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Digital Communication (EC51)

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Einstein College of Engineering

Chapter-1: Introduction The purpose of a Communication System is to transport an information bearing signal from a source to a user destination via a communication channel. MODEL OF A COMMUNICATION SYSTEM(ANALOG)

Information Source and Input Transducer

I/P Signal

TRANSMITTER CHANNEL

Destination and Output Transducer

O/P Signal

RECEIVER

Fig. 1.1: Block diagram of Communication System. The three basic elements of every communication systems are Transmitter, Receiver and Channel. The Overall purpose of this system is to transfer information from one point (called Source) to another point, the user destination. The message produced by a source, normally, is not electrical. Hence an input transducer is used for converting the message to a time – varying electrical quantity called message signal. Similarly, at the destination point, another transducer converts the electrical waveform to the appropriate message. The transmitter is located at one point in space, the receiver is located at some other point separate from the transmitter, and the channel is the medium that provides the electrical connection between them. The purpose of the transmitter is to transform the message signal produced by the source of information into a form suitable for transmission over the channel. The received signal is normally corrupted version of the transmitted signal, which is due to channel imperfections, noise and interference from other sources.The receiver has the task of operating on the received signal so

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Einstein College of Engineering

as to reconstruct a recognizable form of the original message signal and to deliver it to the user destination. Communication Systems are divided into 3 categories: 1. Analog Communication Systems are designed to transmit analog information using analog modulation methods. 2. Digital Communication Systems are designed for transmitting digital information using digital modulation schemes, and 3. Hybrid Systems that use digital modulation schemes for transmitting sampled and quantized values of an analog message signal. ELEMENTS OF DIGITAL COMMUNICATION SYSTEMS: The figure 1.2 shows the functional elements of a digital communication system. Source of Information: 1. Analog Information Sources. 2. Digital Information Sources. Analog Information Sources → Microphone actuated by a speech, TV Camera scanning a scene, continuous amplitude signals. Digital Information Sources → These are teletype or the numerical output of computer which consists of a sequence of discrete symbols or letters. An Analog information is transformed into a discrete information through the process of sampling and quantizing. Digital Communication System

Source of Information Source Encoder Channel Encoder

Modulator

Channel

Received Signal

User of Information Source Decoder Channel Decoder Demodulator

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:-If a source set is having hundred symbols. CHANNEL ENCODER / DECODER: Error control is accomplished by the channel coding operation that consists of systematically adding extra bits to the output of the source coder.e. Aim of the source coding is to remove the redundancy in the transmitting information. shorter is the codeword.2: Block Diagram of a Digital Communication System SOURCE ENCODER / DECODER: The Source encoder ( or Source coder) converts the input i. the source decoder converts the binary output of the channel decoder into a symbol sequence. 4 . symbol sequence into a binary sequence of 0‟s and 1‟s by assigning code words to the symbols in the input sequence. so that bandwidth required for transmission is minimized. 2. Based on the probability of the symbol code word is assigned. Convolution Coding: The information bearing message stream is encoded in a continuous fashion by continuously interleaving information bits and error control bits. actual output data rate compared to the minimum achievable rate) At the receiver. The important parameters of a source encoder are block size.Einstein College of Engineering Fig 1. Higher the probability. where „r‟ is dependent on „k‟ and error control capabilities desired. Block Coding: The encoder takes a block of „k‟ information bits from the source encoder and adds „r‟ error control bits. The decoder for a system using fixed – length code words is quite simple. but the decoder for a system using variable – length code words will be very complex. then the number of bits used to represent each symbol will be 7 because 27=128 unique combinations are available. There are two methods of channel coding: 1. For eg. These extra bits do not convey any information but helps the receiver to detect and / or correct some of the errors in the information bearing bits.e. Ex: Huffman coding. average data rate and the efficiency of the coder (i. code word lengths.

DEMODULATOR: The extraction of the message from the information bearing waveform produced by the modulation is accomplished by the demodulator. The important parameters of the channel are Signal to Noise power Ratio (SNR). efficiency. to match the frequency spectrum of transmitted signal with channel characteristics. The effect of distortion. to provide the capability to multiplex many signals. Optical fibre. non-ideal frequency response. the signal often suffers amplitude and phase distortion as it travels over the channel. unpredictable electrical signals referred to as noise. The output of the demodulator is bit stream. The different channels are: Pair of wires. Radio channel. usable bandwidth. Coaxial cable. The important parameter is the method of demodulation. This is because the disturbance must be large enough to change the pulse from one state to the other. Also. Advantages of Digital Communication 1. MODULATOR: The Modulator converts the input bit stream into an electrical waveform suitable for transmission over the communication channel. The important parameters of coder / decoder are: Method of coding. error control capabilities and complexity of the circuit. Error detection and possible correction is also performed by the channel decoder. Modulator can be effectively used to minimize the effects of channel noise. the signal power decreases due to the attenuation of the channel. noise and interference is less in a digital communication system. CHANNEL: The Channel provides the electrical connection between the source and destination. Satellite channel or combination of any of these.Einstein College of Engineering The Channel decoder recovers the information bearing bits from the coded binary stream. 5 . The signal is corrupted by unwanted. amplitude and phase response and the statistical properties of noise. The communication channels have only finite Bandwidth.

Signal processing functions like encryption. The different types of signals such as data. Channels for Digital Communications The modulation and coding used in a digital communication system depend on the characteristics of the channel. 6 . Disadvantages of Digital Communication: 1. Error detecting and Error correcting codes improve the system performance by reducing the probability of error. In addition the other characteristics are whether the channel is linear or nonlinear. 2. TV can be treated as identical signals in transmission and switching in a digital communication system. 4. Combining digital signals using TDM is simpler than combining analog signals using FDM. 6.Einstein College of Engineering 2. System Synchronization:.Digital detection requires system synchronization whereas the analog signals generally have no such requirement. The Hardware implementation is more flexible than analog hardware because of the use of microprocessors. 8. We can avoid signal jamming using spread spectrum technique.Digital transmission requires a large system bandwidth to communicate the same information in a digital format as compared to analog format. Large System Bandwidth:. telephone. Digital circuits are more reliable and cheaper compared to analog circuits. and how free the channel is free from the external interference. VLSI chips etc. Regenerative repeaters can be used at fixed distance along the link. 5. compression can be employed to maintain the secrecy of the information. The two main characteristics of the channel are BANDWIDTH and POWER. 3. 7. to identify and regenerate a pulse before it is degraded to an ambiguous state.

Einstein College of Engineering Five channels are considered in the digital communication. Telephone channel: It is designed to provide voice grade communication. Also good for data communication over long distances. surrounded by a concentric layer called cladding that is also made of glass. But for the transmission of data and image transmissions. it is continually refracted into the core by the cladding. a high SNR of about 30db. But closely spaced repeaters are required. since the phase delay variations are important an equalizer is used to maintain the flat amplitude response and a linear phase response over the required frequency band. coaxial cables. The antennas are placed on towers at sufficient height to have the transmitter and receiver in line-of-sight of each other. The refractive index of the glass in the core is slightly higher than refractive index of the glass in the cladding. Compared to coaxial cables. The main advantages of the coaxial cable are wide bandwidth and low external interference. Microwave radio: A microwave radio. The channel has a band-pass characteristic occupying the frequency range 300Hz to 3400hz.8 kilobits per second have been achieved over the telephone lines. consists basically of a transmitter and a receiver that are equipped with antennas. and approximately linear response. Transmission rates upto16. optical fibers. operating on the line-of-sight link. That means the difference between the refractive indices of the core and cladding helps guide the propagation of the ray of light inside the core of the fiber from one end to the other. The operating frequencies range from 1 to 30 GHz. With repeaters spaced at 1km intervals the data rates of 274 megabits per second have been achieved. microwave radio. and satellite channels. optical fibers are smaller in size and they offer higher transmission bandwidths and longer repeater separations. which is insulated from each other by a dielectric. Hence if a ray of light is launched into an optical fiber at the right oblique acceptance angle. 7 . namely: telephone channels. For the transmission of voice signals the channel provides flat amplitude response. Coaxial Cable: The coaxial cable consists of a single wire conductor centered inside an outer conductor. Optical Fibers: An optical fiber consists of a very fine inner core made of silica glass.

