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DSP6

The document discusses changing the sampling rate of a discrete-time signal. It describes downsampling a signal by an integer factor, which reduces the sampling rate by regularly omitting samples. It also describes upsampling a signal by an integer factor, which increases the sampling rate by interpolation. Changing the sampling rate by a non-integer factor can be achieved by combining upsampling and downsampling.

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

DSP6

The document discusses changing the sampling rate of a discrete-time signal. It describes downsampling a signal by an integer factor, which reduces the sampling rate by regularly omitting samples. It also describes upsampling a signal by an integer factor, which increases the sampling rate by interpolation. Changing the sampling rate by a non-integer factor can be achieved by combining upsampling and downsampling.

Uploaded by

udz76852
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PPTX, PDF, TXT or read online on Scribd
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Changing the Sampling Rate

Content and Figures are from Discrete-Time Signal Processing, 2e by Oppenheim, Shafer, and Buck, ©1999-2000 Prentice Hall Inc.
Changing the Sampling Rate
• A continuous-time signal can be represented by its samples as

xn  x c nT 
• We can use bandlimited interpolation to go back to the
continuous-time signal from its samples
• Some applications require us to change the sampling rate
– Or to obtain a new discrete-time representation of the same
continuous-time signal of the form

x' n  x c nT' where T  T'


• The problem is to get x’[n] given x[n]
• One way of accomplishing this is to
– Reconstruct the continuous-time signal from x[n]
– Resample the continuous-time signal using new rate to get x’[n]
– This requires analog processing which is often undersired
Sampling Rate Reduction by an Integer Factor: Downsampling
• We reduce the sampling rate of a sequence by “sampling” it

x d n  xnM  x c nMT 


• This is accomplished with a sampling rate compressor

• We obtain xd[n] that is identical to what we would get by


reconstructing the signal and resampling it with T’=MT
• There will be no aliasing if
 
  N
T' MT
Sampling Rate Reduction by an Integer Factor: Downsampling
• We reduce the sampling rate of a sequence by “sampling” it

x d n  xnM  x c nMT 


• This is accomplished with a sampling rate compressor

Anti-aliasing Filter

• We obtain xd[n] that is identical to what we would get by


reconstructing the signal and resampling it with T’=MT
• There will be no aliasing if
 
  N
T' MT
Frequency Domain Representation of Downsampling
• Recall the DTFT of x[n]=xc(nT)
   2k  
Xe j 1 
  X c  j 
T k     T
 
T 
• The DTFT of the downsampled signal can similarly written as
   2r      2r  
  1  1 
Xd e j
  c X  j
T' r     T'
   
T'   MT
 X  j
c    
r     MT MT  

• Let’s represent the summation index as

r  i  kM
where -   k   and 0  i  M - 1
1 M 1  1     2k 2i  
X d e  
j
   X c  j     
M i 0  T k     MT T MT  
• And finally

1 M 1  j M  M  
  2 i 

 
X d e j  X e
M i0  
 
Frequency Domain Representation of Downsampling
• Recall the DTFT of x[n]=xc(nT)
   2k  
Xe j 1 
  X c  j 
T k     T
 
T 
• The DTFT of the downsampled signal can similarly written as
   2r      2r  
  1  1 
Xd e j
  c X  j
T' r     T'
   
T'   MT
 X  j
c    
r     MT MT  

• Let’s represent the summation index as

r  i  kM
where -   k   and 0  i  M - 1
1 M 1  1     2k 2i  
X d e  
j
   X c  j     
M i 0  T k     MT T MT  
• And finally

1 M 1  j M  M  
  2 i 

 
X d e j  X e
M i0  
 
Frequency Domain Representation of Downsampling: No Aliasing

𝜋
2
Important Points

• Downsampling in time domain means you are squeezing the


signal in time domain.

• This squeezing in time is reflected as expansion in frequency


domain, so spectrum expands and scaled down.
Application of downsampling
• Communication system (where specs of hardware mismatch)

• Audio/Image Processing
Increasing the Sampling Rate by an Integer Factor: Upsampling
• We increase the sampling rate of a sequence interpolating it

xi n  xn / L   x c nT / L 


• This is accomplished with a sampling rate expander

• We obtain xi[n] that is identical to what we would get by


reconstructing the signal and resampling it with T’=T/L
• Upsampling consists of two steps
– Expanding
xn / L  n  0,L,2L,... 
x e n     xk n  kL 
– Interpolating
 0 else k  
Frequency Domain Representation of Expander
• The DTFT of xe[n] can be written as

      j n
 
 
X e e     xk n  kL e
j
  xk e  jLk  X e jL
n   k    k  
• The output of the expander is frequency-scaled
Frequency Domain Representation of Interpolator
• To get interpolated signal we apply the following LPF

• The extrapolator output is given as

• The DTFT of the desired interpolated signals is


Interpolator in Time Domain
• xi[n] in a low-pass filtered version of x[n]
• The low-pass filter impulse response is
sin n / L 
hi n 
n
• Hence the interpolated signal is written as

sin  n  kL  / L 
xi n   xk 
k    n  kL 
• Note that

• Therefore, the filter output can be written as

xi n  xn / L   x c nT / L   x c nT' for n  0,L,2L,...


Changing the Sampling Rate by Non-Integer Factor
• Combine upsampling and downsampling for non-integer
factors
Upsampling Downsampling

• The two low-pass filters can be combined into a single one

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