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Unit3 A In its simplest form, a conventional op-amp consists of a differential amplifier (bipolar or FFT) followed by offset compensation and output stages, as shown in Figure 3.3. All of those elements are integrated on a single chip and housed in an IC package. The differential amplifier has a high-impedance (constant-current)“tail” to give it a high input impedance and a high degree of common-mode signal rejection. It also has a highimpedance collector (or drain) load, to give it a large amount of signal-voltage gain (typicallyabout 100 dB).在最简单的常规运放里,包括差动放大级和失调补偿级以及输出级,如图3.3(图略)所示,所有的器件都组合在一块芯片上,装在集成组件内。差动放大级有一个高阻抗的“尾巴”(恒流源)以提供高输入阻抗和对共模信号的深度抑制,同时,它还具有一个高阻抗的集电极或漏极负载以提供高的信号电压增益(典型数据约为100dB)。Unit6 A The signal-processing block at the transmitter conditions the source for more efficient transmission.For example, in an analog system, the signal processor may be an analog low-pass filter that is used to restrict the bandwidth of m(t). In a hybrid system, the signal processor may be an analog-to-digital converter (ADC). This produces a “digital word” that represents samples of the analog input signal. In this case, the ADC in the signal processor is providing source coding of the input signal. In addition, the signal processor may also add parity bits to the digital word to provide channel coding so that error detection and correction can be used by the signal processor in the receiver to reduce or eliminate bit errors that are caused by noise in the channel.The signal at the output of the transmitter signal processor is a baseband signal because it has concentrated nearf =0. consequently, they are said to be baseband signals.发送机中的信号处理单元对信源进行调理,以便更有效地传送。譬如说,在模拟系统中,信号处理器可以是一个模拟低通滤波器,以限制m(t)的带宽。在混合系统中,信号处理器可以是一个模-数变换器(ADC),他用数字来表示模拟输入信号的样本值。也就是说,ADC对输入信号进行源编码。此外,信号处理器还可以给数码字加入奇偶校验码位,即提供信道编码,使得接收机的信号处理器可以进行误码检测和纠错,以减少或消除由信道噪声所引起的误码。发送机中信号处理器的输出信号仍是基带信号,因为它的频谱也集中在f=0附近。 In 1948, C. E. Shannon (also of Bell Telephone Laboratories) published a paper in the Bell System Technical Journal relating the information capacity of a communications channel in bits-per-second (bps) to bandwidth and signal-to-noise ratio. Mathematically stated, the Shannon limit for information capacity is where I= information capacity (bps), B= bandwidth (Hz), S/N= signal-to-noise power ratio (unit-less). For a standard voice-band communications channel with a signal-to-noise power ratio of 1000(30 dB) and a bandwidth of 2.7 kHz, the Shannon limit for information capacity is I= 26.9 kbps.1948年,C.E.Shannon(也属于贝尔电话实验室)在贝尔系统技术杂志上发表了一篇论文,它揭示了一个通信信道每秒通过的带宽的信息容量与信噪比之间的关系。数学上表达式为,信息容量的香农极限是 I=B log2(1+S/N) 其中,I=信息容量(比特/秒),B=带宽(赫兹),S/N=信号噪声功率比(较小的单位)。一个标准的语音通信的信号噪声功率比是1000(30dB),带宽是2.7kHz,信息容量的香农极限是I=26.9kbps.Unit9 A The first essential step in analog-to-digital (A/D) conversion (as shown in Figure 9.1) is to sample an analog signal. This step is performed by a sample and hold circuit, which samples at regular intervals called sampling intervals. The length of the sampling interval is the same as the sampling period, and the reciprocal of the sampling period is the sampling frequency fs. According to the Nyquist theorem, a signal with a maximum frequency of WHz (called a band-limited signal) must be sampled at least 2W samples per second to ensure accurate recording. When this minimum is not respected, distortion called aliasing occurs. Aliasing causes high frequency signals to appear as lower frequency signals. To be sure aliasing will not occur, sampling is always preceded by low pass filtering. The low pass filter, called the anti-aliasing filter, removes all frequencies above half the selected sampling rate.