Sampling and Multirate Techniques For Complex and Bandpass Signals
Sampling and Multirate Techniques For Complex and Bandpass Signals
M. Renfors, TUT/DCE
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Contents
Complex signals and systems
o
o
o
o
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discrete-time
x (t ) = x R (t ) + jx I (t )
complex
signals
are
x ( k ) = x R ( k ) + jx I ( k )
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f
0
f
0
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+ j , < 0
Hilbert filters can be used to construct analytic signals with
only positive (or negative) frequency content:
input spectrum
I
INPUT
OUTPUT
HT
output spectrum
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0
FREQUENCY
TRANSLATION
FILTERING
X( f ) + jX( f )
f
W/2
f
0
W/2
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f
FILTERING
f
0
f
0
fc
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Frequency Translation
One key operation in communications signal
processing is the frequency translation of a signal
spectrum from one center frequency to another.
Conversions between baseband and bandpass
representations (modulation and demodulation) are
special cases of this.
We consider two different ways to do the frequency
translation: mixing and multirate operations, i.e.,
decimation and interpolation.
In case of multirate operations, we assume for
simplicity that the following two sampling rates are
used:
low sampling rate:
fs
fs = 1
T
=1
NT
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jLOk
I
cos(Lok)
fc
sin(Lok)
fc+fLO
in which case the multiplying sequence is +1, -1, +1, -1, ...
This case can be applied to a real signal without
producing a complex result. Converts a lowpass signal
to a highpass signal, and vice versa.
f LO = f s / 4 = 1
4T
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f
0
Complex
bandpass
filter:
f
fc
( )
e j2f T
c
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fs/4
fs/4
fs/2
fs/2
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h0
h4
h3
h1
...
T
h5
T
h0
h1
h2
...
h3 h4
h5
I
Q
h0
Q
h1
h2
h3
h4
h5
...
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Contents
Complex signals and systems
o
o
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o
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Sampling Theorem
The sampling theorem says that a (real or complex)
lowpass signal limited to the frequency band [-W, W] can
represented completely by discrete-time samples if the
sampling rate (1/T) is at least 2W.
In case of a complex signal, each sample is, of course, a
complex number.
In general, discrete-time signals have periodic spectra,
where the continuous-time spectrum is repeated around
frequencies 1 T , 2 T , 3 T ,
2fs
fs
f
0
fs
2fs
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fs
fs
fs
f
0 fs=W
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n/NT1/T
1/NT
COMPLEX
BP-FILTER
RESPONSE
f
0
n/NT
b)
f
0
1/NT
f
0
n/NT
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n/NT1/T
f
0
1/NT
n/NT
f
0
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f
n1 fS /N- fS
n1 fS /N
f
0
f
0
n2 fS /N
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e j
(iii)
e j
BPF
e j
Filter
Mixer
Total
Case (i)
100
4
104
Case (ii)
100/6
2
18.7
Case (iii)
100/6
4/6
17.3
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fcfLO
fc
fc+fLO 0 fcfLO
fc+fLO
cos(LOt)
n
,
NT
NT
+1
2 NT
or
n
1
,
2 NT
NT
NT
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y(m)
N
Nf s
fs
X( f )
fs
fs/2
fs/2
fs
W( f)
k=3
k=2
k=1
k=0
k=0
k=1
k=2
k=3
f
0
Y ( f )| k= 2
k=2
k=2
f
0
Y ( f )| k= 3
k=3
k=3
f
0
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k=3
k=2
y(m)
k=1
f s /N
k=0
k=0
k=1
k=2
k=3
0
X BP( f )
3fs/(2N) fs/N
fs /N
3fs/(2N)
Y( f)
fs/N
fs /N
X BP( f )
2fs/N 3fs/(2N)
3fs/(2N) 2fs/N
Y( f)
fs/N
f
0
fs / N
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Taking the real part effectively reduces the rate of realvalued samples by two. It produces mirror images, in
contrast to the periodic images produced by
decimation by two. In both cases, the new spectral
components may fall on top of the existing spectral
components.
If this operation follows, e.g., an FIR filter, considerable
computational simplifications can be made by
combining the real part- operation with the filter in a
cleaver way. There is no sense to compute samples
that are thrown away by the real part operation!!.
