPF);(2) high-filter (HPF);(3) belt-filter (BPF);(4) to prevent filter (BSF).The following chart dotted line for the ideals of the filter frequency characteristics :A1(f) A2(f)10 c2f c2 f(a) (b)A3(f) A4(f)0 c1c2f c1c2 f(c) (d)(a)LPF (b)HPF (c)BPF (d)BSF2、 MATLAB introducedMATLAB is a matrix laboratory (Matrix Laboratory) is intended. In addition to an excellent value calculation capability, it also provides professional symbols terms, word processing, visualization modeling, simulation and real-time control functions. MATLAB as the world's top mathematical software applications, with a strong engineering computing, algorithms research, engineering drawings, applications development, data analysis and dynamic simulation, and other functions, in aerospace, mechanical manufacturing and construction fields playing an increasingly important role. And the C language function rich, the use of flexibility, high-efficiency goals procedures. High language both advantages as well as low level language features. Therefore, C language is the most widely used programming language. Although MATLAB is a complete, fully functional programming environment, but in some cases, data and procedures with the external environment of the world is very necessary and useful. Filter design using Matlab, could be adjusted with the design requirements and filter characteristics of the parameters, visual simple, greatly reducing the workload for the filter design optimization.In the electricity system protection and secondary computer control, many signal processing and analysis are based on are certain types Yeroskipou and the second harmonics of the system voltage and current signals (especially at D process), are mixed with a variety of complex components, the filter has been installed power system during the critical components. Current computer protection and the introduction of two digital signal processing software main filter. Digital filter design using traditional cumbersome formula, the need to change the parameters after recalculation, especially in high filters, filter design workload. Uses MATLAB signal processing boxes can achieve rapid and effective digital filter design and simulation.MATLAB is the basic unit of data matrix, with its directives Biaodashi mathematics, engineering, commonly used form is very similar, it is used to solve a problem than in MATLAB C, Fortran and other languages End precision much the same thing. The popular MATLAB 5.3/Simulink3.0 including hundreds of internal function with the main pack and 30 types of tool kits (Toolbox). kits can be divided into functional tool kits and disciplines toolkit. MATLAB tool kit used to expand the functional symbols terms, visualization simulation modelling, word processing and real-time control functions. professional disciplines toolkit is a stronger tool kits, tool kits control, signal processing tool kit, tool kits, etc. belonging to such communicationsMATLAB users to open widely welcomed. In addition to the internal function, all the packages MATLAB tool kits are readable document and the document could be amended, modified or users through Yuanchengxu the construction of new procedures to prepare themselves for kits.3、 Digital filter designDigital filter design of the basic requirementsDigital filter design must go through three steps :(1) Identification of indicators : In the design of a filter, there must be some indicators. These indicators should be determined on the basis of the application. In many practical applications, digital filters are often used to achieve the frequency operation. Therefore, indicators in the form of general jurisdiction given frequency range and phase response. Margins key indicators given in two ways. The first is absolute indicators. It provides a function to respond to the demands of the general application of FIR filter design. The second indicator is the relative indicators. Its value in the form of answers to decibels. In engineering practice, the most popular of such indicators. For phase response indicators forms, usually in the hope that the system with a linear phase frequency bands human. Using linear phase filter design with thefollowing response to the indicators strengths:①it only contains a few algorithms, no plural operations;②there is delay distortion, only a fixed amount of delay; ③the filter length N (number of bands for N-1), the volume calculation for N/2 magnitude.(2) Model approach : Once identified indicators can use a previous study of the basic principles and relationships, a filter model to be closer to the target system.(3) Achieved : the results of the above two filters, usually by differential equations, system function or pulse response to describe. According to this description of hardware or software used to achieve it.4、 Introduced FPGA Programmable logic device is a generic logic can use a variety of chips, which is to achieve ASIC ASIC (Application Specific Integrated Circuit) semi-customized device, Its emergence and development of electronic systems designers use CAD tools to design their own laboratory in the ASIC device. Especially FPGA (Field Programmable Gate Array) generated and development, as a microprocessor, memory, the figures for electronic system design and set a new industry standard (that is based on standard product sales catalogue in the market to buy). Is a digital system for microprocessors, memories, FPGA or three standard building blocks constitute their integration direction.Digital circuit design using FPGA devices, can not only simplify the design process and can reduce the size and cost of the entire system, increasing system reliability. They do not need to spend the traditional sense a lot of time and effort required to create integrated circuits, to avoid the investment risk and become the fastest-growing industries of electronic devices group. Digital circuit design system FPGA devices using the following main advantages (1) Design flexibleUse FPGA devices may not in the standard series device logic functional limitations. And changes in system design and the use of logic in any one stage of the process, and only through the use of re-programming the FPGA device can be completed, the system design provides for great flexibility.(2) Increased functional densityFunctional density in a given space refers to the number of functional integration logic. Programmable logic chip components doors several high, a FPGA can replace several films, film scores or even hundreds of small-scale digital IC chip illustrated in the film. FPGA devices using the chip to use digital systems in small numbers, thus reducing the number of chips used to reduce the number of printed size and printed, and will ultimately lead to a reduction in the overall size of the system.(3) Improve reliabilityPrinting plates and reduce the number of chips, not only can reduce system size, but it greatly enhanced system reliability. A higher degree ofintegration than systems in many low-standard integration components for the design of the same system, with much higher reliability. FPGA device used to reduce the number of chips required to achieve the system in the number printed on the cord and joints are reduced, the reliability of the system can be improved.(4) Shortening the design cycleAs FPGA devices and the programmable flexibility, use it to design a system for longer than traditional methods greatly shortened. FPGA device master degrees high, use printed circuit layout wiring simple. At the same time, success in the prototype design, the development of advanced tools, a high degree of automation, their logic is very simple changes quickly. Therefore, the use of FPGA devices can significantly shorten the design cycle system, and speed up the pace of product into the market, improving product competitiveness.(5) Work fastFPGA/CPLD devices work fast, generally can reach several original Hertz, far larger than the DSP device. At the same time, the use of FPGA devices, the system needed to achieve circuitclasses and small, and thus the pace of work of the entire system will be improved.