The document discusses digital baseband modulation and waveform coding techniques. It covers topics like formatting of textual and analog data, pulse code modulation (PCM), quantization, sampling, and delta modulation. Formatting involves character coding using ASCII or EBCDIC. Analog signals are converted to digital form through sampling and quantization. In PCM, an analog signal is sampled, quantized into discrete levels, and encoded into binary digits. Other covered topics include pulse amplitude modulation, pulse width modulation, pulse position modulation, and oversampling.

1. Digital Baseband Modulation & Waveform coding Techniques Or Source Coding Techniques V. S. Hendre Department of E&TC, TCOER, Pune 1 UNIT-I

2. UNIT-I: Digital Baseband Modulation Techniques and Waveform Coding Techniques Base band system, Formatting textual data, messages, characters & symbols, Formatting analog information, Sources of corruption, PCM, Uniform and Non uniform quantization, Baseband modulation, Noise consideration in PCM systems, DPCM, DM,ADM, LPC. V. S. Hendre Department of E&TC, TCOER, Pune 2

3. INTRODUCTION Formatting: is to insure that the message is compatible with Digital Signal Processing Transmit Formatting: is a transformation from source information to digital symbols. Source coding: data compression + formatting Formatting Character Coding Sampling Quantization Pulse Code Modulation (PCM) V. S. Hendre Department of E&TC, TCOER, Pune 3

4. INTRODUCTION V. S. Hendre Department of E&TC, TCOER, Pune 4 Formatting Character Coding Sampling Quantization Pulse Code Modulation (PCM) Source Coding Predictive Coding, Block Coding Variable Length Coding Synthesis Coding Lossless Compression Lossy compression Baseband Signaling Line Codes/Data Formats RZ,NRZ, Phase encoded, Multilevel binary, PAM, PPM ,PWM

5. Baseband Systems 5 Signal Source / information source Signal Sampling circuit Source quantiser encoder Channel encoder modulator Communication channel AWGN V. S. Hendre Department of E&TC, TCOER, Pune

6. 6 Digital info. Format Textual info. source Pulse modulate Transmit Encode Sample Quantize Analog info. Channel Pulse waveforms Bit stream Format Analog info. Low-pass filter Decode Demodulate/ Detect Receive Textual info. sink Digital info. Baseband Systems V. S. Hendre Department of E&TC, TCOER, Pune

7. Formatting Textual Data (Character Coding) Original or baseband data is either textual or analog. If data is alphanumeric text, it will be character encoded with some standard formats. These formats are: ASCII (American Standard Code for Information Interchange) EBCDIC: Extended Binary Coded Decimal Interchange Code. V. S. Hendre Department of E&TC, TCOER, Pune 7

8. ASCII Format (7 bit) V. S. Hendre Department of E&TC, TCOER, Pune 8

9. EBCDIC Format V. S. Hendre Department of E&TC, TCOER, Pune 9

10. Messages, Characters & Symbols Textual message is first encoded in digital form by using ASCII or EBCDIC format. This digital sequence of bits is called as bit stream or baseband signal. Groups of ‘K’ bits can be combined to form new digits or Symbols. Total no of symbols =M=2K . A system using a symbol size of M is called as M-ary System. M= 2- Binary System, M=3-Trinary System M=4 –Quaternary System, M=5- 5ary system V. S. Hendre Department of E&TC, TCOER, Pune 10

11. Ex: Messages, Characters & Symbols V. S. Hendre Department of E&TC, TCOER, Pune 11

12. Formatting Analog Information If information is in Analog form then we can not be character encoded it as in textual form. Here we need to convert it in Digital form by using the processes of Sampling & Quantization V. S. Hendre Department of E&TC, TCOER, Pune 12

