BASEBAND COMMUNICATIONS

Baseband signalling and baseband receivers

Baseband Requirements
In the case of baseband the data bits are transmitted directly as pulses. These pulses may have to be shaped so as to minimise the effect of noise on the received pulses, to minimise the distortions introduced by the transmission medium, to minimise the bandwidth required for transmission of the signals and hence to maximise the throughput of data across the medium, to prevent Inter Symbol Interference (ISI), to reduce crosstalk with other channels using the same medium, etc., etc. In addition various line coding systems may be used to ensure that there is at least one voltage transition per pulse to assist in clock synchronisation and data recovery at the receive end to eliminate long term DC voltages on the medium to minimise the bandwidth requirements, and maximise throughput. In general the pulse shape, and coding system is optimised for the particular application and medium, but it may not be possible to optimise all requirements simultaneously. If one bit of data is transmitted per pulse then the pulse is defined as binary (the pulse has only two levels, or only two shapes , corresponding to 1 and 0). It may be possible to combine a number of data bits into a single multi-level, (or multiple shaped) pulse prior to transmission to give, for example, 3 levels (ternary), 4 levels (quaternary), 5 levels (quinary), or in general, M levels (M-ary) coding prior to transmission. This permits more than 1 bit of data to be transmitted per pulse, so that a higher bit rate is achieved. However because there are more voltage levels they are inevitably closer together than in the case of binary so that a lower level of noise/distortion picked up on the transmission medium can cause a level to be misidentified on reception, leading to errors on the received signal. Therefore, if the transmission medium is sufficiently quiet, M-ary coding can be used to increase throughput, where the value of m depends on the noise on the medium.

Bit Rate and Symbol Rate

In a baseband transmission system pulses are transmitted across the medium. Each pulse may have a different amplitude and/or shape so that it carries more than one bit of data or information. In this case each pulse is called a symbol and the rate at which pulses are transmitted across the medium is the symbol rate (measured in symbols/sec or Baud - - not Baud/sec). Each symbol carries X bits so the bit rate is X * Baud bits/sec. Example: A binary system represents a 1 by +V volts and 0 by -V volts. In this case each pulse (each symbol) carries one data bit, and the bit rate is the same as the symbol rate. Example: In binary encoded ASCII consider each letter, number etc as a symbol. Each symbol is 8 bits long to accommodate all letters, graphics etc. The symbol can have 256 possible values. Say 1000 letters are transmitted across the medium. The symbol rate is 1000 symbols/sec = 1000Baud and the bit rate is 8000 bits/sec.;
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Analogue Telephone Line

At the local exchange a voltage is applied, via inductors and resistors to the copper pair. This allows the transmitting equipment to be powered - to sink current. The variations in current correspond to Telephone RX TX change in the voltage signal on the line. line The receiver terminal reads this voltage. In most cases the receiving terminal is allowed to take a DC feed from the line itself. The exchange power supply - battery- acts like a very a large capacitor which would short circuit any AC signal on the line. The power supply must be isolated from the line by an inductance. At the same the AC signal must be able to go from the transmitter to the receiver, bypassing the power supply via a capacitance. This means that there is no DC path from the transmitter to the receiver- from one telephone, or any other device connected to the network, to another device - and any communication that requires the transmission of a DC voltage will not work. The above diagram shows an analogue telephone exchange (about 20 years obsolete). All exchanges are now digital and the incoming signal passes first into an Analogue to Digital Converter (ADC). All switching is done using the 8 bit symbols generated , which is converted back to analogue in a Digital to Analogue Converter (DAC) on the outgoing line. But through the ADC, switching and DAC there is still the same DC isolation between the two sides. The bandwidth of an analogue telephone line connection is 300 Hz to 3.4 kHz. A square wave or any pulse train with very fast rise times will be distorted if it is sent along a telephone line. Therefore an analogue telephone line is not suitable for sending digital pulses as all components outside the 300 - 3.4 kHz range will be attenuated or removed. This bandwidth limitation is partly due to the copper pair to the exchange but mainly to the anti-aliasing filters in the local exchange which are part of the A-to-D process, sampled at 8,000 samples per second. (In the past some analogue telephone lines also had loading coils (inductors) on the line to give a flat frequency response - obsolete for 30 years now). On a digital telephone line all analogue filters are removed so the usable bandwidth of the copper pair itself is much greater and can extend to a few Megahertz. These lines are suitable for pulse transmission e.g. ISDN, ADSL, HDSL. The challenge is to send high speed data on a band limited network.
Telephone Exchange