(6) Increased system performance confidentialityMany FPGA devices have encryption functions in the system widely used FPGA devices can effectively prevent illegal copying products were others(7) To reduce costsFPGA device used to achieve digital system design, if only device itself into the price, sometimes you would not know it advantages, but there are many factors affecting the cost of the system, taken together, the cost advantages of using FPGA is obvious. First, the use of FPGA devices designed to facilitate change, shorten design cycles, reduce development costs for system development; Secondly, the size and FPGA devices allow automation needs plug-ins, reducing the manufacturing system to lower costs; Again, the use of FPGA devices can enhance system reliability, reduced maintenance workload, thereby lowering the cost of maintenance services for the system. In short, the use of FPGA devices for system design to save costs.FPGA design principles :FPGA design an important guiding principles : the balance and size and speed of exchange, the principles behind the design of the filter expression of a large number of certification.Here, "area" means a design exertion FPGA/CPLD logic resources of the FPGA can be used to the typical consumption (FF) and the search table (IUT) to measuremore general measure can be used to design logic equivalence occupied by the door is measured. "pace" means stability operations in the chip design can achieve the highest frequency, the frequency of the time series design situation, and design to meet the clock cycle -- PADto pad, Clock Setup Time, Clock Hold Beijing, Clock-to-Output Delay, and other characteristics of many time series closely related. Area (area) and speed (speed) runs through the two targets FPGA design always is the ultimate design quality evaluation criteria. On the size and speed of the two basic concepts : balance of size and speed and size and speed of swap.One pair of size and speed is the unity of opposites contradictions body. Requirements for the design of a design while the smallest, highest frequency of operation is unrealistic. More scientific goal should be to meet the design requirements of the design time series (includes requirements for the design frequency) premise, the smallest chip area occupied. Or in the specified area, the design time series cushion greater frequency run higher. This fully embodies the goals of both size and speed balanced thinking. On the size and speed requirements should not be simply interpreted as raising the level and design engineers perfect sexual pursuit, and should recognize that they are products and the quality and cost of direct relevance. If time series cushion larger design, running relatively high frequency, that the design Jianzhuangxing stronger, more quality assurance system as a whole; On the other hand, the smaller size of consumption design is meant to achieve in chip unit more functional modules, the chip needs fewer, the entire system has been significantly reduced cost. As a contradiction of the two components, the size and speed is not the same status. In contrast, meet the timetables and work is more important for some frequency when both conflicts, the use of priority guidelines.Area and the exchange rate is an important FPGA design ideas. Theoretically, if a design time series cushion larger, can run much higher than the frequency design requirements, then we can through the use of functional modules toreduce the consumption of the entire chip design area, which is used for space savings advantages of speed; Conversely, if the design of a time series demanding, less than ordinary methods of design frequency then generally flow through the string and data conversion, parallel reproduction of operational module, designed to take on the whole "string and conversion" and operate in the export module to chip in the data "and string conversion" from the macro point of view the whole chip meets the requirements of processing speed, which is equivalent to the area of reproduction - rate increase.For example. Assuming that the digital signal processing system is 350Mb/s input data flow rate, and in FPGA design, data processing modules for maximum processing speed of 150Mb/s, because the data throughput processing module failed to meet requirements, it is impossible to achieve directly in the FPGA. Such circumstances, they should use "area-velocity" thinking, at least three processing modules from the first data sets will be imported and converted, and then use these three modules parallel processing of data distribution, then the results "and string conversion," we have complete data rate requirements. We look at both ends of the processing modules, data rate is 350Mb/s, and in view of the internal FPGA, each sub-module handles the data rate is 150Mb/s, in fact, all the data throughput is dependent on three security modules parallel processing subsidiary completed, that is used by more chip area achieve high-speed processing through "the area of reproduction for processing speed enhancement" and achieved design.FPGA is the English abbreviation Field of Programmable Gate Array for the site programmable gate array, which is in Pal, Gal, Epld, programmable device basis to further develop the product. It is as ASIC (ASIC) in the field of a semi-customized circuit and the emergence of both a customized solution to the shortage circuit, but overcome the original programmable devices doors circuit few limited shortcomings.FPGA logic module array adopted home (Logic Cell Array), a new concept of internal logic modules may include CLB (Configurable Logic Block), export import module IOB (Input Output Block) and internal links (Interconnect) 3. FPGA basic features are :(1) Using FPGA ASIC design ASIC using FPGA circuits, the chip can be used,while users do not need to vote films production.(2) FPGA do other customized or semi-customized ASIC circuits throughout the Chinese specimen films.3) FPGA internal capability and rich I/O Yinjue.4) FPGA is the ASIC design cycle, the shortest circuit, the lowest development costs, risks among the smallest device5) FPGA using high-speed Chmos crafts, low consumption, with CMOS, TTL low-power compatibleIt can be said that the FPGA chip is for small-scale systems to improve system integration, reliability one of the bestCurrently FPGA many varieties, the Revenue software series, TI companies TPC series, the fiex ALTERA company series FPGA is stored in films from theinternal RAM procedures for the establishment of the state of its work, therefore, need to programmed the internal Ram. Depending on the different configuration, users can use a different programming methodsPlus electricity, FPGA, EPROM chips will be read into the film, programming RAM 中 data, configuration is completed, FPGA into working order. Diaodian, FPGA resume into white films, the internal logic of relations disappear, FPGA to repeated use. FPGA's programming is dedicated FPGA programming tool, using generic EPROM, prom programming device can. When the need to modify functional FPGA, EPROM can only change is. Thus, with a FPGA, different programming datato produce different circuit functions. Therefore, the use of FPGA very flexible.There are a variety of FPGA model : the main model for a parallel FPGA plus a EPROM manner; From the model can support a number of films FPGA; serial prom programming model could be used serial prom FPGA programming FPGA; The external model can be engineered as microprocessors from its programming microprocessors.Verilog HDL is a hardware description language for the algorithm level, doors at the level of abstract level to switch-level digital system design modelling. Modelling of the target figure by the complexity of the system can be something simple doors and integrity of electronic digital systems. Digital system to the levels described, and in the same manner