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INTRODUCTION
DIGITAL TRANSMISSION is the transmittal of digital signals between two or more points in a communications
system.
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.
Transmission of digitally encoded analog signal requires significantly more bandwidth than simply transmitting
the original analog signal.
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.
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.
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
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
𝑉𝑚𝑎𝑥
𝐷𝑅 = 20𝑙𝑜𝑔 𝑉𝑚𝑖𝑛
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
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.
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.
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)
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
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.
10-10. VOCODERS
When digitizing speech signals only, special voice encoders/decoders called VOCODERS are often used.
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.
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.
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
REYES, REISHEL
CODE : N160