13. Sampling for Low Pass Signals Sampling is the process of taking a periodic sample of the waveform to be transmitted. Sampling of signal is the fundamental operation in digital comm. It is the process of conversing an analog signal (continuous time) into discrete time signal. Statement for Low pass Sampling Theorem: A continuous time band limited signal can be completely represented in it sample form and recovered back if the sampling freq fs ≥ 2 w. when fs- sampling freq and w- is the max freq present in the signal. V. S. Hendre Department of E&TC, TCOER, Pune 13 Where fs = sampling frequency fm(max) = maximum frequency of the modulating signal

14. Sampling for Low Pass Signals V. S. Hendre Department of E&TC, TCOER, Pune 14

15. Proof for Sampling Theorem V. S. Hendre Department of E&TC, TCOER, Pune 15

16. V. S. Hendre Department of E&TC, TCOER, Pune 16 V (volt) f (Hz) fs 2fs 3fs fm(max) fs+fm(max) fs-fm(max) Sampling Three basic condition of sampling process: Sampling at fs=2fm(max)

17. V. S. Hendre Department of E&TC, TCOER, Pune 17 V (volt) Guardband f (Hz) fs 2fs fm(max) fs-fm(max) fs+fm(max) Sampling Sampling at fs>2fm(max) This sampling rate creates a guard band between fm(max) and the lowest frequency component fs-fm(max) of the sampling harmonics.

18. V. S. Hendre Department of E&TC, TCOER, Pune 18 V (volt) Aliasing distortion f (Hz) 2fs fs 3fs fs-fm(max) fm(max) fs+fm(max) Sampling Sampling at fs<2fm(max) Aliasing: the distortion produced by the overlapping components from adjacent bands Aliasing occurs when a signal is sampled below its Nyquist rate

19. Sampling V. S. Hendre Department of E&TC, TCOER, Pune 19 Aliasing effect in Time Domain

20. Sampling V. S. Hendre Department of E&TC, TCOER, Pune 20 Sampling Rate: Practical Consideration Voice Signals: Fmmax: 3.4 KHz Nyquist Criteria: 2 x 3.4K =6.8KHz Practical Sampling Rate: 8KHz. 2. High quality Music System: Max. Bandwidth : 20KHz Nyquist Criteria; 2 x 20K = 40 KHz Practical Sampling Rate:44.1 Ksamples/sec 3. Studio Quality Audio : Sampling Rate: 48.0 Ksamples/sec Thus by an engineer’s version, Nyquist sampling Rate is

21. Sampling V. S. Hendre Department of E&TC, TCOER, Pune 21 Sampling Rate: Practical Consideration Voice Signals: Fmmax: 3.4 KHz Nyquist Criteria: 2 x 3.4K =6.8KHz Practical Sampling Rate: 8KHz. 2. High quality Music System: Max. Bandwidth : 20KHz Nyquist Criteria; 2 x 20K = 40 KHz Practical Sampling Rate:44.1 Ksamples/sec 3. Studio Quality Audio : Sampling Rate: 48.0 Ksamples/sec Thus by an engineer’s version, Nyquist sampling Rate is

23. This is so because signal processing performed with high performance analog equipment is typically much more costly than using digital signal processingequipment to perform the same task. Without Oversampling 1. The signal passes through a high performance analog lowpass filter to limit its bandwidth. 2. The filtered signal is sampled at the Nyquist rate for the (approximated) bandlimited signal. 3. The samples are processed by an analog-to-digital converter that maps the continuous-valued samples to a finite list of discrete output levels.

24. Why Over Sample? V. S. Hendre Department of E&TC, TCOER, Pune 23 With Oversampling 1. The signal is passed through a low performance (less costly) analog low-pass filter (prefilter) to limit its bandwidth. 2. The pre-filtered signal is sampled at the (now higher) Nyquist rate for the (approximated) bandlimited signal. 3. The samples are processed by an analog-to-digital converter that maps the continuous-valued samples to a finite list of discrete output levels. 4. The digital samples are then processed by a high performance digital filter to reduce the bandwidth of the digital samples. 5. The sample rate at the output of the digital filter is reduced in proportion to the bandwidth reduction obtained by this digital filter.