模/数(转换换)(见图9.1)(图略)关键性的第一步是对模拟信号进行采样。这一步是由采样保持电路完成的,采样保持电路按照固定的“采样间隔”进行采样.“采样间隔”的长度和“采样周期”的长度相同,而“采样周期”的倒数就是“采样频率fs”.根据奈奎斯特采样定理,为了确保准确记录信号,最高频率WHz的信号(称为带限信号)每秒内必须采样至少2W个样本。如果不能满足这个最低要求,就会出现“混叠”失真。混叠会使高频信号出现在低频段。为了保证不出现混叠,在采用之前总要进行低通滤波。这个低通滤波器(称做“抗混叠滤波器”)将频率超过所选采样频率一半的信号全部滤除。Once digital signal processing is complete, digital-to-analog (D/A) conversion (as shown in Figure 9.2) must occur. This process begins by converting each digital code into an analog voltage that is proportional in size to the number represented by the code. This voltage is held steady through zero order hold until the next code is available, one sampling interval later. This creates a staircase-like signal that contains frequencies above WHz. These signals are removed with a smoothing low pass filter, the last step in D/A conversion.数字信号处理完成之后,就必须进行数/模(D/A)转换(见图9.2)(图略)。首先,将码字转换为和码字所代表的数字的大小成正比的模拟电压。这个电压值在零阶保持器中保持到下一个码字的出现,即需要保持一个采样间隔。这样,信号就是阶梯状的,信号中包含着频率超过WHz的成分。D/A转换的最后一步是使用平滑滤波器滤除那些频率超过WHz的成分。Unit17 AA digital image is made up of many rows and columns of pixels. For gray scale images, each pixel is assigned a number that represents the gray shade assigned to that pixel. The larger the number of pixels in an image, and the larger the number of available gray scale levels, the better the resolution of the image. Figure 17.1 is an 17-bit images, with 217256 possible gray scale levels. The number of row and column are 303228.一幅数字图像是由多行和多列像素组成的。对灰度图而言,每个像素被赋予一个代表该像素灰度级的数值。图像的像素越大,可用灰度级越多,图像分辨力也越好。图17.1(图略)是一幅8比特图像,它有256个可能的灰度级。行列数为303X228。Unit19 A Figure 1.2 shows a block schematic of a typical digital optical fiber link. Initially, the input digital signal from the information source is suitably encoded for optical transmission. The laser drive circuit directly modulates the intensity of the semiconductor laser with the encoded digital signal. Hence a digital optical signal is launched into the optical fiber cable. The avalanche photodiode (APD) detector is followed by a front-end amplifier and equalizer or filter to provide gain as well as linear signal processing and noise bandwidth reduction.Finally, the signal obtained is decoded to give the original digital information. However, at this stage it is instructive to consider the advantages provided by lightwave communication via optical fibers in comparison with other forms of line and radio communication which have brought about the introduction of such systems in many areas throughout the world.图19.2(图略)是一个典型的数字光纤联接示意图。起初,信源的数字信号经适当编码后进行光传输。激光器的驱动单元电路通过这些已编码的数字信号来直接调制激光器的发光强度,然后将数字光纤信号注入光纤。前置放大器与均衡器或滤波器跟在雪崩二极管检测器的后面以获得增益、线性信号处理及噪声带宽的抑制。最后,信号通过解码得到原始信号。在本文中,通过对比光纤通信与其他形式的有线、无线通信系统的优点,对我们很有启发。而后者已经在许多地方做了介绍。3.BIn the binary system of representation the base is 2, and only two numerals 0 and 1 are required to represent a number. The numerals 0 and 1 have the same meaning as in the decimal system, but a different interpretation is placed on the position occupied by a digit. In the binary system the individual digits represent the coefficients of powers of two rather than ten as in the decimal system. For example, the decimal number 19 is written in the binary representation as 10011 since二进制示数系统中的基数是二,且只有0和1两个数被用以示数。0和1在这里与在十进制中具有一样的意义,但是每个数位表示的不一样。二进制系统中每个数字表示二的幂系数,而十进制中表示十的幂系数。