a A Pi Digital Transmission CHAPTER OUTLINE | Introduction Jed Puke Moduaion 103 PCM 04 PCM Sampling {5 Signal:to-Quantization Noise Ratio 106 Linear versus Nonlinear PCM Codes {07 Idle Channel Noise 108 Coding Methods OBJECTIVES Define digital transmission @ List and describe the advant: tages of digital transmission H Briefly describe pulse width modulation, pulse Position modulation, and pulse amplitude modu: lation Define and describe pulse code modulation Explain flat-top and natural sampling Describe the Nyquist samplin; Describe folded binary codes in dynamic range Explain PCM coding effici Senibe signal-to-quantization noise ratio Plain the difference between linear and nonlin- ar PCM codes — and disadvan- theorem 10-9 Companding 10-10 Vocoders 10-11 PCM Line Speed 10-12 Delta Modulation POM 10-13 Adaptive Delta Modulation PCM 10-14 Differential PCM 10-15 Pulse Transmission 10-16 Signal Power in Binary Digital Signals Describe idle channel noise Explain several common coding methods Define companding and explain analog and digital companding Define digizal compression Describe vocoders Explain how to determine PCM line speed ML Describe delta modulation PCM. Describe adaptive delta modulation Define and describe differential pulse code modulation Describe the composition of digital pulses @ Explain intersymbol interference Explain eye patterns Explain the signal power distribution i binary digital signals 389yr ‘As stated previously, digital ransmission isthe transmittal of digital signals between yy points in a communications system, The signals can be binary or any other form of ggg Gigital pulses. The original source information may be in digital form, or it could be. that have been converted to digital pulses prior to transmission and converted back to in the receiver, With digital transmission systems, a physical facility, such as a pur of jal cable, or an optical fiber cable, is required to interconnect the various points within y ite down the cable. Digital pulses cannot be » The pulses are contained in and propagat : through a wireless transmission system, such as Earth's atmosphere or free space (vacuum "AT&T developed the first digital transmission system for the purpose of carrying encoded analog signals, such as the human voice, over metallic wire cables between te ‘Today, digital transmission systems are used to carry not only digitally encoded voice ang nals but also digital source information directly between computers and computer networks fy transmission systems use both metallic and optical fiber cables for their transmission medig. 40-1-1 Advantages of Digital Transmission The primary advantage of digital transmission over analog transmission is noise immunity ty nals are inherently less susceptible than analog signals to interference caused by noise with digital signals itis not necessary to evaluate the precise amplitude, frequency, or phasejg tain its logic condition. Instead, pulses are evaluated during a precise time interval, and determination is made whether the pulse is above or below a prescribed reference level Digital signals are also better suited than analog signals for processing and combi technique called multiplexing, Digital signal processing (DSP) is the processing of analog using digital methods and includes bandlimiting the signal with filters, amplitude equalzatg, phase shifting, It is much simpler fo store digital signals than analog signals, andthe tray rate of digital signals can be easi different types of equipment. Th addition, digital transmission systems are more resistant to analog systems to additie mig because they use signal regeneration rather than signal amplification. Noise produced in elestonicg: cuits is additive (ie., it accumulates); therefore, the signal-to-noise ratio deteriorates each tmeanig Jog signal is amplified. Consequently, the number of circuits the signal must pass through mie total distance analog signals can be transported. However, digital regenerators sample noisy signi then reproduce an entirely new digital signal with the same signal-to-noise ratio asthe orgie) mitted signal. Therefore, digital signals can be transported longer distances than analog signals FFinally, digital signals are simpler to measure and evaluate than analog signals. Thereoi is easier to compare the error performance of one digital system to another digital system. Also} digital signals, transmission errors can be detected and corrected more easily and more than is possible with analog signals. 