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• A continuous representation of a continuous event. • An analog connection is one which continuously varies in amplitude and frequency. • The amplitude of the signal is representation of its loudness while the frequency represents its tone or pitch. • A digital signal is defined at discrete times only represented by fixed states “digits”. • Most usually it is represented by a binary signals with ‘1s’ or ‘0s’ represented by a positive voltage or zero voltage, or by two different carrier frequencies or phases. In optical fiber system ‘1s’ or ‘0s’ may be represented by ‘light on’ or ‘light off’. There are three notable advantages to digital transmission that make it extremely attractive to the telecommunication system engineer when compared to its analog Counterpart. Noise does not accumulate on a digital system as it does on an analog system. Noise accumulation stops at each regenerative repeater where the digital signal is fully regenerated. Noise accumulation was the primary concern in analog network design. The digital format lends itself ideally to solid-state technology and, in particular, to integrated circuits. It is theoretically compatible with digital data, telephone signaling, and computers. • Quality of signals with analog systems varies as overall distance of the circuit varies since used in analog systems along with the original signal. • With digital systems only one of certain number of possible states exist (1 or 0) which if identified can be used to recreate the original signal from an degraded input. is the key benefit of digital transmission systems. • By adopting digital systems the noise performance of a long distance telephone channel is as good as that of a short distance channel. • Higher Data Rate Possible: – With digital computers around, digital lines can transfer data at much higher rates than analog lines. • Improvements using Digital Radio Systems: – Since digital systems free from noise, they are best suited for radio systems. • Digital Exchanges and ISDN: – Digital exchanges have tremendous advantages over analog exchanges, these can be even enhanced by ISDN customers (no conversion required) • Lesser Maintenance • Extra Security: – With digital systems, it is possible to use different codes by ‘scrambling’ the signals at transmitter and unscrambling at the receiver end. • Space Saving and Economical: – Digital systems are physically small and relatively cheap thanks to advancements in LSI circuits • Higher Capacity – Digital systems can use the available bandwidth of the channel much more efficiently as compared to analog systems. In other words, better multiplexing schemes are available and being developed. • The aim of any transmission system is to produce at the output, an exact replica of any signal which is applied to the input. • In AM or FM, a system carrier is continuously varied by the signal. • It is not necessary to continuously send the information and only “samples” at certain levels are sufficient to represent it fully. – Example MOVIE. • Initially invented by A.H. Reeves in 1937 • Pulse Code Modulation is the representation of a signal by a series of digital pulses firstly by sampling the signal, quantizing it and then encoding it. • The PCM signal itself is a succession of discrete, numerically encoded binary values derived from digitizing the analog signal. PCM Steps • Nyquist Sampling Theorem: If a signal is sampled at a rate that is at least twice the highest frequency that it contains, the original signal can be completely reconstructed. • Since the bandwidth of the telephone lines is 300 to 3400 Hz, a sampling rate of 8 kHz is used which is easily above twice the highest frequency component within this range. • 8000 samples per second, or 8 kHz, sampling period 125 s • Within one sampling period, samples of several telephone channels can be sequentially accommodated. This process is called TDM. • Alias distortion occurs if Shannon’s criterion is not satisfied Pulse Amplitude Modulation • Sampling is the process of determining the instantaneous voltage at given intervals in time. PAM is the technique used to produce a pulse when the signal is sampled. • The pulse's amplitude is equal to the level at the time in which the analog signal was sampled. The amplitude of the pulses in a PAM signal contains the intelligence or modulating voltage. • The higher the sampling rate, the closer the recovered signal approaches the original