0% found this document useful (0 votes)
113 views

Waveform Coding Techniques: 1) Pulse Code Modulation or PCM

The document discusses various waveform coding techniques including PCM, DPCM, and DM. [1] PCM directly samples and quantizes an analog signal into digital form but requires high bandwidth. [2] DPCM and DM reduce bandwidth by encoding differences between predicted and actual signal values, with DM using only 1 bit per sample. [3] Choosing the right technique involves tradeoffs between complexity, bandwidth, and distortion.

Uploaded by

anu
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
0% found this document useful (0 votes)
113 views

Waveform Coding Techniques: 1) Pulse Code Modulation or PCM

The document discusses various waveform coding techniques including PCM, DPCM, and DM. [1] PCM directly samples and quantizes an analog signal into digital form but requires high bandwidth. [2] DPCM and DM reduce bandwidth by encoding differences between predicted and actual signal values, with DM using only 1 bit per sample. [3] Choosing the right technique involves tradeoffs between complexity, bandwidth, and distortion.

Uploaded by

anu
Copyright
© © All Rights Reserved
Available Formats
Download as PDF, TXT or read online on Scribd
You are on page 1/ 35

Waveform Coding Techniques

1) Pulse Code Modulation or PCM


• Standard method for conversion of speech and video signals to digital form
• Robust to noise but demanding in bandwidth and computational requirements
2) Differential Pulse Code Modulation or DPCM
• A method for reducing bandwidth
• Highly computationally complex
3) Delta Modulation or DM
• Simple to implement
• Requires high bandwidth
Now we are ready to describe PCM

Pulse Code Modulation (PCM)

PCM is a discrete time, discrete amplitude waveform encoding process by means of which
an analog signal is directly represented by a sequence of coded pulses
A/D Converter
The transmitter consists of – a ) Anti-aliasing LPF
b) Sampler
c) Analog-to-Digital (A/D) Converter

Quantizer + encoder
a) The anti-aliasing LPF will attenuate the high frequencies and bandlimit the signal
and thus prevent aliasing
b) The bandlimited signal is sampled at the sampler
The sampling frequency is chosen to be slightly higher than twice the maximum
frequency in the signal
c) Output of the sampler is then passed on to an analog to digital converter where the
discrete time continuous amplitude output of the sampler is quantized and encoded

• The most important feature of a PCM systems is its ability to control the effects of
distortion and noise produced by transmitting a PCM signal through the channel

• This capability is accomplished by reconstructing the PCM signal through a chain of


regenerative repeaters, located at sufficiently close spacing along the transmission
path.
Regenerative repeater combats the accumulation of noise and distortion by performing three
major functions: equalization, timing, and decision making
i) The equalizer compensates for the effects of channel non-idealities, mainly amplitude and
phase distortions
ii) From the received signal, the timing circuit determines instants of time where the signal-
to-noise ratio is maximum
iii) The decision device compares the equalized pulses received at optimum instants with a
predetermined threshold
Disadvantages of PCM

1. Much more complex compared to PPM, PAM and PDM since the message signal
undergoes a greater number of operations
2. PCM has a large bandwidth requirement

L – number of representation levels at the output of the quantizer


We know that L =2𝑅 where R is the number of bits per sample
Suppose the BW of the message signal be equal to W Hz , the sampling rate
1
𝑓𝑠 = = 2𝑊 Hz (minimum)
𝑇𝑠

Each sample is encoded using R bits


• The pulse transmission rate = 2RW / second , i.e., 2RW pulses are transmitted every
second
1
• Hence the duration of one pulse will be sec
2𝑅𝑊
• In the frequency domain, the bandwidth of the Sinc pulse will be equal to k2RW Hz
where k is a constant usually chosen to be 1 or 2
• As more quantization levels are used, the BW also increases
• BW of the PCM signal is much larger than that of the original message signal ( W Hz)

Variants of PCM , i.e. DM and DPCM are used to overcome these drawbacks of the PCM
Differential Pulse Code Modulation (DPCM)

• When a voice / video signal is sampled at a rate slightly higher than the Nyquist rate, the
sampled signal is found to exhibit high degree of correlation between adjacent samples.

