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UNIT 1

The Principle of PCM

PCM原理

Pcm 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. In these descriptions we shall see how a speech channel of telephone quality may be conveyed as a series of amplitude values, each value being represented, that is, coded as a sequence of 8 binary digits. 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. Practical equipments, however, normally use3 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 64khz. Fig .1-1 illustrates the sampling, quantizing, and coding processes.

PCM的构成依赖于三个环节，即采样、量化和编码。近年来，人们对这三个环节的实现提出了许多不同的方案，我们将对其中的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声信号是如何转换成幅值序列的，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，为了转换频率范围为300HZ—3.4KHZ的话路信号，理论上最小采样频率须为6.8khz。但是，实际设备通常用8khz的采样速率，而如果每个样值用8位码的话，则话路是由一个重复速率为64khz的脉冲流来表示的。图1-1表示了采样、量化、编码的过程。

Reexamination of our simple example shows us that the speech signal of maximum frequency 3.4khz has been represented by a signal of frequency 64khz. However, if only 4-digits per sample value had been used, the quality of transmission would drop, and the repetition rate of the pulses would be reduced to 32khz. Thus the quality of transmission is dependent on the pulse repetition rate, and for digital communication systems these two variables may be interchanged most efficiently.

让我们再研究一下上面提到的简单例子。可以看出，最高频率为3.4khz的话音信号适用64khz的（脉冲流）信号来表示的。但是，如果每个样值中用4位（码）表示，则传输质量会下降，而脉冲的重复速率也将减小到32khz。因而传输质量是取决于脉冲重复速率的。对于数字通信系统，这两个量之间极明显的互相影响着。

Digital transmission provides a powerful method for overcoming noisy environments. Noise can be introduced into transmission patch in many different ways : perhaps via a nearby lightning strike, the sparking of a car ignition system, or the thermal low-level noise within the communication equipment itself. It is the relationship of the true signal to the noise signal, known as the signal-to-noise ratio, which is of the most interest to the communication engineer. Basically, if the signal is very large compared to the noise level, then a perfect message can take place; however, this is not always the case. For example, the signal received from a satellite, located in far outer space, is very weak and is at a level only slightly above that of the noise. Alternative examples may be found within terrestrial systems where, although the message signal is strong, so is the noise power.

数字传输对于克服噪声环境的影响提供了一个强有力的手段。噪声可以以多种不同方式

进入传输信道，比如说因为附近的闪电、汽车点火系统的打火或因通信设备本身低电平的热噪声所致。正是这种被称为信噪比的东西，即真实信号与噪声的关系引起了通信工程师的极大的兴趣。从本质上讲，如果信号比噪声电平大得多，则信息的传输是完美的。但是，实际情况并不总是这样，例如，从位于遥远太空中的卫星接收到的信号极其微弱，其电平仅比噪声稍高一点。地面系统则是另一类例子，尽管信号很强，噪声也很强。

If we consider binary transmission, the complete information about a particular message will always be obtained by simply detecting the presence or absence of the pulse. By comparison, most other forms of transmission systems convey the message information using the shape, or level of the transmitted signal; parameters that are most easily affected by the noise and attenuation introduced by the transmission path. Consequently there is an inherent advantage for overcoming noisy environments by choosing digital transmission.

研究二进制信号的传输可见，只要简单的区判别脉冲的有和无，完美就获得了一条消息的全部信息。相比之下，许多其他形式的传输系统是利用被传信号的波形或电平的高低来传送信息的，而这些参数又极易受到传输途径中的噪声和衰耗的影响。因此选择数字传输系统在克服噪声环境的影响方面有其固有的优势。

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-sped 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 is illustrated in fig. 1-2. The multiplexing principle that is used is known as work interleaving. Since the words, or 8-digit sequences, are interleaved in time.