Bandwidth of a Pulse, Inter-Symbol Interference

Minimum Bandwidth Requirements
Any signal can be treated as if it were made up of an infinite number of frequency components. We have seen the spectrum of a square wave already in which case the (significant) frequency components include much higher frequencies than the fundamental frequency. To transmit the waveform without significant distortion would therefore require a channel with a considerable bandwidth, together with a suitable phase change characteristic. A pulse train does not have to be -T/2 +T/2 received undistorted in order to make +V correct decisions about its binary states. Time In fact, in the case illustrated, so long as the fundamental component, at l/(2T) Hz, of the square wave corresponding to the bit stream ...01010101 can be Period = 2T
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transmitted, then correct decisions can be made about the binary states. It is possible, in theory at least, to transmit 1/T symbols per second over a channel of bandwidth 1/(2T) Hz. Put another way, it is theoretically possible to signal at a rate of 2B symbols per second over an ideal bandlimited channel of bandwidth B Hz. For example a 64 kbit/s data stream, could be sent and recovered over a 32 kHz bandlimited channel. This is an important general rule (due to Nyquist) for digital waveforms. Example: A primary ISDN signal has a bit rate of 2.048 Mbit/s. What would be the minimum theoretical bandwidth required to transmit this signal? Answer: Minimum bandwidth = 2.048 Mbit/s = 1.024 Mbit/s. Example: A spectrum analyser and antenna is used to record the radiation pattern from a TDM system which contains clock generators. There are two peaks, at 153.088 MHz and 154.112 MHz. (a) What is the frequency of the clock generator which is responsible for these peaks? (b) What is the order of these harmonics? (c) How might these peaks be reduced without affecting the performance of the system? Answer: (a) A clock generator outputs a square wave which contains only odd harmonics. Therefore these frequencies are odd multiples of the clock frequency, and the difference between them is twice the clock frequency (or perhaps 4 or six times - but much less likely). The difference is 154.112 - 153.088 MHz = 1.024 MHz. Therefore the clock is 512 kHz. (b) 154.122 MHz is the 301st harmonic (154.122/.512 = 301). The other one is the th 299 harmonic (c) The output from the clock generator should be filtered e.g. by using a ferrite bead in series, or a small capacitor to earth. Track layout is important, keeping all tracks as short as possible. Or the circuit could be screened - effectively put in a metal box.

Inter Symbol Interference

From Fourier analysis the frequency components of a square pulse of duration and amplitude A are given by V(f) = A sin f = A sinc f f A

number of things to note about this it has a maximum of 1 at frequency 0 1.2 the bandwidth is infinite 1 the first zero occurs at sin f = 0, or 0.8 f = 1. The narrower the pulse the 0.6 greater the spread of the central 0.4 lobe. 0.2 0 most of the energy in the pulse (more -0.2 than 90%) is contained in frequencies below f = 1/ A very narrow pulse (small duration , and amplitude A) will require a wide range of frequencies (large 1/) to reproduce it. To transmit a digital waveform without distortion requires a channel with an infinite bandwidth, with a linear phase change characteristic. If a narrow pulse is passed through an ideal low pass filter with cutoff frequency fc << 1/ the time domain representation of the pulse will be (remember sin f = 1 for small fc ) f
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v(t)

