PCM PRINCIPLE

1.0 INTRODUCTION

A long distance or local telephone conversation between two persons could be provided by using a pair of open wire lines or underground cable as early as early as mid of 19th century. However, due to fast industrial development and increased telephone awareness, demand for trunk and local traffic went on increasing at a rapid rate. To cater to the increased demand of traffic between two stations or between two subscribers at the same station we resorted to the use of an increased number of pairs on either the open wire alignment, or in underground cable. This could solve the problem for some time only as there is a limit to the number of open wire pairs that can be installed on one alignment due to headway consideration and maintenance problems. Similarly increasing the number of open wire pairs that can be installed on one alignment due to headway consideration and maintenance problems. Similarly increasing the number of pairs to the underground cable is uneconomical and leads to maintenance problems. It, therefore, became imperative to think of new technical innovations which could exploit the available bandwidth of transmission media such as open wire lines or underground cables to provide more number of circuits on one pair. The technique used to provide a number of circuits using a single transmission link is called Multiplexing.

MULTIPLEXING TECHNIQUES
There are basically two types of multiplexing techniques i. ii Frequency Division Multiplexing (FDM) Time Division Multiplexing (TDM)

The FDM techniques is the process of translating individual speech circuits (300-3400 Hz) into pre-assigned frequency slots within the bandwidth of the transmission medium. The frequency translation is done by amplitude modulation of the audio frequency with an appropriate carrier frequency. At the output of the modulator a filter network is connected to select either a lower or an upper side band. Since the intelligence is carried in either side band, single side band suppressed carrier mode of AM is used. This results in substantial saving of bandwidth mid also permits the use of low power amplifiers. Please refer Fig. 1. FDM techniques usually find their application in analogue transmission systems. An analogue transmission system is one which is used for transmitting continuously varying signals.

2.2

Time Division Multiplexing

Basically, time division multiplexing involves nothing more than sharing a transmission medium by a number of circuits in time domain by establishing a sequence of time slots during which individual channels (circuits) can be transmitted. Thus the entire bandwidth is periodically available to each channel. Normally all time slots1 are equal in length. Each channel is assigned a time slot with a specific common repetition period called a frame interval. This is illustrated in Fig. 2.

Fig. 2 Time Division Multiplexing Each channel is sampled at a specified rate and transmitted for a fixed duration. All channels are sampled one by, the cycle is repeated again and again. The channels are connected to individual gates which are opened one by one in a fixed sequence. At the receiving end also similar gates are opened in unision with the gates at the transmitting end.

The signal received at the receiving end will be in the form of discrete samples and these are combined to reproduce the original signal. Thus, at a given instant of time, only one channel is transmitted through the medium, and by sequential sampling a number of channels can be staggered in time as opposed to transmitting all the channel at the same time as in EDM systems. This staggering of channels in time sequence for transmission over a common medium is called Time Division Multiplexing (TDM). Pulse Code Modulation It was only in 1938, Mr. A.M. Reaves (USA) developed a Pulse Code Modulation (PCM) system to transmit the spoken word in digital form. Since then digital speech transmission has become an alternative to the analogue systems. PCM systems use TDM technique to provide a number of circuits on the same transmission medium viz open wire or underground cable pair or a channel provided by carrier, coaxial, microwave or satellite system Basic Requirements for PCM System

FILTERING:Filters are used to limit the speech signal to the frequency band 300-3400 Hz Sampling:-

In signal processing, sampling is the reduction of a continuous signal to a discrete signal. A common example is the conversion of a sound wave (a continuous signal) to a sequence of samples (a discretetime signal). A sample refers to a value or set of values at a point in time and/or space. A sampler is a subsystem or operation that extracts samples from a continuous signal. A theoretical ideal sampler produces samples equivalent to the instantaneous value of the continuous signal at the desired points.

Sampling Theorem States "If a band limited signal is sampled at regular intervals of time and at a rate equal to or more than twice the highest signal frequency in the band, then the sample contains all the information of the original signal." Mathematically, if fH is the highest frequency in the signal to be sampled then the sampling frequency Fs needs to be greater than 2 fH. i.e. Fs>2fH

Let us say our voice signals are band limited to 4 KHz and let sampling frequency be 8 KHz. Time period of sampling Ts = or Ts = 125 micro seconds If we have just one channel, then this can be sampled every 125 microseconds and the resultant samples will represent the original signal. But, if we are to sample N channels one by one at the rate specified by the sampling theorem, then the time available for sampling each channel would be equal to Ts/N microseconds. 1 sec 8000

Fig. 4 shows how a number of channels can be sampled and combined. The channel gates (a, b ... n) correspond to the switch S in Fig. 3. These gates are opened by a series of pulses called "Clock pulses". These are called gates because, when closed these actually connect the channels to the transmission medium during the clock period and isolate them during the OFF periods of the clock pulses. The clock pulses are staggered so that only one pair of gates is open at any given instant and, therefore, only one channel is connected to the transmission medium. The time intervals during which the common transmission medium is allocated to a particular channel is called the Time Slot for that channel. The width of.this time slot will depend, as stated above, upon the number of channels to be combined and the clock pulse frequency i.e. the sampling frequency. In a 30 channel PCM system. TS i.e. 125 microseconds are divided into 32 parts. That is 30 time slots are used for 30 speech signals, one time slot for signalling of all the 30 chls, and one time slot for synchronization between Transmitter & Receiver. The time available per channel would be Ts/N = 125/32 = 3.9 microseconds. Thus in a 30 channel PCM system, time slot is 3.9 microseconds and time period of sampling i.e..the interval

between 2 consecutive samples of a channel is 125 microseconds. This duration i.e. 125 microseconds is called Time Frame. The signals on the common medium (also called the common highway) of a TDM system will consist of a series of pulses, the amplitudes of which are proportional to the amplitudes of the individual channels at their respective sampling instants. This is illustrated in Fig. 5

i Fig 5 : PAM Output Signals The original signal for each channel can be recovered at the receive end by applying gate pulses at appropriate instants and passing the signals through low pass filters. (Refer Fig. 6).

