10-1.

INTRODUCTION
 DIGITAL TRANSMISSION is the transmittal of digital signals between two or more points in a communications
system.

10-1-1. ADVANTAGES OF DIGITAL TRANSMISSION

 The primary advantage of digital transmission over analog transmission is NOISE IMMUNITY.
 Digital signals are also better suited than analog signals for processing and combining using a technique called
MULTIPLEXING.
 DIGITAL SIGNAL PROCESSING (DSP) is the processing of analog signals using digital methods and includes
bandlimiting the signal with filters, amplitude equalization and phase shifting.
 Digital transmission systems are more resistant to analog systems to additive noise because they use signal
REGENERATION rather than signal amplification.

10-1-2. DISADVANTAGES OF DIGITAL TRANSMISSION

 Transmission of digitally encoded analog signal requires significantly more bandwidth than simply transmitting
the original analog signal.

10-2. PULSE MODULATION

 PULSE MODULATION consists essentially of sampling analog information signals and then converting those
samples into discrete pulses and transporting the pulses from a source to a destination over a physical
transmission medium.
 The four predominant methods of pulse modulation include PULSE WIDTH MODULATION (PWM), PULSE
POSITION MODULATION (PPM), PULSE AMPLITUDE MODULATION (PAM) and PULSE CODE MODULATION (PCM).
 PWM is sometimes called PULSE DURATION MODULATION (PDM) or PULSE LENGTH MODULATION (PLM), as the
width (active portion of the duty cycle) of a constant amplitude pulse is varied proportional to the amplitude of
the analog signal at the time the signal is sampled.
 With PPM, the position of a constant-width pulse within a prescribed time slot is varied according to the
amplitude of the sample of the analog signal.
 With PAM, the amplitude of a constant-width, constant-position pulse is varied according to the amplitude of
the sample of the analog signal.
 With PCM, the analog signal is sampled and then converted to a serial n-bit binary code for transmission.

10-3. PCM
 PCM is the only digitally encoded modulation technique that is commonly used for digital transmission.
 The term PULSE CODE MODULATION is somewhat of a misnomer, as it is not really a type of modulation but
rather a form of digitally coding analog signals.
 FOR THE SIMPLIFIED BLOCK DIAGRAM OF A SINGLE-CHANNEL, SIMPLEX PCM TRANSISSION SYSTEM
 SAMPLE-AND-HOLD CIRCUIT periodically samples the analog input signal and converts those samples to
a multilevel PAM signal.
 ANALOG-TO-DIGITAL CONVERTER (ADC) converts the PAM samples to parallel PCM codes.
 REPEATERS are placed at prescribed distances to regenerate the digital pulses.
 SERIAL-TO-PARALLEL CONVERTER converts serial pulses received from the transmission line to parallel
PCM codes.
 DIGITAL-TO-ANALOG CONVERTER (DAC) converts the parallel PCM codes to multilevel PAM signals.
 CODER/DECODER performs the PCM encoding and decoding functions.
10-4. PCM SAMPLING
 NATURAL SAMPLING is when tops of the sample pulses retain their natural shape during the sample interval,
making it difficult for an ADC to convert the sample to a PCM code.
 FLAT-TOP SAMPLING is the most common method used for sampling voice signals in PCM systems, which is
accomplished in a sample-and-hold circuit.
 The sampling process alters the frequency spectrum and introduces an error called APERTURE ERROR.

10-4-1. SAMPLING RATE

 The NYQUIST SAMPLING THEOREM establishes the minimum sampling rate (fs) that can be used for a given PCM
system.
 The MINIMUM SAMPLING RATE (fs) is equal to twice the highest audio input frequency.

 If fs is less than two times fa, an impairment called ALLIAS or FOLDOVER DISTORTION occurs.

10-4-2. QUANTIZATION AND THE FOLDED BINARY CODE

 QUANTIZATION is the process of converting an infinite number of possibilities to a finite number of conditions.
 FOLDED BINARY CODE is a type of code where the codes on a bottom half of the table are a mirror image of the
codes on the top half, except for the sign bit.
 QUANTIZATION INTERVAL or QUANTUM is the magnitude difference between adjacent steps.
 If the magnitude of the sample exceeds the highest quantization interval, OVERLOAD DISTORTION (also called
PEAR LIMITING) occurs.
 The magnitude of a quantum is also called RESOLUTION.
 QUANTIZATION ERROR (Qe)/ QUANTIZATION NOISE (Qn) is any round-off errors in the transmitted signal are
reproduced when the code is converted back to analog in the receiver.
 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.

𝑠𝑎𝑚𝑝𝑙𝑒 𝑣𝑜𝑙𝑡𝑎𝑔𝑒
𝑟𝑒𝑠𝑜𝑙𝑢𝑡𝑖𝑜𝑛

10-4-3. DYNAMIC RANGE

 DYNAMIC RANGE(DR) is the ratio of 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.