2 A fc

sin 2f c t 2f c t

2 A fc sinc 2fct

This has its first zero at sin 2fct = 1 or t = 1/2fc, and further crossings at n/2fc. Most of the energy of the pulse is contained in the time range -1/2fc < t < 1/2fc and the pulse duration c = 1/fc. Assume that the presence of a pulse indicates 1 and the absence is 0. The pulse spreading or dispersion causes overlap of pulses into adjacent pulse time slots and may result in an error at the point where t the receiver decides which symbol has been transmitted, especially when other impairments are present (such as noise, interference). This pulse overlap and the difficulty of discriminating between symbols at Intersymbol the receiver is called inter symbol interference (ISI). Interferencce Assume the transmission h(t) channel behaves like an H(f) ideal brick wall low pass filter, with transfer -1/2fc 1/2fc function as in the diagram, t with cutoff at fc. An input -fc fc pulse will produce a pulse in the time domain as shown. If two pulses are transmitted through this ideal channel they overlap at reception as in the diagram below. Ideally if the pulse spacing is such that the pulse centre (peak voltage) falls exactly at the first zero of the following pulse it is possible to detect the pulses without mutual interference. If the receiver sampling happens exactly in the middle of the received pulse where the interference contribution of the other pulse is zero (it is going through a zero crossing at that sampling instant), then the receiver has no problem deciding if a pulse was there or not and the effect of ISI is minimised. If the pulses are closer together it will be difficult (or impossible) to distinguish one pulse from the following one. (Note that the same criterion is used in astronomy to decide if the images of two stars are resolved by a telescope - Rayleigh resolution criterion)) The cut off frequency of the medium is fc. Each 1/2fc received pulse is described by v(t) = A fc sinc 2fct. For minimum ISI the first zero of a pulse falls on the centre of the succeeding pulse - the time from pulse centre to centre is 1/2fc so that the pulse rate is 2fc. This is the fastest rate at which pulses can be reliably transmitted a cross a medium If pulses are transmitted at a faster rate than that, ISI may occur and the presence or absence of a pulse may be mistaken and a symbol misidentified. To transmit pulses with a separation T requires a bandwidth of 1/2T. If multi-level (M-ary, see later section) coding is used so that each pulse carries more than one bit of data, where the data are used to encode the amplitude of the transmitted pulse, it will be necessary to determine the amplitude of each pulse on reception. Using the above resolution criterion, with no ISI at the centre of the pulse it is possible to measure the amplitude at the centre of the received pulse and hence determine the value of the encoding bits. The ideal brick wall type of frequency response cannot be achieved. (A filter like this would spread the pulse over an infinite duration, both before and after the pulse. The earlier lobes of the pulse would arrive at the receiver before the start of the transmission of the pulse - lack of causality, implying the ability to predict the generation of the pulse in the future.) Also the sampling instant will not always be exactly at the middle of the peak lobe of the sinc response. (See later section on the structure of a baseband receiver in which other methods are used to detect a pulse
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in the presence of noise). Real transmission channels will not have a sharp cut off frequency so that the transmitted pulse will not be described precisely by the sinc function - the duration between first zeros will be slightly longer but in general there will be less energy in the side lobes. Taking all this into account, the shape of the transmitted pulses can be controlled minimise so as to ISI (see Raised Cosine Filters). One of the design objectives is to minimise the ISI contribution to the system performance degradation. Some practical filter with a suitable realisable frequency response is required, which shapes the impulses such that each pulse still contains zero crossings at the sampling instant of the adjacent pulse.

Transmission Systems
Problems are encountered when a digital signal is sent through a channel. This shows the basic stages in a digital signal transmission. Non-return-to-zero (NRZ) is assumed. The transmission medium might be a coaxial cable or a copper twisted pair used in local area networks or digital telephone systems. The principles apply to systems using other media and/or codes. Typical Retiming Noise Digital Information waveforms Extractor clocked at fc E at the Received - may be source Digital points and/or channel Channel Threshold Retiming Information coded labelled A Transmitter Equaliser e.g. Co-ax Detector Sector A C D F Copper pair B to F in the system are shown. The original waveform A is attenuated and a noise component is added. Because of finite system response time and propagation delays, the transition between voltage levels become indistinct. To counteract the distortion the 1 0 1 1 0 1 0 0 Binary information system includes an equaliser. which A NRZ Line code from Transmitter reshapes the received waveform, so B Received signal from TX channel that the relationship of the equaliser output C to the original binary symbols is much clearer, e.g. if a copper cable C Output from equaliser to picks up 50 Hz noise from the mains or other electromagnetic interference D Output from threshold detector the equaliser must remove it. Or in the case of pulse spreading leading to ISI, E Output from timing extractor or reflections in the case of radio signals leading to multiple receptions, F Output from re-timing circuit the equaliser eliminates these also. The input to the equaliser must be 1 0 1 1 0 1 0 protected from over-voltages such as induced lightning and other transients. Passing the equalised waveform through a threshold detector (e.g. a Schmitt trigger) generates a binary signal very similar to the transmitted one. If the threshold settings are too small then noise will trigger the detector. If the settings are too large then the data may not trigger the detector. It is important that the slew rate of the comparator used in the detector is fast enough for the data rate. If the noise levels are sufficiently low, and the equaliser and threshold detector are set correctly, the only difference between waveforms A and D is that the transitions are not perfectly in step. The transitions of D will correspond to the thresholdcrossings of waveform C which will not precisely mirror those of the original binary waveform. This gives timing irregularities (jitter) requiring re-timing of the received waveform, else the jitter build up to cause error over a long link or multiple links.
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A regular timing reference signal F - the data clock - is derived from the received waveform itself using a timing extraction circuit (maybe based on a Phase Locked Loop). The clock signal and the output from the threshold detector are processed to give a regenerated digital signal F whose transitions now coincide with the clock transitions. Waveforms C and F shows that the combined effect of threshold detection and retiming is equivalent to sampling waveform C near its peaks and troughs to determine the appropriate binary states. So even in the presence of noise, regenerated signal F can be an almost perfect (delayed) replica of the transmitted signal, provided only that the noise is not sufficient to cause an incorrect decision to be made at the threshold detector.

