Pulse-Code Modulation (PCM)
PCM : The most popular form of TDM uses pulse-code modulation (PCM) (see Sec. 7-4), in which multiple channels of digital data are transmitted in serial form. Each channel is assigned a time slot in which to transmit one binary word of data. The data streams from the various channels are interleaved and transmitted sequentially
When PCM is used to transmit analog signals, the signals are sampled with a multiplexer as described previously for PAM, and then converted by an A/D converter into a series of binary numbers, where each number is proportional to the amplitude of the analog signal at the various sampling points. These binary words are converted from parallel to serial format and then transmitted.
At the receiving end, the various channels are demultiplexed and the original sequential binary numbers recovered, stored in digital memory, and then transferred to a D/A converter that reconstructs the analog signal. (When the original data is strictly digital, D/A conversion is not required, of course.)
Any binary data, multiplexed or not, can be transmitted by PCM. Most long-distance space probes have on-board video cameras whose output signals are digitized and transmitted back to earth in binary format. Such PCM video systems make possible the transmission of graphic images over incredible distances. In computer multimedia presentations, video data is often digitized and transmitted by PCM techniques to a remote source.
Fig. 10-21 shows a general block diagram of the major components in a PCM system, where analog voice signals are the initial inputs. The voice signals are applied to A/D converters, which generate an 8-bit parallel binary word (byte) each time a sample is taken. Since the digital data must be transmitted serially, the A/D converter output is fed to a shift register, which produces a serial data output from the parallel input. In telephone systems, a codec takes care of the A/D parallel-to-serial conversion. The clock oscillator circuit driving the shift register operates at the desired bit rate.
The multiplexing is done with a simple digital MUX. Since all the signals to be transmitted are binary, a multiplexer constructed of standard logic gates can be used. A binary counter drives a decoder that selects the desired input channel. The multiplexed output is a serial data waveform of the interleaved binary words. This baseband digital signal can then be encoded and transmitted directly over a twisted pair of cables, a coaxial cable, or a fiber-optic cable. Alternatively, the PCM binary signal can be used to modulate a carrier. A form of phase modulation known as phase-C shift keying (PSK) is the most commonly used.
Fig. 10-22 shows the details of a four-input PCM multiplexer. Typically, the inputs to such a multiplexer come from an A/D converter. The binary inputs are applied to a shift register, which can be loaded from a parallel source such as the A/D converter or another serial source. In most PCM systems, the shift registers are part of a codec.
The multiplexer itself is the familiar digital circuit known as a data selector. It is made up of gates 1 through 5. The serial data is applied to gates 1 to 4; only one gate at a time is enabled, by a 1-of-4 decoder. The serial data from the enabled gate is passed through to OR gate 5 and appears in the output.
Now, assume that all shift registers are loaded with the bytes to be transmitted. The 2-bit counter AB is reset, sending the 00 code to the decoder. This turns on the 00 output, enabling gate 1. Note that the decoder output also enables gate 6. The clock pulses begin to trigger shift register 1, and the data is shifted out 1 bit at a time. The serial bits pass through gates 1 and 5 to the output. At the same time, the bit counter, which is a divide-by-8 circuit, keeps track of the number of bits shifted out.
When 8 pulses have occurred, all 8 bits in the shift register 1 have been transmitted. After counting 8 clock pulses, the bit counter recycles and triggers the 2-bit counter. Its code is now 01. This enables gate 2 and gate7. The clock pulses continue, and now the contents of shift register 2 are shifted out 1 bit at a time. The serial data passes through gates 2 and 5 to the output. Again, the bit counter counts 8 bits and then recycles after all bits in the shift register 2 have been sent. This again triggers the 2-bit counter, whose code is now 10. This enables gates 3 and 8. The contents of shift register 3 are now shifted out. The process repeats for the contents of shift register 4.
When the contents of all four shift registers have been transmitted once, one PCM frame has been formed (see Fig. 10-22). The data in the shift registers is then updated (the next sample of the analog signal converted), and the cycle is repeated.
If the clock in Fig. 10-22 is 64 kHz, the bit rate is 64 kilobits per second (kbps) and the bit interval is 1/64,000 = 15.625 μs. With 8 bits per word, it takes 8 x 15.625=125 μs to transmit one word. This means that the word rate is 1/125 x 10-6 = 8 kbytes/s. If the shift registers get their data from an A/D converter, the sampling rate is 125 μs or 8 kHz. This is the rate used in telephone systems for sampling voice signals. Assuming a maximum voice frequency of 3 kHz, the minimum sampling rate is twice that, or 6 kHz, and so a sampling rate of 8 kHz is more than adequate to accurately represent and reproduce the analog voice signal.