Unit 4 MicroelectronicsThe materials that make up our universe are composed of over one hundred basic and individual types of matter called elements. Ninety-two of these elements occur naturally and the remainders are man-made. Each element has a separate identify of its own, that is, no two elements have the same physical and chemical properties, nor can an element be subdivided by ordinary physical and chemical means into simple elements.1 Examples of elements are gold, mercury and oxygen.构成宇宙的物质是由一百多种基本的不同类型的物质元素组成的。其中92种元素是天然形成的,其他的则是人造的。每种元素都各有其自己的标识,也就是说,元素的物理和化学特性是独一无二的,一种元素也不可以用物理和化学手段再分成简单的元素。例如金,汞和氧元素。Electrons whirl around nuclei in much the same manner that the planets of our solar system travel around the sun. The difference between an atom of one element and that of another is in the number of protons and neutrons contained in the nucleus.2 All the elements in the universe, and therefore all matter, is composed of protons, neutrons, and electrons.电子围绕着原子核旋转,其方式类似于太阳系中行星围绕太阳旋转。一种元素的原子和另外一种元素的原子的差别在于其原子核中的质子数和中子数。宇宙中所有元素,也就是说所有物质,都是由质子,中子和电子组成的。Unit 5 Auxiliary StorageAt first, these disk drives used 14 inch disks, but now they use 5 1/4inch, 3 1/2inch,2 1/2inch, and 1 4/5inch diameters. In contrast to the floppy disk drive, hard disk drives hold from 80 megabytes to gigabytes and terabytes of information. When purchasing a hard disk, consider storage capacity and seek time, a measure of a hard disks access speed. The smaller the numbers, the faster the disk. In the past, 65 milliseconds was the standard access time, but today the standard is less than 7 milliseconds.起初,这些磁盘驱动采用14英寸磁盘,但现在采用的是5 1/4英寸,2 1/2英寸和1 4/5直径的磁盘。与软盘驱动截然不同的是,硬盘驱动保存着兆字节到千兆字节、万兆字节的信息。购买硬盘时要考虑储存容量和寻道时间及存取速度。这些数字越小,磁盘就运行的越快。在过去,标准存取时间为65毫秒,而现在的标准为低于7毫秒。Storing data on hard disks is similar to storing data on diskettes. In order to read or write data on the surface of the spinning disk platter, the disk drives are designed with access arms, or actuators.1 The access arms, or actuators, contain one or more read/write heads per disk surface. As the disk rotates at a high rate of speed, usually 3,600 revolution per minute, the read/write heads move across its surface. These read/write heads float on a cushion of air and do not actually touch the surface of the disk.2 The distance between the head and the surface varies from approximately ten to twenty millionths of an inch. If some form of contamination is introduced or if the alignment of the read/write heads is altered by something accidentally jarring the computer, the disk head can collide and damage the disk surface, causing a loss of data. This event is known as a head crash. Because of the time needed to repair the disk and to reconstruct the data that was lost, head crashes can be extremely costly to users in terms of both time and money.硬盘上的数据存储与软盘上的类似。为了在旋转的磁盘盘片表面读写数据,磁盘驱动器被设计成具有存取和传动装置。存取和传动装置由每个盘片表面的一个或多个读写头构成。当盘片以通常为3600转每分的速度高速旋转时,读写头移过盘片表面。这些读写头在高压气流上漂浮,移动时并不与盘片表面真正接触。读写头与盘片表面的距离约为一万到两千万之一英寸。如果有污染物介入,或读写头队列被电脑的意外震动改变,读写头会碰撞并损坏盘片表面,引起数据丢失。这种情况叫做磁头碰撞。由于修复磁盘并重建数据需要一定的时间,磁头碰撞会话费用户相当多的金钱和时间。Unit 6 Electronic CommunicationsThe signal-processing block at the transmitter conditions the source for more efficient transmission.4 For example, in an analog system, the signal processor may be an analog low-pass filter that is used to restrict the bandwidth of m(t). In a hybrid system, the signal processor may be an analog-to-digital converter (ADC). This produces a “digital word” that represents samples of the analog input signal. In this case, the ADC in the signal processor is providing source coding of the input signal. In addition, the signal processor may also add parity bits to the digital word to provide channel coding so that error detection and correction can be used by the signal processor in the receiver to reduce or eliminate bit errors that are caused by noise in the channel.5 The signal at the output of the transmitter signal processor is a baseband signal because it has concentrated nearf =0.