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M. Renfors, TUT/DCE

TLT-5806/IQ/1

21.9.2010

Complex and Bandpass Signals

Markku Renfors

Department of Communications Engineering

Tampere University of Technology, Finland

markku.renfors@tut.fi

www.cs.tut.fi/tlt

M. Renfors, TUT/DCE

TLT-5806/IQ/2

21.9.2010

Contents

Complex signals and systems

o

o

o

o

Analytic signals and Hilbert transform

Frequency translation by mixing

Complex bandpass filters

signals

o Sampling theory for complex signals

o Multirate processing of real and complex bandpass signals

o Combining mixing and multirate operations for frequency

translation

issues

o

o

o

o

o

Quadrature sampling

Second-order sampling

Sampling jitter and quantization noise

Sigma-delta ADC

o Polyphase structures for decimation and interpolation

o Real and complex filter banks

M. Renfors, TUT/DCE

TLT-5806/IQ/3

21.9.2010

In telecommunications signal processing, it is common to

use the notion of complex signals.

Continuous- and

denoted here as

discrete-time

x (t ) = x R (t ) + jx I (t )

complex

signals

are

x ( k ) = x R ( k ) + jx I ( k )

more than two separate real signals carrying the real and

imaginary parts.

A complex linear time-invariant system is represented by

two real impulse responses

h( k ) = hR ( k ) + jhI ( k )

H (e j ) = H R (e j ) + jH I ( e j )

complex signal, four separate real filters need to be

implemented:

y ( k ) = x ( k ) h( k ) = ( x R ( k ) + jx I ( k )) (hR ( k ) + jhI ( k ))

= x R ( k ) hR ( k ) x I ( k ) hI ( k ) + j ( x R ( k ) hI ( k ) + x I ( k ) hR ( k ))

M. Renfors, TUT/DCE

TLT-5806/IQ/4

21.9.2010

Real Part -Operation

Taking the real part of the complex signal produces mirror

images of all the spectral components:

Taking real part

with any of the original spectral components:

Taking real part

f

0

f

0

M. Renfors, TUT/DCE

TLT-5806/IQ/5

21.9.2010

Analytic Signal and Hilbert Transform

Hilbert transformer is generally defined as an allpass linear

filter which shifts the phase of its input signal by 90

degrees. The (anticausal) impulse and frequency

responses can be formulated as

discrete-time

continuous-time

n even

0,

1

hHT (n) =

hHT (t ) =

t

2 /( n), n odd

j, f 0

j, 0 <

j

H HT ( f ) =

H HT (e ) =

j

f

,

0

+

<

+ 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

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/6

21.9.2010

Real signal spectrum, the corresponding analytic signal

spectrum, and the analytic signal spectrum translated to be

centered at 0-frequency:

X( f )

f

0

FREQUENCY

TRANSLATION

FILTERING

X( f ) + jX( f )

f

W/2

f

0

W/2

symmetric, transition band around 0-frequency.

One practical example is VSB (vestigial side-band)

modulation, the baseband model of which is obtained by a

Hilbert filter with symmetric transition band.

M. Renfors, TUT/DCE

TLT-5806/IQ/7

21.9.2010

Real bandpass signal and the corresponding ideal analytic

bandpass signal:

f

FILTERING

f

0

practical phase splitter with finite attenuation (e.g., for

improving

image

attenuation

in

case

of

I/Q

downconversion):

f

0

fc

allpass filter which approximates 90 phase shift in the

passband and stopband with sufficient accuracy,

depending on the stopband attenuation requirement.

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/8

21.9.2010

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

M. Renfors, TUT/DCE

TLT-5806/IQ/9

21.9.2010

y (k ) = e j 2 f LO kT x(k ) = e jLO k x(k )

Special case with

real input signal:

jLOk

I

cos(Lok)

fc

sin(Lok)

fc+fLO

by f LO .

Important special cases are:

f LO = f s / 2 = 1

2T

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

+1, j, -1, -j, +1, j, ...

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/10

21.9.2010

Certain types of complex filters based on Hilbert

transformers can be designed using standard filter design

packages, like Parks-McClellan routine for FIR filters.

Another way to get complex bandpass filters is through

frequency translations:

Real

prototype

filter:

f

0

Complex

bandpass

filter:

f

fc

function:

H e j H e j ( c )

H ( z ) H ze j 2f cT

( )

T

e j2f T

c

than in the general case, see the special cases of the

previous page.