Transmission Line Impairments

It was assumed that the transmitter sends a rectangular pulse. A transmission line acts like a filter so that its transfer function ensures that a rectangular pulse can be quite distorted. The distortion can mean that pulses can become overlapped (ISI) thus causing receiver errors. We therefore need to model the effects of transmitting pulses through a transmission line.

Linearity
In telecoms engineering there are several definitions of linearity, two are; 1. A linear system obeys the principle of superposition. If input x1(t) produces output y2(t) and input x2(t) produces output y2(t) then input [x1(t) + x2(t)] produces output [y1(t) + y2(t)]. 2. A linear system preserves frequency. If the input is a sinusoid then the output is a sinusoid of the same frequency, but generally differing in amplitude and phase. Many elements in a telecoms system are designed to be as nearly linear as possible. A sufficiently close approximation to linearity means, for example, that the principle of superposition can be used to model the effect of a channel or other system element on a sequence of pulses, given knowledge of the effect on one pulse. a) Rectangular pulse response of a first order lowpass V input pulse filter, where the duration of the pulse is pulse response approximately equal to the filter time constant. b) Response of the same filter to a binary waveform. (a) Time The figure shows the response to a single pulse, and the 1 0 1 1 0 1 0 0 superimposed pulse responses corresponding to an input pulse train (binary waveform). The response to a single pulse takes longer to decay than the duration of a symbol period so that the output waveform accumulates (b) a DC offset. Without further processing this would cause problems for threshold detection. Even in the positions corresponding to a 0 there can be a considerable output voltage. This figure, on the other hand, shows a more desirable V input pulse pulse response for a telecommunications channel. It pulse response shows the possible response of a channel to (a) a rectangular pulse and (b) a binary waveform. The clear (a) Time distinction between 1 and 0 means that the system response to a bit stream could be decoded without 1 0 1 1 0 1 0 0 difficulty. Alternatively a linear channel or component model can be based on the second definition of linearity. Any real (b) signal can be described in terms of its frequency content - or frequency spectrum. Any linear system can be completely specified by its
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frequency response function (transfer function) - a description of amplitude and phase shifts introduced by the system for all frequencies. The frequency response H(f) of a first-order lowpass system as a function of frequency f is H(f) = where k is the low-frequency or DC gain and fc is the 3 dB cut-off frequency. In general, H(f) has a complex value for any value of k, f and fc. For a linear system with a frequency response function H(f) an input sinusoid x(t) = A sin t produces an output B sin (t + ), where B/A = |H(f)| is the amplitude ratio and = Arg H(f) is the phase shift. Gain dB Frequency response functions are usually plotted as separate 0 graphs of amplitude ratio and phase shift e.g. first order -10 lowpass system such as a simple RC low pass filter, which -20 has a 20 dB per decade roll off (or 6 dB per octave roll off), -30 and a phase shift of 45 degrees at the cutoff frequency -40 In this case the amplitude ratio is in decibels - i.e. 20 log | Phase Shift degrees H(f)| and that the figure is drawn with DC gain k of 1 (= 0 0 dB). Scaling allows the curve to be used for other values of k.
-45 -90 .01 fc .1 fc