10-1 INTRODUCTION e ly changed to adapt to different environments and to interfice yg 101-2 Disadvantages of Digital Transmission The transmission of digitally encoded analog signals requires significantly more ‘bandwidth that simply transmitting the original analog signal. Bandwidth is one of the most important ase any communications system because itis costly and limited. ‘Also, analog signals must be converted to digital pulses prior to transmission and) back to their original analog form at the receiver, thus necessitating additional encoding aid ing circuitry. In addition, digital transmission requires precise time synchronization clocks in the transmitters and receivers, Finally, digital transmission systems are incompa ‘older analog transmission systems, 10-2 PULSE MODULATION Pulse modulation consists essentially of sampling analog information signals and then those samples into discrete pulses and transporting the pulses from a source (0 0 des 390 Electronic Communications SystemsTransmission Bbit word i B:bit word in [LIT B:bitword, ML. FIGURE 10-1. Pulse modulation: (a) analog signal; (b) sample pulse; (c) PWM; (d) PPM; (e) PAM, (f) POM 39110-3 PCM nest amplitude sample produces a pulse 10 the far right, ang ‘within the prescribed time slot. The hig! ot ple produces pulse 10 ant-postion pulse is varied according g lowest amplitude samy r shes he amplitude of a constant width, cOnSUDLBOST ET DY ret canbe heamplitude of the sample ofthe analog signal PAM is shown in Figure 10-16 516 10 be ee ial the amplitude of a pulse coincides with the amplitude ci is fat a ! a Se i jamal more than the waveforms for PWM 0 Ten Sampled and te cone rect ach code has the same number of bits and requires the same length of time for trans. mission. PCM is shown in Figure 10- If. M, and PCM, alth with PSK, QAM, and PCM, although ii ‘PAM is used as an intermediate form of modulation ON cules Senet I-purpose cor seldom used by itself, PWM and PPM are used in speci Pt Te eure ci I transmission ifitary but are seldom used for commercial digital Ercan ee ‘of pulse modulation and, consequent, ill be discussed in more detail in sub. sequent sections of this chapter 937 while working for AT&T at its Paris labo. .d early in its development, it Was not until the hat PCM became prevalent. In the United States today, PCM is te preferred method of communications within the public switched telephone mework because with PCM itis easy to combine digitized voice and digital data into a single, high- speed digital signal and propagate it over either metallic or optical fiber cables. PCA isthe only digitally encoded modulation technique shown in Figure 10-1 that is com. monly used for digital transmission. The term pulse ‘code modulation is somewhat of a misnomer, as itis not really a type of modulation but rather a form of digitally coding analog signals. With PCM, the pulses are of fixed length and fixed amplitude. PCM is a binary system where pulse or lack of pulse within a prescribed time slot represents cither a logic | or a logic 0 condition PWM, PPM, ‘and PAM are digital but seldom binary, as a pulse does not represent a single binary digit (bit). Figure 10-2 shows a simplified block diagram of a single-channel, simplex (one-way only) PCM. system. The bandpass filter limits the frequency of the analog input signal to the standard voice-band frequency range of 300 Hz to 3000 Hz. The sample-and-hold circuit periodically samples the analog input signal and converts those samples to a multilevel PAM signal. The analog-to-digital converter (ADC) converts the PAM samples to parallel PCM codes, which are converted to serial binary data in the parallel-to-serial converter and then outputted onto the transmission line as serial digital pulses. The transmission line repeaters are placed at prescribed distances to regenerate the digital pulses. In the receiver, the serial-to-parallel converter converts serial pulses received from the trans- mission line to parallel PCM codes. The digital-to-analog converter (DAC) converts the parallel PCM codes to multilevel PAM signals. The hold circuit is basically a low-pass filter that converts the PAM signals back to its original analog form Figure 10-2 also shows several clock