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CHAPTER - 6

CHAPTER - 6. Digital Communication. Part -1 Introduction to Digital Modulation Sampling theorem Pulse Analog Modulation Delta Modulation. Prepared by Dr M.Murugappan. Introduction to Digital Modulation Sampling theorem Pulse Analog Modulation Pulse Amplitude Modulation

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CHAPTER - 6

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  1. CHAPTER - 6 Digital Communication • Part -1 • Introduction to Digital Modulation • Sampling theorem • Pulse Analog Modulation • Delta Modulation Prepared by Dr M.Murugappan

  2. Introduction to Digital Modulation Sampling theorem Pulse Analog Modulation • Pulse Amplitude Modulation • Pulse Width Modulation • Pulse Position Modulation • Pulse Code Modulation Delta Modulation Line coding Amplitude Shift Keying (ASK) Frequency Shift Keying (FSK) Time-Division Multiplexing (TDM) Frequency Division Multiplexing (FDM) Topics Covered in this chapter

  3. Introduction • Transfer of information in digital form • Digital Communication Vs Data Communication • In the early 90’s, telecommunication networks is changing towards digital world. • With the rapid advancement in the field of microprocessor, several telecommunication components such as transmission line and switching have used digital signals in their operation. • Therefore, information signals must be changed to digital form so that it can be transmitted through this network. • Several techniques requiring full coding of the original signal will be used: - • Pulse Code Modulation (PCM) • Delta Modulation (DM)

  4. Digital Modulation • Advantages : • Immunity to noise • Easy storage and processing: • Easy to measure • Regeneration - Ability to recover the noise corrupted signal to the original value • Enables encryption • Data from several sources can be integrated and transmitted using the same digital communication system • Error correction detection can be utilized • Disadvantages : • Requires a bigger bandwidth • Analog signal need to be changed to digital first • Not compatible to analog system • Need synchronization MP, DSP, RAM, ROM, Computer Voice : Analog : 4 kHz Digit : 2 x 4 kHz x 8 bit = 64 kb/s BWmin 32 kHz

  5. A process of taking samples of information signal at a rate of Nyquist’s sampling frequency. Nyquist’s Sampling Theorem : Sampling The original information signal can be reconstructed at the receiver with minimal distortion if the sampling rate in the pulse modulation system equal to or greater than twice the maximum information signal frequency. fs >= 2fm (max)

  6. Two basic techniques used to perform the sampling function: • Natural sampling • Flat-top sampling Types of Sampling

  7. Tops of the sample pulses retain their natural shape during the sample interval. Frequency spectrum of the sampled output is different from an ideal sample. Amplitude of frequency components produced from narrow, finite-width sample pulses decreases for the higher harmonics • Requiring the use of frequency equalizers Natural Sampling

  8. Natural Sampling

  9. Common used in PCM systems. Accomplish in a sample-and-hold circuit • To periodically sample the continually changing analog input voltage & convert to a series of constant-amplitude PAM voltage levels. The input voltage is sampled with a narrow pulse and then held relatively constant until the next sample is taken. Flat-top Sampling

  10. Sampling process alters the frequency spectrum & introduces aperture error. The amplitude of the sampled signal changes during the sample pulse time. Advantages: • Introduces less aperture distortion • Can operate with a slower ADC Flat-top Sampling

  11. Flat-top Sampling

  12. m(t) ms(t) X m(t) ms(t) t t s(t) Digital signal s(t) t Ts where Sampling Theorem (1/3) multiplier Fourier series for impulse train : Nyquist theorem states that: Therefore :

  13. s(t) A t Ts Sampling Theorem (2/3) • Sampling process shown previously uses an ideal pulse signal • However, it is quite difficult to generate an ideal pulse signal practically • The usual pulse signal generated is as shown below:  - pulse width Ts – pulse period cos where

  14. (a) Sampling frequency=> fs1 < 2fm (max) ms(f) Aliasing f fm fs1 2fs1 3fs1 ms(f) (b) Sampling frequency=> fs2 > 2fm (max) f fm fs2 2fs2 3fs2 Sampling Theorem (3/3) The choice of sampling frequency, fs must follow the sampling theorem to overcome the problem of aliasing and loss of information • Shannon sampling theorem=> fs  2fm • Nyquist frequency • fs = 2fm= fN A band limited signal that has a maximum frequency, fmax can be regenerated from the sampled signal if it is sampled at a rate of at least 2fmax .

