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[[Category:ECE]]
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<font size="4">Downsampling </font>
[[Category:ECE438]]
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[[Category:signal processing]] 
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<center><font size= 4>
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A [https://www.projectrhea.org/learning/slectures.php slecture] by [[ECE]] student Yerkebulan Yeshmukhanbetov
Frequency domain view of the relationship between a signal and a sampling of that signal
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A [https://www.projectrhea.org/learning/slectures.php slecture] by [[ECE]] student Yerkebulan Yeshmukhanbetov
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Partly based on the [[2014 Fall ECE 438 Boutin|ECE438 Fall 2014 lecture]] material of [[User:Mboutin|Prof. Mireille Boutin]].  
will be edited soon!
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</center>
Partly based on the [[2014_Fall_ECE_438_Boutin|ECE438 Fall 2014 lecture]] material of [[user:mboutin|Prof. Mireille Boutin]].
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----
</center>
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----
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<br>
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== <font size="3"></font><font size="3"></font>Outline  ==
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#Introduction
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#Definition of Downsampling<br>
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#Derivation of DTFT&nbsp;of downsampled signal<br>
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#Example
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#Decimator
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#Conclusion<br>
  
 
----
 
----
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==Outline==
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== Introduction  ==
#Introduction
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#Derivation
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This slecture provides definition of downsampling, derives DTFT of&nbsp; downsampled signal and demonstrates it in a frequency domain. Also, it explains process of decimation and why it needs a low-pass filter.
#Example
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#Conclusion
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----
 
----
  
==Introduction==
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== Definition of Downsampling<br>  ==
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Downsampling is an operation which involves throwing away samples from discrete-time signal. Let&nbsp; ''x[n]'' be a digital-time signal shown below: <br>
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[[Image:Xofn.jpg]]<br>
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&nbsp;then y[n] will be produced by downsampling ''x [n]''&nbsp; by factor ''D'' = 3. So, ''y [n] = x[Dn]''.
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[[Image:Yofn.jpg]]<br>
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As seen in above graph, ''y [n]'' is obtained by throwing away some samples from x [n]. So, ''y [n]'' is a downsampled signal from
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''x [n]''.<br>
  
In this slecture I will discuss about the relations between the original signal <math> X(f) </math> (the CTFT of <math> x(t) </math>  ), sampling continuous time signal <math> X_s(f) </math> (the CTFT of <math> x_s(t) </math> ) and sampling discrete time signal <math> X_d(\omega) </math> (the DTFT of <math> x_d[n] </math> )  in frequency domain and give a specific example showing the relations.
 
 
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==Derivation==
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== Derivation of DTFT&nbsp;of downsampled signal  ==
  
The first thing which need to be clarified is that there two different types of sampling signal: <math> x_s(t) </math> and <math> x_d[n] </math>. <math> x_s(t) </math> is created by multiplying a impulse train <math> P_T(t) </math> with the original signal <math> x(t) </math> and actually <math> x_s(t) </math> is  <math> comb_T(x(t)) </math> where T is the sampling period.
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Let ''x (t) ''be a continuous tim''e ''signal. Then ''x<sub>1</sub> [n] = x (T<sub>1</sub>n) ''and''&nbsp; x<sub>2</sub> [n] = x (T<sub>2</sub>n)''. And ratio of sampling periods would be
However the <math> x_d[n] </math> is <math> x(nT) </math> where T is the sampling period.
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Now we first concentrate on the relationship between <math> X(f) </math> and <math> X_s(f) </math>.  
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D = T<sub>2</sub>/T<sub>1</sub>, &nbsp; which is an integer greater than 1. From these equations we obtain realtionship between ''x<sub>1</sub> [n]'' and ''x<sub>2</sub> [n]''. <br>
  
We know that <math> x_s(t) = x(t) \times P_T(t) </math>, we can derive the relationship between <math> x_s(t) </math> and <math> x(t) </math> in the following way:
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<math>\begin{align}
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x_2 [n] = x(T_2 n) = x(DT_1 n) = x_1 [nD]
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\end{align}</math>  
  
