Interpolation and Pulse Shaping in Real-Time Digital Signal Processing

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Interpolation and Pulse Shaping
7 - 2
Outline
 
Discrete-to-continuous conversion
Interpolation
Pulse shapes
Rectangular
Triangular
Sinc
Raised cosine
Sampling and interpolation demonstration
Conclusion
7 - 3
Data Conversion
 
Analog-to-Digital Conversion
Lowpass filter has
stopband frequency
less than ½ 
f
s
 to reduce
aliasing due to sampling
(enforce sampling theorem)
Digital-to-Analog Conversion
Discrete-to-continuous
conversion could be as
simple as sample and hold
Lowpass filter has stopband
frequency less than ½ 
f
s  
to
reduce artificial high frequencies
7 - 4
Discrete-to-Continuous Conversion
 
Input
: sequence of samples 
y
[
n
]
Output
: smooth continuous-time function obtained
through interpolation (by “connecting the dots”)
If  
f
0
 < ½ 
f
s
 , then
would be converted to
 
 
Otherwise, aliasing has occurred, and the converter would
reconstruct a cosine wave whose frequency is equal to the
aliased positive frequency that is less than ½ 
f
s
7 - 5
Discrete-to-Continuous Conversion
 
General form of interpolation is sum of weighted
pulses
 
Sequence
 y
[
n
] converted into continuous-time signal that is an
approximation of 
y
(
t
)
Pulse function
 p
(
t
) could be rectangular, triangular, parabolic,
sinc, truncated sinc, raised cosine, etc.
Pulses overlap in time domain when pulse duration is greater
than or equal to sampling period 
T
s
Pulses generally have unit amplitude and/or unit area
Above formula is related to discrete-time convolution
7 - 6
Interpolation From Tables
 
Using mathematical tables of
numeric values of functions to
compute a value of the function
Estimate 
f
(1.5) from table
Zero-order hold
: take value to be 
f
(1)
to make 
f
(1.5) = 1.0 
(“stairsteps”)
Linear interpolation
: average values of
nearest two neighbors to get 
f
(1.5) = 2.5
Curve fitting
: fit four points in table to
polynomal 
a
0
 + 
a
1
 
x
 + 
a
2
 
x
2
 + 
a
3
 
x
3
which gives 
f
(1.5) = 
x
2
 = 2.25
x
0
1
2
3
1
4
9
7 - 7
Rectangular Pulse
 
Zero-order hold
Easy to implement in hardware or software
 
 
The Fourier transform is
 
 
In time domain, no overlap between 
p
(
t
) and adjacent pulses
p
(
t - T
s
) and 
p
(
t + T
s
)
In frequency domain, sinc has infinite two-sided extent; hence,
the spectrum is not bandlimited
7 - 8
Sinc Function
 
 
 
 
 
 
Even function (symmetric at origin)
Zero crossings at
Amplitude decreases proportionally to 1/x
7 - 9
Triangular Pulse
 
Linear interpolation
It is relatively easy to implement in hardware or software,
although not as easy as zero-order hold
 
 
Overlap between 
p
(
t
) and its adjacent pulses 
p
(
t - T
s
) and
p
(
t + T
s
) but with no others
Fourier transform is
How to compute this?  
Hint:
 Triangular pulse is equal to 1 / 
T
s
times the convolution of rectangular pulse with itself
In frequency domain, sinc
2
(
f
 
T
s
) has infinite two-sided extent;
hence, the spectrum is not bandlimited
7 - 10
Sinc Pulse
 
Ideal bandlimited interpolation
 
 
 
In time domain, infinite overlap between other pulses
Fourier transform has extent 
f
 
 [-
W
, 
W
], where
P
(
f
) is ideal lowpass frequency response with bandwidth 
W
In frequency domain, sinc pulse is bandlimited
Interpolate with infinite extent pulse in time?
Truncate sinc pulse by multiplying it by rectangular pulse
Causes smearing in frequency domain (multiplication in time
domain is convolution in frequency domain)
7 - 11
Raised Cosine Pulse: Time Domain
 
Pulse shaping used in communication systems
 
 
 
 
W
 is bandwidth of an
ideal lowpass response
 
 [0, 1]
 rolloff factor
Zero crossings at
t
 = 
 
T
s
 , 
 2 
T
s
 , …
See handout G in reader on raised cosine pulse
ideal lowpass filter
impulse response
Attenuation by 1/t
2
 for
large t to reduce tail
7 - 12
Raised Cosine Pulse Spectra
 