signals and sample pulses that will be explained in later sections of tis chapter. An integrated circuit that performs the PCM encoding and decoding fune- tions is called a codec (coder/decoder), which is also described in a later section of this chapter. is credited with inventing PCM in I its of PCM were recognize 1t of solid-state electronics, tI ‘Alex H. Reeves i ratories. Although the meri mid-1960s, with the advent 10-4 PCM SAMPLING 392 The function of a sampling circuit in a PCM transmitter is to periodically sample the continually changing analog input voltage and convert those samples to a series of constant-amplitude pulses that can more easily be converted to binary PCM code, For the ADC to accurately convert a voltage 0 8 binary code, the voltage must be relatively constant so that the ADC can complete the conversion before the voltage level changes, If not, the ADC would be continually attempting to follow the changes and may never stabilize on any PCM code r hk Electronic Communications SystemsLine speed clock Conversion clock PCM Receiver IRAE 102 Simplfed block diagram ofa singlechannl, simplex PCM ransmission system (@) Input wavelorm (by Sample (o) Output aa waveform FIGURE 10-3 Natural sampling: (a) input ‘analog signal; (b) sample pulse; (c) sampled LJ courput Essentially, there are two basic techniques used to perform the sampling function: natural sam= pling and flat-top sampling, Natural sampling is shown in Figure 10-3. Natural sampling is when tops of the sample pulses retain their natural shape during the sample interval, making it ifficult for an ADC to convert the sample to.a PCM code, With natural sampling, the frequency spectrum of the sampled output is different from that of an ideal sample, The amplitude ofthe frequency components tal Transmission ae?FIGURE 104 Flettop sampling: (a) inp Fhaiog signal: (6) sample pulse; (c) sompleg () a L ‘output wavelorm ‘creases for the higher harmonics in a (sin xy wiring the use of frequency equalizers de width sample pulses xd from narrow, finite-widl es Fanner. This alters the information frequency spectrum rea wery by a low-pass filter. 2 a -and-hold circuit. The purpose of a sample-and-hold cir. re oteon St Papi PAM voltage levels. With flat-top sampling, the input vo. See «oan = wulse and then held relatively constant until the next sample is taken ee Seep cataling ‘As the figure shows, the sampling process alters the frequency a err yecuces an error cilled ‘aperture error, which is when the amplitude of the sampled Se rins ee the sample pulse time. This prevents the recovery circuit in the PCM receiver Pre roduc the original analog signal voltage. The magnitude of error depends on how Pees ccmucing he ing taken and the width (duration) of voltage hile the sample is bei much the analog signal voltage changes w! a dura the sample pulse. Flat-top sampling, however, introduces less aperture distortion than natural sam. produce ing and can operate with a slower analog-to-digital converter. a : eceae fosacon he schematic diagram of a sample-and-hold circuit. The FET acts as a sim- mpedance path to deposit the analog sample pple analog switch. When turned on, Q, provides a low ‘voltage across capacitor C;. The time that Q, is on is called the aperture or acquisition time. Essentially, , is the hold circuit. When Q, is off, C, does not have a complete path to discharge through and, there- fore, stores the sampled yoltage. The storage time of the capacitor is called the A/D conversion time because it is during this time that the ADC converts the sample voltage to a PCM code. The acquisi- tion time should be very short to ensure that a minimum change occurs in the analog signal while iti being deposited across C;. If the input to the ADC is changing while it is performing the conversion, ‘aperture distortion results. Thus, by having a short aperture time and keeping the input to the ADC rel id-hold circuit can reduce aperture distortion, Flat-top sampling intro- atively constant, the sample- duces less aperture distortion than natural sampling and requires a slower analog-to-digital converter. Figure 10-5b shows the input analog signal, the sampling pulse, and the waveform developed across C;. It is important that the output impedance of voltage follower Z, and the on resistance of (Q; be as small as possible. This ensures that the RC charging time