fc 10 fc 100 fc

Amplitude distortion and phase distortion

An ideal transmission channel passes all frequency components of a signal with their amplitude and phase relationships unchanged The simplest frequency domain model of such behaviour would be a constant amplitude ratio and zero phase shift for Transmitted Signal all frequencies of interest. For example a Time square wave would be unaffected by the Phase and Pulse has spread transmission channel. In practice, the Amplitude Distortion due to phase delay higher order harmonics will be greatly for harmonics attenuated by the transmission channel. Also the phase shift will be different for Amplitude Distortion each harmonic. For example, the Only fundamental harmonic may have a phase shift of 45 degrees whereas the fifth harmonic could have a phase shift of 80 degrees. This will cause components of the pulse to be delayed or stretched. Specifications for digital receiver systems usually include limits for phase delay.

Eye Patterns
In a practical system, the transmitted digital signal is subject to many impairment such as jitter, noise, attenuation delay, ISI etc.. The eye diagram is a convenient way to assess the performance of digital transmission by observing the baseband signal on an oscilloscope. The time base of the oscilloscope is set to trigger at the data signal rate 1/T. The resulting pattern looks like the human eye hence the name. On an analogue scope, the persistence of the cathode ray tube blends together all signal waveforms. The eye pattern displays traces for all allowed signal waveforms in a given sampling interval with a resulting binary or M-ary display (depending on what type of wave format is used. in general a system using an m-level digital signal will have m-1 eyes) at the prescribed sampling time. The sampling time is at the point where the opening is greatest.
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The eye opening is used as an indication of system health As noise, inter symbol interference, jitter is added by the channel, the eye opening will decrease. The height of this opening indicates what margin exists over noise. Example: An eye diagram has an eye closing from top to bottom of 50%. What is the S/N? Answer: The linear signal to noise ratio is 2. This is measured on an oscilloscope which is a voltage measurement. Therefore S/N = 20 log 10 2 = 6 dB.

Equalisation
Distortion (dispersion)during transmission process results in the received signal being corrupted by ISI. Multi-path transmission of radio signals produces echoes or ghosting. Equalisation can overcome these effects. Usually, the transmission characteristics are unknown and may vary with time and the equaliser must be updated with each new channel connection, as in the case of a voice-band modem operating in a switched network, and must also adapt the filters automatically to track changes in transmission characteristics with time, as with GSM mobile networks. Automatic adaptive equalisation is used in digital radio applications, in mobile communications and in voice-band modems for speeds higher than 2400 b/s. The transversal filter consists of a delay line tapped at T second intervals, where T is the symbol width. The diagram shows (underlined) the time at which each sample was input into the t +(N-1)T t + NT t - (N-1)T t - NT equaliser. Each tap is T T T T input connected (through an vin(t) amplifier) to a summing cN cN-1 c-N-1 c-N device that provides the output. The summed output output vout(t) of the equaliser is sampled at the symbol rate and fed to a decision device. The tap gains, or tap coefficients (ci), are set to subtract the effects of interference from symbols that are adjacent in time to the desired symbol. There are (2N + 1) taps with coefficients c-N, c-N+1, , c+N-1 ,c+N . Samples of the equaliser output are expressed in terms of the input vin(t) and tap coefficients ,cn as vout(t) =
k = N

+N

vin(t +kT)

Example: Assume a three tap transversal filter as equaliser with tap coefficients c-1 = -0.25, c0 = 1.0 and c1 = -.25. The input pulse has been distorted so that in successive time intervals (t - 2T) to (t + 2T) its voltage is 0.01, -0.1, 1, 0.5, 0.1. (In this case N = 1.) What is the output pulse voltage in successive time intervals? Solution: vout(t - T) = = vout(t ) = = vout(t + T) =
k = 1

+1

vin((t - T) +kT)

c-1 * vin(t - T - T)+ c0 * vin(t - T) + c1 * vin(t - T +T) -.25*.01 - 1*0.1 - 0.25*1 = -0.3525 =
k = 1

+1

vin(t +kT)

c-1 * vin(t - T)+ c0 * vin(t) + c1 * vin(t +T) 0.25*0.1 + 1*1 - 0.25*0.5 = 0.9 =
k = 1

+1

vin(t + T +kT) =

.225

Output at (t - T), (t) and (t + T) is -.3525, 0.9, 0.225.