signal. • Ideally, an infinite sampling rate would be desirable in terms of reproducing the original signal. • This is not practical, however, due to the bandwidth limitation on the large amounts of data that would need to be transmitted. Quantization • Instead of transmitting the exact amplitude of the sampled signal, only certain discrete value closest to the true one is transmitted. • At the receiving end the signal value will have a value slightly different from any of the specified discrete steps due to noise and distortions encountered in the transmission channel. • If the disturbance is negligible, it will be possible to tell accurately which discrete value was transmitted and the original signal can be approximately reconstructed. Quantization • Quantizing is a process by which analog samples (from a pulse amplitude modulated (PAM) signal) are classified into a number of adjacent quantizing intervals. • Each interval is represented by a single value called the Quantized Value. • This process introduces an error in the magnitude of the samples resulting in quantizing noise • However, once the information Is in quantized form, it can be sent over reasonable distance without further loss in quality through regeneration of the binary levels involved to counter distortion. Quantization the permitted range of values of an analog signal divided into quantizing intervals. Linear Quantization . : the difference between the input signal and the quantized output signal Linear Quantization.. Example Linear Quantization.. Example • Consider sample 2, the actual amplitude of the signal is +1.7V. • This is assigned level 2 (same for any voltage between 1 & 2), which is transmitted as line code 101. • At the receiving end 101 is converted to a pulse of +1.5V (the middle value of the decision level at the encoder) • This produces an error of 0.2V between original input and output signals. Linear Quantization • Errors occur on every sample except where the sample size exactly coincides the mid-point of the decision level. • If smaller steps are taken the quantization error will be less. However, increasing the steps will complicate the coding operation and increase bandwidth requirements. • Quantizing noise depends on step size and not on signal amplitude. Non-Linear Quantization • With linear quantization, the signal to noise ratio is large for high levels but small for low level signals. • Therefore, non-linear quantization is used. Non-Linear Quantization • The quantizing intervals are not of equal size. • Small quantizing intervals are allocated to small signal values (samples) and large quantization intervals to large samples so that the signal-toquantization distortion ratio is nearly independent of the signal level. • S/N ratios for weak signals are much better but is slightly less for the stronger signals. Non-Linear Quantization a process in which compression is followed by expansion. . A-Law • 13 piece-wise linear segments • A=87.6, for x>0 • Y 1 ln Ax 1 ln A Y Ax 1 ln A for x 1 1 A for 0 x where x = normalized input level, Y = normalized quantized steps, ln = natural logarithm 1 A (most significant bit) tells the distant-end receiver if that sample is a positive or a negative voltage identify the segment. , shown in the figure as XXXX, identify where in the segment that voltage line is located The 13-segment approximation of the A-law curve used with E1 PCM equipment µ-Law • µ-law used in North America and Japan: Y= sgn(x) ln(1+µx) ---------------ln(1+µ) where µ=255. ENCODING • PCM signal to be transmitted is obtained by encoding the quantizing intervals. • Allocation of 8-bit word is done to each individual sample. • An 8-digit binary code is used for 128 positive and 128 negative quantizing intervals. • First bit used for all PCM words for all positive intervals is ‘1’ and for negative intervals is ‘0’ MULTIPLEXING • The 8-bit PCM words of a number of telephone signals can be transmitted consecutively in repeated cycles. • A PCM word of one telephone signal is followed by PCM words of of all other telephone signals arranged in consecutive order. • This results in PCM TIME DIVISION MULTIPLEX signal MULTIPLEXING • Multiplexing function is carried out fully electronically. • A switch moves from one input to other. • The PCM-TDM signal is then available at the output of the switch. • The time interval within which a PCM word is transmitted is known as Time Slot. • A bit train containing one PCM word each from all inputs is known as