constant of the capacitor is kept Very short, allowing the capacitor to charge or discharge rapidly during the short acquisition time. The rapid drop in the capacitor voltage immediately following cach sample pulse is due to the red tnbution of the charge across Cy, The inter-clectrode cupacitance between the gate and drain ofthe FET 6 placed in series with C, when the FET i off, thus acting as a capacitive voltage-divider net work, Also, note the gradual discharge across the capacitor during the conversion time, This i called droop and is caused by the capacitor discharging through its own leakage resistance and the input impedance of voltage follower Z,, Therefore, it is important that the input impedance of 23 Electronic Communications Systemsand the leakage resistance of C, isolate the sample-and-hold circui Example 10-1 For the sample-and-hold circuit shown in Fi be as high as possible. Essentially, voltage followers Z, and Z; it (Q, and C,) from the input and output circuitry. igure 10-5a, determine the largest-value capacitor that can be used. Use an output impedance for Z, of 10.0, an on resistance for Q, of 10 2, an acquisition time of 10 us, a maxi- ‘um peak-to-peak input voltage of 10 V, a maximum output curent from Z, of 10 mA, and an accuracy of 1%. Solution The expression for the current through a capacitor is Rearranging and solving for C yields ‘where C = maximum capacitance (farads) i = maximum output current from Z), 10 mA dy = maximum change in voltage across Cj, which equals 10 V dt = charge time, which equals the aperture time, 10 us _ GOmA\O js) 10v Therefore, Ce =10 nF ‘The charge time constant for C when Q, is on is a iC ‘where t= one charge time constant (seconds) wutput impedance of Z, plus the on resistance of Q, (ohms) “apacitance value of C; (Varads)Rearranging and solving for C gives us Cue ar andent on the accurac ized as follows: 0 depe ‘are summari ‘The charge time of capacitor Cy is al its required RC time constant i 46r 69 a 928 For an accuracy of 1%, JOHS_ — 108.7nF 416(20) a maximum capacitance of 10 nF was required. To sat To satisfy both requirements, the smaller-value capacitor mye To satisfy the output current limitations of 2), accuracy requirements, 108.7 nF was required. be used. Therefore, C can be no larger than 10 nF. 10-4-1 Sampling Rate ‘The Nyquise sampling theorem establishes the minimum sampling rate (J,) that can be used f siven PCM system. Fora sample to be reproduced accurately in a PCM receiver, each cycle ofan analog input signal (f,) must be sampled at least twice. Consequently, the minimum sampling = ic aqualtotwice the highest audio input frequency. If less than two times fn impairment alias or foldover distortion occurs. Mathematically, the minimum Nyquist sampling rate is ay 2h (1041) minimum Nyquist sample rte (hert2) ‘f= maximum analog input frequency (hertz) ‘A sample-and-hold circuit is a nonlinear device (mixer) with two inputs: the sampling pulse and Pe ee Corseroenty oclinew: muting (eterodyning) occurs between these aaa Figure 10-6a shows the frequency-domain representation of the output spectrum ia . Sumple-and-holdcireait. ‘The output includes the two original inputs (the audio and the funded where Frequency —> ‘Shaded areas indicate spectral foldover htt m - + sency —» 7 Braet) NUS, trequerey) (b) FIGURE 10-6 Output 6 pectrum for @ sample-and-hold circuit: (a) no aliasing; (b) aliasing distortion Electronic Communications Systems+ their su A 0 on) and tte equ (f= I the hamonics of slated sine waves, Each of eee TEV naetga OOS =f fy 8000) uenci INE WaVes ig arnt tS made up of a series of harmonically Hiri te ages res is Modulated by the analog signal and pro- seat least twice 7 EMEA is separated at Und cach ofthe harmonics of f, Each sum and nother amides note Ot he wae Fegicn et HPCEWS Gale rege by As ong 08 another harmonic, and aliasing does eens fom one harmonic wil spill into the sidebands of frequency greater than 2 Modulates Ta Fieure 10-6b shows the results when an analog input sideband of another harmonic. The far SMe fTequencies fom onc ara ence ina names “aliasing” orders disoniay at Folds over is an alias of the input signal (hence the te remoney ay Si fequency fom the fst harmonic folds Example 10.2 through filtering or any other technique. Fora PCM system with a maxim um audi the alias frequency produced fasten ey ‘of 4 kHz, determine