𝑉𝑚𝑎𝑥
𝐷𝑅 = 𝑉𝑚𝑖𝑛

 Dynamic range is generally expressed as a dB value

𝑉𝑚𝑎𝑥
𝐷𝑅 = 20𝑙𝑜𝑔 𝑉𝑚𝑖𝑛

10-4-4. CODING EFFICIENCY

 CODING EFFICIENCY is a numerical indication of how efficiently a PCM code is utilized.

 CODDING EFFICIENCY is the ratio of the minimum number of bits required to achieved a certain dynamic range
to the actual number of PCM bits used.
 𝑚𝑖𝑛𝑖𝑚𝑢𝑚 𝑛𝑢𝑚𝑏𝑒𝑟 𝑜𝑓 𝑏𝑖𝑡𝑠(𝑖𝑛𝑐𝑙𝑢𝑑𝑖𝑛𝑔 𝑠𝑖𝑔𝑛 𝑏𝑖𝑡)
𝑐𝑜𝑑𝑖𝑛𝑔 𝑒𝑓𝑓𝑖𝑐𝑖𝑒𝑛𝑐𝑦 = 𝑎𝑐𝑡𝑢𝑎𝑙 𝑛𝑢𝑚𝑏𝑒𝑟 𝑜𝑓 𝑏𝑖𝑡𝑠(𝑖𝑛𝑐𝑙𝑢𝑑𝑖𝑛𝑔 𝑠𝑖𝑔𝑛 𝑏𝑖𝑡)

10-5. SIGNAL-TO-QUANTIZATION NOISE RATIO

 The worst possible SIGNAL VOLTAGE-TO-QUANTIZATION NOISE VOLTAGE RATIO (SQR) occurs when the input
signal is at its minimum amplitude (101 or 001).

𝑟𝑒𝑠𝑜𝑙𝑢𝑡𝑖𝑜𝑛 𝑉𝑙𝑠𝑏
𝑆𝑄𝑅 = = =2
𝑄𝑒 𝑉𝑙𝑠𝑏/2

 For linear PCM codes (all quantization intervals have equal magnitudes), the SIGNAL POWER-TO-QUANTIZING
NOISE POWER RATIO (also called SIGNAL-TO-DISTORTION RATIO or SIGNAL-TO-NOISE RATIO) is determined by

𝑉 2 /𝑅
𝑆𝑄𝑅(𝑑𝐵) = 10𝑙𝑜𝑔 𝑞2
( )/𝑅
𝑅

10-6. LINEAR VERSUS NONLINEAR PCM CODES

 With LINEAR CODING, the accuracy (resolution) for the higher-amplitude analog signals is the same as for the
lower-amplitude signals, and the SQR for the lower-amplitude signals is less than for the higher-amplitude
signals.
 With NONLINEAR ENCODING (NONUNIFORM ENCODING), the step size increases with the amplitude of the input
signal.

10-7. IDDLE CHANNEL NOISE

 During times when there is no analog input signal, the only input to the PAM sampler is random, thermal noise
also called as IDLE CHANNEL NOISE, that is converted to a PAM sample just as if it were a signal.
 MIDTREAD QUANTIZATION is a way to reduce idle channel noise wherein the first quantization interval is made
larger in amplitude than the rest of the steps.
 The lowest-magnitude positive and negative codes have the same voltage range as all the other codes ( + or -
one half the resolution) is called MIDRISE QUANTIZATION.

10-8. CODING METHODS

10-8-1. LEVEL-AT-A TIME CODING

 LEVEL-AT-A TIME CODING compares the PAM signal to a ramp waveform while a binary counter is being
advanced at a uniform rate.
 LEVEL-AT-A TIME CODING is generally limited to low-speed application.

10-8-2. DIGIT-AT-A TIME CODING

 This type of coding determines each digit of the PCM code sequentially.
 DIGIT-AT-A TIME CODING is analogous to a balance where known reference weights are used to determine an
unknown weight.
 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 for high
speed application.

10-9. COMPANDING
 COMPANDING is the process of compressing and then expanding.

10-9-1. ANALOG COMPANDING

 μ LAW COMPANDING. In the United States and Japan, μ law companding is used. The compression
characteristics for μ law is
𝑉𝑚𝑎𝑥 ln(1 + 𝜇 𝑉𝑖𝑛⁄𝑉𝑚𝑎𝑥)
𝑉𝑜𝑢𝑡 =
ln(1 + 𝜇)
where: Vmax = maximum uncompressed analog input amplitude (volts)

Vin = amplitude of the input signal at a particular instant of time (volts)

μ = parameter used to define the amount of compression (unitless)

Vout = compressed output amplitude (volts)

 A-LAW COMPANDING. In Europe, the ITU-T has established A- LAW COMPANDING to be used to
approximate true logarithmic companding.