Einstein College of Engineering Under normal atmospheric conditions. The bandwidth is normally a difference between two numbers. for a signal s(t). In general. are real numbers (there are infinite such values) within a finite range. Satellite Channel: A Satellite channel consists of a satellite in geostationary orbit. as in “my Internet connection has 1 Mbps of bandwidth”. But during meteorological variations. meaning it can transmit data at 1 megabit per second.40 GHz to 2. a severe degradation occurs in the system performance. T1≤ t ≤ T2. the signal is analog. with uplink the uplink frequency higher than the down link frequency. If a range of 2. a digital signal implies a discrete-time. Satellite can be viewed as repeater in the sky. a microwave radio channel is very reliable and provides path for high-speed digital transmission. the more data you can fit in at a given moment. Both link operate at microwave frequencies. and a down link to another ground station. say. The range of frequencies contained in a composite signal is its bandwidth.48 GHz is used by a device. if a composite signal contains frequencies between 1000 and 5000. It permits communication over long distances at higher bandwidths and relatively low cost.08 GHz (or more commonly stated as 80MHz). the energy of the signal is finite. or 4000. Smin ≤ S(t) ≤ Smax. then the bandwidth would be 0. Bandwidth: Bandwidth is simply a measure of frequency range. Energy signal: If.the signal is called an energy signal. discrete-amplitude signal. The term bandwidth is often used for something we should rather call a data rate. A digital signal s(t). Usually. Geometric representation of Signals: Analog signal: If the magnitudes of a real signal s(t) over its range of definition.It is easy to see that the bandwidth we define here is closely related to the amount of data you can transmit within it . can assume only any of a finite number of values. the same signal may have large power.The voltage generated by 8 .e. i. an uplink from ground station. However.the more room in frequency space. its bandwidth is 5000 1000. For example. on the contrary.

represent power signals. While electrical signals. T1≤ t ≤ T2.Einstein College of Engineering lightning (which is of short duration) is a close example of physical equivalent of a signal with finite energy but very large power. on the contrary. Continuous time signal: Assuming the independent variable „t‟ to represent time. if s(t) is defined for all possible values of t between its interval of definition (or existence). „T‟ indicates the period of the signal and 1/T is its frequency of repetition. Then the signal s(t) is a continuous time signal. usually deterministic. for entire range of t over which the signal s(t) is defined and T is a constant. then s(t) represents a deterministic signal. described at t = t1 is sufficient for determining the signal at t = t2 at which the signal also exists. A set of sample values represent a discrete time signal. it is a discrete-time signal. Mathematically. derived from physical processes are mostly energy signals. will have a finite power but may have finite or infinite energy. Deterministic and random signals: If a signal s(t). s(t) is said to be periodic or repetitive. 9 . Power signal: A power signal.several mathematical functions.If a signal s(t) is defined only for certain values of t over an interval T1≤ t ≤ T2. Periodic signal: If s(t) = s(t + T).

fs is equal to the Nyquist rate or greater (fs ≥ 2W). Sampling operation is performed in accordance with the sampling theorem.“If a band –limited signal g(t) contains no frequency components for ׀f > ׀W. The process by which the continuous-time signal is converted into a discrete–time signal is called Sampling. g(t) sδ (t) -2Ts -Ts 0 1Ts 2Ts 3Ts 4Ts gδ(t) -2Ts -Ts 0 Ts 2Ts 3Ts 4Ts Fig : Sampling process 10 .Einstein College of Engineering Chapter-2 SAMPLING: A message signal may originate from a digital or analog source. SAMPLING THEOREM FOR LOW-PASS SIGNALS:Statement:. If the sampling rate. If the message signal is analog in nature. then it has to be converted into digital form before it can transmitted by digital means. then it is completely described by instantaneous values g(kT s) uniformly spaced in time with period Ts ≤ 1/2W. the signal g(t) can be exactly reconstructed.

Σ δ(f. the impulse train in time domain.Σ δ(f.nTs) -------------1. is an impulse train in frequency domain: F{Σ δ(t. Part – II The signal x(t) can be accurately reconstructed (recovered) from the set of uniform instantaneous samples by passing the samples sequentially through an ideal (brick-wall) lowpass filter with bandwidth B. when interpreted appropriately.n/Ts) = fs.1 where x(nTs) = x(t)⎢t =nTs .nTs) ---------.X(f)*Σ δ(f.δ(t.e. (1. {x(nTs)}≡ xs(t) = Σ x(t).2. xs(t) = x(t).Einstein College of Engineering Proof:Part .Σ δ(t.5 This equation.2 Now.1.nfs) --------------1. -------------1.T of Σ δ(t. δ(t) is a unit pulse singularity function and „n‟ is an integer. the equivalent set of instantaneous uniform samples {x(nTs)} may be represented as.Σ δ(f. we know that the F.nTs)} = X(f)*[fs. we can write using Eq. i.nfs) -----1.The continuous-time signal x(t) is multiplied by an (ideal) impulse train to obtain {x(nTs)} and can be rewritten as.I If a signal x(t) does not contain any frequency component beyond W Hz.nTs). then the signal is completely described by its instantaneous uniform samples with sampling interval (or period ) of Ts < 1/(2W) sec.4 If Xs(f) denotes the Fourier transform of the energy signal xs(t). gives an intuitive proof to 11 . If x(t) represents a continuous-time signal. from the theory of Fourier Transform.4) and the convolution property: Xs(f) = X(f)* F{Σ δ(t.nfs)] = fs. where W ≤ B < fs – W and fs = 1/(Ts).3 Now.nTs)} = (1/Ts). let X(f) denote the Fourier Transform F(T) of x(t).

Let us note that while writing Eq. centered at f = 0 and will reject all its replicas at f = n fs. This implies that xs(t). This implies that the shape of the continuous time signal xs(t). centered at discrete frequencies n. the Fourier Transform of the sampled signal xs(t) consists of infinite number of replicas of X(f).Einstein College of Engineering Nyquist‟s theorems as stated above and also helps to appreciate their practical implications. 1.1. With this setting. we assumed that x(t) is an energy signal so that its Fourier transform exists.1.W i. As seen from Fig. when this condition is satisfied.2. an ideal lowpass filter (with brick-wall type response) with a bandwidth W ≤ B < (fs – W). the spectra of xs(t). contains all information about x(t) and thus represents x(t).fs. (fs – W) > W and (– fs + W) < – W.5).(1. As indicated by dotted lines in Fig. centered at discrete frequency values n.(1. when fed with xs(t). Part – I of the sampling theorem is about the condition fs > 2. 12 . for n ≠ 0. will be retained at the output of the ideal filter.fs.2. the spectrum of x(t) is present in xs(t) without any distortion. 1. the appropriately sampled version of x(t). then Eq. if we assume that x(t) has no appreciable frequency component greater than W Hz and if fs > 2W.2. will allow the portion of Xs(f).e. 1.5) implies that Xs(f). -∞ < n < ∞ and scaled by a constant fs= 1/Ts Fig. centered at f = 0 and f = ± fs do not overlap and hence.1 indicates that the bandwidth of this instantaneously sampled wave xs(t) is infinite while the spectrum of x(t) appears in a periodic manner. The second part suggests a method of recovering x(t) from its sampled version xs(t) by using an ideal lowpass filter.

exact reconstruction is possible in which case the signal g(t) may be considered as a low pass signal itself. fs 4B 3B 2B B 0 B 2B 3B 4B 5B fu Fig 2. then g(t) can be exactly reproduced from it‟s samples by an ideal Band-Pass filter with the response.Einstein College of Engineering Sampling of Band Pass Signals: Consider a band-pass signal g(t) with the spectrum shown in figure 2. Upper cutoff frequency and Bandwidth. fs ≥ 2fu. H(f) defined as H(f) = 1 0 fl < | f | <fu elsewhere If the sampling rate.2: G(f) B Band width = B Upper Limit = fu Lower Limit = fl -fu -fl 0 fl fu f B Fig 2. g(kTs) if the sampling rate fs is (2fu/m) where m is an integer defined as ((fu / B) -1 ) < m ≤ (fu / B) If the sample values are represented by impulses.3: Relation between Sampling rate.2: Spectrum of a Band-pass Signal The signal g(t) can be represented by instantaneous values. 13 .