the minimum sample rate and ; : were allowed a Solution MENUS sampling trem Equi lo ‘ ig ee 1), we have LF. theres Ifa S-kHz audio frequen A fore, f, = 8 kHz irodaced. can be seca tha te an AB andold Circuit, the output spectrum shown in Figure 10-7 is into the original audio spectrum, 1 EM Produces an alias frequency 3 Kile a tes tae acetal ‘Alias frequency { Site [+s atte ° ete Skitz FIGURE 10-7 Output spectrum for Example 15-2 Table 10-1 Three Bit POM Code ee Sign Magnitude Decimal Value 1 u 8 1 10 + 1 on +1 1 00 +0 0 00 0 0 01 =I 0 10 =2 0 ul 3 Digital Transmission on‘ Folded Binary Code 10-42 Quantization ee ais ‘an infinite number of possi tee oe heae ite number of amplitude possibilities. Thus, convertin ay i nations requires quantiza condi ith a limited number of com on, tp analog signal to. PCM code with 2 Mitt ne amplitudes of Mato ames mag essence, quantization is Hepes “sine wave with a peak amplitude o s between 4.5 "blemumterof levels Forexample 2st wa oT PCM code could have only ible and SV passing trae Pgcombinations Obviously. © convert samples Of sine aye bits, wh u which a PCM requires some rounding o 1 is subdivided into a smaller number of subrange, With ee sienna ‘able 10-2 is athree-bit sign-magnitude code yun San eee ie sinations (four positive and four negative). The leftmost bitis the sign bit (] = oi a ciao rightmost bits represent magnitude. This type of code is called a fngey ano ainsee the codes on the bottom half of the table are a mirror image of the codes on ty Oia one aca an gn bit If the negative codes were folded over on top of the positive odes, top al Oe sy, With a folded binary code, each voltage level has one code assigneg Te aoa sree hich has twocodes, 100 (+0) and 00 (0). The magnitude difference betwegy Lae ication interval ot quantum. For the code shown in Table 10-9 the steps is called the quanti: ir iiss ticral is LV. Therefore, for this code, the maximum signal magnitude that can je =3 V (011), and the minimum signal magnitude is +1 V (101) or ~j y Fp re apne ee cco tha bight quantization inter, veroad ada (also called peak limiting) occurs. ‘Assigning PCM codes to absolute magnitudes is called quantizing. The magnitude of a quan. tum is aso called the resolution. The resolution is equal to the voltage of the minimum step sice which is equal to the voltage of the least significant bit (V4) of the PCM code. The resolution is the ‘minimum voltage other than 0 V that can be decoded by the digital-to-analog converter in the receiver. The resolution for the PCM code shown in Table 10-2 is | V. The smaller the magnitude of quantum, the better (smaller) the resolution and the more accurately the quantized signal wil resemble the original analog sample. In Table 10-2, each three-bit code has a range of input voltages that will be converted to that code, For example, any voltage between +0.5 and +1.5 will be converted to the code 101 (+1 Vp, Each code has a quantization range equal to + or — one-half the magnitude of a quantum except the codes for +0 and —0. The 0-V codes cach have en input range equal to only one-half g quantum (0.5 V), ies 104 finite numb, Table 10-2 Three-Bit POM Code Sign Magnitude Decimal value Quantization range a *2.5.V to 43.5 V +2 +1-5 V to 42.5 V +h +05 V to 41.5 V eee eH, OV to +0.5 V OV to -0.5 v “0-5 V to -1.5 y “15 V to -2.5 V “2-5 V to -3.5V Electronic Communications SystemsFIGURE 10-8 (a) Analog input si erica ignal;(b) sample pulse; (c) PAM signal: Figure 10-8 shows an analog input signal, the sampling pulse, the corresponding quantized sig- nal (PAM), and the PCM code for each sample. The likelihood of a sample voltage being equal to ‘one of the eight quantization levels is remote. Therefore, as shown in the figure, each sample volt- age is rounded off (quantized) to the closest available level and then cenverted to its corresponding PCM code. The PAM signal in the transmitter is essentially the same PAM signal produced in the receiver. Therefore, any round-off errors in the transmitted signal are reproduced when the code is converted back to analog in the receiver. This error is called the quantization error (Q,). The quan- tization error is equivalent to additive white noise as it alters the signal amplitude, Consequently, quantization error is also called quantization noise (Q,). The maximum magnitude for the quantiza- tion error is equal