  15. Sampling analog information signal Converting samples into discrete pulses Transport the pulses from source to destination over physical transmission medium. Pulse Modulation (1/2)

  16. Four (4) Methods 1. PAM 2. PWM 3. PPM 4. PCM Pulse Modulation (2/2) Analog Pulse Modulation Digital Pulse Modulation

  17. Four (4) Methods 1. PAM 2. PWM 3. PPM 4. PCM Pulse Modulation (2/2) Analog Pulse Modulation Digital Pulse Modulation

  18. Basic scheme of PCM system • Quantization • Quantization Error • Companding • Block diagram & function of TDM-PCM communication system Pulse Code Modulation (PCM)

  19. The most common technique for using digital signals to encode analog data is PCM. Example:To transfer analog voice signals off a local loop to digital end office within the phone system, one uses a codec. Basic scheme of PCM system

  20. Because voice data limited to frequencies below 4000 Hz, a codec makes 8000 samples/sec. (i.e., 125 microsecond/sample). If a signal is sampled at regular intervals at a rate higher than twice the highest signal frequency, the samples contain all the information of the original signal. Definition: PCM is essentially analog-to-digital conversion of a special type where the information contained in the instantaneous samples of an analog signal is represented by digital words in a serial bit stream. Basic scheme of PCM system

  21. PCM Block Diagram PAM PCM • Most common form of analog to digital modulation • Four step process • Analog Signal is sampled • Sampled signal is quantized • The quantized values are encodeda as stream of bits (binary) • Signal is digitally encoded for transmission (Encoded)

  22. Pulse Code Modulation

  23. Pulse Code Modulation

  24. Analog signal is sampled. • Converted to discrete-time continuous-amplitude signal (Pulse Amplitude Modulation) Pulses are quantizedand assigned a digital value. • A 7-bit sample allows 128 quantizing levels. PCM uses non-linear encoding, i.e., amplitude spacing of levels is non-linear • There is a greater number of quantizing steps for low amplitude • This reduces overall signal distortion. This introduces quantizing error (or noise). PCM pulses are then encoded into a digital bit stream. 8000 samples/sec x 7 bits/sample = 56 Kbps for a single voice channel. Pulse Code Modulation (PCM)

  25. PCM Example

  26. Quantization • A process of converting an infinite number of possibilities to a finite number of conditions (rounding off the amplitudes of flat-top samples to a manageable number of levels).

  27. Quantization Analog input signal Sample pulse PAM signal PCM code

  28. Quantization (Example)

  29. A difference between the exact value of the analog signal & the nearest quantization level. Quantization Error

  30. Types of Quantization Midtread Midrise

  31. Types of Quantizer 1. Uniform type : The levels of the quantized amplitude are uniformly spaced. 2. Non-uniform type : The levels are not uniform.

  32. Resolution: Its defined as the ratio of maximum or full scale voltage to the quantization level. Dynamic Range:Its defined as the ratio of the maximum input or output voltage level to the smallest voltage level that can be quantized by the converters. Resolution & Dynamic Range • n = number of bits in the PCM code • n = number of bits in the PCM code

  33. Dynamic Range (DR) • DR is typically represented in terms of decibels (dB). For a binary system, each bit can have two logic levels, either a logical low or logical high. • Where • DR = absolute value of dynamic range • Vmax = the maximum voltage magnitude • Vmin = the quantum value (resolution) • n = number of bits in the PCM code

  34. Signal to Noise Ratio (SNR) • The SNR for a digitizing system is given as: • Where n= No of bits used for quantizing the signal • This relationship is based on the ratio of the RMS quantity of the maximum input to the RMS quantization noise. • Optimal Characteristics: (i) Quantization error to be reduced or (ii) input signal amplitude to be increased • Signal to Quantization Noise (SQR) is given as: • WhereL = No of Quantization levels = 2n n = No of bits used for sampling

