AGV观光车运行控制系统设计外文文献.doc

通信技术简述摘要：我们说的话不是数字信号。数字信号是一种由“1”和“0”组成的、能用数学方法处理的语言。我们讲出的话是现实世界中的模拟信号。我们天天遇到的现实世界信号都是模拟信号-如声音、光、温度、和压力。数字信号是模拟信号的数值表示。在数字世界里，对这些信号进行处理可能会更容易、成本更低。在现实世界中，我们可以通过“模数转换”将信号转换数字信号，然后对信号进行处理。如果需要的话，用“数模转换器”将信号转换回到模拟世界中去。关键词：数字信号，模拟信号1 数字通信与模拟通信根据传输信号的不同，通信可分为模拟通信和数字通信。当一个通信系统传输处理的信号是模拟信号时，这个系统就是模拟通信系统。反之，数字通信系统就是传输、处理数字信号的系统。传统的电话通信系统中，用户线上传送的信号在整个通话期间不断地随着用户声音的变化而变化。这个变化的信号无论在时间上或是幅度上都是连续的，这种信号称为模拟信号，数字信号与模拟信号不同，它是一种时间上离散的、幅度取值有限的信号形式。常见的数字信号是幅度取值只有两种（用0和1代表）的波形，称为“二进制信号”。一般地，数字信号可以用二进制、M进制（M）2）来表示，显然，M进制信号就是有M种幅度符号可选的数字信号。由于模拟信号的频谱较窄，模拟通信系统的信道利用较高。但因为连续信号中混入噪声后很难清除，使得输出的还原信号产生波形失真，模拟通信系统的抗干扰能力只能算是差强人意。现代通信系统大多是数字通信系统，因此，模拟信号必须首先变为数字信号后才能进入数字通信系统传输，这个模拟转换为数字信号的过程就称为“模/数转换”。它一般包括三个过程。(1) 抽样：就是对连续的模拟信号进行离散化处理；(2) 量化：将模拟信号抽样值变换到最接近的数字值，把幅度上连续的抽样信号变为离散的；(3) 编码：把变化后的离散信号用一组二进制代码表示，形成数字信号。接收端对收到信号的处理则正好上述过程的逆过程，也就是把收到的数字信号还原为原始模拟信号，即“数/模转换”。2 数字通信系统的组成信源既可以输出模拟信号（如音频信号），也可以输出数字信号（如打字机的输出信号），这是一类时间上离散、且总的个数有限的信号。数字通信系统中，信源输出信息（无论模拟的还是数字的）都将被转换为二进制数子序列，这一转换过程就叫做信源编码或数据压缩，通常希望表示信源输出信息的符号个数越少越好，换句话说，信源编码应用较少的符号有效（或无冗余地）表示信源输出信息。信源编码后二进制数字序列，常称为信息系列，将被送入信道编码器进行信道编码。这是因为信号在信道传输过程中将受到各种噪声和干扰的影响，信道编码在传输的二进制信息序列中增加一些冗余，以便接收端可利用这些冗余客服噪声干扰的不利影响，正确恢复原始信号。因此说，信道编码中增加的冗余可提高接受信号的可靠性和正确性，帮助接收机解调还原所传的信息序列。一般地，信道编码时一次性地将K比特信息序列按一定的规则和比例n/k变换为一个n比特的信息序列，这个比例n/k的倒数就成为该信道编码的编码效率。信道编码输出的二进制序列经过数字调制器调制后被送入信道进行传输。鉴于目前所用的信道绝大部分都是传输信号的，可以说数字调制器的作用就是将二进制信息序列转换为适合信道输入的信号波形。数字通信系统的接收端，解调器将收到的波形、失真波形解调还原为二进制或M进制符号序列后，再送入信道解码器，借助其中的信息冗余，按照事先所知的信道编码规则重构原始信息序列。常用接收端恢复的信息序列中的错误发生概率来衡量解调与解码的性能，具体地，就是解码器输出序列中平均发生一个比特误码的概率-误码率。实际上，误码率不但与接收端解调器、解码器的性能有关，也同样与发送端所选编码的特性、传输波形的类型、发送信号的功率以及信道的性能等有关。最后，若输出端要求输出模拟信号，信道解码器输出序列送入信源解码器，利用已知的信源编码方式还原输出原始信号。信道解码错误以及信源编码/解码差错都将导致信源解码输出仅仅是发送端信源输出原始信息的一个近似，一般常用这两者之间的差异表示数字通信系统的失真。3 数模转换和模数转换精要对模拟信号进行采样是“模数转换”的关键一步。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，最高频率为W Hz的信号（称为“带限信号”）每秒内必须采集至少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较低频段。为了保证不会出现“混叠”，在采样前要进行低通滤波。这个低通滤波器（称作“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。经过一段短暂的采集时间之后（期间采集到一个样本），采样保持电路将这个样本一直保持到下一个采样间隔的开始。A/D转换器需要利用这个保持时间来生成模拟样本对应的码字。A/D转换器为每个模拟样值选择一个量化电平。一个N位转换器可以在2N个可选量化电平当中进行选择。量化电平数越多，量化误差（量化电平和实际电平之差）越小。最大量化误差不会超过“量阶”Q的一半。“量阶”的计算公式Q=R/2N,其中R是模拟信号的满量程范围，N是转换器所用的位数。相对于量化误差的信号强度使用“动态范围”和“信噪比”来度量。数字信号是用一组竖线表示的，竖线顶部的圆圈标出样本选用的量化电平。A/D转换器的比特率为Nfs，这里的fs是采样率。最后，要为每个数字化样本分配一个码字，这样就完成了A/D转换过程。A/D转换的结果就是得到一串数字比特流。数字信号处理就是对这些码字集合进行处理。总结一下，A/D转换过程由抗混合滤波、采样、量化、和数字化这四步组成。数字信号处理完成之后，就必须进行D/A转换。首先，将码字转换为和码字所代表数字的大小成正比的模拟电压。这个电压值要在零阶保持器中保持到下一个码字出现，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W Hz的成分。D/A转换的最后一步就是：使用平滑滤波器滤除那些频率超过W Hz的成分。对于采样信号中的任意频率f，其因采样而产生的镜像频率将出现在无数个频率处（kfsfHz）。当采样率低于要求的奈奎斯特率（即fs2) signals, clearly, the Mary signal means a type of digital signal which has M amplitudes for selection. Comparing with a digital signal, the bandwidth transmitting an analog signal is relatively narrow, and the efficiency of channel bandwidth usage for analog communication system is higher than that of the digital communication system. Its difficult for analog signals to distinguish the noise from original signals, and the capability of anti-interference in analog communication system is just passable.The majority of modern communication systems are digital communication systems, so the analog signals have to be converted into corresponding digital signals before they are processed and transmitted in digital communication system. The converting process is called Analog/Digital conversion, which usually contains the following three steps. (1)Sampling: transforming analog signal into the time-discrete and amplitude-continuous signal;(2)Quantizing: transforming the