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It is seen that the shape of the sharpened by the equaliser. Selection of tap coefficients is based on the minimisation of either peak distortion (using zero forcing equaliser) or mean square distortion.

Line Coding
Requirements
Digital data can be transmitted by various line codes. The following properties are desirable : The pulse stream has no DC component. (DC wont pass through a pulse transformer, or a capacitor, and so is blocked by most communication links). It should be relatively easy to recover the data clock at the receive end (self synchronising). The line coding scheme should be bandwidth efficient to permit as many bits as possible to be transmitted in a given time and yet fit into the channel bandwidth. The line code should be robust in the presence of noise - have as big a difference as possible between a 1 and 0. It should be possible to recognise a line coding error, sometimes called a line violation. (In some protocols, a line violation is deliberately generated to mark the start of a frame)

Signalling Formats
1 0 1 1 0 0 0 1 Unipolar NRZ Bipolar NRZ

Unipolar Non Return to Zero (NRZ)

1 is represented by a pulse of constant amplitude for the entire duration of the bit interval, and 0 by no pulse. NRZ indicates that the assigned amplitude is maintained for the entire bit period. This is the format of TTL or RS323 signalling. Advantages: It is the simplest to implement. Requires only one power supply. Disadvantages It has a DC voltage equal to half the voltage used to represent a 1. Requires DC coupling If there is a long stream of 1s or 0 there are no voltage transitions and timing information will be lost.

Unipolar RZ Biplolar RZ AMI

Manchester

Bipolar NRZ
Pulses of equal positive and negative amplitudes represent 1 and 0. The assigned amplitude level is maintained for the bit interval. Advantages: It is simple to implement. Has no DC. More immune to noise than NRZ - goes from +v to -v rather than from +v to 0v. Disadvantages Has the same timing information problem as NRZ. Requires double power supply. Twice the bandwidth of NRZ.

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Unipolar Return to Zero (RZ)

1 is represented by a positive pulse that returns to zero before the end of the bit interval and 0 by the absence of a pulse. Advantages: It is simple to implement. (Derived from U-NRZ AND Clock) Single power supply. Has timing information in the case of a long string of 1s. Disadvantages; Has an average DC voltage requiring DC coupling. Loses timing information for a long string of 0s - like NRZ.

Bipolar RZ (B-RZ)
Positive and negative pulses of equal amplitude are used for 1 and 0. The pulse returns to 0 before the end of the bit interval. Advantages: Has timing information in the case of a long string of 1s and 0s. Better noise immunity then Unipolar RZ Disadvantages; Wider bandwidth than NRZ

Alternate Mark Inversion (AMI) RZ Signalling

Positive and negative pulses (of equal amplitude) are used for alternative 1. No pulse is used for 0. The pulse returns to 0 before the end of the bit interval. A version of this in PCM (High Density Bipolar 3 (HDB3). Advantages: An advantage of AMI is that it is easy to recognise a line violation. Has timing information in the case of a long string of 1s. Loses timing information for a series of 0 hence the inclusion of parity violating 0 bits in PCM. Used in synchronous systems where a small amount of clock drift is allowed for a short period of time but will be corrected. Mean power level is low. No DC voltage. Disadvantages; Loses timing information for a long string of 0s - like NRZ. The variant used in PCM has additional transitions to handle this. Relatively complex to recover

Manchester or Biphase Coding

1 is represented by a positive pulse followed by a negative pulse - with each pulse being of equal amplitude and duration of half a pulse. For 0 the polarities of these pulses are reversed. Advantages: Fairly simple to implement and to recover. (= NRZ X-OR Clock) An advantage of this coding is that it is easy to recover the original data clock regardless of whether there are strings of 1s or 0s. It is used in LANs and other asynchronous systems for this reason - because the receiving machine does not know when to expect a packet of data to arrive it must be able to recover the timing information very fast. No DC voltage. Relatively immune to noise, because the code for a 1 equals the inverse of the code for 0 - the most different it can possibly be. Disadvantages; Wider bandwidth than NRZ.
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Baseband Receiver
Assume that the coding used is Bipolar Non Return to Zero (B-NRZ) with each bit duration being T. In the transmitted data a 1 is represented by +V and 0 by -V. In each case the 1