Accordingly each component may be sampled at the rate of 2W samples per second. Then the sampling rate is fs = 2fu / m = 240K / 2 = 120KHz.Einstein College of Engineering Example-1 : Consider a signal g(t) having the Upper Cutoff frequency. In this scheme. The ratio of upper cutoff frequency to bandwidth of the signal g(t) is fu / B = 100K / 20K = 5. The ratio of upper cutoff frequency to bandwidth of the signal g(t) is fu / B = 120K / 50K = 2. gI(t) is obtained by multiplying g(t) with cos(2πfct) and then filtering out the high frequency components. Lower Cutoff frequency fl = 70KHz. having band limited to (-W < f < W). This form of sampling is called quadrature sampling.. Let g(t) be a band pass signal. fc. of bandwidth „2W‟ centered around the frequency. the band pass signal is split into two components. gI(t) and quadrature phase gQ(t) signals are low–pass signals. The in-phase component. one is in-phase component and other is quadrature component. m is an integer less than (fu /B). fu = 120KHz and the Quadrature Sampling of Band – Pass Signals: This scheme represents a natural extension of the sampling of low – pass signals. Parallelly a quadrature phase component is obtained by multiplying g(t) with sin(2πfct) and then filtering out the high frequency components. (fc>W). These two components will be low–pass signals and are sampled separately. Then the sampling rate is fs = 2fu / m = 200K / 5 = 40KHz fu = 100KHz and the Example-2 : Consider a signal g(t) having the Upper Cutoff frequency. Lower Cutoff frequency fl = 80KHz. g(t) = gI(t).4 Therefore we can choose m = 2. cos(2πfct) – gQ(t) sin(2πfct) The in-phase. ie. 14 . The band pass signal g(t) can be expressed as. Therefore we can choose m = 5..

Einstein College of Engineering g(t)cos(2πfct) LPF g(t) cos (2πfct) g(t) sin(2πfct) LPF ½ gI (t) sampler ½ gI (nTs) ½gQ(t) -½ gQ(nTs) sampler sin (2πfct) Fig 2.4: Generation of in-phase and quadrature phase samples G(f) -fc 0 fc 2W-> f a) Spectrum of a Band pass signal. GI(f) / GQ(f) -W 0 W f b) Spectrum of gI(t) and gQ(t) Fig 2.5 a) Spectrum of Band-pass signal g(t) b) Spectrum of in-phase and quadrature phase signals 15 .

In both the natural sampling and flat-top sampling methods.6: Reconstruction of Band-pass signal g(t) Sample and Hold Circuit for Signal Recovery. gI(nTs) Reconstruction Filter + Cos (2πfct) gQ(nTs) Reconstruction Filter Σ g(t) Sin (2πfct) Fig 2. where τ is the pulse duration and Ts is the sampling period. the signal power at the output of the reconstruction filter is correspondingly small. Since this ratio is very small. the signals gI(t) and gQ(t) are obtained. To overcome this problem a sample-and-hold circuit is used . the spectrum of the signals are scaled by the ratio τ/Ts.Einstein College of Engineering RECONSTRUCTION: From the sampled signals gI(nTs) and gQ(nTs). multiply the signals g I(t) by cos(2πfct) and sin(2πfct) respectively and then add the results. SW AMPLIFIER Input g(t) Output u(t) a) Sample and Hold Circuit 16 . To reconstruct the original band pass signal.

s(t) Fig: 2. The output xs(t) of the sampler consists of segments of x(t) and hence x s(t) can be considered as the product of x(t) and sampling function s(t). The Sample-and-Hold circuit consists of an amplifier of unity gain and low output impedance. during which time the capacitor charges up to a voltage level equal to that of the input sample. The switch is timed to close only for the small duration of each sampling pulse.7 Sample Hold Circuit with Waveforms. Thus the sample-and-hold circuit produces an output waveform that represents a staircase interpolation of the original analog signal. Natural Sampling: In this method of sampling. 17 . it is assumed that the load impedance is large. the capacitor retains the voltage level until the next closure of the switch. When the switch is open .Einstein College of Engineering b) Idealized output waveform of the circuit Fig: 2. an electronic switch is used to periodically shift between the two contacts at a rate of fs = (1/Ts ) Hz. staying on the input contact for C seconds and on the grounded contact for the remainder of each sampling period. xs(t) = x(t) . a switch and a capacitor.8 Natural Sampling – Simple Circuit.

.9 Natural Sampling – Waveforms.. … n≠0 1 X(f) f -W 0 +W Message Signal Spectrum Xs(f) C0 C2 C1 C1 C2 f -2fs -fs -W 0 +W fs 2fs Sampled Signal Spectrum (fs > 2W) 18 .Einstein College of Engineering Fig: 2.X(f) + C1 [X(f-f0) + X(f+f0)] + C2 [X(f-f0) + X(f+f0)] + . Applying Fourier transform Using x(t) x(t) cos(2πf0t) X(f) ½ [X(f-f0) + X(f+f0)] FT Xs(f) = Co.

. The quantization Process has a two-fold effect: 1.Einstein College of Engineering Fig:2. Since the spectrum is not distorted it is possible to reconstruct x(t) from the sampled waveform xs(t). independently of earlier analog samples applied to the input. the peak-to-peak range of the input sample values is subdivided into a finite set of decision levels or decision thresholds that are aligned with the risers of the staircase. Analog Signal Discrete Samples ( Quantized ) 0 Ts 2Ts 3Ts Time 19 . the output is assigned a discrete value selected from a finite set of representation levels that are aligned with the treads of the staircase. But the message term is scaled by „Co”. Quantization: The process of transforming Sampled amplitude values of a message signal into a discrete amplitude value is referred to as Quantization.10 Natural Sampling Spectrum The signal xs(t) has the spectrum which consists of message spectrum and repetition of message spectrum periodically in the frequency domain with a period of f s. and 2. A quantizer is memory less in that the quantizer output is determined only by the value of a corresponding input sample.

the origin lies the middle of the tread portion in Mid –Tread type where as the origin lies in the middle of the rise portion in the Mid-Rise type.Einstein College of Engineering Fig:2. Uniform Quantizer 2. Types of Uniform Quantizers: ( based on I/P . Mid-Rise type Quantizer 2. 20 . the quantization levels are uniformly spaced.Uniform Quantizer Uniform Quantizer: In Uniform type. Mid – Rise type: Quantization levels – even number. Non. whereas in nonuniform type the spacing between the levels will be unequal and mostly the relation is logarithmic. Mid – tread type: Quantization levels – odd number.O/P Characteristics) 1. Types of Quantizers: 1.11 Typical Quantization process. Mid-Tread type Quantizer In the stair case like graph.

Einstein College of Engineering Output 7Δ/2 5Δ/2 3Δ/2 Δ/2 Δ 2Δ 3Δ 4Δ Input Fig:2.12 Input-Output Characteristics of a Mid-Rise type Quantizer Output 2Δ Δ Δ/2 3Δ/2 Input Fig:2.13 Input-Output Characteristics of a Mid-Tread type Quantizer 21 .

. ± 2 Δ.14 MODEL OF NON UNIFORM QUANTIZER 22 .” q(t) = x(t) – y(t) Quantization noise is produced in the transmitter end of a PCM system by rounding off sample values of an analog base-band signal to the nearest permissible representation levels of the quantizer. The use of a non – uniform quantizer is equivalent to passing the baseband signal through a compressor and then applying the compressed signal to a uniform quantizer.. . yt). . The resultant signal is then transmitted.Einstein College of Engineering Quantization Noise and Signal-to-Noise: “The Quantization process introduces an error defined as the difference between the input signal. Quantization levels are 0. x(t) and the output signal. L = Number of quantization levels X = Quantizer input Y = Quantizer output The output y is given by Y=Q(x) which is a staircase function that befits the type of mid tread or mid riser quantizer of interest. Consider a memory less quantizer that is both uniform and symmetric. ± Δ. The Quantization error.. Non – Uniform Quantizer: In Non – Uniform Quantizer the step size varies. UNIFORM QUANTIZER COMPRESSOR EXPANDER Fig: 2. . As such quantization noise differs from channel noise in that it is signal dependent. . If Δ is small. ±3 Δ . Q is a random variable and will have its sample values bounded by [-(Δ/2) < q < (Δ/2)]. This error is called the Quantization Noise. . the quantization error can be assumed to a uniformly distributed random variable. Let „Δ‟ be the step size of a quantizer and L be the total number of quantization levels.

Einstein College of Engineering At the receiver, a device with a characteristic complementary to the compressor called Expander is used to restore the signal samples to their correct relative level. The Compressor and expander take together constitute a Compander. Compander = Compressor + Expander Advantages of Non – Uniform Quantization : 1. Higher average signal to quantization noise power ratio than the uniform quantizer when the signal pdf is non uniform which is the case in many practical situation. 2. RMS value of the quantizer noise power of a non – uniform quantizer is substantially proportional to the sampled value and hence the effect of the quantizer noise is reduced.

