Lecture6 Sampling
Lecture6 Sampling
Lalan Kumar
Assistant Professor
Multichannel Signal Processing Lab
Department of Electrical Engineering
Indian Institute of Technology Delhi
n=∞
X
xs (t) = xc (t) δ(t − nT ) (1)
n=−∞
n=∞
X
= xc (nT )δ(t − nT ) (2)
n=−∞
• A fundamental question:
k=∞
1 X
Xs (jΩ) = Xc (jΩ − kΩs ) (4)
T
k=−∞
• Xs (jΩ) can be interpreted as superimposed version of periodically repeated
Xc (jΩ)
SAMPLING AND DATA RECONSTRUCTION 9 / 38
Frequency Domain Representation of Sampling : CD Converter
• Let xc (t) be a band-limited signal with Xc (jΩ) = 0|Ω| > ΩN , Then xc (t)
is uniquely determined by its samples
x[n] = xc (nT )n = · · · − 2, −1, 0, 1, 2 . . . , if Ωs = 2π
T
> 2ΩN
• Where ΩN is called the Nyquist frequency and 2ΩN is called the Nyquist
rate.
• Ωs must exceed Nyquist rate for ideal reconstruction of xc (t).
Application fN = ΩN /2π fs = Ωs /2π
Biomedical <500 Hz 1 kHz
• Telephone speech <4 kHz 8 kHz
Music <20 kHz 44.1 kHz
Ultasonic <100 kHz 250 kHz
Radar <100 MHz 200 MHz
Table: Typical Bandwidths and sampling frequencies in signal processing applications
Figure: DC Block
• Having Xs (jΩ), the signal can be recovered from Figure 2 using Hr (jΩ) as
Hr (jΩ) = T |Ω| < Ωc
SAMPLING AND DATA RECONSTRUCTION 13 / 38
= 0, o/w
DC Converter : Reconstruction of Band-Limited Signal from Samples
Ωs π
Ωc = = (6)
2 T
• To get hr (t), we take inverse Fourier transform of Hr (jΩ).
•
Z π
T sin( πt
Z ∞
1 1 T ) t
hr (t) = H(jΩ)e jΩt dΩ = Te jΩt dΩ = T
= sinc( )
2π −∞ 2π − π πt T
T
(7)
∞
X t − nT
xr (t) = x[n]sinc( ) (8)
n=−∞
T
• Eq.8 is Interpolation. The sinc is interpolating function for xr (t),
interpolating from the discreter sequence x[n].
SAMPLING AND DATA RECONSTRUCTION 15 / 38
DC Converter : Reconstruction of Band-Limited Signal from Samples
• Q. xr (t) = xc (t) ?
• Using interpolation formula at t=kT:
∞
X
xr (kT ) = x[n]sinc(k − n) (9)
n=−∞
But
sinc(k − n) = δ(k − n) (10)
• Putting Eq.10 in Eq.9, we have
or
n=∞
X
Xr (jΩ) = x[n]Hr (jΩ)e −jΩnT
n=−∞
n=∞
X
= Hr (jΩ) x[n]e −jΩnT
n=−∞
= Hr (jΩ)X (e jω )
ω=ΩT
x[n] = xc (nT )
k=∞
1 X
X (e jω ) = Xc (j(Ω − kΩs ))
T Ω= ω
T
k=−∞
= Xs (jΩ)
• For D/C converter: Having just Hr
k=∞
X t − nT
yr (t) = y [n]sinc( ) (11)
T
k=−∞
• Let H(e jω ) be the frequency response of an LTI discrete time system, i.e.
Y (e jω ) = H(e jω )X (e jω ) . Substituting in (12), we have
Ωs > 2ΩN
2π
> 2ΩN
T
π
ΩN <
T
• So, Xc (jΩ) = 0 for |Ω| ≥ π
T
, a band limited.
• Hr (jΩ) of the lowpass filter selects only the k=0 term in (15).
• (
H(e jΩT )Xc (jΩ), |Ω| < Tπ
Yr (jΩ) =
0, |Ω| ≥ Tπ
Let
(
H(e jΩT ), |Ω| < π
T
Heff (jΩ) = π
(16)
0, |Ω| ≥ T
• This means the effective continuous time system is also LTI, whose
frequency response is Heff (jΩ).
• In the derivation, we assumed the discrete time system is LTI (time
domain convolution and frequency domain multiplication) and also we
assumed no aliasing occured.
• Q. What are sufficient conditions for the effective continuous system to be
LTI?
1. The discrete time system has to be LTI.
2. Input must be bandlimited.
3. Sampling rate must be high enough above Nyquist rate so that there is no
aliasing occuring.
4. If input xc (t) is not precisely bandlimited → a low-pass filter or anti-aliasing
filter is needed to process.
•
• now passing the signal through a low pass reconstruction filter, bandwidth
of the reconstruction filter is taken to be Tπ
• Comparing Figure (a) and (f), it is clear that Heff acts a low pass filter.
• By varying the sampling period T (in C/D part), an equivalent continuous
time low pass filter (Heff ) with variable cut-off frequency can be
implemented.
• The cut-off frequency for such LPF Ωc = ωTc where ωc is the cut-off
frequency of the DT system.
