Avanthi Institute of Engineering & Technology: Key Paper

AVANTHI INSTITUTE OF ENGINEERING & TECHNOLOGY
Gunthapally (Vi), Hayathnagar (M) R.R. Dist., A.P.

DEPARTMENT OF ELECTRONICS & COMMUNIUCATION ENGINEERING
Department : E.C.E Academic Year : 2012-13
Course Name : BTech ECE Course Code :56027
Name of the Faculty: S.John Babu Designation :Asst Prof
Class :IIIyear IIsem ECE Document :2nd mid Key Paper
B.Tech III Year II Sem Course File

1.a) A Difference equation describing a filter is given below y(n)-3/4y(n-1)+y(n-2)=x(n)+1/2x(n-1)
Draw direct form-I and Direct form II structures ?

b) H(Z)=
2 1
1
15 . 0 2 . 0 1
1

z z
z
draw the cascade and parallel realization?


2. a) What is sub band coding? How is it achieved with the help of multirate Dsp?
In signal processing, Sub-band coding (SBC) is any form of transform coding that breaks a
signal into a number of different frequency bands and encodes each one independently. This
decomposition is often the first step in data compression for audio and video signals.


To enable higher quality compression, one may use subband coding. First, a digital filter bank
divides the input signal spectrum into some number (e.g., 32) of subbands. The psychoacoustic model
looks at the energy in each of these subbands, as well as in the original signal, and computes masking
thresholds using psychoacoustic information. Each of the subband samples is quantized and encoded so
as to keep the quantization noise below the dynamically computed masking threshold. The final step is
to format all these quantized samples into groups of data called frames, to facilitate eventual playback by
a decoder.Decoding is much easier than encoding, since no psychoacoustic model is involved. The
frames are unpacked, subband samples are decoded, and a frequency-time mapping reconstructs an
output audio signal.

b) Derive the relationship b/w the spectrums of input and output signals in Interpolation?

Taking the DFT of N samples of zero-inserted signal we have Note from this equation that within a
frequency interval of to 2 or (0 to FS Hz) there are L repetition of the spectrum of X(k), this is because
stretching the signal by a factor of L shrinks its spectrum by a factor of 1/L.


3. a) Design IIR filter by using Impulse Invariance method?
The Impulse Invariant Method

In the impulse invariant method, the impulse response of the digital filter, h[ n] , is made
(approximately) equal to the impulse response of an analog
filter, h
c
(t ) , evaluated at t nT
d
, where T
d
is an (abitrary) sampling period. Specifically
h[n] = T
d
h
c
( nT
d
)

| e 2t k |

H (e
je
) = H
c

j + j

|
T
d

k =\
T
d.
and aliasing would occur if
H

c
(
j
O)is not bandlimited to t / T
d
(in rad/s).

If
H

c
(
j
O)is bandlimited to t / T
d
, then

H (e
je
) = H
c
( je / T
d
)
.

In this case, it is straight forward to specify the prototype analog filter.
However, all the commonly used prototype analog filters used in the impulse invariant design
method are indeed non-band limited. So there is aliasing. However, the aliasing can be minimized if
we over-design the analog filter (especially in the stop band). The picture below illustrates the design
procedure. We first specify the digital filter as shown in the first diagram. Then we map the digital
frequency e on to the analog frequency O = e / T
d
and make
H
c
( jO) = H (e
j

OT
d
)=
Notice from the diagram that we can only control
the magnitude of the responses because of the nature of the analog filters used.


H (e
je
)

1

1o
1

o
e
p
e
s

t

Digital filter specifications
b) If Ha(s)=
) 2 )( 1 (
1
+ s s
, find the Co-responding H(z) using Impulse Invariance method
for sampling frequency of 5 sam/sec? ?


4. a) design IIR filter by using Bilinear transformation and explain about frequency
warping?
The Bilinear Transformation Method
In the impulse invariant method, aliasing occurs when the prototype analog filter is transformed
back into the digital filter. To reduce the distortion introduced by aliasing, we start off by
tightening the specifications on the digital filter. This is somewhat cumbersome and may lead to
several iterations before the optimal filter is found.
Aliasing occurs because points in theO axis separated by 2t / T
d
are
mapped into the same digital frequency e . In the Bilinear transformation
method, there is a one-to-one correspondence betweene and t . So aliasing is avoided in
transforming the prototype analog filter back into the digital filter.


b) Design IIR Filter by using step invariance method?
Infinite impulse response (IIR) is a property of signal processing systems. Systems with this
property are known as IIR systems or, when dealing with filter systems, as IIR filters. IIR
systems have an impulse response function that is non-zero over an infinite length of time. This
is in contrast to finite impulse response (FIR) filters, which have fixed-duration impulse
responses. The simplest analog IIR filter is an RC filter made up of a single resistor (R) feeding
into a node shared with a single capacitor (C). This filter has an exponential impulse response
characterized by an RC time constant. Because the exponential function is asymptotic to a limit,
and thus never settles at a fixed value, the response is considered infinite.

Digital filters are often described and implemented in terms of the difference equation that
defines how the output signal is related to the input signal:

where:
- is the feedforward filter order
- are the feedforward filter coefficients
- is the feedback filter order
- are the feedback filter coefficients
- is the input signal
- is the output signal.
A more condensed form of the difference equation is:

which, when rearranged, becomes:

To find the transfer function of the filter, we first take the Z-transform of each side of the above
equation, where we use the time-shift property to obtain:

We define the transfer function to be:

Considering that in most IIR filter designs coefficient is 1, the IIR filter transfer function
takes the more traditional form:

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AVANTHI INSTITUTE OF ENGINEERING & TECHNOLOGY

Gunthapally (Vi), Hayathnagar (M) R.R. Dist., A.P.

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