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1、图解:译文“蓝色”UNIT 1 The Principle of PCMPCM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
2、 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 oc
3、cupying 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 codin
4、g 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300HZ3.4KHZ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 ex
5、ample 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 o
6、f 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
7、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 equipmen
8、t 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 al
9、ways 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
10、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 particula
11、r 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 attenuatio
12、n 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 th
13、is 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 assoc
14、iated 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-d
15、igit 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
16、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 t