25. V. S. Hendre Department of E&TC, TCOER, Pune 24 Communication System Continuous Wave Digital Wave Analogue Pulse Modulation Digital Pulse Modulation PAM PWM PPM Analogue Pulse Modulation Chart

26. Analog Pulse Modulation (APM) V. S. Hendre Department of E&TC, TCOER, Pune 25 In APM, the carrier signal is in the form of pulse form, and the modulated signal is where one of the characteristics either (amplitude, width, or position) is changed according to the modulating/audio signal. Three common techniques of APM: Pulse amplitude modulation (PAM) Pulse Width Modulation (PWM) Pulse Position Modulation (PPM)

27. Waveforms for PAM, PWM and PPM V. S. Hendre Department of E&TC, TCOER, Pune 26 Modulating signal carrier signal PAM (dual polarity) PWM PPM

28. Pulse Amplitude Modulation (PAM) V. S. Hendre Department of E&TC, TCOER, Pune 27 It is very similar to AM The amplitude of a carrier signal is varied according to the amplitude of the modulating signal. Two type PAM Dual- polarity PAM Single -polarity PAM

29. Pulse Width Modulation (PWM) V. S. Hendre Department of E&TC, TCOER, Pune 28 The technique of varying the width of the constant amplitude pulse proportional to the amplitude of the modulating signal. PWM gives a better signal to noise performance than PAM

30. Pulse Position Modulation (PPM) V. S. Hendre Department of E&TC, TCOER, Pune 29 PPM is when the position of a constant width and constant amplitude pulse within prescribed time slot is varied according to the amplitude of the modulating signal.

31. Basic Techniquesa) Variable Length Codingb) Fixed Length Coding PCM, DM, ADM, DPCM etc. PCM-Linear Pulse code modulation Need-Analog PAM signaldigital V. S. Hendre Department of E&TC, TCOER, Pune 30

33. 2)Regenerative repeaters-increases SNR

34. (occuresamplitude & phase distortion)

35. 3)Encryption-privacy & security

36. 4) Uniform representation of signal

38. V. S. Hendre Department of E&TC, TCOER, Pune 33 Digital info. Format Textual info. source Pulse modulate Transmit Encode Sample Quantize Analog info. Channel Pulse waveforms Bit stream Format Analog info. Low-pass filter Decode Demodulate/ Detect Receive Textual info. sink Digital info.

40. Operation of quantisation V. S. Hendre Department of E&TC, TCOER, Pune 35 X(t) Xq(t) VH 7 q7 Quantization level  qo o VL =(VH-VL)/Q, Q:no of levels-signal is divided (Q=8), Q=2N, N=bits/sample

41. V. S. Hendre Department of E&TC, TCOER, Pune 36

42. V. S. Hendre Department of E&TC, TCOER, Pune 37 Whenever x(t) is in the range 0, xq(t) maintains the constant level qo xq(t) makes a quantum jump of step size  Quantized signal-approximation of original signal approximated signal is practically indistinguishable form original signal Quantization removes additive noise  /2

43. Qunatization example V. S. Hendre Department of E&TC, TCOER, Pune 38 Quant. levels boundaries x(nTs): sampled values xq(nTs): quantized values amplitude x(t) 3.1867 2.2762 1.3657  0.4552 -0.4552 -1.3657 -2.2762 -3.1867 Ts: sampling time Actual Sample value t

44. PCM-conversion V. S. Hendre Department of E&TC, TCOER, Pune 39 PCM Sequence

45. V. S. Hendre Department of E&TC, TCOER, Pune 40 Output Xq(nTs) Representation levels Transfer characteristics of quantizer/quantizer curve 7/2 5/2 Maximum quantization error /2 3/2  -X(nTs) Input X(nTs) /2  Decision levels 3 4 0 2 -/2 Overload levels -3/2 -5/2  Peak to peak excursion of the signal Quantization error () /2 Input X(nTs)  -/2