发送机中的信号处理单元对信号源进行调理,以便更好的传送。譬如说,在模拟系统中,信号的处理可以使一个模拟低通滤波器,以限制m(t)的带宽。在混合的系统中,信号处理器可以是一个模-数变换器(ADC)它用数字来表示模拟输入信号的样本值。也就是说,ADC对输入信号进行原编码。此外,信号处理器还可以给数码字加入奇偶校验位,即提供信道编码,使得接收机的信号处理器可以进行误码检验和纠错,以减少或消除由信道噪声所引起的误码。发送机中的信号处理器的输出信号仍是基带信号,因为他的频谱也是集中在f=0附近。In 1948, C. E. Shannon (also of Bell Telephone Laboratories) published a paper in the Bell System Technical Journal relating the information capacity of a communications channel in bits-per-second (bps) to bandwidth and signal-to-noise ratio. Mathematically stated, the Shannon limit for information capacity is where I= information capacity (bps), B= bandwidth (Hz), S/N= signal-to-noise power ratio (unit-less). For a standard voice-band communications channel with a signal-to-noise power ratio of 1000(30 dB) and a bandwidth of 2.7 kHz, the Shannon limit for information capacity is I= 26.9 kbps.1948年,C. E. Shannon在贝尔系统技术杂志上发表了一篇论文,他揭示了一个通信信道每秒通过带宽的信息容量与信噪之间的关系。数学上表达式为,信息容量的香农极限是其中。I=信息容量(比特/秒),B=带宽(赫兹),S/N=信号噪声功率比(较小单位)。一个标准的语音通信信道的信号噪声功率比是1000(30dB)带宽是2.7KHz,信息容量的香农极限是I=26.9kbps。BPCM is dependent on three separate operations: sampling, quantizing, and coding. Many different schemes for performing these three functions have evolved during recent years, and we shall describe the main ones.1 In these descriptions we shall see how a speech channel of telephone quality maybe conveyed as a series of amplitude values, each value being represented, that is, coded, as a sequence of 8 binary digits.2 Furthermore, we shall prove that a minimum theoretical sampling frequency of order 6.8 kilohertz (kHz) is required to convey a voice channel occupying the range 300 Hz to 3.4 kHz.3 Practical equipments, however, normally use a sampling rate of 8 kHz, and if 8-digits per sample value are used, the voice channel becomes represented by a stream of pulses with a repetition rate of 64 kHz. Figure 6.2 illustrates the sampling, quantizing, and coding processes.脉冲编码调制依靠三个独立的运作:采样,量化,编码。近年来,人们对这三个环节的实现提出了许多不同的方案,我们将对其中一些主要的方案进行讨论。在这些讨论中,我们会看到话路中的语音信号是如何转换成一个幅值序列的,而每个幅值又被编码,即以8位二进制数的序列表示。而且我们将证明,为了变换频率范围为300hz3.4khz的话路信号,理论上的最小采样频率为6.8khz。但实际设备通常采用8KHz的采样速度,而如果采用8位每样本的值,则会出现重复速率为64kHz的脉冲流。采样,量化和编码的过程如图6.2所示。So far in this discussion we have assumed that each voice channel has a separate coder, the unit that converts sampled amplitude values to a set of pulses; and decoder, the unit that performs the reverse operation. This need not be so, and systems are in operation where a single codec (i.e., coder and its associated decoder) is shared between 24, 30, or even 120 separate channels. A high-speed electronic switch is used to present the analog information signal of each channel, taken in turn, to the codec. The codec is then arranged to sequentially sample the amplitude value, and code this value into the 8-digit sequence. Thus the output to the codec may be seen as a sequence of 8 pulses relating to channel 1, then channel 2, and so on. This unit is called a time division multiplexer (TDM), and 15 illustrated in Figure 6.3. The multiplexing principle that is used is known as word interleaving. Since the words, or 8-digit sequences, are interleaved in time.目前我们在这个讨论中假定每个话路有单独的编码器,也就是将采样幅值转换成脉冲序列的单元;并有单独的解码器,也就是执行相反的操作单元。其实并不需要这样,系统靠单一的编译码器(编码器和其关联的解码器)运行,它有24,30,甚至120个单独的信道。高速电子开关用以表示每个信道的模拟信息信号,并把信号依次送入编译码器。而后编译码器顺序地进行幅值采样,并把幅值编排成8位码序列。这样编译码器的输出看起来就像一串关于信道1,信道2等的8位脉冲序列。这个单元叫做时分多路器。图6.3说明了15个信道的时分多路器工作原理。采用的分路原理叫字交叉。