described in Hin-time series modelling.Verilog HDL language with the following description of capacity : design behaviour characteristics, design data flow characteristics, composition and structure designed to control and contain the transmission and waveform design a certification mechanism. All this with the use of a modelling language. In addition, Verilog HDL language programming language interface provided by the interface in simulation, design certification from the external design of the visit, including specific simulation control and operation.Verilog HDL language grammar is not only a definition, but the definition of each grammar structure are clear simulation, simulation exercises. Therefore, the use of such language to use Verilog simulation models prepared by a certification. From the C programming language, the language inherited multiple operating sites and structures. Verilog HDL provides modelling capacity expansion, many of the initial expansion would be difficult to understand. However, the core subsets of Verilog HDL language very easy to learn and use, which is sufficient for most modelling applications. Of course, the integrityof the hardware description language is the most complex chips from theintegrity of the electronic systems described.historyVerilog HDL language initially in 1983 by Gateway Design Automation companies for product development simulator hardware modelling language. Then it is only a dedicated language. Since their simulation, simulation devices widely used products, Verilog HDL as a user-friendly and practical language for many designers gradually accepted. In an effort to increase the popularity of the language activities, Verilog HDL language in 1990 was a public area. Open Verilog International (OVI) is to promote the development of Verilog international organizations. 1992, decided to promote OVI OVI standards as IEEE Verilog standards. The effort will ultimately succeed, a IEEE1995 Verilog language standard, known as IEEE Std 1364-1995. Integrity standards in Verilog hardware description language reference manual contains a detailed description.Main capacity:Listed below are the main Verilog hardware description language ability*Basic logic gate, and, for example, or have embedded in the language and nand * Users of the original definition of the term (UDP), the flexibility. Users can be defined in the original language combinations logic original language, the original language of logic could also be time series* Switches class infrastructure models, such as the nmos and pmos also be embedded in the language* Hin-language structure designated for the cost of printing the design and trails Shi Shi and design time series checks.* Available three different ways to design or mixed mode modelling. These methods include : acts described ways - use process of structural modelling; Data flow approach - use of a modelling approach Fuzhi expression; Structured way - using examples of words to describe modular doors and modelling.* Verilog HDL has two types of data : data types and sequence data line network types. Line network types that the physical links between components and sequence types that abstract data storage components.* To describe the level design, the structure can be used to describe any level module example * Design size can be arbitrary; Language is design size (size) impose any restrictions* Verilog HDL is no longer the exclusive language of certain companies but IEEE standards. * And the machine can read Verilog language, it may as EDA tools and languages of the world between the designers* Verilog HDL language to describe capacity through the use of programming language interface (PLI) mechanism further expansion. PLI is to allow external functions of the visit Verilog module information, allowing designers and simulator world Licheng assembly* Design to be described at a number of levels, from the switch level, doors level, register transfer level (RTL) to the algorithm level, including thelevel of process and content* To use embedded switching level of the original language in class switch design integrity modelling* Same language can be used to generate simulated incentive and certification by the designated testing conditions, such as the value of imports of the designated*Verilog HDL simulation to monitor the implementation of certification, the certification process of implementing the simulation can be designed to monitorand demonstrate value. These values can be used to compare with the expectations that are not matched in the case of print news reports.* Acts described in the class, not only in the RTL level Verilog HDL design description, and to describe their level architecture design algorithm level behavioural description* Examples can use doors and modular structure of language in a class structure described* Verilog HDL mixed mode modelling capabilities in the design of a different design in each module can level modelling* Verilog HDL has built-in logic function, such as*Structure of high-level programming languages, such as conditions of expression, and the cycle of expression language, language can be used* To it and can display regular modelling* Provide a powerful document literacy* Language in the specific circumstances of non-certainty that in the simulator, different models can produce different results; For example, describing events in the standard sequence of events is not defined.5、In troduction of DSPToday, DSP is w idely used in the modern techno logy and it has been the key part of many p roducts and p layed more and mo re impo rtant ro le in our daily life.Recent ly, Northw estern Po lytechnica lUniversity Aviation Microelect ronic Center has comp leted the design of digital signal signal p rocesso r co re NDSP25, w h ich is aim ing at TM S320C25 digital signal p rocesso r of Texas Inst rument TM S320 series. By using top 2dow n design flow , NDSP25 is compat ible w ith inst ruct ion and interface t im ing of TM S320C25.Digital signal processors (DSP) is a fit for real-time digital signal processing for high-speed dedicated processors, the main variety used for real-time digital signal processing to achieve rapid algorithms. In today's digital age background, the DSP has become the communications, computer, and consumer electronics products, and other fields based device.Digital signal processors and digital signal processing is inseparably, we usually say "DSP" can also mean the digital signal processing (Digital Signal Processing), is that in this digital signal processors Lane. Digital signal processing is a cover many disciplines applied to many areas and disciplines, refers to the use of computers or specialized processing equipment, the signals in digital form for the collection, conversion, recovery, valuation, enhancement, compression, identification, processing, the signals are compliant form. Digital signal processors for digital signal processing devices, it is accompanied by a digital signal processing to produce. DSP development process is broadly divided into three phases : the 20th century to the 1970s theorythat the 1980s and 1990s for the development of products. Before the emergence of the digital signal processing in the DSP can only rely on microprocessors (MPU) to complete. However, the advantage of lower high-speed real-time processing can not meet the requirements. Therefore, until the 1970s, a talent made based DSP theory and algorithms. With LSI technology development in 1982 was the first recipient of the world gave birth to the DSP chip. Years later, the second generation based on CMOS工艺 DSP chips have emerged. The late1980s, the advent of the third generation of DSP chips. DSP is the fastest-growing 1990s, there have been four successive five-generation and the generation DSP devices. After 20 years of development, the application of DSP products has been extended to people's learning, work and all aspects of life and gradually become electronics products determinants.中文翻译。