to one-half a quantum (0,5 V for the code shown in Table 10-2) ‘The first sample shown in Figure 10-8 occurs at time f,, when the input voltage is exactly +2 V. ‘The PCM code that corresponds to +2 V is 110, and there is no quantization error. Sample 2 occurs at time f, when the input voltage is —1 V. The corresponding PCM code is 001, and again there is ‘no quantization error. To determine the PCM code for a particular sample voltage, simply divide the voltage by the resolution, convert the quotient to an n-bit binary code, and then add the sign bit. For sample 3 in Figure 10-9, the voltage at is approximately +2.6 V. The folded PCM code is sample voltage resolution There is no PCM code for +2.6; therefore, the magnitude of the semple is rounded off to the nearest valid code, which is 111, or +3 V. The rounding-off process results in a quantization error of 0.4 V, ‘The quantized signal shown in Figure 10-8e at best only roughly resembles the original analog. input signal. This is because with a three-bit PCM code, the resolution israther poor and also because 399FIGURE 10-3 PAM: (a) input signal; (b) sample pulse: (c) PAM signal there are only three samples taken of the analog signal. The quality of the PAM signa improved by using a PCM code with more bits, reducing the magnitude of a quantum and imp: ing the resolution. The quality can also be improved by sampling the analog signal at a faster p Figure 10-9 shows the same analog input signal shown in Figure 10-8 except the signal is being, ae pled at a much higher rate. As the figure shows, the PAM signal resembles the analog input si rather closely. Figure 10-10 shows the input-versus-output transfer function for a linear analog-to-digital verter (sometimes called a linear quantizer). As the figure shows for a linear analog input signal jj a ramp), the quantized signal is a staircase function. Thus, as shown in Figure 10-7e, the ma quantization error is the same for any magnitude input signal. Example 10-3 aie boiz nape 2 low sfc and PCM code for the analog sample voltage of +1.07 V. Solution To determine the quantized level, simply divide the sample voltage by resolution and then. answer off to the nearest quantization level: +107 IV 07=1 ‘The quantization error is the difference between the original sample voltage and the quantized level, or Q.= 107-1 =007 From Table 10-2, the PCM code for +1 is 101, 10-4-3 Dynamic Range The number of PCM bits transmitted per sample is determined by several variables, includi - imum allowable input amplitude, resolution, and dynamic range, Dynamic range (DR) is the raid Electronic Communications Systems® FIGURE 10-10 Linear inputversus-output transfer curve: (a) linear transfer function: {b) quantization: (c) G, the largest possible magnitude to the smallest possible magnitude (other than 0 V) that can be decoded by the digital-to-analog converter in the receiver. Mathematically, dynamic range is ¥, fc lén 10-2} Von ey where DR = dynamic range (unitless ratio) the quantum value (resolution) the maximum voltage magnitude that can be discerned by the DACs in. the receiver Equation 10-2 can be rewritten as DR=——*\_ resolution Oc) For the system shown in Table 10-2, av. DR=—o= We ‘ansmission 401“The number of bits used for a PCM code depends on the dynamic ‘between dynamic range and the number of bits ina PCM code is a —-1=DR and for a minimum number of bits a where _n = number of bits in a PCM code, excluding the sign bit DR = absolute value of dynamic range Why 2" — 1? One positive and one negative PCM code is used for 0 V, which is not dynamic range. Therefore, =DR 2 =DR+1 To solve for the number of bits (1) necessary to produce a dynamic range of 3, convert log 2" = log(DR + 1) nog 2 = log(DR + 1) log(3+1) _ 0.602 log? 0.301 For a dynamic range of 3, a PCM code with two bits is required. Dynamic range can bee in decibels as V, DRiaw) = zatg{ f=) or Rip = 20 log(2" — 1) where n is the number of PCM bits, For values of n > 4, dynamic range is approximated Rp) = 20-103(2") =20n log(2) =6n Equation 10-7 indicates that there is approximately 6 dB dynamic range for each magnitude Jinear PCM code. Table 10-3 summarizes dynamic range for PCM codes with n bits for valk upto 16. 