Transmitted Signal

Received Signal

or 0 lasts for a time T. The received signal will also have this form but will also contain noise and distortion. The peak noise voltage may be greater than V. It is seen that the received signal may be highly distorted so as to virtually hide the original signal. Dump Switch SW1 Recovery of the data is based on the idea that the noise in random, varies rapidly in the long term and has a mean value of zero, but the signal has a defined set of values which are held for the duration of Sample Switch SW2 _ a bit. The structure of a circuit to recover v(t) Incoming Data v(T) the signal is shown. Plus Noise The amplifier is an integrator. The input signal is the wanted signal with unwanted added noise. At the start of a pulse i.e. at time t = 0, the capacitor is discharged by closing dump switch SW1 momentarily. At the end of a pulse, i.e. at time t = T, the output is sampled by closing sample switch SW2 momentarily to look at the final voltage. The capacitor is then discharged again for the next pulse. The value of a sample is the voltage output of the integrator at time T.

Signal recovery in presence of noise

The output of the integrator circuit is: v(t) = -1/RCvin(t) dt = -1/vin(t) dt. with time constant defined so that = RC The input signal is the wanted signal s(t) with unwanted added noise n(t) i.e. vin(t) = s(t) + n(t). At the start of a pulse i.e. time t = 0, when SW1 discharges the capacitor, the output is v(t) = 0. At the end of a pulse, i.e. time t = T, when the output is sampled by closing SW2 the output is v(t) = -v(T) i.e. v(T) = 1

vin(t) dt

1 s(t) dt

n(t) dt

or v(T) = So(T) + No(T) The signal voltage vin(t) is either +V or -V (assume +V) so that the sample voltage due to the signal is VT S0(T) = 1

s(t) dt =

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The sample voltage due to the noise is

N0(T) =

n(t) dt

The value of the latter needs to be calculated by looking at the operation of an integrator. We can consider the noise to be made up of a wide range of frequency components (white noise is made up of all frequencies, all of equal amplitude). Look at a single component of frequency fn and amplitude an. The output Nn(T) from the integrator at time T due to this frequency is Nn(T) = = 1 an

an sin (2 fn t) dt

an

1 1 cos (2 fn t) 2 fn

T 0

1 1 (1 - cos (2 fn T)) 2 fn 1 . fn

It is seen that the amplitude of this component of the output is proportional to

The integrator acts like a low pass filter, filtering out the higher noise frequencies. The integrator (a circuit like this is actually called a matched filter) causes the signal voltage to increase linearly over the integration period T whereas the noise voltage stays very small. The noise is random and is as often positive as negative so that over a long time its average value is 0 and it integrates to 0, but the RMS of the integrated noise voltage does increase with the square root of the integration time. This gives: S0(T) = kT N0(T) = k

If the pulse period is made long enough we can have S0(T) > N0(T) and can distinguish the signal from the noise. The form of the output from the integrator is shown for an input pulse train which contains a great deal of noise. The output voltage when the sample switch closes at time T is the sum of the integral of the signal input voltage and the integral of the noise input Received Signal voltage. If this is greater than 0 then the pulse is interpreted as a 1. If it is less than 0 the pulse is interpreted as a 0. If the integration period (the length of the pulse) is Signal Output sufficiently long it should happen that the integral of the signal input voltage will be very much greater than the integral of the noise input voltage at the sample Noise Output time (the end of the integration period) so that the probability that the signal output will be so distorted as to cause a mis-identification of a 1 or 0 should be very small.

Effect of timing errors

For this circuit to operate correctly the integration MUST start exactly at the start of a pulse period, and the sampling MUST take place exactly at the end of pulse period. For this to happen the integrator period must be precisely matched to, and be in phase with, the pulse period. The integrator and associated circuitry is then called a matched filter for the particular type of pulse coding (B-NRZ). For other coding systems the matched filter will be similar in principle, but different in detail. If the data is used to modulate a carrier (ASK, FSK, BPSK, QPSK, QAM etc.), then the
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matched filter contains a module such as a balanced modulator to act on the carrier prior to integrator. As seen above the output from the integrator at the end of a pulse period is: S0(T) = 1

V dt

VT

where +V represents 1 and -V represents 0. Now let the timing in the receiver be off by a small amount so that the dump switch across the capacitor is late in closing by an amount T. This means that T before the end of the pulse the input signal changes polarity so that the output of the integrator at the end of a pulse period is: 1 S0(T) = = or
T T