数字信号处理

数字信号处理

数字信号处理数字信号处理(Digital signal processing,DSP)是一门广泛应用于信号处理领域的技术。

传统的信号处理技术是指将连续信号进行分析和处理,而数字信号处理则是指将连续信号通过采样和量化的方式转化为离散信号,然后对这些离散信号进行数字化的运算和处理。

数字信号处理的基本原理是将模拟信号转换为数字信号,然后按照数学模型进行数字信号的处理,最后再通过数字信号转换回模拟信号。

数字信号处理在现代通信、音频、视频、图像、控制等领域得到了广泛的应用,几乎每个人都在日常生活中体验到了数字信号处理的便捷性和高效性。

一、数字信号处理的基础1.离散时间系统:数字信号处理中的离散时间系统(discrete time system)是指使用离散的时序来描述的系统,该系统输入和输出的信号都是离散信号。

离散时间系统有多种类型,包括差分方程系统、线性时不变系统(LTI)和非线性时变系统(NLTV)等。

2.数字信号:数字信号是时域离散和幅度量化的信号,可以通过采样和量化的方式将连续信号转变为离散信号。

数字信号可以用一系列的数字来表示,由于数字信号处于离散状态,因此操作数域也是离散的。

3.频域:频域是指信号在频率上的展示,包括信号的功率谱、频谱和相位谱等等。

数字信号处理中,频域变换是一种将时域信号转换为频域信号的变换,常见的频域变换包括傅里叶变换、快速傅里叶变换和Z变换等。

4.量化:量化是将模拟信号转化为数字信号的必要步骤,它将连续和无限的模拟信号转化为离散和有限的数字信号。

量化方法包括线性量化和非线性量化两种,其中非线性量化更适用于高动态范围(HDR)信号等应用场合。

二、数字信号处理的应用数字信号处理在通讯、音频、视频、图像等领域得到广泛应用。

下面是其中几个应用领域的浅析。

1.通信:数字信号处理在通信领域中最广泛的应用之一是数字调制和解调。

数字调制将数字信号转化为模拟信号,然后发送到接收端。

在接收端,通过数字解调将模拟信号转化为数字信号。

电子信息与通信工程专业英语课文翻译3.2

电子信息与通信工程专业英语课文翻译3.2

3.2 数字信号处理1 简介数字信号处理是21世纪用于科学和工程领域最强大的技术之一,它使一个广阔的领域发生了革命性的改变:通信,医学影像,雷达或声纳,高保真音乐复制,石油勘测,以上只是列举几个。

每个领域都有它自身独特的算法(algorithm),数学运算(mathematic),专用工艺(specialized technique)。

数字信号处理在计算机科学方面有别于其他领域,因为他采用一种特殊的数据类型:信号。

现代社会中,我们的身边充满各种类型的信号。

有些信号是天然形成的,但大多数是人为制造的。

有些信号是必要的(语音),有些是宜人的(音乐),而有些信号在某个特定的场合是不需要或不必要的。

在大多数情况下,这些信号来源于人对真实世界的感觉,比如地震的震动,视觉图像,声音波形等。

数字信号处理是一种数学工具,是一种用来处理那些将上述信号转换成数字形式后的信号的算法和技术。

这包括一系列目的,如:视觉图像的优化处理,语音识别和生成,数据压缩存储和传输等。

在工程范围内,信号是信息的载体,既有益又有害。

信号处理中最简单的形式是从一连串相互矛盾的信息中提取和增强有用信息。

信息的有用和无用往往只是主管和客观的区别。

因此信号处理往往依赖于应用程序。

傅里叶分析和滤波器设计是信号处理时常用的方法。

他们的原则简单描述如下。

2 傅里叶分析函数的傅里叶表示,即将函数表示成正弦和余弦信号的叠加,这种方法已经广泛用于微分方程的解析法和数值法求解过程以及通信信号的分析和处理。

傅里叶变换的效用在于它能够在时域范围内分析它的频率内容。

变换的第一步是将时域上的函数转换为时域表示。

(The transform works by first translating a function in the time domain into a function in the frequency domain)。

然后就可以分析信号的频率内容了。

因为变换函数的傅里叶系数代表各个正弦和余弦函数在各自对应频率区间的分配。

数字信号处理 英文 教材

数字信号处理 英文 教材

数字信号处理英文教材
数字信号处理是一门涉及数学、工程学和计算机科学的学科,其英文教材有很多,以下是一些经典的数字信号处理英文教材:
1. "Digital Signal Processing" by John G. Proakis and Masoud Salehi
2. "Digital Signal Processing: Principles, Algorithms, and Applications" by Robert J. Schroeder and Ray P. Kailath
3. "Digital Signal Processing: A Practical Guide for Applications Engineers" by Robert A. Monzingo and John W. Cocke
4. "Digital Signal Processing: A Computer-Based Approach" by John
G. Proakis and Dimitris G. Manolakis
5. "Digital Signal Processing: Theory, Algorithms, and Practicalities" by John G. Proakis and Gilbert H. Walker
这些教材都是非常经典的数字信号处理教材,被广泛应用于数字信号处理领域。

它们涵盖了数字信号处理的基本概念、原理、算法和应用,适合初学者和有一定基础的读者。

数字信号处理的简单介绍文献翻译及英文原文

数字信号处理的简单介绍文献翻译及英文原文

单位代码01学号070110105分类号密级文献翻译数字信号处理的简单论述院(系)名称信息工程学院专业名称通信工程学生姓名徐治明指导教师赵春雨2011年 4 月 5 日英文译文数字信号处理的简单论述冈萨雷斯(美国)一、数字信号处理的概述数字信号处理是将信号以数字方式表示并处理的理论和技术。

数字信号处理与模拟信号处理是信号处理的子集。

数字信号处理的目的是对真实世界的连续模拟信号进行测量或滤波。

因此在进行数字信号处理之前需要将信号从模拟域转换到数字域,这通常通过模数转换器实现。

而数字信号处理的输出经常也要变换到模拟域,这是通过数模转换器实现的。

数字信号处理的算法需要利用计算机或专用处理设备如数字信号处理器(DSP)和专用集成电路(ASIC)等。

数字信号处理技术及设备具有灵活、精确、抗干扰强、设备尺寸小、造价低、速度快等突出优点,这些都是模拟信号处理技术与设备所无法比拟的。

数字信号处理的核心算法是离散傅立叶变换(DFT),是DFT使信号在数字域和频域都实现了离散化,从而可以用通用计算机处理离散信号。

而使数字信号处理从理论走向实用的是快速傅立叶变换(FFT),FFT的出现大大减少了DFT的运算量,使实时的数字信号处理成为可能、极大促进了该学科的发展。

世界上三大DSP芯片生产商:1.德克萨斯仪器公司(TI) 2.模拟器件公司(ADI) 3.摩托罗拉公司(Motorola).这三家公司几乎垄断了通用DSP芯片市场。