到目前为止，在这个讨论中，我们一直假定每个话路各有一个编码器和解码器。前者是将幅度采样值变换成脉冲，而后者则施行相反的变换，这种设置并非必须。在实际运行的PCM系统中，一个编、译码器为24路、30路，甚至120路所共用（注：在当代的PCM设备中，编、译码器系分路设备，即每个话路各有一套。）一个高速的电子开关被用来将每一话路的模拟信号依次的送往编、译码器。然后编、译码器再顺序采样幅值并把这个幅值编成8位码序列。这样，编解码器输出的8位码序列就分别对应于话路1、话路2，等等。这种设备称为时分复用（TDM）,如图1-2所示。由于8位码的码字序列按时间顺序插接在一起，所以上面所用的复用原则称为码字插接。

At the receive terminal a demultiplexer is arranged to separate the 8-digit sequences into the appropriate channels. The reader may ask, how does the demultiplexer know which group of 8-digits relates to channel 1,2, and so on? Clearly this is important! The problem is easily overcome by specifying a frame format, where at the start of each frame a unique sequence of pulses called the frame code, or synchronization word, is placed so at to identify the start of the frame. A circuit of the demultiplexer is arranged to detect the synchronization word, and thereby it knows that the next group of 8-digits corresponds to channel 1. The synchronization word reoccurs once again after the last channel has been received.

接收端设置了分路设备将8位码序列分配到相应的话路中。读者也许会问，分路设备怎么会知道哪一组8位码对应于第1路、第2路及其他各路呢？显然这是很重要的。这个问题

是很容易解决的。我们只要制定一个帧格式，即在每一帧的开始放置一个被称作帧码或同步字的独特码序列以标志每帧的起始，而用分路设备的一个电路去检测同步字，从而就知道下一个8位码组对应于话路1。当最后一个话路的码字收到之后，同步码又再次出现。

UNIT 2

Asynchronous Serial Data Transmission

异步串行数据传输

By far the most popular serial interface between a computer and its CRT terminal is the asynchronous serial interface. This interface is so called because the transmitted data and the received data are not synchronized over any extended period and therefore no special means of synchronizing the clocks at the transmitter and receiver is necessary. In fact ,the asynchronous serial data link is a very old form of data transmission system and has its origin in the era of teleprinter.

在计算机及其显示器之间最为常见的串行接口是异步串行接口。这个接口之所以如此称呼，是因为无论在多长的时间区间里发送的数据和接收的数据是不同步的，因而没有必要采用特殊的手段使发送器和接收器的时钟同步。实际上，异步串行数据链路是一种古老的数据传输方式，它起源于电传打字机的时代。

Serial data transmission systems have been around for a long time and are found in the telephone (human speech), Morse code, semaphore, and even the smoke signals one used by native Americans. The fundamental problem encountered by all serial data transmission systems is how to split the incoming data steam into individual units (i.e., bits) and how to group these units into characters. For example, in Morse code the dots and dashes of a character are separated by an intersymbol space, while the individual characters are separated by an intercharacter space, which is three times the duration of an intersymbol space.

串行数据传输系统已有很长的历史了，电话（人类语音）、莫尔斯电码、旗语，甚至土著美洲人从前用过的烟火信号都可以视为传些数据传输。所有的串行数据传输系统面临的首要的问题都是如何把如数的数据流分开为单独的码元（即比特），以及怎样把这些码元组合成字符。例如，在莫尔斯电码中，字的点、划是由符号间的空格来分开的，而字符之间又是由字符间的空格分开的，字符间空格的时长是点、划间空格的三倍。

First we examine how the data stream is divided into individual bits and the bits grouped into characters in an asynchronous serial data link. The key to the operation of this type of link is both simple and ingenious. Fig.2-1 gives the format of data transmitted over such a link.

首先我们研究一下在异步串行数据链路中数据流是怎样分成单独码元的，以及码元是如何组成字符的。这类系统运行的核心原理既简单有精巧。图2-1绘出了在这个链路中传送的数据格式。

An asynchronous serial data link is said to be character-oriented, as information is transmitted in the form of groups of bits called characters. These characters are invariable units comprising 7 or 8 bits of ―information‖ plus 2 to 4 control bits and frequently correspond to ASCII-encoded characters. Initially ,when no information is being transmitted, the line is in an