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Example of Down-Conversion:
I/Q-Demodulation
N
4
PHASE
SPLITTER
MATCHED
FILTER
I
Q
f
0
f
0
2/T
f
0
1/T
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Contents
Complex signals and systems
o
o
o
o
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fcW/2
T/H
f
fc
kfsfs/2
f
kfs
kfs+fs/2
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Quadrature Sampling
In this case we are sampling the complex analytic signal
obtained by a phase-splitter:
I
T/H
fs
90
fcW/2
T/H
ANALYTIC
BANDPASS
SIGNAL
fc
kfs
(k+1)fs
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Second-Order Sampling
Quadrature sampling can be approximated by the following
structure:
I
T/H
fs
=1/4fc
T/H
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4 fc
f
2 = 1 +
rads
1
fc 2
( fc + f )
We are actually dealing with phase imbalance and the
image rejection formula for quadrature mixing can be
utilized. The resulting image rejection is:
f
1 cos
1 cos
fc 2
R=
=
1 + cos
f
1 + cos
fc 2
fc
0.0001
0.001
0.01
0.1
Phase
imbalance
0.009o
0.09o
0.9o
9o
Image rejection
82.1dB
62.1 dB
42.1 dB
22.1 dB
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REQUIRED
DYNAMIC
RANGE
f
STRONG NEIGHBORING
CHANNELS
WEAK DESIRED
CHANNEL
Sampling
The sampling to get a discrete time signal is done usually
with a track-and-hold circuit (T/H).
In practical sampling clocks and sampling circuits, there
are unavoidable random variations in the sampling
instants, sampling aperture jitter. In bandpass sampling,
the requirements for aperture jitter become very hard.
DCE - Tampere University of Technology. All rights reserved.
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(dB)
where n
is the number of bits
CFdB is the Crest Factor in dB
B
is the useful signal bandwidth
f s is the sampling rate
k
2 for baseband, 1 for bandpass sampling.
The crest factor for a voltage signal is defined as the ratio of the
peak absolute value and the RMS-value. The maximum SNR is
achieved when the signal utilizes the A/D-converters full voltage
range, which is assumed to be symmetric around DC. For a
sinusoidal signal, the crest factor is 3 dB, and for a bandpass
signal, the crest factor is 3 dB higher than that of the equivalent
baseband signal.
The last term takes into account the processing gain due to
oversampling in relation to the useful signal band. When the
quantization noise outside that useful signal band is filtered away,
the overall quantization noise power is reduced by the factor fs/kB.
Considering the choice between (real) baseband and bandpass
sampling, the difference is due to the 3 dB higher crest factor in
the bandpass sampling case. The k parameter is just due to the
definition of bandwidth. The bandwidths of B in baseband model
2B in bandpass model contain the same amount of signal power
and quantization noise power.
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Aperture Jitter
Aperture jitter is the variation in time of the exact
sampling instant, that causes phase modulation and
results in an additional noise component in the
sampled signal.
Aperture jitter is caused both by the sampling clock
and the sampling circuit.
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SNRaj = 20 log10
2 f maxTa
where fmax is the maximum frequency in the sampler input
and Ta is the RMS value of the aperture jitter.
This model is derived for a sinusoidal input signal, but
applied also more generally. In critical test cases of the
wideband sampling receiver application, the blocking signal
is often defined as a sinusoidal signal, and the model is
expected to work reasonably well.
The processing gain due to oversampling effects in the
same way as in case of quantization noise.
Example of Sampling Jitter Effects
110
- 14 bits
- 1 ps RMS jitter
100
SNR in dB
90
80
70
60
50
40
0
10
DCE - Tampere University of Technology. All rights reserved.
10
10
Frequency in MHz
10
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f
fc
f
fc
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Sigma-Delta Modulator - 1
e[n]
u[n]
x[n]
y[n]
DAC
z 1 1
U ( z) =
1
1 z 1 z 1
Y ( z) = U ( z) + E ( z)
L 1
z 1
( X ( z) Y ( z))
Y ( z)
1
1 z
Y ( z ) = z X ( z ) + (1 z
1
) E (z)
L
STF = z 1
NTF = (1 z 1 ) L
DCE - Tampere University of Technology. All rights reserved.