Encoding: Encoding is the final step in what we call analog-to-digital (A/D) conversion.Sometimes, encoding is considered as a separate stage that follows the A/D stage. The encoding stage takes a bit stream of 1‟s and 0‟s and converts it into voltages appropriate for transmission on a physical channel. Encoding schemes are usually divided into: Source encoding, Channel encoding and Line encoding . Source coding: Source coding (sometimes called entropy encoding) refers to the process of compressing data. This is typically done by replacing long binary codes (named codewords) that occur frequently by shorter ones, and those that occur less frequently by longer codes. For example, a 4-bit sequence “0110” occurring frequently can be mapped into the shorter 2-bit “01” sequence, while another 4-bit sequence “1011” occurring less frequently can be mapped to the longer 7-bit sequence “0011011”. This makes sure that shorter sequences occur more often in the bit stream. In information theory, Shannon's noiseless coding theorem places an upper and a lower bound on the expected compression ratio. Examples of source codes currently in use are: Shannon codes, Huffman codes, run-length coding, arithmetic coding, Lempel-Ziv coding, MPEG video coding, etc. Channel coding: A channel code is a broadly used term mostly referring to error correcting codes. Such codes are used to protect data sent over the channel from corruption even in the presence of noise. In other words, channel codes

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Einstein College of Engineering

can improve the signal-to-noise ratio (SNR) of the received signal. The most obvious example of such codes is the simple parity bit system. The theory behind designing and analyzing channel codes is called Shannon’s noisy channel coding theorem. It puts an upper limit on the amount of information you can send in a noisy channel using a perfect channel code. This is given by the following equation: where C is the upper bound on the capacity of the channel (bit/s), B is the bandwidth of the channel (Hz) and SNR is the Signal-to-Noise ratio (unitless). Examples of channel codes currently in-use include: Hamming codes, Reed-Solomon codes, convolutional codes (usually decoded by an iterative Viterbi decoder), Turbo codes, etc.

**Differential Pulse Code Modulation (DPCM)
**

For the signals which does not change rapidly from one sample to next sample, the PCM scheme is not preferred. When such highly correlated samples are encoded the resulting encoded signal contains redundant information. By removing this redundancy before encoding an efficient coded signal can be obtained. One of such scheme is the DPCM technique. By knowing the past behavior of a signal up to a certain point in time, it is possible to make some inference about the future values. The transmitter and receiver of the DPCM scheme is shown in the below fig.respectively. Transmitter: Let x(t) be the signal to be sampled and x(nTs) be it‟s samples. In this scheme the input to the quantizer is a signal e(nTs) = x(nTs) - x^(nTs) ----- (3.31)

where x^(nTs) is the prediction for unquantized sample x(nTs). This predicted value is produced by using a predictor whose input, consists of a quantized versions of the input signal x(nTs). The signal e(nTs) is called the prediction error. By encoding the quantizer output, in this method, we obtain a modified version of the PCM called differential pulse code modulation (DPCM). Quantizer output, v(nTs) = Q[e(nTs)] = e(nTs) + q(nTs) where q(nTs) is the quantization error.

---- (3.32)

Predictor input is the sum of quantizer output and predictor output, u(nTs) = x^(nTs) + v(nTs) ---- (3.33) 24

Einstein College of Engineering

Using 3.32 in 3.33,

u(nTs) = x^(nTs) + e(nTs) + q(nTs) ----(3.34) u(nTs) = x(nTs) + q(nTs) ----(3.35)

The receiver consists of a decoder to reconstruct the quantized error signal. The quantized version of the original input is reconstructed from the decoder output using the same predictor as used in the transmitter. In the absence of noise the encoded signal at the receiver input is identical to the encoded signal at the transmitter output. Correspondingly the receive output is equal to u(nTs), which differs from the input x(nts) only by the quantizing error q(nTs).

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Einstein College of Engineering Sampled Input x(nTs) Σ e(nTs) Quantizer v(nTs) Output + ^ x(nTs) Σ Predictor u(nTs) Block diagram of DPCM Transmitter Input v(nTs) u(nTs) Σ Output Decoder b(nTs) x^(nTs ) Predictor Block diagram of DPCM Receiver. 26 .

The difference between the input and the approximation is quantized into only two levels. DM provides a staircase approximation to the over sampled version of an input base band signal. Delta Modulation is the one-bit (or two-level) versions of DPCM. so as to permit the use of a simple quantizing strategy for constructing the encoded signal. it is increased by δ. if the approximation falls below the signal at any sampling epoch. Delta modulation (DM) is precisely such as scheme. Provided that the signal does not change too rapidly from sample to sample.Einstein College of Engineering Delta Modulation (DM) Delta Modulation is a special case of DPCM. ±δ corresponding to positive and negative differences. Output +δ 0 -δ Input Input-Output characteristics of the delta modulator. In DPCM scheme if the base band signal is sampled at a rate much higher than the Nyquist rate purposely to increase the correlation between adjacent samples of the signal. The symbol δ denotes the absolute value of the two representation levels of the one-bit quantizer used in the DM. we find that the stair case approximation remains within ±δ of the input signal. Thus. respectively. 27 . Let the input signal be x(t) and the staircase approximation to it is u(t). namely.

The out-of –band quantization noise in the high frequency staircase waveform u(t) is rejected by passing it through a low-pass filter with a bandwidth equal to the original signal bandwidth. Simple design for both Transmitter and Receiver u(nTs) Input b(nTs) Delay Ts u(nTs-Ts) Σ Low pass Filter Block diagram for Receiver of a DM system 28 . No need for Word Framing because of one-bit code word.Einstein College of Engineering Sampled Input x(nTs) Σ e(nTs) One . 2.Bit Quantizer b(nTs) Output + ^ x(nTs) Σ Delay Ts u(nTs) Block diagram for Transmitter of a DM system In the receiver the stair case approximation u(t) is reconstructed by passing the incoming sequence of positive and negative pulses through an accumulator in a manner similar to that used in the transmitter. Delta modulation offers two unique features: 1.

the size is adapted to the level of the input signal. In this way. and (2) granular noise. There are several types of ADM. In particular. Conversely. a discrete set of values is provided for the step size. Block Diagram of ADM Transmitter. The resulting method is called adaptive delta modulation (ADM). 29 . Adaptive Delta Modulation: The performance of a delta modulator can be improved significantly by making the step size of the modulator assume a time-varying form. during a steep segment of the input signal the step size is increased. In this ADM. depending on the type of scheme used for adjusting the step size. when the input signal is varying slowly.Einstein College of Engineering Disadvantage of DM: Delta modulation systems are subject to two types of quantization error: (1) slope –overload distortion. the step size is reduced.

Applications 1. the number of bits used to encode each sub-band is varied dynamically and shared with other sub-bands. Indeed. The coder is capable of digitizing speech at a rate of 16 kb/s with a quality comparable to that of 64 kb/s PCM. In particular. This periodicity permits pitch prediction. sub-bands with little or no energy may not be encoded at all. Light wave Transmission Link 30 . noise shaping is accomplished by adaptive bit assignment. and therefore a further reduction in the level of the prediction error that requires quantization. Hierarchy of Digital Multiplexers 2. In adaptive sub band coding (ASBC). it exploits the quasi-periodic nature of voiced speech and a characteristic of the hearing mechanism known as noise masking. compared to differential pulse code modulation without pitch prediction. Periodicity of voiced speech manifests itself in the fact that people speak with a characteristic pitch frequency. Adaptive sub-band coding is a frequency domain coder. To accomplish this performance. The number of bits per sample that needs to be transmitted is thereby greatly reduced. in which the speech signal is divided into a number of sub-bands and each one is encoded separately.Einstein College of Engineering Block Diagram of ADM Receiver Adaptive Sub-band Coding: PCM and ADPCM are both time-domain coders in that the speech signal is processed in the time-domain as a single full band signal. such that the encoding accuracy is always placed where it is needed in the frequency – domain characterization of the signal. without a serious degradation in speech quality.

