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TRƯỜNG ĐẠI HỌC KHOA HỌC TỰ NHIÊN
KHOA ĐIỆN TỬ - VIỄN THÔNG

TOPIC 5:
Multirate Digital Signal Processing

Names of the members of group 7:

Đà m Tuấ n Khô i - 20207044
Phạ m Vă n Biên – 20207023
Đinh Bử u Hiền – 20207032
Trầ n Bá ch Cườ ng – 20207026

LECTURE IN CHARGE:
Trầ n Thị Thả o Nguyên

Hồ Chí Minh city - 2022

 TABLE OF CONTENTS

Chapter 1: An Introduction to Multirate Digital Signal Processing

 What is Multirate Digital Signal Processing?..............................3
 Why Multirate DSP?....................................................................3

Chapter 2: Decimation and Interpolation by an Intergral Factor

 Decimation……………………………………………………8
 Interpolation………………………………………………….16
Chapter 3: Applications of Multirate DSP……………………….22

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 INTRODUCTION

Multirate systems have gained popularity since the early 1980s and they are
commonly used for audio and video processing, communications systems,
and transform analysis to name but a few. In most applications multirate
systems are used to improve the performance, or for increased computational
efficiency. The two basic operations in a multirate system are decreasing
(decimation) and increasing (interpolation) the sampling-rate of a signal.
Multirate systems are sometimes used for sampling-rate conversion, which
involves both decimation and interpolation.

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Chapter 1: An Introduction to Multirate Digital Signal Processing

a) What is Multirate DSP?

The folowing two definitions (from Proakis & Manolakis) best define Multirate
Digital Signal Processing.

Sampling Rate Conversion: The process of converting a digital signal from a

given sampling rate to a different sampling rate is called Sampling Rate
Conversion.

Multirate DSP Systems: Systems that employ multiple sampling rates in the
processing of digital signals are called Multirate Digital Signal Processing
Systems.

b) Why Multirate DSP?

 Sampling rate conversion in Communication Systems where the receivers and
transmitter may have a different sampling rate.

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  Signals can be acquired from different sources sampled at different sample
rates-for processing the signals to make decisions the best way is to bring them
all to a common sampling rate.

Suppose that, we hace the values f [0], f [1], f [2], f [3], … sampled with sampling
period Tx from a signal f (t). This situation is depicted below:

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 Figure: Continuous Time signal f (t) sampled at fx = 1/Tx

In the picture above, we illustrated Tx > Ty, but the other Ty > Tx can aslo be true.

Let’s state what we want to do:

Given the values of the function at the orange locations in the above picture, we want to
predict the values at the green locations.

An Intutitive Way

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  An intutitive way of thinking about this is by reconstructing the continuos time
signal and sampling it at the required rate (fy).
 To reconstruct the signal, we first pass it through a low pass filter with cut-off
frequency fx.
 From Sampling theory, the highest frequency in the reconstructed signal is at most
fx/2.

Before we sample again at sampling rate fy, we need to consider two cases:

 fy > fx
 fx > f y

Case 1: fy > fx

+ In this case, since,

fy > fx => fy > 2(fx/2) (1)

The Nyquist Sampling Condition is satisfied, therefore, we can sample at the rate fy with
no aliasing effects.

Case 2: fx > fy

+ In this case, we see that,

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 fy < 2(fx/2) (2)

The Nyquist Sampling condition is not satisfied, therefore to prevent aliasing, we first
use an anti-aliasing filter (with cut off frequency fy/2) before reconstruction, so that
maximum frequency of the signal is fy/2.

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Chapter 2: Decimation and Interpolation by an Intergral Factor
We perform Multirate operations on a given discrete time siganl x[n], sampled form a
continuos time signal at a sampling frequency of fx to get a new sequence y[n] which is a
sampled version of the same continuos time signal sampled at a different rate, fy. In this
class we study two special cases:

1. Decimation by a Factor D (a special case of fx < fy where fy = fx/D) given a

discrete time signal sampled at fx, we want to find the discrete time signal sampled
from the same continuous time signal sampled at fx/D. We do this in two steps.
2. Interpolation by a Factor L (a special case of fx > fy wherw fy = Lfx) given a
discrete time signal sampled at fx, we want to find the discrete time signal sampled
from the continuous time signal sampled at Lfx.

a) Decimation

+ Using an Impulse train-first step:

We decimate (kill!) D – 1 samples between 0 and D, i.e, we set them all to zero. We
continue doing to all samples between kD to (k + 1)D, for all k = 1, 2, …
Mathematically,this can be done by multiplying the signal x[n] with an impulse train
of the form:

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Since p[n] is periodic with period D, we use Discrete Fourier Series to write p[n] as:

We soon see why this is convenient.

By substituting p[n] in the above, for each k =0, 1, 2, … , D- 1, we have,

Substituting the above back in 4 we have,

The signal we obtain after first step is:

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 (5)

Multirate DSP – First step – Example Decimation by a factor of 3

Figure: Obtaining x [n]p[n] from x [n]

After the first step, we downsample the signal.

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Downsampling by a factor D

For a discrete time signal x[n], the following mathematical expression best
describes downsampling by a factor D,

In this, x˜[m] is the downsampled signal and the process of going from x [n] to x˜[m] is
called Decimation by a factor of D.