1 V dt

T T

-V dt

V =(
-

T T

dt -

T T

dt) = -

V T T T ( |0 - | T T ) = V S0(T) = (T- 2T)

-

V (T- T -T)

V (T- 2T)

This is less than the output would be if there was no timing error so that a smaller value of noise is now more likely to cause a data error. In this diagram assume that the output from Input data Stream (B-NRZ) the integrator at the end of a data pulse is 8v. The integrated output noise would have to be greater than 8v for a data error to be made. The next part shows the effect on the integrator output if T = 0.2 * T (error of 20%). This reduces the output signal by 40% to a value of 4.8v. A smaller level of noise will Output now cause data errors. No timing error Finally, yf the timing error is 40% the integrator output is reduced by 89% to 0.64v. Very little noise is now required to cause a data error.

Output with Timing error T Output - large Timing error T

M-ary Coding
The utilisation of bandwidth can be made more efficient by adopting a M-ary format for the representation of the input binary data . A binary code consists of two symbols- 1 and 0. A quaternary (i.e. 4 level) code would consist of 4 symbols. The 4 symbols could be assigned to 00, 01, 10 and 11 for example. This would allow us to half the symbol rate on a transmission line compared to one bit per symbol for the
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same throughput of data, or to double the number of bits transmitted in a given time. Note that binary data rate is measured in bits/second whereas the symbol rate is measured in Baud. (Symbols per second). One M-ary line coding is 2B1Q line code used on ISDN basic rate lines between a customer premises and the local 11 exchange. In this case the baud rate is half the bit rate. (There 10 are 4 possible symbols, each of which requires 2 bits. If the Time probability of each symbol is then the information in each symbol is log 2 (1/) = log 2 4 = 2 bits, so that the information 01 rate is 2 * baud rate = bit rate). View the binary sequence 00 11100001 as a sequence of dibits ; 11 10 00 01. Each dibit symbol is assigned one of 4 levels. By increasing the number of levels there will be a trade-off between noise performance and bandwidth. Example: Explain how a ternary line coding system can code 3 bits per symbol. Answer: At 3 bits per symbol, 23 = 8, therefore we need at least 8 symbols. A single ternary pulse would only allow one of 3 symbols to be represented. Two ternary pulses in a particular order, however would allow for 9 combinations of levels. This code could be abbreviated as 3B2T. It is not as efficient as 2B1Q but one advantage is that zero volts is one of the levels and the wave form would resemble a binary bipolar format, Note that 4B3T coding is also used on ISDN lines in some countries.

Noise Immunity for M-ary Coding

Consider the following cases, in each of which the code format used is Binary-NRZ. Case 1: A binary system, with two code symbols 1 and 0. Assume that 1 is represented by a voltage = +5V, and 0 by a voltage of -5V. There is a 10V separation between these levels. At first approximation if the noise voltage picked up by the signal is less than 5V then a 1 will never be mistaken for a 0 , and vice versa. We can say that the system is immune to noise of less than 5V. Case 2: A quaternary system, with four code symbols 11, 10, 01 and 00. Assume that 11 is represented by +5V , and 00 by a voltage of -5V, so that the full voltage range of the system is still as in case 1.We now have to put the other symbols into this range. We would have 10 represented by +1.667V, and 01 represented by -1.667V. This means that there is 3.333V separation between all levels. At a first approximation if the noise voltage is less than 1.667V one symbol will never be mistaken for another one. The noise immunity of this system which carries 2 bits per symbol is now 1.667V. Case 2 will not work in an environment which has as much noise as Case 1. Case 3: A 8-ary system, with eight code symbols from 000 to 111. Assume that 111 is represented by +5V , and 000 by a voltage of -5V, so that again the full voltage range of the system is still as in Case 1.In this case there is 10/7 = 1.43V separation between all levels. As before, at a first approximation if the noise voltage is less than 0.71V one symbol will never be mistaken for another one. The noise immunity of this system which carries 3 bits per symbol is now 0.71V. Case 3 requires an even less noisy environment than Case 2. The same process can be continued for any number of bits per symbol. In general, it can be seen that to increase the number of bits transmitted per symbol it is necessary to operate in a less noisy environment. In all cases, the background noise
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level is the limiting factor in deciding the rate at which data can be transmitted across a network.

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