Pulse Frame. Receiver Side • Demultiplexing PCM-TDM signal i.e. the PCM words of the telephone signals are distributed to the individual lines. • Decoding PCM words in the PCM signal i.e. a signal amplitude is allocated to each word, which is equal to the midpoint value of the particular quantizing interval. Result is a PAM signal. • Reproducing the original analog telephone signal from the PAM signal through a LPF. Line Code • A code chosen for use within a communication systems for transmission purposes. • A line code may differ from the code generated at a user terminal, and thus may require translation Line Codes • Objectives: – Better spectrum (no DC component) – Noise immunity – Error detection – Clocking capability • No added complexity Line Codes for PCM • Unipolar NRZ -- Stays positive and does not return to level 0 during binary 1 cell. • Bipolar NRZ -- 2 non zero voltages i.e. positive for 1 and negative for 0 and does not return to 0. • Unipolar RZ -- there is always a return to level 0 between individual bits during binary 1 cell. • Bipolar RZ -- 2 non zero voltages i.e. positive for 1 and negative for 0 and returns to level 0 as well. Line Codes for PCM • CMI -- 1s represented by alternate + and states and 0s always represented by a - state during first half and + in second half of bit interval. • AMI -- 1s represented by alternate + and states and 0s always represented by zero voltage. Line Codes for PCM • HDB3 -- used to eliminate series of more than 3 0s in the AMI. – The last zero of 4 consecutive zeroes is replaced by a violation (V) pulse that violates the AMI rule. – The first zero may be replaced by a 1 to prevent two V’s to have the same polarity. 0000 ==> X00V, X is so chosen the V’s polarities alternate. Line Codes 1 0 1 1 0 0 0 0 1 1 AMI 1 1 0 0 0 0 0 0 0 0 0 0 0 V B 0 V B 1 1 0 0 0 0 B8ZS - - +0 + 0 HDB3 0 0 V B 0 0 V B 0 0 V 0 1 0 Regenerative Repeater • The advantage of PCM lies chiefly in the fact that it is a digital process. • it is much easier for a receiver to distinguish between a 1 and a 0 than to reproduce faithfully a continuous wave signal. • Transmission media carrying PCM signals employ regenerative Repeaters that are spaced sufficiently close to each other (approximately 2kms) to prevent any ambiguity in the recognition of the binary PCM pulses Regenerative Repeater • The regenerative repeater conditions the received (attenuated and distorted) pulses through preamplifiers and equalizer circuits. • The signal is then compared against a voltage threshold • Above the threshold is a logic 1, and below the threshold is a logic 0. The resulting signal is said to be threshold detected. Regenerative Repeater • Timing circuits within the regenerative repeater are synchronized to the bit rate of the incoming signal. • The threshold detected signal is sampled at the optimum time to determine the logic level of the signal. • The resulting code is used to regenerate and retransmit the new equivalent signal Regenerative Repeater Formats for 30-channel PCM systems (E1) • A time slot: 8 bits • A frame lasts 125 s and is divided into 32 slots, numbered slot 0 to slot 31, transmission rate 2.048 Mbps • Time slot 0: frame alignment and service bits • Time slot 16 for multiframe alignment and signaling, the remaining 30 slots for data transmission (voice channel) • A multiframe consists of 16 frames (2ms) numbered frame 0 to 15 Formats for 30-channel PCM systems (E1) • TS 0 for even frames: Y0011011 for odd frames: Y1ZXXXX TS 16 for frame 0: 0000XZXX (0000 multiframe alignment signal, Y:international use, Z: frame alignment loss indicator, X: not used) • TS 16 for frames 1 to 15: signaling for 30 channels PCM 30 Pulse Frame Formats for 24 channel PCM systems (T1) • Used in North America and Japan (DS1) • A frame lasts 125 s, 24 time slots each having 8 bits • The 8th bit in every six frames is used for signaling. • 1 bit at the start of every frame included for frame and multiframe alignment purpose Formats for 24 channel PCM systems (T1) • A multiframe consisting of 12 frames, frame alignment word 101010 on odd frames, multiframe alignment word 001110 on even frames • Transmission rate (1+24* 8)/125 = 1.544 Mbps PCM24 Pulse Frame T1 Format for CCS • Pulse Frames are not combined to for multifranmes. • The first bits in every even pulse frames are used for Signaling (S-bits). SUMMARY Higher-Order Digital Multiplexing (CCITT) • T1 and E1 are the primary order of digital multiplexing • Higher orders can be formed • Example: second order multiplexing (2 to 8 Mbps) -- Four low rate bit streams E1 (tributary) are multiplexed in a bit-by-bit manner to E2 PLESIOCHRONOUS HIERARCHY 5760 Ch 397.2 Mbps 2.048 Mbps Intentionally Left Blank PULSE-CODE MODULATION SYSTEM OPERATION Simplified functional block diagram of a PCM codec or channel bank. Low-pass filter The voice channel to be transmitted is passed through a 3.4-kHz low-pass filter. Sampling circuit The output of the filter is fed to a sampling circuit. The sample of each channel of a set of n channels is released in turn to the pulse amplitude modulation (PAM) highway. Channel gating The release of samples is under control of a channel gating pulse derived from the transmit clock. To accommodate the 32channels, the gate is open 125/32 μsec, or 3.906 μsec. Coder The input to the coder is the PAM highway. The coder accepts a sample of each channel in sequence and then generates the appropriate 8-bit signal character corresponding to each sample presented. Digit combiner The coder output is the basic PCM signal that is fed to the digit combiner where framing-alignment signals are inserted in the appropriate time slots, as well as the necessary supervisory signaling digits corresponding to each channel Digit separator On the receiver side, it delivers the PCM signal to four locations to carry out the following processing functions: (1) Timing recovery (2) Decoding (3) Frame alignment (4) Signaling (supervisory) Timing recovery Timing recovery keeps the receive clock in synchronism with the far-end transmit clock. receive clock The receive clock provides the necessary gating pulses for the receive side of the PCM codec. Frame-alignment circuit The frame-alignment circuit senses the presence of the frame-alignment signal at the correct time interval, thus providing the receive terminal with frame alignment Decoder The decoder, under control of the receive clock, decodes the code character signals corresponding to each channel. The output of the decoder is the reconstituted pulses making up a PAM highway. Signaling processor Responsible for processing the control information associated with the corresponding voice channels. Intentionally Left Blank Transmission Limitations Transmission medium for PCM could be wire pair, coaxial cable, fiberoptic cable, and wideband radio media. Each medium has transmission limitations brought about by impairments. In one way or another each limitation is a function of and As loss increases signal-to-noise ratio suffers, directly impacting bit error performance Dispersion is another impairment that limits circuit length over a particular medium, especially as transmission rate increases •Displacement of the ideal sampling instant. This leads to a degradation in system error performance •Slips in timing recovery circuits manifesting itself in degraded error performance. •Distortion of the resulting analog signal after decoding at the receive end of the circuit. The sources of timing jitter may be classified as or Systematic jitter sources lead to jitter which degrades the bit stream in the same way at each repeater in the chain. Systematic sources include: •Intersymbol interference •Clock threshold effects •Finite pulse width Nonsystematic jitter sources causes timing degradations which are random from repeater to repeater. Nonsystematic jitter sources such as: • Mistuning • Crosstalk Thermal and impulse noise are not serious contributors to timing jitter In a long repeater chain, the total accumulated jitter is dominated by components produced by systematic sources. Wire-pair systems have repeaters every Coaxial cable has repeaters approximately Fiber-optic systems, depending on design and bit rate, have repeaters every Microwave radio, may have repeaters every Satellite links have the least repeaters, circuit in a long There are three cable characteristics that create this distortion: •Loss •Amplitude distortion (amplitude–frequency response) •Delay distortion Because of the nature of a digital system, impairments like thermal noise need only be considered on a per-repeatersection basis, because noise does not accumulate due to the regenerative process carried out at repeaters and nodes. Crosstalk is a major impairment in PCM wire-pair systems, particularly when “go” and “return” channels are carried in the same cable sheath. Echo is caused by impedance discontinuities in the transmission line, including repeaters and terminations (MDFs, codecs, switch ports) THANKS