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M. Renfors, TUT/DCE

TLT-5806/IQ/11

21.9.2010

Frequency Translated FIR

Frequency translation by fs/4 => Analytic bandpass

filter with passband around fs/4.

fs/4

fs/4

h0, h1, h2, h3, h4, , hN

fs/2

fs/2

M. Renfors, TUT/DCE

TLT-5806/IQ/12

21.9.2010

(i) Real input signal

I

h2

h0

h4

h3

h1

...

T

h5

I

T

h0

h1

h2

...

h3 h4

h5

I

Q

h0

Q

h1

h2

h3

h4

h5

...

coefficient symmetry (of linear phase FIR) in both

cases.

M. Renfors, TUT/DCE

TLT-5806/IQ/13

21.9.2010

Contents

Complex signals and systems

o

o

o

o

Analytic signals and Hilbert transform

Frequency translation by mixing

Complex bandpass filters

signals

o Sampling theory for complex signals

o Multirate processing of real and complex bandpass signals

o Combining mixing and multirate operations for frequency

translation

issues

o

o

o

o

o

Quadrature sampling

Second-order sampling

Sampling jitter and quantization noise

Sigma-delta ADC

o Polyphase structures for decimation and interpolation

o Real and complex filter banks

M. Renfors, TUT/DCE

TLT-5806/IQ/14

21.9.2010

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

original signal is located symmetrically around 0.

Any part of the periodic signal can be considered as the

useful part. This allows many possibilities for multirate

processing of bandpass signals.

In general, the key criterion is that no destructive aliasing

effect occur on top of the desired part of the spectrum.

M. Renfors, TUT/DCE

TLT-5806/IQ/15

21.9.2010

Real signal:

fs

fs

represent the signal.

Complex signal:

fs

f

0 fs=W

The resulting rates of real-valued samples are the

same.

However, the quantization effects may be quite

different. (Recall from the standard treatment of

SSB that Hilbert-transformed signals may be

difficult.)

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M. Renfors, TUT/DCE

TLT-5806/IQ/16

21.9.2010

Sampling rate increase produces a periodic

spectrum, and the desired part of the spectrum is

then separated by an (analytic) bandpass filter.

N

a)

f

0

n/NT1/T

1/NT

COMPLEX

BP-FILTER

RESPONSE

f

0

n/NT

b)

f

0

1/NT

f

0

n/NT

M. Renfors, TUT/DCE

TLT-5806/IQ/17

21.9.2010

Sampling rate decrease produces aliasing, such that

the original band is at one of the image bands of the

resulting final band.

The signal has to be band-limited to a bandwidth of

1 / NT before this operation can be done without

severe aliasing effects.

N

COMPLEX

BP-FILTER

RESPONSE

n/NT1/T

f

0

1/NT

n/NT

f

0

M. Renfors, TUT/DCE

TLT-5806/IQ/18

21.9.2010

Complex Signal

Combining decimation and interpolation, a frequency shift

by n / NT can be realized, where n is an arbitrary integer.

N

f

n1 fS /N- fS

n1 fS /N

f

0

f

0

n2 fS /N

higher than the signal bandwidth, determines the resolution

of the frequency translations based on multirate

operations.

If, for example, a bandpass signal is desired to be

translated to the baseband form, this can be done using

multirate operations if and only if the carrier frequency is a

multiple of the low sampling rate.

Using also simple frequency translations (with coefficients

+1, -1, +j, -j), the resolution is 1/(4NT).

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/19

21.9.2010

for Complex Signals

A general frequency shift of f O = n NT + f can be done

in the following two ways:

(1) Direct frequency conversion by f O using mixing.

(2) Conversion using multirate operations by n NT

followed by a mixing with f (or vice versa).

The differences in these two approaches are due to

the possible filtering operations associated with the

multirate operations, and aliasing/reconstruction

filters in case of mixed continuous-time/discrete-time

processing.

Assuming ideal filtering, these two ways would be

equivalent.

M. Renfors, TUT/DCE

TLT-5806/IQ/20

21.9.2010

Operations

Conversion from bandpass to baseband representation

and decimation to symbol rate, i.e., I/Q-demodulation.