46. V. S. Hendre Department of E&TC, TCOER, Pune 41 Two types of quantization: (a) midtread and (b) midrise. 13

47. V. S. Hendre Department of E&TC, TCOER, Pune 42 Model of quantizing noise Quantization error Quantizing error: The difference between the input and output of a quantizer Maximum quantisation error=

48. V. S. Hendre Department of E&TC, TCOER, Pune 43 Quantization Noise Illustration of the quantization process. 14

49. Transmission Bandwidth N-no of bits/sample Quantization levels Q=2N Signaling rate=r=n.fs BW (PCM)=(1/2) x signaling rate (But ) V. S. Hendre Department of E&TC, TCOER, Pune 44

50. V. S. Hendre Department of E&TC, TCOER, Pune 45 Bandlimits fm-3.3KHz Flat Top PAM Quantized PAM P C M N bit q-level Parallel to serial converter Low pass Filter Sample & hold circuit Quantiser (uniform) Binary encoder Good SNR 8 bit -approximation -rounding off -reduces additive noise fc=fm Fs>>2fm @ 8KHz R=64 kbps Analog Speech signal (300Hz- 3.3 KHz) Pulse Generator Basic Block diagram PCM Transmitter X(t)

51. PCM receiver V. S. Hendre Department of E&TC, TCOER, Pune 46

53. No of bits per Sample: v or N = 8 bits

54. Quantization levels Q=2N = 256

55. Step Size :

56. If Sampling Frequency is 8KHz

59. Low o/p impd.Large load impd. Sample & Hold Circuit

60. V. S. Hendre Department of E&TC, TCOER, Pune 50 Output Xq(nTs) Representation levels Signal to Quantization noise ratio: SNRq 7/2 5/2 Maximum quantization error /2 3/2  -X(nTs) Input X(nTs) /2  Decision levels 3 4 0 2 -/2 Overload levels -3/2 -5/2  Peak to peak excursion of the signal Quantization error () /2 Input X(nTs)  -/2

61. Signal to Quantization noise ratio: SNRq V. S. Hendre Department of E&TC, TCOER, Pune 51 If the range of amplitude is from – Xmax to + Xmax The step size

62. Signal to Quantization noise ratio: SNRq V. S. Hendre Department of E&TC, TCOER, Pune 52

63. Signal to Quantization noise Ratio: SNRq For quantizer Noise power Noise by r.v. & its PDF Mean square value V. S. Hendre Department of E&TC, TCOER, Pune 53

64. V. S. Hendre Department of E&TC, TCOER, Pune 54 Mean square value of r.v. x Putting II) into I), mean square value of noise voltage = At R=1, noise power is normalized  Normalized noise power/ quantization noise power =

66.  S/N = 3 x 22n x P

67. If input signal power is normalised, P≤1,

69. V. S. Hendre Department of E&TC, TCOER, Pune 57 Virtues, Limitations and Modifications of PCM Advantages of PCM 1. Robustness to noise and interference 2. Efficient regeneration 3. Efficient SNR and bandwidth trade-off 4. Uniform format 5. Secure

70. PCM waveforms V. S. Hendre Department of E&TC, TCOER, Pune 58 Criteria for comparing and selecting PCM waveforms: Spectral characteristics (power spectral density and bandwidth efficiency) Bit synchronization capability Error detection capability Interference and noise immunity Implementation cost and complexity

71. Uniform and non-uniform quantisation V. S. Hendre Department of E&TC, TCOER, Pune 59 Uniform (linear) quantizing: step size -uniform No assumption about amplitude statistics and correlation properties of the input. Not using the user-related specifications Robust to small changes in input statistic by not finely tuned to a specific set of input parameters Simply implemented Over complete range of signal max=|/2| Application of linear quantizer: Signal processing, graphic and display applications, process control applications