因为每个字或8为码序列是在一段时间内的交叉存取的。17unitA digital image is made up of many rows and columns of pixels. For gray scale images, each pixel is assigned a number that represents the gray shade assigned to that pixel. The larger the number of pixel in a image, and the larger the number of available gray scale levels, the better the resolution of the image.figure 17.1 is an 8_bit image, with 256 possible gray scale levels. The number of rows and column are 303*228.一幅数字图像是由多行和多列像素组成的。对灰度图而言,每个像素被赋予一个代表该像素灰度级的数值。图像的像素数越大,可用灰度级越多,图像分辨力也越好。图17.1是一幅8比特图像,它有256个可能的灰度级。行列数为303*228。In part due to the Internet, digital image are transmitted from place to place more often than ever. To save time and bandwidth (space), both images and other files are often compressed before being transmitted.loss-less compression means that a file is compacted without losing any information, so that the reconstructed file is identical to the original. Lossy compression means that some information from the original file is irretrievably loss, but generally the reconstructed file is extremely close to the original. The compression ratio is the ratio of the original file size to the compressed file size.部分由于互联网的原因,数字图像比以往更频繁地从一个地方传到另一个地方。为了节省时间和带宽(空间),(与其他文件一样)在传输前经常需要对图像文件进行压缩。无损压缩是不损失任何信息地将文件进行压缩,重建文件和原文件是完全一样的。有损压缩是指原文件当中的某些信息不可恢复地丢失了,但重建文件与原文件非常相近。压缩比是指原文件大小和压缩文件大小之比。Numerous methods have been developed for the compression of digital image data. One of the principle drivers for this development is the television industry where quality image data must be transferred to receivers using relatively simple equipment. The development of high definition television is further focusing the attention of industry and university scientists toward problems of data reduction and digital transmission. The principal evaluation criteria for the analysis of compressed versus uncompressed imagery is whether a person can tell the difference between the images. A more implemental measure is the Root Mean Square (RMS) error between the original image and the image that has been compressed. Compression rates may be generate by determining the size of the compressed image in terms of number of bits per pixel for original image.许多方法已经开发为数字图像数据的压缩。推动这一原则的发展是电视行业质量图像数据必须传输到接收器使用相对简单的设备。高清晰度电视的发展进一步聚焦的关注行业和大学的科学家对数据压缩和数字传输的问题。主要评价标准分析一个人能否分辨压缩和未压缩的图像的区别。一个更有帮助的措施是原始图像和压缩后图像之间的均方根误差。压缩比可以通过原始图像每个像素所需压缩的比特数确定。There are two general types of compression:(1) loss-less, and (2) loss. Loss-less compression means that you can achieve a certain compression factor an be able to exactly reproduces the original image. Loss compression on the other hand allows some loss, but has the potential for much higher compression rates. No matter what technique that you use, the exact rate is very dependent on the complexity of the image that you are analyzing. For example, the normal best can be achieve with loss-less encoding in a rate of 2 bits per pixel. In fact, for some Land-sat scenes with urban areas and many small farms, the factor of 2 bits per pixel may not be able to achieve. The same technique applied to a Land-sat image of the mid-west where large fields occur and few shadows exist in images might produce a much better compression.通常有两种类型的压缩:(1)无损压缩和(2)有损压缩。无损压缩意味着您可以获得确定的压缩因子和能够精确还原原始图像。有损压缩一方面允许一些损失,但有潜在的更高的压缩率。不管你使用什么技术,精确率很大程度上依赖于你正在分析的图像的复杂性,。例如,正常可以实现无损最好与编码率为2位/像素。事实上,对于一些市区和许多小农场场景的卫星图像以2位/像素的压缩系数可能无法实现。同样的技术应用于产生大领域的中西部的卫星图像在图像中存在一些阴影可能产生更好的压缩。The last type of compression considered is fractal compression. Based on Mandelbrot sets which take

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