• As a result, the values of near by samples will be close to each other

• In standard PCM systems, these highly correlated samples when encoded contains lot of
redundant information.

• Removing the redundancy before encoding, we can obtain an efficient coded signal, i.e. the
number of bits to be sent can be reduced → Basic Idea Behind DPCM
How does DPCM achieve this ?

➢ By predicting the message sample 𝑚[𝑛] and then ‘encoding the difference between
the actual value and the predicted value’ Hence the name Differential PCM

Prediction algorithms are available which can predict an estimate of a future value of the
message signal if it knows the past behavior up to that point in time

1
Assume 𝑚(𝑡) is the input message signal sampled at 𝑓𝑠 = to produce the sequence
𝑇𝑠
{𝑚[𝑛]} whose samples are 𝑇𝑠 seconds apart
• The quantizer in DPCM quantizes the difference between the message sample
and its predicted value
• The input to the quantizer is given by Prediction of input
sample 𝑚[𝑛]
Prediction error 𝑒[𝑛] = 𝑚 𝑛 − 𝑚
ෝ 𝑛
Delta Modulation (DM)

• Has lower system complexity compared to standard PCM


• Transmission bandwidth is traded off for reduced system complexity
• DM exploits bandwidth-complexity tradeoff

• In DM, the incoming message signal 𝑚(𝑡) is oversampled (at a rate higher than
Nyquist rate) so that adjacent samples of the signal would be highly correlated or will
have values that are close in range
• This is done to simplify the quantization process
• DM allows to quantize each sample by using only a single bit
• DM produces a staircase approximation to the original message signal
• The difference between the input and the approximation can take only two values ±∆
corresponding to positive and negative differences
• If the approximation falls below the signal at any sampling instant, it is increased by ∆ and if it
lies above the signal, it is decreased by ∆
• If the signal does not change too much from sample to sample, the staircase approximation
always lies within ±∆ of the input signal
Error signal 𝑒[𝑛] = 𝑚 𝑛 − 𝑚𝑞 𝑛 − 1
where 𝑚 𝑛 is the sample at 𝑛𝑇𝑠 , 𝑚𝑞 𝑛 − 1 is the latest approximation and 𝑒[𝑛] is the
error signal representing the difference between the two

+∆ 𝑖𝑓 𝑒 𝑛 > 0
Quantized version of 𝑒[𝑛] 𝑒𝑞 [𝑛] = ∆ sgn 𝑒 𝑛 =ቊ
−∆ 𝑖𝑓 𝑒 𝑛 < 0
Approximation at 𝑛𝑇𝑠 𝑚𝑞 [𝑛] = 𝑚𝑞 [𝑛 − 1] + 𝑒𝑞 [𝑛]

Approximation at (𝑛 − 1)𝑇𝑠

𝑚𝑞 (𝑡) is the staircase approximated signal


At the output of the encoder, for every +∆ , symbol 1 is sent out and for every −∆, symbol
0 is sent out
• Hence instead of quantizing the actual sample values, the difference between the
actual message sample and the staircase approximation is quantized
• Hence there are only two possible output representation levels, i.e. ±∆ which
requires only a one-bit quantizer

𝑒[𝑛] = 𝑚 𝑛 − 𝑚𝑞 𝑛 − 1
𝑒𝑞 [𝑛] = ∆ sgn 𝑒[𝑛]
𝑚𝑞 [𝑛] = 𝑚𝑞 [𝑛 − 1]+ 𝑒𝑞 [𝑛]
• Sampled message signal is applied to a modulator containing a comparator, quantizer

and accumulator

• 𝑧 −1 represents a unit delay, i.e., a delay equal to one sampling period

• One bit quantizer quantizes the error signal to ±∆

• Encoder encodes +∆ by symbol 1 and for every −∆ with symbol 0


𝑚𝑞 [𝑛] = 𝑚𝑞 [𝑛 − 1]+ 𝑒𝑞 [𝑛]
= 𝑚𝑞 [𝑛 − 2]+ 𝑒𝑞 [𝑛-1]+𝑒𝑞 [𝑛]
=σ𝑛𝑖=1 𝑒𝑞 [i]