Assume that

- N=6, f0=4/(6T)+f.

- The required complex bandpass filter is obtained from

an FIR filter of length 50 by frequency translation.

The following three ways are equivalent but lead to

different computational requirements (the required real

multiplication rates at input rate are shown, not exploiting

possible coefficient symmetry):

BPF

LPF

(ii)

(i)

N

N

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

M. Renfors, TUT/DCE

TLT-5806/IQ/21

21.9.2010

Operations (continued)

Notes:

(i) Complex bandpass filter, real inputs

=> 100 real multipliers needed for filter

(ii) Real lowpass filter, complex input to filter

=>100 real multipliers needed for filter

- Decimation can be combined efficiently with the

filter. Utilizing coefficient symmetry is easiest in

this case.

(iii) As (i) but decimation can be included efficiently

with the filter.

- Mixing and LO generation done at lower rate

and thus easier to implement.

Here we have not taken use of the possible

coefficient symmetry, which may reduce the

multiplication rates by 1/2 in all cases.

In general, mixing is a memoryless operation, so upsampling and down-sampling operations can be

commuted with it in block diagram manipulations.

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/22

21.9.2010

Mixing and multirate operations can be done in similar way

for real signals. The difference is that the two parts of the

spectrum, on the positive and negative frequency axis, and

their images, must be accommodated in the spectrum.

(1) Mixing

Mixing produces two translated spectral components (note

that cos(t ) = (e jt + e jt ) / 2 ). The image band appearing

on top of the desired band after mixing must be

suppressed before mixing.

fc

fcfLO

fc

fc+fLO 0 fcfLO

fc+fLO

cos(LOt)

In case of decimation, to avoid destructive aliasing effects,

the signal to be translated must be within one of the

intervals

n

,

NT

NT

+1

2 NT

or

n

1

,

2 NT

NT

NT

the spectrum will be inverted.

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M. Renfors, TUT/DCE

TLT-5806/IQ/23

21.9.2010

w(m)

x(n)

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

M. Renfors, TUT/DCE

TLT-5806/IQ/24

21.9.2010

xBP(n)

x(n)

fs

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

M. Renfors, TUT/DCE

TLT-5806/IQ/25

21.9.2010

Multirate Operations

The transformations between real and complex signal

formats can be seen as multirate operations:

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!!.

filter bank) increases the rate of real-valued samples

by two. Mirror images are removed form the spectrum,

in contrast to the periodic images that are removed in

interpolation.

M. Renfors, TUT/DCE

TLT-5806/IQ/26

21.9.2010

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

as long as possible, because after converting to complex

form, all subsequent signal processing operations require

(at least) double computational capacity compared to the

corresponding real algorithms.

M. Renfors, TUT/DCE

TLT-5806/IQ/27

21.9.2010

Contents

Complex signals and systems

o

o

o

o

Analytic signals and Hilbert transform

Frequency translation by mixing

Complex bandpass filters

signals

o Sampling theory for complex signals

o Multirate processing of real and complex bandpass signals

o Combining mixing and multirate operations for frequency

translation

issues

o

o

o

o

o

Quadrature sampling

Second-order sampling

Sampling jitter and quantization noise

Sigma-delta ADC

o Polyphase structures for decimation and interpolation

o Real and complex filter banks

M. Renfors, TUT/DCE

TLT-5806/IQ/28

21.9.2010

Down-conversion can also be implemented by sampling a

bandpass signal. Any part of the periodic spectrum can be

selected for further processing.

fs

fcW/2

T/H

f

fc

kfsfs/2

f

kfs

kfs+fs/2

aliasing appears on top of the desired band.

In general, the feasible sampling frequencies are

determined from W, B (useful signal bandwidth), and fs.

Minimum sampling frequency is B+W, which is adequate in

the case where the center frequency of the desired signal

is kfs fs/4:

W+B=fs

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M. Renfors, TUT/DCE

TLT-5806/IQ/29

21.9.2010

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

shown periodic replicas are not completely attenuated but

only suppressed to a level that is determined by the

amplitude and phase imbalances in the phase splitter &

sampler & ADC blocks.

The gain and phase imbalance analysis of quadrature

down-conversion applies also to this case.