Pulse shaping used in communication systems
Bandwidth increased
by factor of (1 + 
):
(1 + 
) 
W = 2 W – f
1
f
1
 marks transition from
passband to stopband
 
 
 
 
 
Bandwidth generally scarce in communication systems
7 - 13
Sampling and Interpolation Demo
 
DSP First, Ch. 4, Sampling and interpolation,
http://www.ece.gatech.edu/research/DSP/DSPFirstCD
/
Sample sinusoid 
y
(
t
) to form 
y
[
n
]
Reconstruct sinusoid using
rectangular, triangular, or
truncated sinc pulse 
p
(
t
)
Which pulse gives the best reconstruction?
Sinc pulse is truncated to be four sampling periods
long.  Why is the sinc pulse truncated?
What happens as the sampling rate is increased?
7 - 14
Conclusion
 
Discrete-to-continuous time conversion involves
interpolating between known discrete-time samples
y
[
n
] using pulse shape 
p
(
t
)
 
Common pulse shapes
Rectangular for same-and-hold interpolation
Triangular for linear interpolation
Sinc for optimal bandlimited linear interpolation but impractical
Truncated raised cosine for practical bandlimited interpolation
Truncation causes smearing in frequency domain
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Discrete-to-continuous conversion, interpolation, pulse shaping techniques, and data conversion in real-time digital signal processing are discussed in this content. Topics include types of pulse shapes, sampling, continuous signal approximation, interpolation methods, and data conversion processes from analog to digital and digital to analog. Concepts such as aliasing, pulse overlap, and different interpolation techniques like zero-order hold and linear interpolation are also covered.

  • Signal Processing
  • Interpolation
  • Pulse Shaping
  • Digital Conversion
  • Real-Time Processing

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  1. EE445S Real-Time Digital Signal Processing Lab Spring 2014 Interpolation and Pulse Shaping Prof. Brian L. Evans Dept. of Electrical and Computer Engineering The University of Texas at Austin Lecture 7

  2. Outline Discrete-to-continuous conversion Interpolation Pulse shapes Rectangular Triangular Sinc Raised cosine Sampling and interpolation demonstration Conclusion 7 - 2

  3. Data Conversion Analog-to-Digital Conversion Lowpass filter has stopband frequency less than fs to reduce aliasing due to sampling (enforce sampling theorem) Digital-to-Analog Conversion Discrete-to-continuous conversion could be as simple as sample and hold Lowpass filter has stopband frequency less than fs to reduce artificial high frequencies Lecture 4 Lecture 8 Analog Lowpass Filter Quantizer Sampler at sampling rate of fs Lecture 7 Discrete to Continuous Conversion Analog Lowpass Filter fs 7 - 3

  4. Discrete-to-Continuous Conversion Input: sequence of samples y[n] Output: smooth continuous-time function obtained through interpolation (by connecting the dots ) [n y ] If f0 < fs , then ] [ = A n y ( ) + cos 2 f T n 3 4 5 6 7 0 s n would be converted to ~ = A t y ~t y 1 2 ( ) ( ) 0 + ( ) cos 2 t f Otherwise, aliasing has occurred, and the converter would reconstruct a cosine wave whose frequency is equal to the aliased positive frequency that is less than fs 7 - 4

  5. Discrete-to-Continuous Conversion General form of interpolation is sum of weighted pulses = n ~ = ( ) [ ] ( sn T ) y t y n p t Sequence y[n] converted into continuous-time signal that is an approximation of y(t) Pulse function p(t) could be rectangular, triangular, parabolic, sinc, truncated sinc, raised cosine, etc. Pulses overlap in time domain when pulse duration is greater than or equal to sampling period Ts Pulses generally have unit amplitude and/or unit area Above formula is related to discrete-time convolution 7 - 5

  6. Interpolation From Tables Using mathematical tables of numeric values of functions to compute a value of the function x 0 1 2 3 f(x) 0.0 1.0 4.0 9.0 Estimate f(1.5) from table Zero-order hold: take value to be f(1) to make f(1.5) = 1.0 ( stairsteps ) Linear interpolation: average values of nearest two neighbors to get f(1.5) = 2.5 Curve fitting: fit four points in table to polynomal a0 + a1x + a2x2 + a3x3 which gives f(1.5) = x2 = 2.25 ~x f ( ) 9 4 1 x 0 1 2 3 7 - 6