数字信号处理的经典书籍是麻省理工学院奥本海姆编著的《Discrete Time Signal Processing》,有中译本《离散时间信号处理》由西安交通大学出版。

现在是第二版。

二、特征和分类信号(signal)是一种物理体现,或是传递信息的函数。

而信息是信号的具体内容。

模拟信号(analog signal):指时间连续、幅度连续的信号。

数字信号(digital signal):时间和幅度上都是离散(量化)的信号。

《数字信号处理(英文)》Maria Elena Angoletta

《数字信号处理(英文)》Maria Elena Angoletta

Bandwidth: indicates rate of change of a signal. High bandwidth signal changes fast.
Warning: formal description makes use
of “negative” frequencies !
0.3
Digital
Discrete function Vk of discrete sampling variable tk, with k = integer: Vk = V(tk).
0.3
0.1 0 -0.1 -0.2 0 2 4 6 time [ms] 8 10
Voltage [V]
Voltage [V]
• General purpose processors (GPP), -controllers.
Hardware
• Digital Signal Processors (DSP). • Programmable logic ( PLD, FPGA ).
Fang
• Reproducibility.
M. E. Angoletta - DISP2003 -
From analog to digital domain
4 / 30
DSPing: aim & tools
Applications
• Predicting a system‟s output. • Implementing a certain processing task. • Studying a certain signal.
M. E. Angoletta - DISP2003 -
From analog to digital domain

科技英语 5数字信号处理器原文与翻译

科技英语 5数字信号处理器原文与翻译

Words and Expressionsfollow v.遵循memory n.存储器register n.寄存器access v.访问overlap v. 重叠pipelining n. 流水线操作multiplier n. 乘法器accumulator n. 累加器shifter n.移位器reference n. 寻址mantissa n.尾数exponent n. 指数cycle n. 机器周期customize v.定制,用户化package v.封装digital signal processor 数字信号处理器von Neumann architecture 冯·诺伊曼结构shared single memory 单一共享存储器program instruction 程序指令harvard architecture 哈佛结构fetch from 从…获取circular buffer 循环缓冲区,环形缓冲区address generator 地址产生器fixed point 定点floating point 浮点binary point 二进制小数点available precision 可用精度dynamic range 动态范围scale range 量程smallest Resolvable Difference 最小分辨率scientific notation 科学计数法assembly language 汇编语言multi-function instructions 多功能指令parallel architecture 并行结构looping scheme 循环机制sampling frequency 采样频率on-chip memory 片内存储器well-matched 非常匹配software tools 软件开发工具low level programming language 低级编程语言high level programming language 高级编程语言third party software 第三方软件board level product 板级产品data register 数据寄存器ALU=Arithmetic Logical Unit 运算逻辑单元program sequencer 程序定序器peripheral sections 外设single integrated circuit 单片集成电路cellular telephone 蜂窝电话printed circuit board 印刷电路板licensing agreement 专利使用权转让协定custom devices 定制器件extra memory 附加存储器stand alone 单机third party developer 第三方开发商multimedia operations 多媒体操作merged into 融合calculation-intensive algorithm运算密集型算法Unit 5 Digital Signal ProcessorsDigital signal processing tasks can be performed by all processors. Specialized digital signal processors(DSPs), however, perform these tasks most efficiently and most quickly. While traditional processors follow the Von Neumann architecture[]1model, which assumes a shared single memory to be used for both program instructions and data, DSPs use the Harvard or modified Harvard architecture []2, which includes multiple program and data memories, along with multiple buses to access them. This arrangement means that much less waiting is required when instructions or numbers are fetched from memory. In fact at least one of each can be fetched simultaneously. Such overlapping of tasks is called pipelining. In addition to multiple memories and buses, all DSPs have fast multipliers, accumulators, and shifters, and many have hardware support for circular buffers. Address generators can speed up accesses to memory locations referenced by registers.DSPs are available in two major classes: fixed point and floating point. The fixed point class represents real numbers in a fixed number of bits. The position of the binary point (similar to the decimal point) can be controlled by the programmer, and determines the range of numbers that can be represented. As the range increases, though, the available precision goes down, since fewer bits lie to the right of the binary point. In 16 bits, the formats 16.0, 15.1, 14.2, 13.3, 12.4, 11.5, 10.6, 9.7, 8.8, 7.9, 6.10, 5.11, 4.12, 3.13, 2.14, and 1.15 are possible. The dynamic range, calculated as 20log (Full Scale Range/Smallest2= 96.3 dB.Resolvable Difference), remains the same for all 16-bit formats, 20log16Figure 6.3 Van Neumann architectureFigure 6.4 Harvard architectureFloating point DSPs represent real numbers using a mantissa and an exponent , similar to scientific notation : Many combine mantissa and exponent into a 32-bit number. The dynamic range for floating point devices is calculated from the largest and smallest multipliers E 2, where E is the exponent. Thus, for a representation that uses 24 bits for the mantissa and 8 bits for the signed exponent, the dynamic range is 20 log (1281272/2-) = 1535.3 dB. A large dynamic range means the system has great power to represent a wide range of input signals, from very small to very large.Assembly language is the command language for DSPs. DSPs often have specialized instructions that make programming for common DSP tasks more convenient and more efficient. For example, most DSPs offer multi-function instructions that exploit their parallel architecture . Other constructs that are frequently offered are efficient looping schemes , since so many DSP operations involve a great deal of repetition.Choosing a DSP for a particular application is not always easy. The first decision is on whether tochoose a fixed point or a floating point device []3. Generally, fixed point devices are cheaper and quicker,but floating point devices are more convenient to program and more suited to calculation-intensive algorithms . Second, the data width of the DSP determines how accurately it can represent numbers. Speed is another issue, not only how many cycles occur in each second, but also how many instructions execute in each cycle and how much work each of these instructions accomplishes. One way to assess the minimum requirements for the DSP is to estimate how many instructions must be executed for each received sample. When this number is multiplied by the sampling frequency , the minimum required number of instructions per second is obtained.The specific hardware and software features offered by a particular DSP can make one choice betterthan another, as can the amount of on-chip memory available []4. Sometimes DSPs are chosen becausewell-matched supporting hardware, particularly A/D and D/A converters, is obtainable. Frequently, the quality and convenience of the software tools, for both low level and high level programming languages, are also major factors, as is the availability of third party software. As always, cost is a factor. In fact, quite often, the DSP that is fastest and offers the most features, but also fits the budget, is the one selected.DSPs can be purchased in three forms, as a core, as a processor, and as a board level product. In DSP, the term "core" refers to the section of the processor where the key tasks are carried out, including the data registers, multiplier, ALU, address generator, and program sequencer. A complete processor requires combining the core with memory and interfaces to the outside world. While the core and these peripheral sections are designed separately, they will be fabricated on the same piece of silicon, making the processor a single integrated circuit.Suppose you build cellular telephones and want to include a DSP in the design. You will probably want to purchase the DSP as a processor, that is, an integrated circuit that contains the core, memory and other internal features. To incorporate this IC in your product, you have to design a printed circuit board where it will be soldered in next to your other electronics. This is the most common way that DSPs are used.Now, suppose the company you work for manufactures its own integrated circuits. In this case, you might not want the entire processor, just the design of the core. After completing the appropriate licensing agreement, you can start making chips that are highly customized to your particular application. This gives you the flexibility of selecting how much memory is included, how the chip receives and transmits data, how it is packaged, and so on.Custom devices of this type are an increasingly important segment of the DSP marketplace.There are several dozen companies that will sell you DSPs already mounted on a printed circuit board. These have such features as extra memory, A/D and D/A converters, EPROM sockets, multiple processors on the same board, and so on. While some of these boards are intended to be used as stand alone computers, most are configured to be plugged into a host, such as a personal computer. Companies that make these types of boards are called Third Party Developers. The best way to find them is to ask the manufacturer of the DSP you want to use. Look at the DSP manufacturer's website; if you don't find a list there, send them an e-mail. They will be more than happy to tell you who are using their products and how to contact them.Keep in mind that the distinction between DSPs and other microprocessors is not always a clear line. For instance, look at how Intel describes the MMX technology addition to its Pentium processor: "Intel engineers have added 57 powerful new instructions specifically designed to manipulate and process video, audio and graphical data efficiently. These instructions are oriented to the highly parallel, repetitivesequences often found in multimedia operations . "In the future, we will undoubtedly see more DSP-like functions merged into traditional microprocessors and microcontrollers. The Internet and other multimedia applications are a strong driving force for these changes. These applications are expanding so rapidly, in twenty years it is very possible that the Digital Signal Processor may be the "traditional" microprocessor.Notes1. “冯·诺伊曼结构”取名字美国杰出的数学家—约翰·冯·诺伊曼(John Von Neumann,1903~1957)。