sampled signal into fully discrete signal, and put the quantized signal with an approximate digit;(3)Coding: using binary code words to describe the quantized digit.The process of receiving signals in the receiving end of a digital communication system is completely the opposite to the process in sending end, which means that the receiving signal must have to be converted into analog signals by a Digital/Analog conversion at last.2 Elements of a Digital Communication SystemThe source output may be either an analog signal, such as audio signal, or a digital signal, such as the output of a teletype machine, which is discrete in time and has a finite number of output characters. In a digital communication system, the messages produced by the source are converted into a sequence of binary digits. Ideally, we should like to represent the source output (message) by as few binary digits as possible. In other words, we seek an efficient representation of the source output that results in little or no redundancy. The process of efficiently converting the output of either an analog or digital source into a sequence of binary digits is called source encoding or data compression. The sequence of binary digits from the source encoder, which is called the information sequence, is passed to the channel encoder. The channel encoder introduces, in a controlled manner, some redundancy in the binary information sequence, which can be used at the receiver to overcome the effects of noise and interference encountered in the transmission of the signal through the channel. Thus, the added redundancy serves to increase the reliability of the received data and improve the fidelity of the received signal. In effect, redundancy in the information sequence aids the receiver in decoding the desired information sequence. Encoding involves taking information bits at a time mapping each n-bit sequence introduced by encoding the data in this manner is measured by the ratio n/k. The reciprocal of this ratio, namely k/n, is called the rate of the code or, simply, the code rate.The binary sequence at the output of the channel encoder is passed to the digital modulator, which serves as the interface to the communications channel. Since nearly all of the communication channels encountered in practice are capable of transmitting electrical signals, the primary purpose of the digital modulator is to map the binary information sequence into signal waveforms.At the receiving end of a digital communication system, the demodulator processes the channel-corrupted transmitted waveform and reduces the waveforms to a sequence of numbers that present estimates of the transmitted data symbol (binary or M-ary). This sequence of numbers is passed to the channel decoder, which attempts to reconstruct the original information sequence from knowledge of the code used by the channel encoder and the redundancy contained in the received data.A measure of how the demodulator and decoder perform is the frequency with which errors occur in the decoded sequence. More precisely, the average probability of a bit-error at the output of the decoder is a measure of the performance of the demodulator-decoder combination. In general, the probability of error is a function of the code characters, the types of waveforms used to transmit the information over the channel, the transmitter power, and the characteristics of the channel. As a final step, when an analog output is desired, the source decoder accepts the output sequence from the channel decoder, and, from knowledge of the source encoding method used, attempts to reconstruct the original from the source. Due to channel decoding errors and possible distortion introduced by the source encoder, and, perhaps, the source decoder, the signal at the output of the source decoder is an approximation to the original source output. The difference between the original signal and the reconstructed signal is a measure of the distortion introduced by the digital communication system. 