M. Renfors, TUT/DCE

TLT-5806/IQ/30

21.9.2010

Second-Order Sampling

Quadrature sampling can be approximated by the following

structure:

I

T/H

fs

=1/4fc

T/H

corresponds exactly to the 90o phase shift. Farther away

from the center frequency this is only approximative, but

for relatively narrowband signals, it works. The nonideality

can be evaluated using the phase imbalance analysis.

M. Renfors, TUT/DCE

TLT-5806/IQ/31

21.9.2010

The second-order sampling concept works perfectly at the

carrier frequency (ignoring the other sources of I/Q

imbalance) but only approximately at other frequencies. At

frequency f c + f , a time-shift of 1 (4 f c ) corresponds to a

phase shift of

1

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

f

0.1 MHz

1 MHz

10 MHz

100 MHz

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

M. Renfors, TUT/DCE

TLT-5806/IQ/32

21.9.2010

Analog to Digital Converter (ADC)

Sampling a wideband signal, containing several channels

is a tempting approach for designing a flexible radio

receiver. However, there are some great challenges to do

this.

The strongest signal in the ADC input signal band should

be in the linear range of the ADC. When the desired signal

is weak, a large ADC dynamic range is needed, the

resolution of the converter has to be many bits, e.g., 14 ...

17 bits.

MAGNITUDE

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.

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M. Renfors, TUT/DCE

TLT-5806/IQ/33

21.9.2010

In general, the maximum S/N-ratio for an A/D-converter is

(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.

M. Renfors, TUT/DCE

TLT-5806/IQ/34

21.9.2010

Lets consider now the case of complex I/Q sampling. In most

cases, half of the overall signal power is captured by both of the

ADCs in the I and Q branches. Thus the signal to quantization

noise power ratio is the same in both ADCs as that of the overall

complex signal. Also the processing gain is the same as in the

real case, if the bandwidths are defined in the same way.

However, the full peak value of the waveform may appear in both

of them. Thus the crest factor should be defined as the ratio of

the peak complex signal magnitude divided by the rms value of

the I or Q signal.

The number of additional bits needed to quantize a wideband

signal, containing strong spectral components in addition to the

wanted signal, can be estimated by:

CF 2 P

B / 6 bits

10log10 B

CFd 2 Pd

where PB is the worst case power in the full band

Pd is the minimum useful signal power

CFB is the crest factor in the worst case test situation

CFd is the crest factor of the desired signal.

Usually, in communications receivers, the worst case power is

determined from the adjacent channel or blocking specifications.

For many types of communications signals (e.g., CDMA, OFDM),

the crest factor is much higher than that of the sinusoid, which

might be a valid assumption in the blocking test situation.

B

M. Renfors, TUT/DCE

TLT-5806/IQ/35

21.9.2010

Practical ADC's have also discrete spectral frequency

components, spurious signals (or spurs), in addition to the

flat quantization noise.

In many applications, the spurious-free dynamic range,

SFDR, is the primary measure of the dynamic range of the

converter.

M. Renfors, TUT/DCE

TLT-5806/IQ/36

21.9.2010

Advanced bandpass sampling approaches could

mean that we are sampling a tens-of-MHz to GHzrange signal with a relatively low sampling rate.

Noise Aliasing

aliased to the signal band. In case of bandpass

sampling, aliasing increases with increasing

subsampling (fc/fs) factor. Basically, the noise figure

depends on the subsampling factor.

Therefore, it is important to have a good noise figure

for the track&hold circuit and/or to have sufficient

amplification in the analog front-end.

M. Renfors, TUT/DCE

TLT-5806/IQ/37

21.9.2010

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.

M. Renfors, TUT/DCE

TLT-5806/IQ/38

21.9.2010

The noise produced by aperture jitter is usually modelled

as white noise, which results in a signal-to-noise ratio of

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

Quantization effect

Joint effect

100

SNR in dB

90

80

70

60

50

40

0

10

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10

10

Frequency in MHz

10

M. Renfors, TUT/DCE

TLT-5806/IQ/39

21.9.2010

It is obvious that the requirements for the T/H-circuit and

A/D-converter are the main bottlenecks for implementing

receiver selectivity with DSP.

One promising A/D-converter technology in this context is

the sigma-delta () principle.

- This principle involves low-resolution, high-speed

conversion in a noise-shaping configuration, together

with decimating noise filtering.