  7. Rectangular Pulse Zero-order hold Easy to implement in hardware or software = 0 p(t) 1 1 t 1 if T t T 1 = ( ) rect p t s s 2 2 T otherwise s - Ts Ts t The Fourier transform is ( ) x sin( T ) sin f x ( ) = = = ( ) sinc T where sinc ( ) s P f T f T x s s s T f s In time domain, no overlap between p(t) and adjacent pulses p(t - Ts) and p(t + Ts) In frequency domain, sinc has infinite two-sided extent; hence, the spectrum is not bandlimited 7 - 7

  8. Sinc Function ( ) x sin x ( ) x = sinc sinc(x) 1 How compute to sinc(0)? As numerator 0, and x x denominato both are r going 3 2 2 3 0 How 0. to handle to it? Even function (symmetric at origin) Zero crossings at Amplitude decreases proportionally to 1/x = , 2 , 3 ... , x 7 - 8

  9. Triangular Pulse Linear interpolation It is relatively easy to implement in hardware or software, although not as easy as zero-order hold = otherwise 0 p(t) 1 | T | t 1 if T t T t = s s ( ) p t s T -Ts Ts t s Overlap between p(t) and its adjacent pulses p(t - Ts) and p(t + Ts) but with no others Fourier transform is How to compute this? Hint: Triangular pulse is equal to 1 / Ts times the convolution of rectangular pulse with itself In frequency domain, sinc2(fTs) has infinite two-sided extent; hence, the spectrum is not bandlimited ( ) = 2 ( ) sinc T P f T f s s 7 - 9

  10. Sinc Pulse Ideal bandlimited interpolation sin t T 1 f 1 = = s = ( ) sinc p t t ( ) rect P f = W T T T 2 sT t s s s T s In time domain, infinite overlap between other pulses Fourier transform has extent f [-W, W], where P(f) is ideal lowpass frequency response with bandwidth W In frequency domain, sinc pulse is bandlimited Interpolate with infinite extent pulse in time? Truncate sinc pulse by multiplying it by rectangular pulse Causes smearing in frequency domain (multiplication in time domain is convolution in frequency domain) 7 - 10

  11. Raised Cosine Pulse: Time Domain Pulse shaping used in communication systems ( ) 2 2 2 16 1 t W T s cos 2 t W t sinc = ( ) p t ideal lowpass filter impulse response Attenuation by 1/t2 for large t to reduce tail W is bandwidth of an ideal lowpass response [0, 1] rolloff factor Zero crossings at t = Ts , 2 Ts, See handout G in reader on raised cosine pulse 7 - 11

  12. Raised Cosine Pulse Spectra Pulse shaping used in communication systems Bandwidth increased by factor of (1 + ): (1 + ) W = 2 W f1 f1 marks transition from passband to stopband 2W W f f P 1 1 | if 0 | f f = W 1 2 sT ( ) 1 | W | = ( ) 1 sin if | | 2 f f W f f1 1 1 4W 2 2 f = 1 1 W 0 otherwise Bandwidth generally scarce in communication systems 7 - 12

  13. Sampling and Interpolation Demo DSP First, Ch. 4, Sampling and interpolation, http://www.ece.gatech.edu/research/DSP/DSPFirstCD/ Sample sinusoid y(t) to form y[n] Reconstruct sinusoid using rectangular, triangular, or truncated sinc pulse p(t) Which pulse gives the best reconstruction? Sinc pulse is truncated to be four sampling periods long. Why is the sinc pulse truncated? What happens as the sampling rate is increased? ~ = n = ( ) [ ] ( sn T ) y t y n p t 7 - 13

  14. Conclusion Discrete-to-continuous time conversion involves interpolating between known discrete-time samples y[n] using pulse shape p(t) = n [n y ] ~ = ( ) [ ] ( sn T ) y t y n p t 3 4 5 6 7 n ~t y 1 2 Common pulse shapes Rectangular for same-and-hold interpolation Triangular for linear interpolation Sinc for optimal bandlimited linear interpolation but impractical Truncated raised cosine for practical bandlimited interpolation Truncation causes smearing in frequency domain ( ) 7 - 14

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