Sixteen unique sequences can be obtained from four bit words. This signal extends in range from (-fc . one to each integer. we measure distance by Euclidean concepts such as lengths. The bandwidth is always a positive quantity so the bandwidth of this signal is fm. distances are measured between two binary words by something called the Hamming distance. k) codes.In the binary world. This number is 3 among the 10 codewords we have chosen. Although the Hamming weight of our chosen code set is 3. A block of k information bits are coded to become a block of n bits. The new signal has doubled in bandwidth. This signal is now called the passband signal.fm ) to (fc + fm. Let‟s say that we want to code the 10 integers. But before we go any further with the details. angles and vectors.Block codes are referred to as (n. let‟s look at an important concept in coding called Hamming distance. Passband Signal . (the ones in the while space) Concept of Hamming Distance: In continuous variables. Hamming Weight: The Hamming weight of this code scheme is the largest number of 1‟s in a valid codeword. The passband signal bandwidth is double that of the baseband signal.The information signal is called the baseband signal. The knowledge of Hamming distance is used to determine the capability of a code to detect and correct errors.The multiplication of this signal with a sinusoid carrier signal translates the whole thing up to fc. We can generalize this to say that the maximum number of error bits that can be detected is 31 . Hamming distance and weight are very important and useful concepts in coding. We assign the first ten of these. the minimum Hamming distance is 1. 0 to 9 by a digital sequence.). The Hamming distance between sequences 001 and 101 is = 1 The Hamming distance between sequences 0011001 and 1010100 is = 4. The Hamming distance is the number of disagreements between two binary sequences of the same size. Block Codes: Block codes operate on a block of bits. Each integer is now identified by its own unique sequence of bits .Einstein College of Engineering Chapter-3 Baseband Coding Techniques Baseband Signal .

k). The number of errors that we can correct is given by Creating block codes: The block codes are specified by (n. number of code symbols per block (n).Einstein College of Engineering t = dmin –1 Where dmin is Hamming distance of the codewords. All Hamming codes are able to detect three errors and correct one. *Hamming code. An R-S code is described by a generator polynomial g(x) and other usual important code parameters such as the number of message symbols per block (k). a simple linear block code *Hamming codes are most widely used linear block codes. * The number of information bits in the block is equal to 2n-n-1 and the number of overhead bits is equal to n. * A Hamming code is generally specified as (2n.Chaudhuri-Hocquenghem (BCH) codes and are very powerful linear non-binary block codes capable of correcting multiple random as well as burst errors. 2n-n-1). used in polynomials is an indeterminate which usually implies unit amount of delay. 32 . They have an important feature that the generator polynomial and the code symbols are derived from the same finite field. So we want to have a code set with as large a Hamming distance as possible since this directly effects our ability to detect errors. For a code with dmin = 3. A parity-check polynomial h (X) of order k also plays a role in designing the code.1. A large number of R-S codes are available with different code rates. The code takes k information bits and computes (n-k) parity bits from the code generator matrix. Reed-Solomon Codes: Reed Solomon (R-S) codes form an important sub-class of the family of Bose. Most block codes are systematic in that the information bits remain unchanged with parity bits attached either to the front or to the back of the information sequence. we can both detect 1 and 2 bit errors. The symbol x. * The size of the block is equal to 2n-1. maximum number of erroneous symbols (t) that can surely be corrected per block of received symbols and the designed minimum symbol Hamming distance (d). This enables to reduce the complexity and also the number of computations involved in their implementation.

t) R-S code has optimum error correcting capability. Closely related to K is the parameter m. t) R-S code is defined as below: Number of encoded symbols per block: n = 2m – 1 Number of message symbols per block: k Code rate: R= k/n Number of parity symbols per block: n – k = 2t Minimum symbol Hamming distance per block: d = 2t +1. *The constraint length parameter. k.Einstein College of Engineering For positive integers m and t. A simple convolutional encoder is shown below(fig 3. k. how many k-bit stages are available to feed the combinatorial logic that produces the output symbols. The output encoded bits are obtained by modulo-2 addition (EXCLUSIVE-OR operation) of the input information bits and the contents of the shift registers which are a few previous information bits. is expressed as a ratio of the number of bits into the convolutional encoder (k) to the number of channel symbols output by the convolutional encoder (n) in a given encoder cycle. *Convolutional codes are commonly described using two parameters: the code rate and the constraint length. *The encoded bits depend on the current k input bits and a few past input bits. 33 . t) R-S code is bounded by the corresponding finite field GF(2m). which can be thought of as the memory length of the encoder. denotes the "length" of the convolutional encoder. an (n. as n – k = 2t. Moreover.1). The code rate. k/n. k. Convolutional codes: *Convolutional codes are widely used as channel codes in practical communication systems for error correction. The information bits are fed in small groups of k-bits at a time to a shift register. i. It can be noted that the block length n of an (n.e. K. a primitive (n. * The main decoding strategy for convolutional codes is based on the widely used Viterbi algorithm.

n=2 and r=1/2 The operation of a convolutional encoder can be explained in several but equivalent ways such as. is depicted in the state diagram. b) tree diagramrepresentation. The transition of an encoder from one state to another. by a) state diagram representation. c) trellis diagram representation. 3. 3. 3. a) State Diagram Representation: A convolutional encoder may be defined as a finite state machine.2 shows the state diagram of the encoder in Fig. as caused by input bits.Fig. Contents of the rightmost (K-1) shift register stages define the states of the encoder.1 has four states. 34 . the encoder in Fig. A new input bit causes a transition from one state to another.Einstein College of Engineering Fig 3. So.1 A convolutional encoder with k=1.1.

3 shows the tree diagram for the encoder in Fig.2 State diagram representation for the encoder in Fig. The encoded bits are labeled on the branches of the tree. Representing convolutional codes compactly: code trellis and state diagram: State diagram 35 . Fig.1 b) Tree Diagram Representation: The tree diagram representation shows all possible information and encoded sequences for the convolutional encoder. 3. Given an nput sequence.35. the encoded sequence can be directly read from the tree. 6. 3.1.Einstein College of Engineering Fig 3.

1 1. 1 1. 0 1. encoded word v=(1 1. 1 0. below for u=(1 1 1 0 1).Einstein College of Engineering Inspecting state diagram: Structural properties of convolutional codes: • Each new block of k input bits causes a transition into new state • Hence there are 2k branches leaving each state • Assuming encoder zero initial state. 0 1. 1 1) is produced: • 36 . 1 0. encoded word for any input of k bits can thus be obtained. For instance.

encoder state diagram for (n.L)=(2.1.Einstein College of Engineering .2) code .note that the number of states is 2L+1 = 8 Distance for some convolutional codes: 37 .k.

3.3. corresponding input and output bits etc. as is present in the corresponding tree diagram.Einstein College of Engineering Fig.1. is very convenient for describing the behavior of the corresponding decoder. All state transitions at each time step are explicitly shown in the diagram to retain the time dimension.3 A tree diagram for the encoder in Fig. are labeled in the trellis diagram. which describes the operation of the encoder.1 c) Trellis Diagram Representation: The trellis diagram of a convolutional code is obtained from its state diagram. It is interesting to note that the trellis diagram. Fig. 38 .4 shows the trellis diagram for the encoder in Figure 3. 3. Usually. especially when the famous „Viterbi Algorithm (VA)‟ is followed. supporting descriptions on state transitions.

3.4) • Generator matrix G: first 4-by-4 identical matrix • Message information vector p • Transmission vector x • Received vector r and error vector e • Parity check matrix H 39 .Einstein College of Engineering Fig.1 Hamming Code Example: • H(7. Trellis diagram for the encoder in Fig.4.3.

syndrome vector z=zeros • If there is one error at location 2 • New syndrome vector z is 40 .Einstein College of Engineering Error Correction: • If there is no error.

and the absence of a pulse (i. Two possibilities for the pulse p(t) 41 . These codes are explained here: 1. The DC component in a line code is called the bias or the DC coefficient.. zero-bias. zero-DC. Another advantage is to get rid of DC shifts. a good recovery of the clock is guaranteed. MLT-3 and Duobinary encoding.Such codes are called DC balanced. Manchester. and if enough transitions exist. or DC equalized. The selection of a proper line code can help in so many ways: One possibility is to aid in clock recovery at the receiver. A clock signal is recovered by observing transitions in the received bit sequence. zero voltage) to represent a binary 0. It has the advantage of being compatible with TTL logic.Einstein College of Engineering CLASSIFICATION OF LINECODES Line coding: Line coding refers to the process of representing the bit stream (1‟s and 0‟s) in the form of voltage or current variations optimally tuned for the specific properties of the physical channel being used. Unipolar coding uses a positive rectangular pulse p(t) to represent binary 1. Unfortunately. Unipolar (Unipolar NRZ and Unipolar RZ): Unipolar is the simplest line coding scheme possible.Some common types of line encoding in common-use nowadays are unipolar. and the signal is said to be self-clocking.e. bipolar. polar. This is why most line codes try to eliminate the DC component before being transmitted on the channel. most longdistance communication channels cannot transport a DC component.

Polar (Polar NRZ and Polar RZ): In Polar NRZ line coding binary 1‟s are represented by a pulse p(t) and binary 0‟s are represented by the negative of this pulse -p(t) (e..Einstein College of Engineering exist3: Non-Return-to-Zero (NRZ) rectangular pulse and Return-to-Zero (RZ) rectangular pulse. which means it creates a significant DC-component at the receiver (see the impulse at zero frequency in the corresponding power spectral density (PSD) of this line code The disadvantage of unipolar RZ compared to unipolar NRZ is that each rectangular pulse in RZ is only half the length of NRZ pulse.. This means that unipolar RZ requires twice the bandwidth of the NRZ code.Using the assumption that in a regular bit 42 . The difference between Unipolar NRZ and Unipolar RZ codes is that the rectangular pulse in NRZ stays at a positive value (e. A drawback of unipolar (RZ and NRZ) is that its average value is not zero. while the pule in RZ drops from +5V to 0V in the middle of the bit time. Polar (NRZ and RZ) signals .g.g. +5V) for the full duration of the logic 1 bit. -5V).