- In case of lowpass and bandpass sampling with suitable

fixed center frequency, this principle can be combined

nicely with the selectivity filtering part of the receiver.

Noise filtering in basic ADC:

f

fc

f

fc

M. Renfors, TUT/DCE

TLT-5806/IQ/40

21.9.2010

Sigma-Delta Modulator - 1

e[n]

u[n]

x[n]

y[n]

DAC

sequence e[n] models the quantization noise of a single-bit

(or few-bit) ADC included in the loop. In an Lth-order

modulator, the first integrator is repeated L-1 times.

The system has different transfer functions for input signal

and quantization noise. They can be analyzed as follows:

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

from this expression as:

STF = z 1

NTF = (1 z 1 ) L

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/41

21.9.2010

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

Doubling the oversampling ratio Fs/2B, the noise is

decreased by the factor 3(2L+1) in dB.

The number of quantization bits can be reduced by

increasing the oversampling ratio.

Noise can be filtered out by decimating digital filters.

M. Renfors, TUT/DCE

TLT-5806/IQ/42

21.9.2010

Contents

Complex signals and systems

o

o

o

o

Analytic signals and Hilbert transform

Frequency translation by mixing

Complex bandpass filters

signals

o Sampling theory for complex signals

o Multirate processing of real and complex bandpass signals

o Combining mixing and multirate operations for frequency

translation

issues

o

o

o

o

o

Quadrature sampling

Second-order sampling

Sampling jitter and quantization noise

Sigma-delta ADC

o Polyphase structures for decimation and interpolation

o Real and complex filter banks

M. Renfors, TUT/DCE

TLT-5806/IQ/43

21.9.2010

Polyphase Decomposition

Any FIR filter transfer function can be expressed as

L 1

H ( z ) = z i H i ( z L )

i =0

length M / L or M / L , where M is the length of the

original FIR filter.

Example: Polyphase structure with L=2:

=

H0(z2)

z 1

H1(z2)

above form. Especially, the Nth-band IIR filters have

polyphase decomposition where the polyphase branches

are allpass filters.

M. Renfors, TUT/DCE

TLT-5806/IQ/44

21.9.2010

Complex Bandpass Filters

Including a frequency shift by kfs/L, the polyphase filter can

be represented as

L 1

H ( z ) = ( z e jk 2 / L ) i H i ( z L )

i =0

shifting do not appear in the polyphase branches, but only

in the separate unit delay chain.

If this part of the structure is on the output side, then just

replicating this structure with different values of k, we

obtain bandpass filters with the same input but different

center frequencies, and sharing the main part of the

computations.

In this way we can implement filter banks that split the

input signal into a number of narrower signal bands.

Polyphase structures are mostly considered in multirate

signal processing applications, as discussed on the

following pages. Here we want to point out that they can

be useful even in cases where sampling rate conversion is

not involved, e.g., for implementing complex bandpass

filters and filter banks.

M. Renfors, TUT/DCE

TLT-5806/IQ/45

21.9.2010

Decimation and Interpolation Filters

The polyphase structure with sampling rate reduction by

two looks as follows:

2

H0(z2)

z 1

H1(z2)

developed as follows:

z 1

H0(z)

H1(z)

model the operation of connecting even-indexed input

samples to the upper branch and odd-indexed ones two

the lower branch.

M. Renfors, TUT/DCE

TLT-5806/IQ/46

21.9.2010

Decimation by L:

H0(z)

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

M. Renfors, TUT/DCE

TLT-5806/IQ/47

21.9.2010

Filter Banks

As mentioned already, we can easily embed in the

polyphase structure the complex coefficients needed for

frequency translation. In the decimation case, the

bandpass polyphase structure looks as follows:

H0(z)

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.

complex multipliers and output adder with different values

of k. If all (or even several) channels are need, this can be

done very efficiently using FFT operation. Such a structure

is generally called DFT filter bank.

The decimation case corresponds to analysis bank, in

which the wideband signal is divided into L subchannels

with sample rate fs/L.

Likewise, the interpolation case corresponds to a synthesis

bank, in which several narrowband signals are combined

into a single wideband signal.

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/48

21.9.2010

In critically sampled L-channel filter banks, the sampling

rate conversion factor is L. For every L input samples of a

(decimating) analysis bank, one sample is generated to

each of the low-rate subbands.