  35. A CD audio laser disk system has a frequency bandwidth of 20 Hz to 20 kHz. What is the minimum sample rate required to satisfy the Nyquist Sampling rate? Example 1

  36. A CD audio laser disk system has a frequency bandwidth of 20 Hz to 20 kHz. What is the minimum sample rate required to satisfy the Nyquist Sampling rate? Solution: Example 1

  37. A digitizing system specifies 55 dB of dynamic range. How many bits are required to satisfy the dynamic range specification? What is the signal to noise ratio for the system? What is SQR for the system? Example 2

  38. A digitizing system specifies 55 dB of dynamic range. How many bits are required to satisfy the dynamic range specification? What is the signal to noise ratio for the system? What is SQR for the system? Solution: DR= 6.02 dB/bit or 6.02 X n n = DR/6.02 n= 55/6.02 = 9.136 10 bit 9 bit provide a DR of 54.18 dB. However, 10 bits will provide a DR of 60.2 dB (10 X 6.02) S/N = [1.76+6.02n] dB = [1.76+6.02(10)] dB S/N = 61.96 dB SQR (or) (S/N)q = 10 log 3L2 = 10 log 3(2n)2 (S/N)q = 10 log 3(210)2 = 64.97 dB Example 2

  39. Quantization levels not evenly spaced Reduces overall signal distortion Can also be done by Companding Non-Linear Coding

  40. Companding • The process of compressing and then expanding. • The higher amplitude analog signals are compressed • prior to transmission and then expanded in receiver. • Improving the DR of a communication system.

  41. Companding Functions

  42. Methods of Companding • For the compression, two laws are adopted: the -law in US and Japan and the A-law in Europe. • -law • A-law • The typical values used in practice are: =255 and A=87.6. • After quantization the different quantized levels have to be represented in a form suitable for transmission. This is done via an encoding process. Vmax= Max uncompressed analog input voltage Vin= amplitude of the input signal at a particular of instant time Vout= compressed output amplitude A, = parameter define the amount of compression

  43. Methods of Companding A-law μ-law

  44. PCM LINE SPEED • The data rate at which serial PCM bits are clocked out of the PCM encoder onto the transmission line. • Where • Line speed = the transmission rate in bits per second • Sample/second = sample rate, fs • Bits/sample = no of bits in the compressed PCM code

  45. The most important advantages of PCM are: • Robustness to channel noise and interference. • Efficient regeneration of the coded signal along the channel path. • Efficient exchange between Bt and SNR. • Uniform format for different kind of base-band signals. Advantages of PCM

  46. DELTA MODULATION (DM)

  47. Pulse signal s(t) comparator d(t) e(t) ∑ X m(t) xDM(t) + +Δ - -Δ integrator Delta Modulation (1/3) Analog signal • There are 2 main components in the DM generator circuit, i.e comparator and integrator.

  48. Delta Modulation (2/3) • Comparator will compare the error signal e(t), where e(t) = m(t) - • Output signal from comparator has the following function: • The output from the comparator will be sampled with a pulse signal at a rate of 1/Ts. (follow Pulse signal s(t)) • Next, DM signal will be generated with the equation below: • The DM signal will be feed back, but before that this signal will be integrated first (dirac delta) • This signal (information signal) will determine the error value e(t).

  49. Effects of steep slope Δ Ts t 0001010111111101100010000000 Delta Modulation (3/3) If e(t) < 0 or -∆ , it will be coded as 0 If e(t) > 0 or +∆, it will be coded as 1 A steep slope results in noise in DM signal. To avoid this from happening, it has to follow the following condition:

  50. Its also called “Slope Modulation” or “Tracking ADC”. • Because, the output of DM is follows the contour changes of the input waveform. Simple modulation technique compared to PCM • No parallel to serial converters are required Highly suitable for voice communication Dis-advantage: • Not suitable for “slope overload” – High rate of change of input signal • Continuous variable slope delta (CVSD) modulation Delta Modulation

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