As shown in Fig. 10-22, a sync pulse is added to the end of the frame. This signals the receiver that one frame of four signals has been transmitted and that another is about to begin. The receiver uses the sync pulse to keep all its circuits in step so that each original signal can be accurately recovered.
At the receiving end of the communication link, the PCM signal is demultiplexed and converted back to the original data (see Fig. 10-23). The PCM baseband signal may come in over a cable, in which case the signal is regenerated and reshaped prior to being applied to the demultiplexer. Alternatively, if the PCM signal has modulated a carrier and is being transmitted by radio, the RF signal will be picked up by a receiver and then demodulated. The original serial PCM binary waveform is recovered and fed to a shaping circuit to clean up and rejuvenate the binary pulses. The original signal is then demultiplexed by means of a digital demultiplexer using AND or NAND gates. The binary counter and decoder driving the demultiplexer are kept in step with the receiver through a combination of clock recovery and sync pulse detector circuits similar to those used in PAM systems. The sync pulse is usually generated and sent at the end of each frame. The demultiplexed serial output signals are fed to a shift register for conversion to parallel data and sent to a D/A converter and then a low-pass filter. The shift register and D/A converter are usually part of a codec. The result is a highly accurate reproduction of the original voice signal.
Keep in mind that all the multiplexing and demultiplexing circuits are usually in an integrated form. In fact, both MUX and DEMUX circuits are combined on a single chip to form a TDM transceiver that is used at both ends of the communications link. The individual circuits are not accessible; however, you do have access to all the inputs and outputs that allow you to perform tests, measurements, and troubleshooting as required.
Benefits or Advantages of PCM
PCM is reliable, inexpensive, and highly resistant to noise. In PCM, the transmitted binary pulses all have the same amplitude and, like FM signals, can be clipped to reduce noise. Further, even when signals have been degraded because of noise, attenuation, or distortion, all the receiver has to do is to determine whether a pulse was transmitted. Amplitude, width, frequency, phase shape, and so on do not affect reception. Thus PCM signals are easily recovered and rejuvenated, no matter what the circumstances. PCM is so superior to other forms of pulse modulation and multiplexing for the transmission of data that it has virtually replaced them all in communication applications.
A special PCM system uses 16 channels of data, one whose purpose is identification (ID) and synchronization. The sampling rate is 3.5 kHz. The word length is 6 bits. Find (a) the number of available data channels, (b) the number of bits per frame, and (c) the serial data rate.
a. 16 (total no. of channels) – 1 (channel used for ID) =15 (for data)
b. Bits per frame = 6 x 16 = 96
c. Serial data rate 5 sampling rate 3 no. bits/frame = 3.5 kHz x 96 = 336 kHz
Digital Carrier Systems
The most widespread use of TDM is in the telephone system. All modern telephone systems use digital transmission via PCM and TDM. The only place where analog signals are still used is in the local loop—the connection between a telephone company’s central office (CO) and the subscriber’s telephone, known as the customer premises equipment (CPE). All local and long-distance connections are digital. Years ago, the telephone companies developed a completely digital transmission system called the T-carrier system. It is used throughout the United States for all telephone calls and for the transmission of computer data including Internet access. Similar systems are used in Japan and Europe.
The T-carrier system defines a range of PCM TDM systems with progressively faster data rates. The physical implementations of these systems are referred to as T-1, T-2, T-3, and T-4. The digital signals they carry are defined by the terms DS1, DS2, DS3, and DS4. It begins with the T-1 system, which multiplexes 24 basic DS1 digital voice signals that are then multiplexed into larger and faster DS2, DS3, and DS4 signals for transmission. Usually, T-1 transmission is by way of a dual twisted-pair cable or coaxial cable. Wireless transmission is also common today. The T-2, T-3, and T-4 systems use coaxial cable, microwave radio, or fiber-optic cable transmission.
The most commonly used PCM system is the T-1 system developed by Bell Telephone for transmitting telephone conversations by high-speed digital links. The T-1 system multiplexes 24 voice channels onto a single line by using TDM techniques. Each serial digital word (8-bit words, 7 bits of magnitude, and 1-bit representing polarity) from the 24 channels is then transmitted sequentially.