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Sigma-Delta Modulator - 2
NTF attenuates noise from the desired signal band. The inband quantization noise variance becomes
e2 =
SQ ( f ) NTF ( f ) df
2
2
=
12 f s
2
B sin ( f / f s ) df 12 f s
B
2L
2 2L 2B
=
12 ( 2 L + 1) f s
( f / f s )
2L
df
2 L +1
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Contents
Complex signals and systems
o
o
o
o
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Polyphase Decomposition
Any FIR filter transfer function can be expressed as
L 1
H ( z ) = z i H i ( z L )
i =0
=
H0(z2)
z 1
H1(z2)
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H ( z ) = ( z e jk 2 / L ) i H i ( z L )
i =0
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H0(z2)
z 1
H1(z2)
z 1
H0(z)
H1(z)
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H1(z)
...
fS
H(z)
fS
fS
fS /L
HL-1(z)
fS /L
(b)
(a)
Interpolation by L:
H0(z)
H1(z)
fS
fS
(a)
LfS
H(z)
...
HL-1(z)
LfS
(b)
LfS
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fS
HL-1(z)
with wi,k = e jk (2 )i / L ,
...
wL-1, k
fS /L
H1(z)
w1, k
i = 1, 2, ..., L 1.
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L-channel
Analysis
Filter
Bank
L-channel
Synthesis
Filter
Bank
Processing
x(n)
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20
40
60
0.1
0.2
0.3
0.4
0.5
0.6
Frequency /
0.7
0.8
0.9
20
40
60
0.1
0.2
0.3
0.4
0.5
0.6
Frequency /
0.7
0.8
0.9
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H0 (z)
Re[]
F0 (z)
H1 (z)
Re[]
F1 (z)
H2M 1 (z)
Re[]
F2M 1 (z)
x
(n)
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2
L + 1
1
+
h p ( n ) exp j n +
k
L
2
2
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Synthesis
Analysis
CMFB
CMFB
x
I (n)
xM 1 (m)
xI (n)
x2M 1 (m)
x
Q (n)
xQ (n)
Synthesis
Analysis
SMFB
SMFB
xM (m)
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F1 ( ) F0 ( ) F0+ ( ) F1+ ( )
FM+ 1 ( )
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Transmultiplexers
A transmultiplexer system includes also analysis and
synthesis banks, but in the reverse order. They are mostly
applied in communication systems for TDM FDM
transmultiplexing and for multicarrier modulation, as an
alternative to the widely adopted IFFT/FFT based OFDM
system.
In a transmultiplexer, a number of low-rate symbol
sequences are modulated to subcarriers and combined
into a wideband signal for transmission. In the analysis
bank of the receiver, the subcarrier signals are again
separated. In an ideal transmultiplexer, in case of an ideal
noise-free channel, the output subcarrier sequences are
exactly the same as the input sequences.
Perfect-reconstruction CMFBs, EMFBs, and MDFT banks
can directly be used to get ideal transmultiplexers.
The main benefits of filter bank based multicarrier
modulation over OFDM are due to the high frequency
selectivity: the system is very robust to narrowband
interferences and the overall spectrum has very sharp
transition bands, allowing to use very narrow guard-bands
in frequency domain. Also, the receiver filter bank can
implement a considerable part of the channel selection
filtering in a flexible manner.
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EMFBs as Transmultiplexer
Real transmultiplexers have real baseband signal, whereas
in the complex case, the wideband baseband signal is
complex, and it can be directly modulated to the RF carrier
in a spectrally efficient way.
CMFBs and SMFBs can again be used as building blocks
for critically-sampled complex transmultiplexers:
X
+
0
(z )
X 1+ ( z )
Synthesis:
Cosine
Modulated
Filter Bank
X 0+ ( z )
X 1+ ( z )
Analysis:
Cosine
Modulated
Filter Bank
I
+
M 1
(z )
X 1 ( z )
Channel
X 0 ( z )
M
M
+
M
Synthesis:
Sine
Modulated
Filter Bank
XM
1 ( z )
+
XM
1 ( z )
X 0 ( z )
X 1 ( z )
Analysis:
Sine
Modulated
Filter Bank
XM
1 ( z )
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I
Re{ . }
X k+ (m )
1
2
CMFB
SMFB
Synthesis
Analysis
Re{ . }
Re{ . }
X k+ (m )
Hlp(z)
X k (m )
SMFB
1
2
Synthesis
CMFB
Analysis
Q
Im{ . }
SMFB
X k (m )
Analysis
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