<div style="margin-left: 3em;">
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Below we derive Discrete-Time Fourier Transform of ''x<sub>2</sub> [n]'' in terms of DTFT of ''x<sub>1</sub> [n]''.
<math>
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\begin{align}
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F(comb_T(x(t)) &= F(x(t) \times P_T(t))\\
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&= X(f)*F(P_T(t))\\
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&= X(f)*\frac{1}{T}\sum_{n = -\infty}^\infty \delta(f-\frac{n}{T})\\
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&= \frac{1}{T}X(f)*P_\frac{1}{T}(f)\\
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&= \frac{1}{T}rep_\frac{1}{T}X(f)\\
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\end{align}
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</math>
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Show this relationship in graph below:
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<math>\begin{align}
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&\mathcal{X}_2(\omega)= \mathcal{F}(x_2 [n]) = \mathcal{F}(x_1 [Dn])\\
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&=  \sum_{n = -\infty}^\infty x_1[Dn] e^{-j \omega n} =  \sum_{m = -\infty}^\infty x_1[m] e^{-j \omega {\frac{m}{D}}}\\
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&=  \sum_{n = -\infty}^\infty s_D[m]* x_1 [m] e^{-j \omega {\frac{m}{D}}}\\
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\end{align}</math>
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<br>
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where <br>
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<math>s_D [m]=\left\{ \begin{array}{ll}
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1,& \text{ if } n \text{ is a multiple of } D,\\
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0, & \text{ else}.
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\end{array}\right. = {\frac{1}{D}} \sum_{k = -\infty}^{D-1} e^{jk {\frac{2 \pi}{D} m}}</math>
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<br>
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<math>\begin{align}
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&\mathcal{X}_2(\omega)= \sum_{m = -\infty}^\infty {\frac{1}{D}} \sum_{k = -\infty}^{D-1} e^{jk {\frac{2 \pi}{D} m}}  x_1[m] e^{-j \omega {\frac{m}{D}}}\\
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&= {\frac{1}{D}} \sum_{k = -\infty}^{D-1} \sum_{m = -\infty}^\infty  x_1[m] e^{-jm ({\frac{\omega - 2 \pi k}{D}})} =  \\
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&=  {\frac{1}{D}} \sum_{k = -\infty}^{D-1} \mathcal{X}_1 ({\frac{\omega - 2 \pi k}{D}}) \\
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\end{align}</math>
  
 
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==example==
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== Example<br>  ==
  
[[Image:Xfcbt.png]]
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<br>
  
[[Image:xsfcbt.png]]
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Let's take a look&nbsp; at&nbsp; an original signal ''X<sub>1</sub> (w)'' and&nbsp;&nbsp;''X<sub>2</sub> (w)'' which is obtained after downsampling X<sub>1</sub>(w) by factor D = 2 in a frequency domain.
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[[Image:Downsamplegraph.jpg]]<br>
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<br>
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From two graphs it is seen that signal is stretched by D&nbsp; in frequency domain and&nbsp; decreased by D in a magnitude after downsampling. Both signals have the frequency of&nbsp;<math>\begin{align}
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2\pi
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\end{align}</math> .
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== Decimator  ==
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As seen in second graph, if&nbsp;<math>\begin{align}
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D2\pi T_1f_{max}
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\end{align}</math> is greater than <math>\begin{align}
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\pi
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\end{align}</math> aliasing occurs. Downsampler is a part of a decimator which also has a low-pass filter to&nbsp; prevent aliasing.&nbsp; LPF eliminates signal components which has&nbsp; frequencies higher than cutoff frequency, which can be found from graphs shown above.<br>
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&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <math>\begin{align}
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& D\omega_c = D 2 \pi T_1 f_{max} < \pi\\
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& {\frac{T_2}{T_1}} 2\pi T_1 f_{max} < \pi \\
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&  2\pi T_2f_{max} < \pi \\
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&f_{max} < {\frac{1}{2T_2}}
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\end{align}</math>
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Thereby, signal needs to be filtered before downsampling if f<sub>max</sub> &gt; 1/(2T<sub>2</sub>) . Complete block diagram of a decimator is shown below:<br>
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[[Image:Decimator cutoff.jpg]]  
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----
 