𝐴𝑉𝑖𝑛
𝑉𝑚𝑎𝑥
𝑉𝑜𝑢𝑡 = 𝑉𝑚𝑎𝑥 0 ≤ 𝑉𝑖𝑛/𝑉𝑚𝑎𝑥 ≤ 1/𝐴
1+𝑙𝑛𝐴
𝐴𝑉𝑖𝑛
1+ln( ) 1 𝑉𝑖𝑛
𝑉𝑚𝑎𝑥
= 1+𝑙𝑛𝐴 𝐴
≤ 𝑉𝑚𝑎𝑥 ≤ 1

10-9-2. DIGITAL COMPANDING

 DIGITAL COMPANDING involves compression in the transmitter after the input sample has been converted
to a linear PCM code and then expansion in the receiver prior to PCM decoding.

10-9-3. DIGITAL COMPRESSION ERROR

12 𝑏𝑖𝑡 𝑒𝑛𝑐𝑜𝑑𝑒𝑑 𝑣𝑜𝑙𝑡𝑎𝑔𝑒 − 12 𝑏𝑖𝑡 𝑑𝑒𝑐𝑜𝑑𝑒𝑑 𝑣𝑜𝑙𝑡𝑎𝑔𝑒
% 𝑒𝑟𝑟𝑜𝑟 = × 100
12 𝑏𝑖𝑡 𝑑𝑒𝑐𝑜𝑑𝑒𝑑 𝑣𝑜𝑙𝑡𝑎𝑔𝑒

10-10. VOCODERS
 When digitizing speech signals only, special voice encoders/decoders called VOCODERS are often used.

10-10-1. CHANNEL VOCODERS

 Digital CHANNEL VOCODERS use bandpass filters to separate the speech waveform into narrower sub-bands.
 10-10-2. FORMANT VOCODERS

 The spectral power of most speech energy concentrates at three or four peak frequencies called
FORMANTS.
 A FORMANT VOCODER simply determines the location of these peaks and encodes and transmits only the
information with the most significant short-term components.

10-10-3. LINEAR PREDICTIVE CODERS

 A LINEAR PREDICTIVE CODER extracts the most significant portions of speech information directly from
the time waveform rather than from the frequency spectrum as with the channel and formant vocoders.

10-11. PCM LINE SPEED

 LINE SPEED is simply the data rate at which serial PCM encoder onto the transmission line.
𝑠𝑎𝑚𝑝𝑙𝑒𝑠 𝑏𝑖𝑡𝑠
𝑙𝑖𝑛𝑒 𝑠𝑝𝑒𝑒𝑑 = ×
𝑠𝑒𝑐𝑜𝑛𝑑 𝑠𝑎𝑚𝑝𝑙𝑒
where: line speed = the transmission rate in bits per second

samples/second = sample rate (fs)

bits/sample = number of bits in the compressed PCM code

10-12. DELTA MODULATION PCM

 DELTA MODULATION uses a single-bit PCM code to achieve digital transmission of analog signals.
 The slope of the analog signal is greater than the delta modulator can maintain and is called SLOPE
OVERLOAD.
 It can be seen that when the original analog input signal has a relatively constant amplitude, the
reconstructed signal has variations that were not present in the original signal and is called GRANULAR
NOISE.

10-13. ADAPTIVE DELTA MODULATION PCM

 ADAPTIVE DELTA MODULATION is a delta modulation system where the step size of the DAC is automatically
varied, depending on the amplitude characteristics of the analog input signal.

10-14. DIFFERENTIAL PCM

 With DPCM, the difference in the amplitude of two successive samples is transmitted rather than the actual
sample.

10-15. PULSE TRANSMISSION

 The secondary lobes are called RINGING TAILS.
 The output frequency spectrum corresponding to a rectangular pulse is referred to as (sin x)/x response and is
given as
𝜔𝑇
sin( 2 )
𝑓(𝜔) = (𝑇)
𝜔𝑇/2
where: R = 2πf (radians)

T = pulse width (seconds)

 Nyquist rate is equal to twice the bandwidth.
𝑅 = 2𝐵
where: R = signaling rate = 1/T
B = specified bandwidth
10-15-1. INTERSYMBOL INTERFERENCE
 INTERSYMBOL INTERFERENCE (ISI) causes crosstalk between channels that occupy adjacent time slots in a time-
division-multiplexed carrier system.
 Special filters called EQUALIZERS are inserted in the transmission path to “equalize” the distortion for all
frequencies, creating uniform transmission medium reducing transmission impairments.
10-15-2. EYE PATTERNS
 An EYE PATTERN is a convenient technique for determining the effects of the degradations introduced into the
pulses as they travel to the regenerator.
 The EYE OPENING is a function of a number of code levels and the intersymbol interference caused by the
ringing tails of any preceding or succeeding pulses.
 ISI degradation is
ℎ
𝐼𝑆𝐼 = 20𝑙𝑜𝑔
𝐻
where: H = ideal vertical opening (cm)

h = degraded vertical opening (cm)

 The JITTER(DATA TRANSITION JITTER) has an effect on the symbol timing (clock) recovery circuit and, if
excessive, may significantly degrade the performance of cascaded regenerative sections.
 CHAPTER 10
DIGITAL TRANSMISSION

NAMES : RAMIREZ, RICKCEL

REYES, REISHEL

CODE : N160

DATE : MARCH 2018