• The system behaves as an LTI system with frequency response given by
Heff (jΩ), bcz o/p is Xc Heff .
• See the above Figure (c), if T were chosen such that ΩN T < ωc then o/p
of the system of figure (7) would be yr (t) = xc (t), i.e. no distortion.
• For ΩN T > ωc , Nyquist requirement for DT processing of CT signal, From
figure (c)
(2π − ΩN T ) > ωc
Rather than the original Nyquist criteria
(2π − ΩN T ) > ΩN T
• We can tolerate some aliasing while sampling as long as the distorted part
are eliminated by the DT system H(e jω ).
• (16) is valid even if there is aliasing in CD converter, as long as H(e jω )
does not pass the aliased component-Prob 4.31, oppeinheim 3rd edition.
• (2π − ΩN T ) < ωc , aliasing while sampling, can not be filtered by the DT
system, and (16) or Yr (jΩ) = Heff (jΩ)Xc (jΩ) is no more valid, bcz the
eqn assumes only zeroth spectrum of Xc (j(Ω − kΩs ))
n
xe (n) = x , n = KL
L
= 0, else
X (Z ) = x0 + x1 Z −1 + x2 Z −2 + . . .
• For L=2, one zero is introduced
Xe (Z ) = x0 + (0 × Z −1 ) + x1 Z −2 + (0 × Z −3 ) + x2 Z −4 . . .
⇒ Xe (Z ) = X (Z 2 )
• In general, Xe (Z ) = X (Z L ).
• Spectrum : Xe (e jω ) = X (e jωL )
• Upsampling in time domain by L leads to shrinking in frequency domain by
L
•
SAMPLING AND DATA RECONSTRUCTION 36 / 38
Up-Sampling
x[n/L], n = kL
xe (n) =
0, otherwise
Xe (e jω ) = X (e jωL )
|ω| ≤ πL
L,
Hi (e jω ) =
SAMPLING AND DATA RECONSTRUCTION 0, πL < |ω| < π 38 / 38
Removing the False contours:
Why the amplitude L ?
1 L
= orT ′ = T /L
T′ T
• Time period is reduced and sampling frequency is increased.
• Upsampling is always followed by an interpolation filter.
xd [n] = x[nN]
X
Xd (Z ) = x[nN]Z −n (22)
• x[n] : x0 x1 x2 . . . → X (Z ) = x0 + x1 Z −1 + x2 Z −2 + . . .
• For N=2 xp [n] : x0 0 x2 . . . → Xp (Z ) = x0 + x2 Z −2 + . . .
• xd [n] : x0 x2 . . . → Xd (Z ) = x0 + x2 Z −1 + x4 Z −2 . . .
• Xd (Z ) = Xp (Z 1/2 ), or in general, Xd (Z ) = Xp (Z 1/N )
• we we dont have relation between Xd (Z ) and X (Z )
j4π j8π
• p[2] = 13 [1 + e − 3 + e − 3 ] = 0
• p[3] = 1, Same, so use the alternative, rather than (23)
SAMPLING AND DATA RECONSTRUCTION 45 / 38
Down-Sampling
N−1
1 X
• ∴ Xp (e jω ) = X (e j(ω−kωs ) ) with ωs = 2π
N
, repeated n shifted.
N
k=0
Points To Be remembered
ω
• X (e j( N ) ) implies Spread in Frequency domain by N (X (e jω/N )) in
downsampling.
• The downsampling ≈ Sampling the DT signal ≈ multiplied by
1
PN−1 −kN
N k=0 WN .
N−1
1 X ω−2kπ
Xd (e jω ) = X (e j( N ) ) (25)
N
k=0
jω
• Periodicity of X (e N ) = 2Nπ, that of Xd (e jω ) remains 2π
• For Xd (e jω ), compute first Xp (e jω ) and then Xd (e jω )= Xp (e jω/N ) , or use
eq.(25) directly
SAMPLING AND DATA RECONSTRUCTION 47 / 38
Down-Sampling
PN−1 2kπ )
• For N=2, downsample by 2 :Xp (e jω ) = 1
N k=0 X (e j(ω− N ), using the
first approach
ω
• Now, Xd (e jω ) =Xp (e j 2 )
ω
• So Xp (e jω ) has periodicity π, then Xp (e j 2 ) has periodicity 2π with
spreaded frequency spectrum
• a. CTFT of xc (t).
2π pi
• b. of x[n] = xc (nT ) where = 4ΩN or ΩN T =
T 2
• c. say downsample by factor of N=3; xd [n] = x[3n]: see aliasing in
Xd (e jω ).
• d. To avoid the aliasing in downsampling by factor of N requires
ΩN TN ≤ π (ΩN T from CT to DT, N from DT to downsampled) where
π
(ΩN T = ωN ) or ωN ≤
N
• The downsampling can be done without aliasing if we are willing to reduce
the bandwidth of signal x[n] before downsampling.
π
• If x[n] is pre-filtered by LPF with ωc = then x̃[n] can be downsampled
N
by factir N without aliasing.
• We have no aliasing but loosing certain part of spectrum due to LPF
(which was the case with aliasing as well).
Lalan Kumar
{lkumar}@[Link]