In NRZI there are two possible pulses. is that it lacks clock information especially when a long sequence of 0‟s or 1‟s is transmitted. A transition from one pulse to the other happens if the bit being transmitted is a logic 1. and hence have better SNR at the receiver.Einstein College of Engineering stream a logic 0 is just as likely as a logic 1. which means that polar signals have more power than unipolar signals. however. and no transition happens if the bit being transmitted is a logic 0. Actually. p(t) and –p(t). The rms value of polar signals is bigger than unipolar signals. Inverted (NRZI): NRZI is a variant of Polar NRZ. polar NRZ signals have more power compared to polar RZ signals. Non-Return-to-Zero.polar signals (whether RZ or NRZ) have the advantage that the resulting Dccomponent is very close to zero. The drawback of polar NRZ. 43 .

44 . and on fiber-based Fast Ethernet at 100-Mbit/s . which means that accurate clock recovery from a data stream is possible.Einstein College of Engineering This is the code used on compact discs (CD). the DC component of the encoded signal is zero. Although transitions allow the signal to be self-clocking. USB ports. In addition. it carries significant overhead as there is a need for essentially twice the bandwidth of a simple NRZ or NRZI encoding POWER SPECTRA OF LINE CODES: • Unipolar most of signal power is centered around origin and there is waste of power due to DC component that is present. Manchester encoding is therefore considered to be self-clocking. Manchester encoding: In Manchester code each bit of data is signified by at least one transition.

Einstein College of Engineering • • • Polar format most of signal power is centered around origin and they are simple to implement. Bipolar format does not have DC component and does not demand more bandwidth. Manchester format does not have DC component but provides proper clocking. but power requirement is double than other formats. 45 .

Einstein College of Engineering CHAPTER –IV BASEBAND RECEPTION TECHNIQUES BASEBAND: RECEIVING FILTER: Correlative receiver 46 .

Einstein College of Engineering Observation Vector x For an AWGN channel and for the case when the transmitted signals are equally likely. the optimum receiver consists of two subsystems 1) .Receiver consists of a bank of M product-integrator or correlators Φ1(t) .Φ2(t) …….ΦM(t) orthonormal function The bank of correlator operate on the received signal x(t) to produce observation vector x Implemented in the form of maximum likelihood detector that operates on observation 47 .

s2… sM . the largest in the resulting set of numbers is selected and a corresponding decision on the transmitted message made.ΦM(t) is assumed to be zero outside the interval 0<t<T.Einstein College of Engineering vector x to produce an estimate of the transmitted symbol mi i = 1 to M. The optimum receiver is commonly referred as a correlation receiver MATCHED FILTER Science each of t he orthonormal basic functions are Φ1(t) .M. A filter whose impulse response is time-reversed and delayed version of the input signal is said to be matched to j (t ) . correspondingly . it must be causal. sk)} k= 1. For a matched filter operating in real time to be physically realizable. we can design a linear filter with impulse response hj(t). The N elements of the observation vector x are first multiplied by the corresponding N elements of each of the M signal vectors s1. The inner products are corrected for the fact that the transmitted signal energies may be unequal. with the received signal x(t) the fitter output is given by the convolution integral yj(t) = xj where xj is the j th correlator output produced by the received signal x(t).. 2 . the optimum receiver based on this is referred as the matched filter receiver.Φ2(t) ……. in a way that would minimize the average probability of symbol error. For causal system 48 . Finally. and the resulting products are successively summed in accumulator to form the corresponding set of Inner products {(x.

PROPERTY 3 The output Signal to Noise Ratio of a Matched filter depends only on the ratio of the signal energy to the power spectral density of the white noise at the filter input. 49 . proportional to the energy spectral density of the input signal. PROPERTY 2 The output signal of a Matched Filter is proportional to a shifted version of the autocorrelation function of the input signal to which the filter is matched. except for a time delay factor.Einstein College of Engineering MATCHED FILTER Φ(t) = input signal h(t) = impulse response W(t) =white noise The impulse response of the matched filter is time-reversed and delayed version of the input signal MATCHED FILTER PROPERTIES PROPERTY 1 The spectrum of the output signal of a matched filter with the matched signal as input is.

one of the M possible signals S1(t).Einstein College of Engineering PROPERTY 4 The Matched Filtering operation may be separated into two matching conditions. We may represents s i(t) by a point in a Euclidean space of dimensions N ≤ M. knowing Si is as good as knowing the transmitted signal Si(t) itself. the output of the correlator define a new vector x called observation vector. The collection of M message points in the N Euclidean space is called a signal constellation. Such a point is referred as transmitted signal point or message point. . Based on the observation vector x. MAXIMUM LIKELIHOOD DETECTOR: Detection of known signals in noise Assume that in each time slot of duration T seconds. .The vectors x and w are sampled values of the random vectors X and W respectively. is applied to a bank of correlators. we represent the received signal s(t)by a point in the same Euclidean space. where w(t) is sample function of the white Gaussian noise process W(t). . and vice versa. the noise vector w represents that portion of the noise w(t) which will interfere with the detected process. The receiver has to observe the signal x(t) and make a best estimate of the transmitted signal si(t) or equivalently symbol mi The transmitted signal si(t). and the spectral amplitude matching that gives this peak value its optimum signal to noise density ratio. the resulting correlator outputs define the signal vector Si. S2(t). . with zero mean and PSD N0/2. . i= 1to M . Then for an AWGN channel a possible realization of sample function x(t) of the received random process X(t) . with a common input and supplied with an appropriate set of N orthonormal basic functions. When the received signal x(t) is applied to the bank o N correlators . namely spectral phase matching that produces the desired output peak at time T. . SM(t) is transmitted with equal probability of 1/M. we refer this point as received signal point. . this vector x differs from the signal vector si by a random noise vector w. The relation between them is as shown in the fig 50 .

Such a channel disperses or spreads a pulse carrying digitized samples passing through it. Optimum transmitter & receiver Probability of error depends on signal to noise ratio As the SNR increases the probability of error decreases An optimum transmitter and receiver is one which maximize the SNR and minimize the probability of error. The maximum likelihood detector provides solution to this problem. spreading will occur and cause signal pulses to overlap. if we transmit digital data which demands more bandwidth which exceeds channel bandwidth. spreading of pulse is very less. When the channel bandwidth is greater than bandwidth of pulse. in away that would minimize the average probability of symbol error in the decision. In short it is called ISI. i.e. communication channel is always band limited. But when channel bandwidth is close to signal bandwidth.Einstein College of Engineering Fig: Illustrating the effect of noise perturbation on location of the received signal point In the detection problem . Inter symbol Interference Generally. ISI causes degradations of signal if left 51 . Similar to interference caused by other sources. the observation vector x is given. digital data is represented by electrical pulse. This overlapping is called Inter Symbol Interference. we have to perform a mapping from x to an estimate of the transmitted symbol.

EYE PATTERN The quality of digital transmission systems are evaluated using the bit error rate. • Eye patterns can be observed using an oscilloscope. Degradation of quality occurs in each process modulation. The received wave is applied to the vertical deflection plates of an oscilloscope and the sawtooth wave at a rate equal to transmitted symbol rate is applied to the horizontal deflection plates. and detection. resulting display is eye pattern as it resembles human eye. This waveform is generally termed as Line codes. This problem of ISI exists strongly in Telephone channels like coaxial cables and optical fibers. First let us have look at different formats of transmitting digital data. Therefore. In this chapter main objective is to study the effect of ISI.Einstein College of Engineering uncontrolled. tail of smeared pulse enter into adjacent symbol intervals making it difficult to decide actual transmitted pulse. 1 0 11 Tb Transmitted Waveform Pulse Dispersion The effect of sequence of pulses transmitted through channel is shown in fig. The Spreading of pulse is greater than symbol duration. The eye pattern is experimental method that contains all the information concerning the degradation of quality. careful analysis of the eye pattern is important in analyzing the degradation mechanism. when digital data is transmitted through band limited channel and solution to overcome the degradation of waveform by properly shaping pulse.e. transmission. • The interior region of eye pattern is called eye opening 52 . as a result adjacent pulses interfere. i. pulses get completely smeared.In base band transmission best way is to map digits or symbols into pulse waveform.

traces from the upper portion of the eye pattern cross traces from lower portion with the result that the eye is completely closed. Any non linear transmission distortion would reveal itself in an asymmetric or squinted eye. When the effected of ISI is excessive. Example of eye pattern: Binary-PAM Perfect channel (no noise and no ISI) 53 . The sensitivity of the system to timing error is determined by the rate of closure of the eye as the sampling time is varied. • • • The width of the eye opening defines the time interval over which the received wave can be sampled without error from ISI The optimum sampling time corresponds to the maximum eye opening The height of the eye opening at a specified sampling time is a measure of the margin over channel noise.Einstein College of Engineering We get superposition of successive symbol intervals to produce eye pattern as shown below.