72. Dis-advantages:Uniform Quantisation V. S. Hendre Department of E&TC, TCOER, Pune 60 1) let n=4 bits Q=2n=24=16 levels =2/q=(2/16)=(1/8) v max=|/2|=(1/16) If signal range=16 V, max=1 V is acceptable But it is very Harmful for signal amplitudes 2, 3 V….Lower Acceptable-signal amplitudes 15, 16… Higher  Non-Uniform quantization

73. V. S. Hendre Department of E&TC, TCOER, Pune 61 1.0 Probability density function 0.5 2.0 1.0 3.0 Normalized magnitude of speech signal 0.0 2) Statistical of speech amplitudes In speech, weak signals are more frequent than strong ones. Using equal step sizes (uniform quantizer) gives low for weak signals and high for strong signals. Adjusting the step size of the quantizer by taking into account the speech statistics improves the SNR for the input range.

74. 2) Statistical of speech amplitudes Another way: Crest Factor = Peak Value / RMS Value For speech or music signals Crest factor is very high.

75. V. S. Hendre Department of E&TC, TCOER, Pune 63 Non-Uniform Quantisation -Uses the input statistics to tune quantizer parameters -Larger SNR than uniform quantizing with same number of levels -Non-uniform intervals in the dynamic range with same quantization noise variance -Application of non-uniform quantizer: Commonly used for speech -for voice-amplitude values-concentrated near zero -variable step size-directly not applicable (generates error) -process:-signal amplification-at low level & -signal attenuation –at high level & - Uniform quantization -overall effect-Non-Uniform Quantization

76. Nonuniform Quantizer Used to reduce quantization error and increase the dynamic range when input signal is not uniformly distributed over its allowed range of values. allowed values input values for most of time time

77. “Compressing-and-expanding” is called “companding.” Nonuniform quantizer Discrete samples Uniform Quantizer digital signals Compressor • • • • Channel • • • • output Decoder Expander received digital signals

78. Compression Techniques

79. Practical Implementation of µ-law compressor

80. Output SNR of 8-bit PCM systems with and without companding.

81. V. S. Hendre Department of E&TC, TCOER, Pune 69 compression+expansion companding Non-uniform quantization….process At the transmitter Uniformly quantizing the “compressed” signal. At the receiver, an inverse compression/expansion characteristic, called “expansion” is employed to avoid signal distortion. Compress Qauntize Expand Channel Transmitter Receiver

82. Companding curve V. S. Hendre Department of E&TC, TCOER, Pune 70

83. Companding Curve V. S. Hendre Department of E&TC, TCOER, Pune 71 O/P. Voltage of Compander Compression Expansion I/P. Voltage of Compander Expansion Compression

84. Effect of companding V. S. Hendre Department of E&TC, TCOER, Pune 72

85. Compression laws Two Laws-’’ Law-United states, Canada, Japan (=225) ‘A’ Law- Europe & India (A=87.6) ’’ Law Defn: V. S. Hendre Department of E&TC, TCOER, Pune 73 W1(t)-input to compressor, allowed value= 1 W2(t)-output of compressor Appli: speech, music signals, PCM systems

86. SNR Performance of PCM with  Law V. S. Hendre Department of E&TC, TCOER, Pune 74 Fixed SNR-irrespective of wide variations of signal levels among individual talkers

87. Compression characteristic for  Law V. S. Hendre Department of E&TC, TCOER, Pune 75 As µ  ∞, Linear Amplification Standard Value of µ=255

88. ‘A’ Law characteristics V. S. Hendre Department of E&TC, TCOER, Pune 76 Compression characteristics As A  ∞, Linear Amplification Standard Value of A=87.6

89. V. S. Hendre Department of E&TC, TCOER, Pune 77 Figure 3.14 Compression laws. (a) m-law. (b) A-law.

90. Noise consideration in PCM systems (Channel noise, quantization noise) V. S. Hendre Department of E&TC, TCOER, Pune 78