Each frame is sampled at an 8-kHz rate, producing a 125-μs sampling interval. During the 125-μs interval between analog samples on each channel, 24 words of 8 bits, each representing one sample from each of the inputs, is transmitted. The channel sampling interval is 125 μs/24 5 5.2 μs, which corresponds to a rate of 192 kHz. This represents a total of 24 x 8 = 192 bits. An additional bit—a frame sync pulse—is added to this stream to keep the transmitting and receiving signals in synchronization with each other. The 24 words of 8 bits and the synchronizing bit form one frame of 193 bits. This sequence is carried out repeatedly. The total bit rate for the multiplexed signal is 193 x 8 kHz = 1544 kHz or 1.544 MHz. Fig. 10-24 shows one frame of a T-1 signal.
The T-1 signal can be transmitted via cable, coaxial cable, twisted-pair cable, or fiber-optic cable; or it can be used to modulate a carrier for radio transmission. For example, for long-distance telephone calls, T-1 signals are sent to microwave relay stations, where they frequency-modulate a carrier for transmission over long distances. Also, T-1 signals are transmitted via satellite or fiber-optic cable.
The T-1 systems transmit each voice signal at a 64-kbps rate. But they are also frequently used to transmit fewer than 24 inputs at a faster rate. For instance, a T-1 line can transmit a single source of computer data at a 1.544-Mbps rate. It can also transmit two data sources at a 722-kbps rate or four sources at a 386-kbps rate and so on. These are known as fractional T-1 lines.
T-2, T-3, and T-4 Systems
To produce greater capacity for voice traffic as well as computer data traffic, the DS1 signals may be further multiplexed into faster signals that carry even more channels. Fig. 10-25 shows how four DS1 signals are multiplexed to form a DS2 signal. The result is a 6.312-Mbps serial digital signal containing 4 x 24 = 96 voice channels. The T-2 systems are not widely used except as a steppingstone to form DS3 signals. As Fig. 10-25 shows, seven DS2 outputs are combined in a T-3 multiplexer to generate a DS3 signal. This signal contains 7 x 96 = 672 voice channels at a data rate of 44.736 Mbps. Four DS3 signals may further be multiplexed to form a DS4 signal. The T-4 multiplexer output data rate is 274.176 Mbps.
The T-1 and T-3 lines are widely used by business and industry for telephone service as well as for digital data transmission. These are dedicated circuits leased from the telephone company and used only by the subscriber so that the full data rate is available. These lines are also used in various unmultiplexed forms to achieve fast internet access or digital data transmission other than voice traffic. The T-2 and T-4 lines are rarely used by subscribers, but they are used within the telephone system itself.
Duplexing is the method by which two-way communications are handled. Remember that half duplexing means that the two stations communicating take turns transmitting and receiving. Mobile, marine, and aircraft radios use half duplexing. Full duplexing means that the two stations can send and receive simultaneously. A full-duplex is certainly preferred, as in phone calls. But not all systems require a simultaneous send/receive capability.
As with multiplexing, there are two ways to provide duplexing—frequency-division duplexing (FDD) and time-division duplexing (TDD). The simplest and perhaps best way to provide full-duplex is to use FDD, which utilizes two separate channels, one for send and another for receive. Fig. 10-26 shows the concept. The communicating parties are called station 1 and station 2. Station 1 uses the channel around f1 for receiving only and the channel around f2 for transmitting. Station 2 uses f1 for transmitting and f2 for receiving. By spacing the two channels far enough apart, the transmitter will not interfere with the receiver. Selective filters keep the signals separated. The big disadvantage of this method is the extra spectrum space required. Spectrum space is scarce and expensive. Yet most cell phone systems use this method because it is the easiest to implement and the most reliable.
Time-division duplexing (TDD) means that signals are transmitted simultaneously on a single channel by interleaving them in different time slots. For example, alternating time slots are devoted to transmitting and receiving. This is illustrated in Fig. 10-27. During time slot t1, station 1 is transmitting (TX) while station 2 is receiving (RX). Then during time slot t2, station 1 is receiving while station 2 is transmitting. Each time slot may contain one data word, such as 1 byte from an A/D converter or a D/A converter. As long as the serial data rate is high enough, a user will never know the difference.
The primary benefit of TDD is that only one channel is needed. It saves spectrum space and cost. On the other hand, the TDD method is harder to implement. The key to making it work is precise timing and synchronization between transmitter and receiver. Special synchronizing pulses or frame sequences are needed to constantly ensure that timing will not result in collisions between transmit and receive. Several of the newer third-generation cell phone systems may use TDD.
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Reference : Electronic communication by Louis Frenzel