Post channel equalization Achieved prior to data transmission by training the filter with the guidance of a training sequence transmitted through the channel so as to adjust the filter parameters to optimum values. Adaptive equalizer can adjust its coefficients continuously during the transmission of data. set of adjustable multipliers connected to the delay line taps and a summer for adding multiplier outputs. Pre channel equalization requires feed back channel causes burden on transmission. Adaptive equalization – It consists of tapped delay line filter with set of delay elements. 54 .Einstein College of Engineering Example of eye pattern: Binary-PAM with noise no ISI EQUALISING FILTER Adaptive equalization • An equalizer is a filter that compensates for the dispersion effects of a channel.

In the adaptive equaliser the Ci's are variable and are adjusted by an algorithm Two modes of operation 1.Einstein College of Engineering The output of the Adaptive equalizer is given by Y(nt)=∑Ci x(nT-iT) Ci is weight of the ith tap Total number of taps are M .Tap spacing is equal to symbol duration T of transmitted signal In a conventional FIR filter the tap weights are constant and particular designed response is obtained. Decision directed mode Mechanism of adaptation 55 . Training mode 2 .

The difference between resulting response y(nT) and desired response d(nT)is error signal which is used to estimate the direction in which the coefficients of filter are to be optimized using algorithms . This training sequence has maximal length PN Sequence. 56 .Einstein College of Engineering Training mode A known sequence d(nT) is transmitted and synchronized version of it is generated in the receiver applied to adaptive equalizer. because it has large average power and large SNR. resulting response sequence (Impulse) is observed by measuring the filter outputs at the sampling instants.

Different Shift keying methods that are used in digital modulation techniques are Amplitude shift keying [ASK] Frequency shift keying [FSK] Phase shift keying [PSK] Fig shows different modulations 57 .Einstein College of Engineering CHAPTER-V BANDPASS SIGNAL TRANSMISSION AND RECEPTION Memoryless modulation techniques Modulation is defined as the process by which some characteristics of a carrier is varied in accordance with a modulating wave. the modulating wave consists of binary data or an M-ary encoded version of it and the carrier is sinusoidal wave. In digital communications.

Einstein College of Engineering Hierarchy of digital modulation technique Digital Modulation Technique Coherent Binary (m) = 2 * ASK * FSK * PSK M .ary Hybrid Non .Coherent Binary (m) = 2 * ASK * FSK * DPSK M .ary M-ary ASK M-ary APK M-ary FSK M-ary QAM M-ary PSK (QPSK) M-ary ASK M-ary FSK M-ary DPSK 58 .

Einstein College of Engineering Implementation of binary ASK: 59 .

Which gives an output Eb volts for symbol 1 and 0 volt for symbol 0. The desired BASK wave is obtained at the modulator output. BANDWIDTH: • d ≥ 0-related to the condition of the line B = (1+d) x S = (1+d) x N x 1/r The BASK system has one dimensional signal space with two messages (N=1. M=2) Region E2 Region E1 Message Point 2 Eb ᶲ(t) Message Point 1 Signal Space representation of BASK signal 0 Eb 2 In transmitter the binary data sequence is given to an on-off encoder. The resulting binary wave [in unipolar form] and sinusoidal carrier are applied to a product modulator.Einstein College of Engineering BINARY AMPLITUDE SHIFT KEYING. 60 .

If x > λ the receiver decides in favour of symbol 1. The correlator output x is compared with threshold λ. Generation and Detection:- 61 .Einstein College of Engineering In demodulator. If x < λ the receiver decides in favour of symbol 0. the received noisy BASK signal x(t) is apply to correlator with coherent reference signal.

These two frequencies are combined using an adder circuit and then transmitted. The correlator outputs are then subtracted one from the other and resulting a random vector „l‟ (l=x1 . If l > 0.Einstein College of Engineering FSK transmitter fig b FSK receiver A binary FSK Transmitter is as shown . the receiver decides in favour of symbol 0. The output of encoder is Eb volts for symbol 1 and 0 volts for symbol „0‟. The detector consists of two correlators. l < 0. for symbol „0‟. When we have symbol 1 the upper channel is switched on with oscillator frequency f1.the incoming binary data sequence is applied to on-off level encoder. The Coherent reference signal ᶲ1(t) & ᶲ2(t) are supplied to upper and lower correlators respectively. because of inverter the lower channel is switched on with oscillator frequency f2 . The transmitted signal is nothing but FSK Bandwidth: 62 .x2). The output „l‟ is compared with threshold of zero volts. required BFSK signal. The incoming noisy BFSK signal x(t) is common to both correlator. the receiver decides in favour of symbol 1.

used up to 1200bps – Used for high-frequency (3 to 30 MHz) radio transmission – used at higher frequencies on LANs that use coaxial cable. m=2] they are represented.Einstein College of Engineering • Limiting factor: Physical capabilities of the carrier • Not susceptible to noise as much as ASK • Applications – On voice-grade lines. Fig. Therefore Binary FSK system has 2 dimensional signal space with two messages S1(t) and S2(t). Signal Space diagram of Coherent binary FSK system. 63 . [N=2 .

Einstein College of Engineering PHASE SHIFT KEYING(PSK): Non Return to Zero Level Encoder Binary Data Sequence Product Modulator Binary PSK Signal 1 (t ) 2 Cos 2f c t Tb Fig(a) Block diagram of BPSK transmitter x(t) Tb dt 0 x1 Decision Device Choose 1 if x1>0 Choose 0 if x1<0 1 (t ) Correlator Threshold λ = 0 Fig (b) Coherent binary PSK receiver In a Coherent binary PSK system the pair of signals S 1(t) and S2(t) are used to represent binary symbol „1‟ and „0‟ respectively. the receiver decides in favour of symbol 0. which is also supplied with a locally generated coherent reference signal The correlator output x1 is compared with a threshold of zero volt. To generate a binary PSK signal we have to represent the input binary sequence in polar form with symbol „1‟ and „0‟ represented by constant amplitude levels To detect the original binary sequence of 1‟s and 0‟s we apply the noisy PSK signal x(t) to a Correlator. the receiver decides in favour of symbol 1. 64 . If x1 > 0. If x1 < 0.

Fig. Signal Space Representation of BPSK QUADRATURE PHASE – SHIFT KEYING(QPSK) In a sense. where Tb: duration of 1 bit. A group of two bits is often called a „dibit‟. Let. E: Energy per Symbol and T: SymbolDuration = 2. So.Eb 0 Message Point 2 S2(t) Decision Boundary Message Point 1 S1(t) Region R1 Eb Fig. QPSK is an expanded version from binary PSK where in a symbol consists of two bits and two orthonormal basis functions are used. Each symbol carries same energy. (a) QPSK Transmitter 65 .Einstein College of Engineering The signal space representation is as shown in fig (N=1 & M=2) Region R2 .* Tb. four dibits are possible.

66 .Einstein College of Engineering Fig. (b) QPSK Receiver Fig. QPSK Waveform In QPSK system the information carried by the transmitted signal is contained in the phase.

The in-phase channel output x1 and the Q-channel output x2 may be viewed as the individual outputs of the two coherent binary PSK systems. Thus the two binary PSK systems may be characterized as follows.Einstein College of Engineering QPSK Receiver:The QPSK receiver consists of a pair of correlators with a common input and supplied with a locally generated pair of coherent reference signals ᶲ1(t) & ᶲ2(t)as shown in fig(b). The signal energy per bit E 2 67 .The correlator outputs x1 and x2 produced in response to the received signal x(t) are each compared with a threshold value of zero. Probability of error:A QPSK system is in fact equivalent to two coherent binary PSK systems working in parallel and using carriers that are in-phase and quadrature. The in-phase channel output : If x1 > 0 a decision is made in favour of symbol 1 x1 < 0 a decision is made in favour of symbol 0 Similarly quadrature channel output: If x2 >0 a decision is made in favour of symbol 1 and x2 <0 a decision is made in favour of symbol 0 Finally these two binary sequences at the in phase and quadrature channel outputs are combined in a multiplexer (Parallel to Serial) to reproduce the original binary sequence.