91. Examples on PCM V. S. Hendre Department of E&TC, TCOER, Pune 79 A low pass signal of 3 KHz B.W. & amplitude over -5 volts to +5 volts range is sampled at Nyquist rate & converted to 8 bit PCM using uniform quantization. The mean squared value of message signal is 2 volt-squared. Calculate i) normalized power for quantization noise ii) Bit transmission rate iii) (S/N)Q in dB Soln: Given : W=3KHz, VL =-5V, VH =5V, N=8 i) Normalized quantization noise:

92. Examples on PCM V. S. Hendre Department of E&TC, TCOER, Pune 80 ii) Bit Transmission Rate: iii) (S/N)Q in dB:

93. Examples on PCM V. S. Hendre Department of E&TC, TCOER, Pune 81 2. A compact disc recording system samples each of the two stereo signals with 16 bit A/D converter at 44.1Kbps. Determine i) output S/N ratio for full scale sinusoid ii) The bit stream of digitized data is augmented by addition of error correcting bits, clock extraction bits etc., these additional bits represents 100% overhead. Determine output bit rate of the system. iii) The CD can record an hours worth of music. Determine no of bits recorded on CD. Soln: i) Output (S/N) = (1.76+6*N) = 97.76 dB ii) Bit rate of single channel: 16*44.1=705.6Kbps For two channels :705.6*2=1.411Mbps For additional 100% overhead Final Bit rate = 1.411*2=2.8224Mbps ii)This o/p bit rate represents 2.8224Mbps bits are coming per second (1 second). So, number of bits recorded in hour (3600 seconds) will be=2.8224Mbps x 3600=1.016 x 10^10 bits.

94. PCM with Noise 82

95. Delta Modulation V. S. Hendre Department of E&TC, TCOER, Pune 83 PCM-drabacks-1)Large signalling rate -2) Larger transmission BW Delta modulation:- 1bit/sample Present sample-compared with previous Result-Increase/Decrease in amplitude Input x(t)approximated , fixed step size  Diffn:x(t) & staircase approximated 2 levels:+ or - If diffn:+ve, increased by one step  & has step with Ts=delay time If diffn:-ve, decreased by one step  & has step with Ts=delay time

96. V. S. Hendre Department of E&TC, TCOER, Pune 84 For reduced step – ‘0’-transmitted For increased step- ‘1’-transmitted  for each sample-one bit transmitted Delta modulator/transmitter

97. Waveform representation V. S. Hendre Department of E&TC, TCOER, Pune 85

98. V. S. Hendre Department of E&TC, TCOER, Pune 86

99. V. S. Hendre Department of E&TC, TCOER, Pune 87

100. V. S. Hendre Department of E&TC, TCOER, Pune 88

101. V. S. Hendre Department of E&TC, TCOER, Pune 89 “start up interval”-interval required to meet approximated signal to input signal “Hunting” of approximated signal:-condition whenever input signal is almost constant or flat Error (kTs) Granular Noise When input signal increases or decreases too rapidly, approximated signal lags behind “Slope Overload error” Advantages:1) transmits only 1 bit/sample signaling rate & transmission BW-reduced 2)transmitter & receiver –implementation –easy Disadvantages:-1)Granular noise,2)slope overlaod Overcome-ADM

102. V. S. Hendre Department of E&TC, TCOER, Pune 90 The modulator consists of a comparator, a quantizer, and an accumulator Two types of quantization errors : Slope overload distortion and granular noise

103. Slope Overload Condition The slope overload distortion will occur if

104. V. S. Hendre Department of E&TC, TCOER, Pune 92

105. Examples on DM V. S. Hendre Department of E&TC, TCOER, Pune 93 A delta modulated system is designed to operate at five times the nyquist rate for a signal with 3Khz B.W. Determine the max. amplitude of a 2KHz input sinusoid for which the delta mod doesn't have slope overload. Quantizing step size is 250mV Soln: Given : W=3KHz, fm =2 KHz, fs =5 * 3 *2= 30KHz,