Multirate DSP – Second Step – Example Decimation by a factor of 3

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 Figure: Obtaining x˜[n] from x [n]p[n]

In this section, we want to relate the z-domain expressions of the signal before and after
Decimation by a factor D. Let X(z) be the z-transform of x[n]. We, therefore, can write,

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And let, Y(z) be the z-transform of x˜[m]. We can write,

Now we use equation (5) to help modify the above:

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Thus, we have,

We put, z = e jω to analyze in the frequency domain.

Thus,

Let us start with a bandlimited signal bandlimited to digital angular frequency π/D.

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From the above, when –n ≤ ω ≤ n, we can see that,

Moreover, it is also periodic with period 2n, therefore it is a valid Discrete Time Fourier
Transform.

When we have an input signal of frequency more than π/D , we first pass it through a
low pass filter with cut off frequency π/D and then do the two steps shown above. We do
this to prevent aliasing, therefore, we call the low pass filter an anti-aliasing filter. Thus
the final structure of the decimation process will look like:

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 Figure: Decimation by a factor D

b) Interpolation

+) Interpolating with Zeros – First Step:

We modify the given signal x[n] by placing L-1 zeros between every two samples.
Mathematically, we create a new function ¯x[m], as,

for k = · · · , −3, −2, −1, 0, 1, 2, 3, · · · . We set the value of 0 for all arguments which
are not a multiple of L.

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 Figure: First step of Interpolation L=4

Z transform Analysis of First Step

Let, X (z ) be the z - transform of x [n].

Also let, X¯ (z ) be z-transform of x¯[m].

Frequency Analysis of First Step

We evalute the frequency response by evaluating on the unit circle, z = ejω. We see that
zL = ejωL. Thus,

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 Figure: Frequency Analysis of Interpolation by L

We observe that we get L copies of the frequency response of the input signal in the
interval [−π, π]. Therefore, we filter out everything outside [− π/L , π/L] on the interval
[−π, π]. Thus, we use the following low-pass filter:

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Figure: Output Discrete Time Fourier Transform

Figure: Interpolation by L

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Chapter 3: Applications of Multirate DSP
Multirate systems are used in a CD player when the music signal is converted from
digital into analogue (DAC). Digital data (16-bit words) are read from the disk at a sampling
rate of 44.1 kHz. If this data were converted directly into an analogue signal, image
frequency bands centred on multiples of the sampling-rate would occur, causing amplifier
overload, and distortion in the music signal. To protect against this, a common technique
called oversampling is often implemented nowadays in all CD players and in most digital
processing systems of music signals. Figure 3 below illustrates a basic block diagram of a CD
player and how oversampling is utilised. It is customary to oversample (or expand) the digital
signal by a factor of x8, followed by an interpolation filter to remove the image frequencies.
The sampling rate of the resulting signal is now increased up to 352.8 kHz. The digital signal
is then converted into an analogue waveform by passing it through a 14-bit DAC. Then the
output from this device is passed through an analogue low-pass filter before it is sent to the
speakers.

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 Figure 3.1 illustrates the procedure of converting a digital waveform into an analogue
signal in a CD player using x8 oversampling. As an example, Figure (a) illustrates a 20 kHz
sinusoidal signal sampled at 44.1 kHz, denoted by x[n]. The six samples of the signal
represent the waveform over two periods. If the signal x[n] was converted directly into an
analogue waveform, it would be very hard to exactly reconstruct the 20 kHz signal from this

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diagram. Now, Figure (b) shows x[n] with an x8 interpolation, denoted by y[n]. Figure (c)
shows the analogue signal y(t), reconstructed from the digital signal y[n] by passing it
through a DAC. Finally, Figure (d) shows the waveform of z(t), which is obtained by passing
the signal y(t) through an analogue low-pass filter.

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 The effect of oversampling also has some other desirable features. Firstly, it causes the
image frequencies to be much higher and therefore easier to filter out. The anti-alias filter
specification can therefore be very much relaxed i.e. the cut- off frequency of the filter for the
previous example increases from [44.1 / 2] = 22.05 kHz to [44.1x8 / 2] = 176.4 kHz after the
interpolation.

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 References
"EEE305”, “EEE801 Part A”: Digital Signal Processing

University of Florida vardhanh71@ufl.edu. October 23, 2018. Harsha Vardhan Tetali

(UF). Multirate DSP. October 23, 2018.

https://dspguru.com/dsp/faqs/multirate/interpolation/?
fbclid=IwAR1nB7bz1ZSuqQQuigztW8wifZhQ-oGuZ-
FfZ0BFC0Z2okAATaG2uBJci7U

https://dspguru.com/dsp/faqs/multirate/decimation/?
fbclid=IwAR04eyoMVaJ3p9y8XP_T3WOKmLUtuRFX0RuAdKTzCR3DVlSPvLqUY
wrq4ZE#:~:text=Loosely%20speaking%2C%20%E2%80%9Cdecimation%E2%80%9D
%20is,without%20the%20lowpass%20filtering%20operation

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ST Họ Và Tên Nhiệm Vụ Được Phân Công Ghi Chú
T
1 Trần Bách Cường Thuyết Trình Nhóm Trưởng
2 Đàm Tuấn Khôi Làm File Word
3 Đinh Bửu Hiền Làm PowerPoint
4 Phạm Văn Biên Tìm Kiếm Tài Liệu,Chỉnh sửa
File Word

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