Unit 09 译文

Unit 09 译文

Unit 9 数字信号和信号处理Unit 9-1第一部分:数字信号处理数字信号处理(DSP)是研究数字表示的信号以及这些信号的处理方法。

数字信号处理和模拟信号处理是信号处理的子领域。

数字信号处理包括音频及语音信号处理、声纳和雷达信号处理、传感器阵列处理、谱估计、统计信号处理、图像处理、通信信号处理、生物医学信号处理等子领域。

数字信号处理的目标通常是测量连续的真实世界的模拟信号或对其滤波,因此,第一步常常是使用模数转换器将信号从模拟形式转换成数字形式。

通常,要求的输出信号为另一个模拟输出信号,这就需要数模转换器。

数字信号处理的算法有时通过使用专用计算机来实现,它们(专用计算机)利用被称为数字信号处理器的专用微处理器(简称DSP)。

这些数字信号处理器实时处理信号,通常是针对具体目的而设计的专用集成电路(ASIC)。

当灵活性和快速开发比大批量生产的成本更重要时,DSP算法也可以用现场可编程门阵列来实现。

数字信号处理域在数字信号处理中,工程师通常在下面几个域的一个域中来研究数字信号:时域(一维信号),空域(多维信号),频域,自相关域以及小波域。

他们按照某些依据来猜测(或试验不同的可能性)那一个域能够最好地表示信号的本质特性来选择在其中进行信号处理的域。

从测量设备得到的样本序列产生(信号的)时域或空域表示,而离散Fourier变换则产生频域表示即频谱。

自相关定义为信号与其自身经过时间或空间间隔变化后的互相关。

信号采样随着计算机应用的增长,数字信号处理的使用和需求日益增多。

为了能够在计算机上使用模拟信号,必须使用模数转换器(ADC)对其进行数字化。

采样通常分两步实现:离散化和量化。

在离散化阶段,信号空间被分割为相等的区间,用相应区间的代表性信号值代替信号本身。

在量化阶段,用有限集中的值来近似代表性的信号值。

为了能够正确地重建被采样的模拟信号,必须满足奈奎斯特-香农采样定理。

定理规定:采样频率必须大于两倍的信号带宽。

数字信号处理课程介绍(英文版)

数字信号处理课程介绍(英文版)

Introduction of Digital Signal Processing CourseCourse cod(work out unified by school)Course name: Digital Signal ProcessingChinese name: 数字信号处理credits: 3 attendance time:Fall of juniorschool target:Technical instituteCourse leader: Xiaoying Hong、lecturer、masterCourse descriptionThe course is one of basic courses for Electrical Engineering and Automation. It can make use of a computer and professional equipment to do acquisition of signal、transform、synthesize、Valuation and discern, etc ,so as to withdraw the information and use. This curriculum mainly specializes more in telling about the basic theory, it also introduces some efficient algorithm,the both software and hardware-implemented about these theories. Through the course study,the students will master the basic concept 、basic method on Digital Signal Processing to lay the foundation for further study.Practical trainingIt’s necessary to carry on the important experiment (hands-on) in the study. On one hand, it can deepen the understanding on basic theory; on the other hand, it can cultivate the ability to solve concrete problems. This course contains 18 classes of experiments (hands-on) which are teaching material appendixes primarily. The educational software is MATLAB.Checking:Homework 10%; Hands-on Lab 20%; final exam70%Prescribed textbook:Digital Signal Processing,Yumei Ding ,Xidian University Press,January 2001,2nd Edition.Bibliography:【1】Digital Signal Processing,Peiqing Cheng,Tsinghua University Press,August 2001,2nd Edition【2】Digital Signal Processing,Guangshu Hu,Tsinghua University Press,August 2003,2nd Edition.。