106. Examples on DM V. S. Hendre Department of E&TC, TCOER, Pune 94 2. A 1KHz signal sampled by 8KHz, is to be encoded by using 1) 12 bit PCM 2) DM system. If 20 cycles of 1 KHz are digitized, state how many bits will be there in digital output signal in each case. State signaling rate and B.W. in each case. Soln: 1) 12 bit PCM: Signaling Rate= v * fs=96Kbps B.W. = 48KHz Tm = 1/1000, Ts =1/800, Samples in one cycle= Tm/Ts = 8 In 20 cycles = 8*20 =160 samples No of bits transmitted = 160 * 12 =1920 2) DM systems Signaling Rate: 8Kbps, B.W.=4KHz No of bits transmitted= 160bits

107. DM receiver V. S. Hendre Department of E&TC, TCOER, Pune 95 *When received binary 1-accumulator adds + to previous o/p *When received binary 0-accumulator subtracts  from previous o/p

108. Adaptive Delta Modulation (ADM) V. S. Hendre Department of E&TC, TCOER, Pune 96 Step size -adaptive to variations of input signal x(t) Step size-reduced for slowly varying signal Step size-increased for steep segment of signal ADM transmitter

109. ADM-waveform V. S. Hendre Department of E&TC, TCOER, Pune 97

110. ADM-receiver V. S. Hendre Department of E&TC, TCOER, Pune 98

111. Why DM is not alternative for PCM for voice Signals? V. S. Hendre Department of E&TC, TCOER, Pune 99 Let us consider 8 bit PCM, N=8, Q=256,

112. V. S. Hendre Department of E&TC, TCOER, Pune 100 Delta-Sigma modulation (sigma-delta modulation) -Delta modulator with integrator -removes draw back of delta modulation -(Input to quantizer-approximation-derivative of input signal-demodulation-error) Beneficial effects of using integrator: 1. Pre-emphasize the low-frequency content 2. Increase correlation between adjacent samples (reduce the variance of the error signal at the quantizer input ) 3. Simplify receiver design Because the transmitter has an integrator , the receiver consists simply of a low-pass filter. (The accumulator in the conventional DM receiver is cancelled by the differentiator )

113. V. S. Hendre Department of E&TC, TCOER, Pune 101 F>nyquist rate 1 Product modulator output +1 -ve i/p +ve i/p -1 Two equivalent versions of delta-sigma modulation system.

114. A single period of the trigonometric sine function, sampled 100 times and represented as a PDM bitstream, is:0101011011110111111111111111111111011111101101101010100100100000010000000000000000000001000010010101 V. S. Hendre Department of E&TC, TCOER, Pune 102

115. V. S. Hendre Department of E&TC, TCOER, Pune 103

116. Applications V. S. Hendre Department of E&TC, TCOER, Pune 104 Data conversion systems Frequency Synthesizers SMPS motor controls Sony’s Super Audio CD (SACD) format

119. -Difference in adjucent samples (present & previous)-encoded-transmitted

121. V. S. Hendre Department of E&TC, TCOER, Pune 107 prediction Unquanitsedi/p signal Quantisation error Quantised version of signal Original sample value Quantisation error (+/-)

122. DPCM receiver V. S. Hendre Department of E&TC, TCOER, Pune 108

123. Comparision V. S. Hendre Department of E&TC, TCOER, Pune 109

124. PCM with Noise V. S. Hendre Department of E&TC, TCOER, Pune 110 The reconstructed message contains two types of noise: 1) Quantization Noise 2) Decoding Noise Decoding Noise: Random Noise added to PCM signal at the receiver causes regeneration errors that appears as erroneous digits in the codeword is called as decoding noise. Expression for Decoding Noise power: Let us consider a binary PCM with uniform quantization. Let ‘v’ no of bits/samples & PCM is having very small bit error probability ‘Pe’. Bit error probability: probability of a particular bit in error When Pe << 1, The Prob. of one error in given word is P= v Pe --------- (i)