10-4-4 Coding Efficiency Coding efficiency is a numerical indication of how efficiently a PCM code is utilized, ciency is the ratio of the minimum number of bits required to achieve a certain dynamiet the actual number of PCM bits used, Mathematically, coding efficieney is coding efficiency = itimum numberof bits (including sam bi) 9) actual number of bits (including sign bit) Electronic Communications Systems‘the coding efficiency for Example 10-4 is coding efficiency 100 = 95.89% Example 10-4 For minimum line speed with an 8-bit PCM for speech signal ranging upto 1 Volt fa Calculate the resolution and quantization error. by. Calculate the coding efficiency for a resolution of 0.01 Volt 4 with the 8-bit PCM. Solution Minimum line speed with an 8-bit PCM is 64 Kbps: a, plied De eet = 0.003922 V ‘Therefore, Resolution = quantization step, q = 0.003922 V Quantization error = 2 = 0.00196 V ey ones arse Dries wv 2010 = 404B or 100 7a eae Minimum numberof bits n° required to achieve his dynamic range is given by los\DR +) _ 6.66 Tog? minimum number of its 49 erefore, efficiency = ce ogine amd actual number of bits 6.66 $88 100 = 83.25% 80 | SIGNAL-TO-QUANTIZATION NOISE RATIO The three-bit PCM coding scheme shown in Figures 10-8 and 10-9 consists of linear codes, which means that the magnitude change between any two successive codes is the same, Consequently, the I Transmission . 403_ resolution _ 5 0 is 4 Fi 10-8, the worst-case (minimum) SQR. ltage (+1 V). Therefore, the minimum SQR is ee SOR win = 95 = 20 log(2) = 6dB 2 From the preceding example, it can be seen that even though the magnitude of tion error remains constant throughout the entire PCM code, the percentage error ¢p decreases as the magnitude of the sample increases. The preceding expression for SQR is for voltage and presumes the maximum g error: therefore, itis of little practical use and is shown only for comparison purposes ang, tate that the SQR is not constant throughout the entire range of sample amplitudes, Tn yea shown in Figure 10:9, the difference between the PAM waveform and the analog input varies in magnitude. Therefore, the SQR is not constant. Generally, the quantization 2 tion caused by digitizing an analog sample is expressed as an average signal power-to-g power ratio, For linear PCM codes (all quantization intervals have equal magnitudes), power-to-quantizing noise power ratio (also called signal-to-distortion ratio or si ratio) is determined by the following formula: SOR w= 10loe a oyR where R = resistance (ohms) v = rms signal voltage (volts) 4 = quantization interval (volts) v/R = average signal power (watts) (@/2)IR = average quantization noise power (watts) If the resistances are assumed to be equal, Equation 10-8a reduces to i SOR = 101o; 4 el 7ia] =10.8=20log~ q 10-6 LINEAR VERSUS NONLINEAR PCM CODES Early PCM systems used linear codes (i.e., the magnitude change between any two sucee uniform). With linear coding, the accuracy (resolution) for the higher-amplitude analo 404 Electronic Communications SystemsNodecodednoite FIGURE 10-12 die channel noise same as for the lower-amplitude signals, andthe SQR for the lower-amplitude signals is less Wat for aa Kisher amptitude signals. With voice transmission, low amplitude signals are-more likely 19 0° caearge-amplitude signals. Therefore, if there were more codes for the lower amplitudes: would ipafeave the accuracy where the accuracy is needed, AS a result, there would be fewer codes available toe the higher amplitudes, which would increase the quantization error forthe larger-amplitude signals {ttus decreasing the SQR), Such a coding technique is called nonlinear or nonuniform encod: With nonlinear encoding, the step size increases with the amplitude of the input signal: Figure 10-11 shows the step outputs from a linear and a nonlinear analog-to-digital comvese Note, with nonlinear encoding, there are more codes atthe bottom of the scale than there are at the {Ps thos ;neteasing the accuracy for the smaller-amplitude signals. Also note that the distance betwesn sae ae to codes is greater fr the higher-amplitude signal, thus increasing the quantization error and reduc tha the SOR, Also, because the ratio of Vu © Vann icreased with nonlinear encoding, the dynamic ‘age is larger than with a uniform linear code. Iis evident that nonlinear encoding is a compromise; SOR is sacrificed for the higher amplitude signals to achieve more accuracy forthe lower-amplitude sig- vhals and to achieve a larger dynamic range, Itis difficult to fabricate nonlinear analog-to-digital con- Verters; consequently, alternative methods of achieving the same results have been devised. 