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==Derivation==
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== Conclusion  ==
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</font>
  
Then we are going to find the relation between <math> X_s(f) </math> and <math> X_d(\omega) </math>
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We know another way to express CTFT of <math> x_s(t) </math>:
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X_s(f) &= F(\sum_{n = -\infty}^\infty x(nT)\delta(t-nT))\\
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&= \sum_{n = -\infty}^\infty x(nT)F(\delta(t-nT))\\
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&= \sum_{n = -\infty}^\infty x(nT)e^{-j2\pi fnT}\\
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\end{align}
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compare it with DTFT of <math> x_d[n] </math>:
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X_d(\omega) &= \sum_{n = -\infty}^\infty x_d[n]e^{-j\omega n}\\
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&= \sum_{n = -\infty}^\infty x(nT)e^{-j\omega n}\\
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\end{align}
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we can find that:
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X_d(2\pi Tf) &= X_s(f)\\
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if <math> f = \frac{1}{T} </math>
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we have that:
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X_d(2\pi ) &= X_s(\frac{1}{T})\\
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from this equation, we can know the relationship between <math> X_s(f) </math> and <math> X_d(\omega) </math> and the relationship is showed in graph as below:
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<font size="3">To summarize, downsampling is a process of removing samples from signal. After downsampling,&nbsp; signal decreases by factor D in the magnitude and stretches by D in frequency domain.&nbsp; In order to downsample a signal, it first should be filtered by LPF to prevent aliasing.&nbsp; Both LPF and downsampler are parts of a decimator. </font>
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==example==
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<br>
  
[[Image:xsfcbt.png]]
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----
  
[[Image:xdwcbt.png]]
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==conclusion==
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== [[Yeshmukhanbetov ECE438 slecture review|Questions and comments]]  ==
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If you have any questions, comments, etc. please post them on [[Yeshmukhanbetov ECE438 slecture review|this page]].
  
So the relationship between <math> X(f) </math> and <math> X_s(f) </math> is that <math> X_s(f) </math> is a a rep of <math> X(f) </math> in frequency domain with period of <math> \frac{1}{T} </math> and magnitude scaled by <math> \frac{1}{T} </math>.
 
the relationship between <math> X(f) </math> and <math> X_d(\omega) </math> is that <math> X_d(\omega) </math> is also a a rep of <math> X(f) </math> in frequency domain with period <math> 2\pi </math> and magnitude is also scaled by <math> \frac{1}{T} </math>, but the frequency is scaled by <math> 2\pi T </math>
 
 
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[[2014_Fall_ECE_438_Boutin_digital_signal_processing_slectures|Back to ECE438 slectures, Fall 2014]]
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[[Category:Slecture]] [[Category:ECE438Fall2014Boutin]] [[Category:ECE]] [[Category:ECE438]] [[Category:Signal_processing]]

Latest revision as of 18:07, 16 March 2015


Downsampling

A slecture by ECE student Yerkebulan Yeshmukhanbetov

Partly based on the ECE438 Fall 2014 lecture material of Prof. Mireille Boutin.




Outline

  1. Introduction
  2. Definition of Downsampling
  3. Derivation of DTFT of downsampled signal
  4. Example
  5. Decimator
  6. Conclusion

Introduction

This slecture provides definition of downsampling, derives DTFT of  downsampled signal and demonstrates it in a frequency domain. Also, it explains process of decimation and why it needs a low-pass filter.


Definition of Downsampling

Downsampling is an operation which involves throwing away samples from discrete-time signal. Let  x[n] be a digital-time signal shown below:

Xofn.jpg

 then y[n] will be produced by downsampling x [n]  by factor D = 3. So, y [n] = x[Dn].