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单位代码01学号070110105分类号密级文献翻译数字信号处理的简单论述院(系)名称信息工程学院专业名称通信工程学生姓名徐治明指导教师赵春雨2011年 4 月 5 日英文译文数字信号处理的简单论述冈萨雷斯(美国)一、数字信号处理的概述数字信号处理是将信号以数字方式表示并处理的理论和技术。

数字信号处理与模拟信号处理是信号处理的子集。

数字信号处理的目的是对真实世界的连续模拟信号进行测量或滤波。

因此在进行数字信号处理之前需要将信号从模拟域转换到数字域,这通常通过模数转换器实现。

而数字信号处理的输出经常也要变换到模拟域,这是通过数模转换器实现的。

数字信号处理的算法需要利用计算机或专用处理设备如数字信号处理器(DSP)和专用集成电路(ASIC)等。

数字信号处理技术及设备具有灵活、精确、抗干扰强、设备尺寸小、造价低、速度快等突出优点,这些都是模拟信号处理技术与设备所无法比拟的。

数字信号处理的核心算法是离散傅立叶变换(DFT),是DFT使信号在数字域和频域都实现了离散化,从而可以用通用计算机处理离散信号。

而使数字信号处理从理论走向实用的是快速傅立叶变换(FFT),FFT的出现大大减少了DFT的运算量,使实时的数字信号处理成为可能、极大促进了该学科的发展。

世界上三大DSP芯片生产商:1.德克萨斯仪器公司(TI) 2.模拟器件公司(ADI) 3.摩托罗拉公司(Motorola).这三家公司几乎垄断了通用DSP芯片市场。

数字信号处理的经典书籍是麻省理工学院奥本海姆编著的《Discrete Time Signal Processing》,有中译本《离散时间信号处理》由西安交通大学出版。

现在是第二版。

二、特征和分类信号(signal)是一种物理体现,或是传递信息的函数。

而信息是信号的具体内容。

模拟信号(analog signal):指时间连续、幅度连续的信号。

数字信号(digital signal):时间和幅度上都是离散(量化)的信号。

数字信号可用一序列的数表示,而每个数又可表示为二制码的形式,适合计算机处理。

一维(1-D)信号: 一个自变量的函数。

二维(2-D)信号: 两个自变量的函数。

多维(M-D)信号: 多个自变量的函数。

系统:处理信号的物理设备。

或者说,凡是能将信号加以变换以达到人们要求的各种设备。

模拟系统与数字系统。

信号处理的内容:滤波、变换、检测、谱分析、估计、压缩、识别等一系列的加工处理。

多数科学和工程中遇到的是模拟信号。

以前都是研究模拟信号处理的理论和实现。

模拟信号处理缺点:难以做到高精度,受环境影响较大,可靠性差,且不灵活等。

数字系统的优点:体积小、功耗低、精度高、可靠性高、灵活性大、易于大规模集成、可进行二维与多维处理随着大规模集成电路以及数字计算机的飞速发展,加之从60年代末以来数字信号处理理论和技术的成熟和完善,用数字方法来处理信号,即数字信号处理,已逐渐取代模拟信号处理。

随着信息时代、数字世界的到来,数字信号处理已成为一门极其重要的学科和技术领域。

三、应用广义来说,数字信号处理是研究用数字方法对信号进行分析、变换、滤波、检测、调制、解调以及快速算法的一门技术学科。

但很多人认为:数字信号处理主要是研究有关数字滤波技术、离散变换快速算法和谱分析方法。

随着数字电路与系统技术以及计算机技术的发展,数字信号处理技术也相应地得到发展,其应用领域十分广泛。

四、数字滤波器数字滤波器的实用型式很多,大略可分为有限冲激响应型和无限冲激响应型两类,可用硬件和软件两种方式实现。

在硬件实现方式中,它由加法器、乘法器等单元所组成,这与电阻器、电感器和电容器所构成的模拟滤波器完全不同。

数字信号处理系统很容易用数字集成电路制成,显示出体积小、稳定性高、可程控等优点。

数字滤波器也可以用软件实现。

软件实现方法是借助于通用数字计算机按滤波器的设计算法编出程序进行数字滤波计算。

离散傅里叶变换的快速算法1965年J.W.库利和T.W.图基首先提出离散傅里叶变换的快速算法,简称快速傅里叶变换,以FFT表示。

自有了快速算法以后,离散傅里叶变换的运算次数大为减少,使数字信号处理的实现成为可能。

快速傅里叶变换还可用来进行一系列有关的快速运算,如相关、褶积、功率谱等运算。

快速傅里叶变换可做成专用设备,也可以通过软件实现。

与快速傅里叶变换相似,其他形式的变换,如沃尔什变换、数论变换等也可有其快速算法。

六、谱分析在频域中描述信号特性的一种分析方法,不仅可用于确定性信号,也可用于随机性信号。

所谓确定性信号可用既定的时间函数来表示,它在任何时刻的值是确定的;随机信号则不具有这样的特性,它在某一时刻的值是随机的。

因此,随机信号处理只能根据随机过程理论,利用统计方法来进行分析和处理,如经常利用均值、均方值、方差、相关函数、功率谱密度函数等统计量来描述随机过程的特征或随机信号的特性。