3 The Essentials of Analog-to-Digital and Digital-to-Analog ConversionThe first essential step in analog-to-digital (A/D) conversion 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 W Hz (called a band-limited signal) must be sampled at least 2 W 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.After a brief acquisition time, during which a sample is acquired, the sample and hold circuit holds the sample steady for the remainder of the sampling interval. This hold time is needed to allow time for an A/D converter to generate a digital code that best corresponds to the analog sample.The A/D converter chooses a quantization level for each analog sample. An N-bit converter chooses among 2N possible quantization levels, The larger the number of levels, the smaller the quantization errors, calculated as the difference between the quantized level and the true sample level. Most quantization errors are limited in size to half a quantization step Q. The quantization step size is calculated as Q=R/2N, where R is the full scale range of the analog signal and N is the number of bits used by the converter. The strength of the signal compared to that of the quantization errors is measured by dynamic range and signal-to-noise ratio.A digital signal is represented by a set of vertical lines with circles at the top to mark the quantization levels selected for each sample. The bit rate for an A/D converter is the N fs, where fs is the sampling rate.Finally, each digital sample is assigned a digital sample is assigned a digital code, which completes the A/D process. The result is a digital bit stream. It is this collection of digital codes that is processed in digital signal processing.To summarize, A/D comprises anti-aliasing, sampling, quantization and digitization.Once digital signal processing is complete, digital-to-analog(D/A) conversion 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 W Hz. These signals are removed with a smoothing low pass filter, the last step in D/A conversion.The images of each frequency f present in a sampled signal appear, through sampling, at the infinite number of frequencies kfs fHz. When the sampling rate is lower than the required Nyquist rate , that is fs2w, images of high frequency signals erroneously appear in the baseband(or Nyquist range) due to aliasing. While this under sampling is normally avoided, it can be exploited. For example, signals whose frequencies are restricted to a narrow band of high frequencies can be sampled at a rate similar to twice the with of the band instead of twice the maximum frequency. All of the important signal characteristics can be deduced from the copy of the spectrum that appears in the baseband through sampling. Depending on the relationship between the signal frequencies and the sampling rate, spectral inversion may cause of the spectrum in the baseband to be inverted from the true spectrum of the signal.If a universal microprocessor solution existed with which every design could be realized, the electronics industry wouldnt be a very competitive place. However, typically in most electronic designs, more than one processor technology can be used to implement the required fun