Fig. 6 : Reconstruction of Original Signal Quantization

The process of measuring the numerical values of the samples and giving them a table value in a suitable scale is called "Quantizing". Of course, the scales and the number of points should be so chosen that the signal could be effectively reconstructed after demodulation. Quantizing, in other words, can be defined as a process of breaking down a continuous amplitude range into a finite number of amplitude values or steps

Quantizing Process Suppose we have a signal as shown in Fig. 7 which is sampled at instants a, b, c, d and e. For the sake of explanation, let us suppose that the signal has maximum amplitude of 7 volts.

In order to quantize these five samples taken of the signal, let us say the total amplitude is divided into eight ranges or intervals as shown in Fig. 7. Sample (a) lies in the 5th range. Accordingly, the quantizing process will assign a binary code corresponding to this i.e. 101, Similarly codes are assigned for other samples also. Here the quantizing intervals are of the same size. This is called Linear Quantizing.

FIG. 7: QUANTIZING-POSITIVE SIGNAL Assigning an interval of 5 for sample 1, 7 for 2 etc. is the quantizing process. Giving, the assigned levels of samples, the binary code are called coding of the quantized samples. Conversion of quantised analogue levels to binary signal is called encoding. To represent 256 steps, 8 level code is required. The eight bit code is also called an eight bit "word". The 8 bit word appears in the form

P
Polarity bit 1 for + ve 'O' for - ve.

ABC
Segment Code

WXYZ
Linear encoding in the segment

The first bit gives the sign of the voltage to be coded. Next 3 bits gives the segment number. There are 8 segments for the positive voltages and 8 for negative voltages. Last 4 bits give the position in the segment. Each segment contains 16 positions. Referring to Fig. 9(b), voltage Vc will be encoded as 1 1 1 1 0101.

FIG. 9 (b) : Encoding Curve with Compression 8 Bit Code

The quantization and encoding are done by a circuit called coder. The coder converts PAM signals (i.e. after sampling) into an 8 bit binary signal. The coding is done as per Fig. 9 which shows a relationship between voltage V to be coded and equivalent binary number N. In a PCM system the channels are sampled one by one by applying the sampling pulses to the sampling gates. Refer Fig. 10. The gates open only when a pulse is applied to them and pass the analogue signals through them for the duration for which the gates remain open. Since only one gate will be activated at a given instant, a common encoding circuit is used for all channels. Here the samples are quantized and encoded. The encoded samples of all the channels and signals etc are combined in the digital combiner and transmitted.

The reverse process is carried out at the receiving end to retrieve the original analogue signals. The digital combiner combines the encoded samples in the form of "frames". The digital separator decombines the incoming digital streams into individual frames. These frames are decoded to give the PAM (Pulse Amplitude Modulated) samples. The samples corresponding to individual channels are separated by operating the receive sample gates in the same sequence i.e. in synchronism with the transmit sample gates.

4.0

CONCEPT OF FRAME

In Fig. 10, the sampling pulse has a repetition rate of Ts sees and a pulse width of "St". When a sampling pulse arrives, the sampling gate remains opened during the time "St" and remains closed till the next pulse arrives. It means that a channel is activated for the duration "St". This duration, which is the width of the sampling pulse, is called the "time slot" for a given channel. Since Ts is much larger as compared to St. a number of channels can be sampled each for a duration of St within the time Ts. With reference to Fig. 10, the first sample of the first channel is taken by pulse 'a', encoded and is passed on the combiner. Then the first sample of the second channel is taken by pulse 'b' which is also encoded and passed on to the combiner, Likewise the remaining channels are also sampled sequentially and are encoded before being fed to the combiner. After the first sample of the Nth channel is taken and processed, the second sample of the first channel is taken, this process is repeated for all channels. One full set of samples for all

channels taken within the duration Ts is called a "frame". Thus the set of all first samples of all channels is one frame; the set of all second samples is another frame and so on. For a 30 chl PCM system, we have 32 time slots. Thus the time available per channel would be 3.9 microsecs. Thus for a 30 chl PCM system, Frame = 125 microseconds Time slot per chl = 3.9 microseconds.

5.0

Structure of Frame

A frame of 125 microseconds duration has 32 time slots. These slots are numbered Ts 0 to Ts 31. Information for providing synchronization between trans and receive ends is passed through a separate time slot. Usually the slot Ts 0 carries the synchronization signals. This slot is also called Frame alignment word (FAW). The signaling information is transmitted through time slot Ts 16. Ts 1 to Ts 15 are utilized for voltage signal of channels 1 to 15 respectively. Ts 17 to Ts 31 are utilized for voltage signal of channels 16 to 30 respectively.