• To reconstruct the approximated value 𝑚𝑞 [𝑛], the decoded signal (𝑒𝑞 [𝑛] ) has to be
continuously added

At the output of the accumulator , the staircase approximation 𝑚𝑞 (𝑡) is reconstructed and the
sharp edges are smoothened out by passing through a LPF
DM is a special case of DPCM with two important differences:
1. Use of one bit quantizer in DM
2. Replacing the prediction filter by a delay element
“ DM is the one-bit version of DPCM”
Delta modulation is subject to 2 types of Quantization errors:
1. Slope Overload Distortion
2. Granular Noise

1. Slope Overload Distortion

When the step-size ∆ is too small and the signal 𝑚(𝑡) changes too fast, i.e., rises
too fast or falls too fast, then the staircase approximation may fall short of the
original signal and may not catch up to it
For overcoming slope overload, the step size ∆ should be chosen such that
Maximum slope
Slope of
of message signal
staircase
approximation

Otherwise ∆ is too small for the approximation to follow a steep segment of the
input waveform 𝑚(𝑡) resulting in 𝑚𝑞 (𝑡) falling behind 𝑚 𝑡 . This condition is called
as slope overload distortion
2. Granular Noise

Occurs when step size ∆ is too large relative to the slope of the input message signal
𝑚(𝑡) , i.e. the signal is relatively flat
This causes the staircase approximation to hunt around the flat segment of the input
waveform
Small step size Slope overload error
Large step size Granular noise

• We need to choose a large enough step size to avoid slope overload distortion/error
and a small enough step size to avoid granular noise
• It requires a delta modulator that is “adaptive” in the sense that the step size is made
to vary/adapt in accordance to the input signal
• We go for Adaptive Delta Modulation (ADM)
• The one we studied just now is called as linear delta modulation to differentiate
between the two
Q)

A sinusoidal voice signal g(t)=cos(6000πt) is to be transmitted using either PCM or DM.


The sampling rate for PCM is 8kHz and for transmission with DM, the step size is decided
to be 31.25mV. Slope overload distortion is to be avoided in DM.

Assuming that the number of quantization levels for PCM is 64, determine the bit rate.

Which scheme is to be chosen for a bandwidth constrained application ? DM or PCM ?


Linear Prediction
A linear predictor is nothing but an FIR discrete time filter which consists of mainly 2
blocks:
• a set of p unit-delay elements, each of which is represented by 𝑧 −1
• a corresponding set of adders used to sum the scaled versions of the delayed
inputs, 𝑚𝑛−1 , 𝑚𝑛−2 , …, 𝑚𝑛−𝑝
• p is called as the prediction order
• The prediction error e[n] is defined as the difference between the actual sample
value m[n] and the predicted value 𝑚[𝑛]

𝑒 𝑛 = 𝑚 𝑛 − 𝑚[𝑛]

• The design objective is to choose the filter coefficients 𝑤1 , 𝑤2 , … 𝑤𝑝 so as to


minimize an index of performance J defined as the mean square error
Substituting for e[n] and expanding, we have

It is assumed that the input signal m(t) is a sample function of a stationary RP M(t)
with zero mean. Hence we get E[m[n]]=0
• Define
In order to find the filter coefficients which minimize this index, differentiate it wrt
the coefficients 𝑤𝑘 and equate to zero

Wiener-Hopf
equations for linear
prediction
These equations can be reformulated into matrix form as
• The diagonal elements of the autocorrelation matrix are equal to mean square
value ( since zero mean assumption, equal to variance)
• The matrix is symmetric about the main diagonal
• The optimal predictor coefficients can be found as

• The corresponding minimum mean square error can be found by substituting


the optimum coefficients into the mean square value equation

You might also like