Einstein College of Engineering - The noise spectral density is N 2 0 The bit errors in the I-channel and Q-channel of the QPSK system are statistically independent . each baud equals 4 bits of information (2 ^ 4 = 16). More phases than amplitudes. 32.M=4. 64. 12 different phases are combined with two different amplitudes. The I-channel makes a decision on one of the two bits constituting a symbol (di bit) of the QPSK signal and the Q-channel takes care of the other bit. there are a total of 16 combinations. Combine ASK and PSK such that each signal corresponds to multiple bits. With 16 signal combinations. 128. 16. Minimum bandwidth requirement same as ASK or PSK 68 . Since only 4 phase angles have 2 different amplitudes. QAM(Quadrature Amplitude Modulation): • QAM is a combination of ASK and PSK Two different signals sent simultaneously on the same carrier frequency ie. 256 As an example of QAM.

Reconstruction Filter are the components of a Digital Communication System.Write the two methods of channel coding? 69 . Modulator. 3. *Security of information. *Flexible implementation of digital hardware system *Coding of digital signal to yield extremely low error rate and high fidelity. What are the advantages of Digital Communication? *Ruggedness to channel noise and other interferences. What are Non Coherent Receivers? Receivers in which the carrier used in the receiver is of different frequency and phase of the transmitted one is called coherent receivers. Encoder. 4. Channel. What are Coherent Receivers? Receivers in which the carrier used in the receiver is of same frequency and phaseof the transmitted one is called coherent receivers.Draw the model of analog communication system Information Source and Input Transducer Destination and Output Transducer I/P Signal TRANSMITTER CHANNEL O/P Signal RECEIVER 6. 5.Einstein College of Engineering Short Questions and Answers: 1. Decoder. Quantizer. 2. What are the components of a Digital Communication System? Sampler. Demodulator.

The in-phase component gI(t) and the quadrature component gQ(t) may be obtained by multiplying the bandpass signal g(t) by cos(2Π fct) and sin(2Πfct) respectively and then suppressing the sumfrequency components by means of appropriate low pass filter. 8. Mathematically it is expressed as. Quadrature sampling is used for uniform sampling of band pass signals Consider g(t) = gI(t) cos(2Πfct) – gQ(t) sin(2Πfct).we find that gI(t)&gQ(t) are both low-pass signals limited to - 70 .Define Bandwidth Bandwidth is simply a measure of frequency range. Define quadrature sampling. State sampling theorem. The range of frequencies contained in a composite signal is its bandwidth.Define Power signal A power signal. 9. Under the assumption that fc>W. it may be completely recovered from its co=ordinates at a sequence of points spaced 1/2W seconds apart 10. The bandwidth is normally a difference between two numbers. If a finite energy signal g(t) contains no frequencies higher than W hertz. If a finite –energy signal g(t) contains no frequencies higher than W hertz.Einstein College of Engineering i)Channel coding ii)Block Coding 7. it is completely determined by specifying its co-ordinates at a sequence of points spaced 1/2W seconds apart. will have a finite power but may have finite or infinite energy.

This type of sampling is called quadrature sampling.Einstein College of Engineering W<f<W.This type of quantization is called as robust quantization. what do you mean by companding? Define compander. 14. The combination of a compressor and expander is called a compander.frequency in the spectrum of the original signal g(t) seemingly taking on the identity of a lower frequency in the spectrum of the sampled signal g(t) is called aliasing or foldover.uniform quantization is otherwise called as robust quantization 16. 11. State NRZ unipolar format 71 . Accordingly each component may be sampled at the rate of 2W samples per second. The conversion of analog sample of the signal into digital form is called quantizing process. 12. ii)Mid riser type quantizer. This is called as companding. Non uniform quantizer is characterized by a step size that increases as the separation from the origin of the transfer characteristics is increased. Non. The signal is compressed at the transmitter and expanded at the receiver.uniform quantization? Step size is not uniform. What is aliasing? The phenomenon of a high. 15. Define quantizing process. What you mean by non. 17. What is meant by prediction error? The difference between the actual sample of the process at the time of interest and the predictor output is called a prediction error. 18. 13. What is the disadvantage of uniform quantization over the non-uniform quantization? SNR decreases with decrease in input power level at the uniform quantizer but non-uniform quantization maintain a constant SNR for wide range of input power levels. Name the types of uniform quantizer? i)Mid tread type quantizer.

M.Einstein College of Engineering In this format binary 0 is represent by no pulse and binary 1 is represented by the positive pulse. State manchester format Binary 0 The first half bit duration negative pulse and the second half Bit duration positive pulse. Why do we go for Gram-Schmidt Orthogonalization procedure ? Consider a message signal m. 23. The justification for this separation lies in the Gram-Schmidt orthogonalization procedure which permits the representation of any set of M energy signals.2. 21. State NRZ bipolar format Binary 0 is reporesented by no pulse and binary one is represented by the alternative positive and negative pulse.. It is defined by r (i. {si(t)}. The task of transforming an incoming message mi=1. 19. 20. into a modulated wave si(t) may be divided into separate discrete time & continuous time operations. State NRZ polar format Binary 1 is represented by a positive pulse and binary 0 is represented by a Negative pulse. as linear combinations of N orthonormal basis functions.…. 24. What is code rate ? Code rate is the ratio of message bits (k) and the encoder output bits (n).e) r= k/N 72 . 22. What is linear code ? A code is linear if the sum of any two code vectors produces another code vector. Binary 1 first half bit duration positive pulse and the second half Bit duration negative pulse.

Block length = n =2 m -1 symbols Message size : k symbols Parity check size : n-k= 2t symbols Minimum distance . 29. Define code word & block length. of bits „n‟ after coding is called block length. of Check bits q Block length n = 2 –1 q No of message bits K = n-q Minimum distance dmin =3 26. The encoded block of „n‟ bits is called code word. Define a random binary sequence. Define code efficiency It is the ratio of message bits in a block to the transmitted bits for that block by the encoder i. These are non binary BCH codes.e Message bits in a block code efficiency= -------------------------------Transmitted bits for the block 25.Einstein College of Engineering 25. Why RS codes are called maximum distance separable codes ? Linear block code for which the minimum distance equals n – k + 1 is called maximum distance separable codes. For RS code minimum distance equals n – k + 1 so it is called as maximum distance separable codes. Give the parameters of RS codes : Reed Solomon codes. The no. A random binary sequence is a sequence in which the presence of a binary 73 . What are the conditions to satisfy the hamming code No. dmin =2t +1 symbols. 27. 28.

What is Signal constellation diagram? Suppose that in each time slot of duration T seconds. The correlator outputs define the signal 74 .What is meant by a matched filter? Matched filter is used for detection of signal in base band and pass band transmission. is applied to a bank of correlators. one s2(t). i = 1. . The noise can be internal to the system or external to the system. A filter whose impulse response is a time reversed & delayed version of some signal . 33. sM(t) is transmitted with equal probability. the optimum receiver based on the detector is referred to as the matched filter receiver.. For example let the two code vectors be X=(101) and Y= (110) These two code vectors differ in second and third bits. . 31. M. .Einstein College of Engineering symbol 1 or 0 is equally probable. 1/M For geometric representation. This process of suppressing channel induced distortion is called channel equalization. Hence at the receiver the distortion must be compensated in order to reconstruct the transmitted symbols.Need for equalization in digital communication The two principal causes of distortion in a digital communication channels are Inter Symbol Interference (ISI) and the additive noise. The ISI can be characterized by a Finite Impulse Response (FIR) filter. What is hamming distance? The hamming distance between two code vectors is equal to the number of Elements in which they differ.then it is said to be matched to correspondingly. The equalization in digital communication scenario is illustrated 32. Therefore the hamming distance between x and Y is two. 30. the signal si (t). . 2..

Hence it is also called synchronous detection. In non coherent detection the local carrier generated at the receiver not be phase locked with the carrier at the transmitter.Distinguish between Coherent and Noncoherent receiver.Einstein College of Engineering vector si. 75 . 35.How do we get eye pattern?What you infer from this? The eye pattern is obtained by applying the received wave to the vertical deflection plates of an oscilloscope and to apply a saw tooth wave at the transmitted symbol rate to the horizontal deflection plate. at a specified sampling time. In coherent detection the local carrier generated at the receiver is phase locked with the carrier at the transmitter. The set of message points corresponding to the set of transmitted signals {si(t))} i=1.. o The height of the eye opening. but it has higher probability of error.M is called a signal constellation. o The width of the eye defines the time interval over which the received waveform can be sampled without error from intersymbol interference. o The sensitivity of the system to timing errors is determined by the rate of closure of the eye as the sampling time is varied. 34. It is simple. defines the noise margin of the system.

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