107 IDLE CHANNEL NOISE During times when there is no analog input signal, the only input to the PAM sampler is random, thermal noise. This noise is called idle channel noise and is converted to a PAM sample just as if it ‘were a signal. Consequently, even input noise is quantized by the ADC. Figure 10-12 shows a way to reduce idle channel noise by a method called midtread quantization. With midtread quantizing, the first quantization interval is made larger in amplitude than the rest of the steps. Consequently, input noise can be quite large and still be quantized as a positive or negative zero code. AS a result, the noise is suppressed during the encoding process. Transmission aosLM code, residual noise that fMuctu ates slightly above somes a .d, consequently, is eliminated. In s, ea ‘could cause the PCM encoder (alt "Zet6'eode/and the minimum + or — code. Consequently, the decoder would A horse. With a folded binary code, most of the residual noise is inherently ej itize PAM signals into 2" levels. These me an There are several coding methods used 10 quantize PAN se a times Tes classified according to whether the coding operal a word at a time. Level-at-a-Time Codi ae of coding ee the ami signal to a ramp waveform while a binary ‘counter ig advanced at a uniform rate, When the ramp waveform equals or exceeds the PAM counter contains the PCM code. This type of coding requires a very fast clock ifthe number ql in the PCM code is large, Level-at-a-time coding also requires that 2" sequential decisions je” for each PCM code generated. Therefore, level-at-a-time coding is generally tim, low-speed applications. Nonuniform coding is achieved by using a nonlinear function reference ramp. 10-8-2 Digit-at-a-Time Coding : q This type of coding determines each digit of the PCM code sequentially. Digit-at-a-time Coding ‘analogous to a balance where known reference weights are used to determine an unknown, Digit-at-a-time coders provide a compromise between speed and complexity. One common king digit-at-a-time coder, called a feedback coder, uses a successive approximation register (SAR), this type of coder, the entire PCM code word is determined simultaneously. 10-8-3 Word-at-a-Time Coding Word-at-a-time coders are flash encoders and are more complex; however, they are more suitable iy high-speed applications, One common type of word-at-a-time coder uses multiple threshold cireu Logic circuits sense the highest threshold circuit sensed by the PAM input signal and produce te approximate PCM code. This method is again impractical for large values of 1 10-9 COMPANDING OB Companding is the process of compressing and then expanding. With companded systems, te higher-amplitude analog signals are compressed (amplified less than the lower-amplitude signa) Prior to transmission and then expanded (amplified more than the lower-amplitude signals) inthe receiver. Companding is a means of improving the dynamic range of a communications system, Figure 10-13 illustrates the process of companding. An analog input signal with a dynamic range of 50 dB is compressed to 25 dB prior to transmission and then, in the receiver, expanded bik to its original dynamic range of 50 dB. With PCM, companding may be accomplished using andl, or digital techniques. Early PCM systems used analog companding, whereas more modern syste use digital companding. i Electronic Communications Systemsresi ay Ge ile ing Saal Input Transmission Output media FIGURE 10-13 Basic companding process arate! nh i 7 a N N PAM ‘Serial 2 = = eS ee See et lea = ee . PCM transmitter eran | OY [seal gine. JI) See Ny aasie Ie om cet sreiog [2—] paral aw ww corer | | env [Sere Pow POM recive t Para! Pow FIGURE 10-14 PCM system with analog companding 10-31 Analog Companding Historically, analog compression was implemented using specially designed diodes inserted in the analog signal path in a PCM transmitter prior to the sample-and-hold circuit. Analog expansion was also implemented with diodes that were placed just after the low-pass filter in the PCM receiver, Figure 10-14 shows the basic process of analog companding, In the transmitter, the dynamic range of the analog signal is compressed, sampled, and then converted to a linear PCM code. In the Transmission sinontoml 407