实际上,经常遇到的随机过程多是平稳随机过程而且是各态历经的,因而它的样本函数集平均可以根据某一个样本函数的时间平均来确定。

平稳随机信号本身虽仍是不确定的,但它的相关函数却是确定的。

在均值为零时,它的相关函数的傅里叶变换或Z变换恰恰可以表示为随机信号的功率谱密度函数,一般简称为功率谱。

这一特性十分重要,这样就可以利用快速变换算法进行计算和处理。

在实际中观测到的数据是有限的。

这就需要利用一些估计的方法,根据有限的实测数据估计出整个信号的功率谱。

针对不同的要求,如减小谱分析的偏差,减小对噪声的灵敏程度,提高谱分辨率等。

已提出许多不同的谱估计方法。

在线性估计方法中,有周期图法,相关法和协方差法;在非线性估计方法中,有最大似然法,最大熵法,自回归滑动平均信号模型法等。

谱分析和谱估计仍在研究和发展中。

数字信号处理的应用领域十分广泛。

就所获取信号的来源而言,有通信信号的处理,雷达信号的处理,遥感信号的处理,控制信号的处理,生物医学信号的处理,地球物理信号的处理,振动信号的处理等。

若以所处理信号的特点来讲,又可分为语音信号处理,图像信号处理,一维信号处理和多维信号处理等。

七、数字信号处理系统无论哪方面的应用,首先须经过信息的获取或数据的采集过程得到所需的原始信号,如果原始信号是连续信号,还须经过抽样过程使之成为离散信号,再经过模数转换得到能为数字计算机或处理器所接受的二进制数字信号。

如果所收集到的数据已是离散数据,则只须经过模数转换即可得到二进制数码。

数字信号处理器的功能是将从原始信号抽样转换得来的数字信号按照一定的要求,例如滤波的要求,加以适当的处理,即得到所需的数字输出信号。

经过数模转换先将数字输出信号转换为离散信号,再经过保持电路将离散信号连接起来成为模拟输出信号,这样的处理系统适用于各种数字信号处理的应用,只不过专用处理器或所用软件有所不同而已。

八、语音信号处理语音信号处理是信号处理中的重要分支之一。

它包括的主要方面有:语音的识别,语言的理解,语音的合成,语音的增强,语音的数据压缩等。

各种应用均有其特殊问题。

语音识别是将待识别的语音信号的特征参数即时地提取出来,与已知的语音样本进行匹配,从而判定出待识别语音信号的音素属性。

关于语音识别方法,有统计模式语音识别,结构和语句模式语音识别,利用这些方法可以得到共振峰频率、音调、嗓音、噪声等重要参数,语音理解是人和计算机用自然语言对话的理论和技术基础。

语音合成的主要目的是使计算机能够讲话。

为此,首先需要研究清楚在发音时语音特征参数随时间的变化规律,然后利用适当的方法模拟发音的过程,合成为语言。

其他有关语言处理问题也各有其特点。

语音信号处理是发展智能计算机和智能机器人的基础,是制造声码器的依据。

语音信号处理是迅速发展中的一项信号处理技术。

九、图像信号处理图像信号处理的应用已渗透到各个科学技术领域。

譬如,图像处理技术可用于研究粒子的运动轨迹、生物细胞的结构、地貌的状态、气象云图的分析、宇宙星体的构成等。

在图像处理的实际应用中,获得较大成果的有遥感图像处理技术、断层成像技术、计算机视觉技术和景物分析技术等。

根据图像信号处理的应用特点,处理技术大体可分为图像增强、恢复、分割、识别、编码和重建等几个方面。

这些处理技术各具特点,且正在迅速发展中。

十、振动信号处理机械振动信号的分析与处理技术已应用于汽车、飞机、船只、机械设备、房屋建筑、水坝设计等方面的研究和生产中。

振动信号处理的基本原理是在测试体上加一激振力,做为输入信号。

在测量点上监测输出信号。

输出信号与输入信号之比称为由测试体所构成的系统的传递函数(或称转移函数)。

根据得到的传递函数进行所谓模态参数识别,从而计算出系统的模态刚度、模态阻尼等主要参数。

这样就建立起系统的数学模型。

进而可以做出结构的动态优化设计。

这些工作均可利用数字处理器来进行。

这种分析和处理方法一般称为模态分析。

实质上,它就是信号处理在振动工程中所采用的一种特殊方法。

十一、地球物理信号处理为了勘探地下深处所储藏的石油和天然气以及其他矿藏,通常采用地震勘探方法来探测地层结构和岩性。

这种方法的基本原理是在一选定的地点施加人为的激震,如用爆炸方法产生一振动波向地下传播,遇到地层分界面即产生反射波,在距离振源一定远的地方放置一列感受器,接收到达地面的反射波。

从反射波的延迟时间和强度来判断地层的深度和结构。

感受器所接收到的地震记录是比较复杂的,需要处理才能进行地质解释。

处理的方法很多,有反褶积法,同态滤波法等,这是一个尚在努力研究的问题。

十二、生物医学信号处理信号处理在生物医学方面主要是用来辅助生物医学基础理论的研究和用于诊断检查和监护。

例如,用于细胞学、脑神经学、心血管学、遗传学等方面的基础理论研究。

人的脑神经系统由约100亿个神经细胞所组成,是一个十分复杂而庞大的信息处理系统。

在这个处理系统中,信息的传输与处理是并列进行的,并具有特殊的功能,即使系统的某一部分发生障碍,其他部分仍能工作,这是计算机所做不到的。

因此,关于人脑的信息处理模型的研究就成为基础理论研究的重要课题。

此外,神经细胞模型的研究,染色体功能的研究等等,都可借助于信号处理的原理和技术来进行。

信号处理用于诊断检查较为成功的实例,有脑电或心电的自动分析系统、断层成像技术等。

断层成像技术是诊断学领域中的重大发明。

X射线断层的基本原理是X射线穿过被观测物体后构成物体的二维投影。

接收器接收后,再经过恢复或重建,即可在一系列的不同方位计算出二维投影,经过运算处理即取得实体的断层信息,从而大屏幕上得到断层造像。

信号处理在生物医学方面的应用正处于迅速发展阶段。

数字信号处理在其他方面还有多种用途,如雷达信号处理、地学信号处理等,它们虽各有其特殊要求,但所利用的基本技术大致相同。

在这些方面,数字信号处理技术起着主要的作用。

摘自:冈萨雷斯(美国),数字信号处理处理(英文版),2009年12月。

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