Yofn.jpg

As seen in above graph, y [n] is obtained by throwing away some samples from x [n]. So, y [n] is a downsampled signal from

x [n].


Derivation of DTFT of downsampled signal

Let x (t) be a continuous time signal. Then x1 [n] = x (T1n) and  x2 [n] = x (T2n). And ratio of sampling periods would be

D = T2/T1,   which is an integer greater than 1. From these equations we obtain realtionship between x1 [n] and x2 [n].

$ \begin{align} x_2 [n] = x(T_2 n) = x(DT_1 n) = x_1 [nD] \end{align} $

Below we derive Discrete-Time Fourier Transform of x2 [n] in terms of DTFT of x1 [n].


$ \begin{align} &\mathcal{X}_2(\omega)= \mathcal{F}(x_2 [n]) = \mathcal{F}(x_1 [Dn])\\ &= \sum_{n = -\infty}^\infty x_1[Dn] e^{-j \omega n} = \sum_{m = -\infty}^\infty x_1[m] e^{-j \omega {\frac{m}{D}}}\\ &= \sum_{n = -\infty}^\infty s_D[m]* x_1 [m] e^{-j \omega {\frac{m}{D}}}\\ \end{align} $


where

$ s_D [m]=\left\{ \begin{array}{ll} 1,& \text{ if } n \text{ is a multiple of } D,\\ 0, & \text{ else}. \end{array}\right. = {\frac{1}{D}} \sum_{k = -\infty}^{D-1} e^{jk {\frac{2 \pi}{D} m}} $


$ \begin{align} &\mathcal{X}_2(\omega)= \sum_{m = -\infty}^\infty {\frac{1}{D}} \sum_{k = -\infty}^{D-1} e^{jk {\frac{2 \pi}{D} m}} x_1[m] e^{-j \omega {\frac{m}{D}}}\\ &= {\frac{1}{D}} \sum_{k = -\infty}^{D-1} \sum_{m = -\infty}^\infty x_1[m] e^{-jm ({\frac{\omega - 2 \pi k}{D}})} = \\ &= {\frac{1}{D}} \sum_{k = -\infty}^{D-1} \mathcal{X}_1 ({\frac{\omega - 2 \pi k}{D}}) \\ \end{align} $


Example


Let's take a look  at  an original signal X1 (w) and  X2 (w) which is obtained after downsampling X1(w) by factor D = 2 in a frequency domain.

Downsamplegraph.jpg


From two graphs it is seen that signal is stretched by D  in frequency domain and  decreased by D in a magnitude after downsampling. Both signals have the frequency of $ \begin{align} 2\pi \end{align} $ .

Decimator

As seen in second graph, if $ \begin{align} D2\pi T_1f_{max} \end{align} $ is greater than $ \begin{align} \pi \end{align} $ aliasing occurs. Downsampler is a part of a decimator which also has a low-pass filter to  prevent aliasing.  LPF eliminates signal components which has  frequencies higher than cutoff frequency, which can be found from graphs shown above.

                             $ \begin{align} & D\omega_c = D 2 \pi T_1 f_{max} < \pi\\ & {\frac{T_2}{T_1}} 2\pi T_1 f_{max} < \pi \\ & 2\pi T_2f_{max} < \pi \\ &f_{max} < {\frac{1}{2T_2}} \end{align} $

Thereby, signal needs to be filtered before downsampling if fmax > 1/(2T2) . Complete block diagram of a decimator is shown below:


Decimator cutoff.jpg




Conclusion

To summarize, downsampling is a process of removing samples from signal. After downsampling,  signal decreases by factor D in the magnitude and stretches by D in frequency domain.  In order to downsample a signal, it first should be filtered by LPF to prevent aliasing.  Both LPF and downsampler are parts of a decimator.






Questions and comments

If you have any questions, comments, etc. please post them on this page.


Back to ECE438 slectures, Fall 2014

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