125. PCM with Noise V. S. Hendre Department of E&TC, TCOER, Pune 111 If we consider that the PCM word bits are given by bv-1, bv-2,……………….. b1, b0, If there is error in mth bit , the decoded codeword is shifted by ±2m For Ex. The transmitted codeword: 00001000 & error occurred at bo bit Received codeword is :00001001 Decoded codeword is shifted by ±20 = ± i.e. by one step. Thus Error in mth bit is given by: m=±2m ……………..(ii)

126. PCM with Noise The random bit error can be obtained by mean square value

127. PCM with Noise 113

128. DM with Noise 114

129. Linear Predictive Coding (LPC) V. S. Hendre Department of E&TC, TCOER, Pune 115 Digital encoding technique-different approach Uses-transversal filter + auxillary components (to synthesize the waveform) Transversal filter-one of the convenient & flexible device used for equalisation Square up corners-for small amplitude higher harmonics (Equalisation N/W:- Cures-linear distortion –amplitude & delay)

130. V. S. Hendre Department of E&TC, TCOER, Pune 116 LPC Transmitter LPC receiver Decoder Encoder

131. Speech model V. S. Hendre Department of E&TC, TCOER, Pune 117 Frequency generators -electrical equivalent to generate sound Because -wide frequency spectrum

132. V. S. Hendre Department of E&TC, TCOER, Pune 118 Complete LPC codeword-@80 bits/sample -1 bit used to switch-voice/unvoiced -6 bits-pitch freqn of voice -few bits-represents error Sampling rate-40-100 Hz, Bit rate=3-8 Kbps

133. Digital Audio Recording V. S. Hendre Department of E&TC, TCOER, Pune 119 Disadvantages of analog audio storage 1)Wear & tear due to constant use & mechanical contacts with magnetic tape 2)Tapes stretch out & produces flutter 3)Dynamic range is limited @ 70dB (range required 100 dB to 120 dB) 4)Soft music is lost & loud music saturates the amplifier

134. Advantages of CD Technology V. S. Hendre Department of E&TC, TCOER, Pune 120 Digital recording Uses plastic disk @ 120mm diameter 20,000 tracks & width of each track-@0.5m Spacing between adjucent tracks-1.6 m Each track has-microscopic PITS -Lands:regions betn the PITS

135. V. S. Hendre Department of E&TC, TCOER, Pune 121 PITS & LANDS pattern on CD Electrical signal

136. V. S. Hendre Department of E&TC, TCOER, Pune 122

137. V. S. Hendre Department of E&TC, TCOER, Pune 123 CD AUDIO RECORDING

138. CD Playback V. S. Hendre Department of E&TC, TCOER, Pune 124 Job:1) Extract framing & sync. Information 2) Extract merging bits 3) Decode EFM signal 4) Extract control word bits Sampling rate conversion -Random error-air bubbles, PIT inaccuracies -Burst error-scratches, fingerprints 16 bits DAC -expensive -Incorrect bit-changed to opposite state -If not possible-incorrect value-cancelled, its value-interpolated between neighbeuring samples

139. ITU-Voice encoding & multimedia standards V. S. Hendre Department of E&TC, TCOER, Pune 125 International Telecommunication Union (united nations) Publishes: Telecommunication Technology :regulatory & standards information ITU standards 1) ITU-T: Telecommunication standardization sector :developes recommendations for wireless n/w. 2)ITU-R: Radio communication standardization sector :develops recommendations for wireless communications 3)ITU-D:standards for developing nations

140. Voice encoding standards V. S. Hendre Department of E&TC, TCOER, Pune 126

141. Multimedia /Multiplexing standards: video, data, multiplexing, signalling & encryption V. S. Hendre Department of E&TC, TCOER, Pune 127