An analysis-synthesis system includes analysis and

synthesis banks, connected by a subband processing

stage.

L

L-channel

Analysis

Filter

Bank

L-channel

Synthesis

Filter

Bank

Processing

x(n)

absence of subband processing, the synthesis bank output

is an exact delayed replica of the analysis bank input. With

proper filter bank design, perfect reconstruction can be

achieved even in the critically sampled case, even though

aliasing takes place in the subband signals. These must

have considerably overlapping transition bands in practical

designs.

Cosine-modulated filter banks (CMFBs) are a common and

efficient solution for real critically-sampled uniform filter

banks.

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/49

21.9.2010

Probably the computationally most efficient filter banks are

FFT (as analysis bank) and IFFT (as synthesis bank). The

benefit of filter banks is better frequency selectivity of the

subbands, at the cost of somewhat higher complexity.

Example: Subband frequency responses for L=16

channels

(a) FFT:

Amplitude in dB

20

40

60

0.1

0.2

0.3

0.4

0.5

0.6

Frequency /

0.7

0.8

0.9

Amplitude in dB

20

40

60

0.1

0.2

0.3

0.4

0.5

0.6

Frequency /

0.7

0.8

0.9

here consist of the prototype filter in polyphase or lattice

form, and a transform block, e.g., discrete cosine/sine

transform, or DFT/FFT.

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/50

21.9.2010

The structure of the real critically sampled filter bank

cannot be generalized to the complex case as such. Using

L-channel banks and sampling rate conversion by L,

together with complex subband signals leads to perfect

reconstruction only in case of the DFT-IDFT pair.

The solution is to use 2M-channel banks, sampling rate

conversion by M, and real subband signals.

Exponentially modulated filter bank (EMFB):

x(n)

H0 (z)

Re[]

F0 (z)

H1 (z)

Re[]

F1 (z)

H2M 1 (z)

Re[]

F2M 1 (z)

x

(n)

M. Renfors, TUT/DCE

TLT-5806/IQ/51

21.9.2010

In the EMFB case, the sub-channel impulse responses of

the synthesis bank are obtained from a low-pass prototype

filter hp(n) as follows:

f k ( n ) = f kc ( n ) j f ks ( n ) =

2

L + 1

1

+

h p ( n ) exp j n +

k

L

2

2

positive/negative sides of the spectrum, respectively. The

analysis bank impulse responses are related to the

synthesis bank as follows:

hk ( n ) = f kc ( N 1 n ) j f ks ( N 1 n ) = hkc ( n ) j hks ( n )

Critically-sampled perfect-reconstruction EMFB systems

can be obtained by using the same prototype filters as for

CMFBs.

M. Renfors, TUT/DCE

TLT-5806/IQ/52

21.9.2010

EMFBs can be implemented efficiently using CMFBs and

SMFBs as building blocks (also an efficient FFT-based

structure is available):

x0 (m)

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)

M. Renfors, TUT/DCE

TLT-5806/IQ/53

21.9.2010

EMFBs use so-called odd channel stacking arrangement:

FM 1 ( )

F1 ( ) F0 ( ) F0+ ( ) F1+ ( )

FM+ 1 ( )

or /2, and the other side of the spectrum is empty. This is

also important for perfect reconstruction, since now there is

room for the mirror image produced by the real-part

operation. Then, only the transition bands of the two

spectral components are overlapping. What basically

happens is that a real subband signal is constructed from

its analytic version.

Also even channel stacking arrangement can be used in

complex filter banks. In this case, the subbands are

centered at k/M, and after decimation they are centered at

0 or . In this case, the real part operation cannot be done

as such for the subband signals, because the mirror image

would fall on top of the decimated subband signal.

One well known class of complex filter banks using even

channel stacking are the modified DFT (MDFT) filter banks.

MDFT banks use slightly more complicated processing of

the subband signals to achieve perfect reconstruction. This

leads to more complicated relation between the original

spectrum and the subband signal spectra.

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/54

21.9.2010

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.

M. Renfors, TUT/DCE

TLT-5806/IQ/55

21.9.2010

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 )

conversion factor M. In the critically-sampled case, the

subchannel signals are real. In the modulation sense, VSB

modulation or offset-QAM modulation take place for the

subcarrier signals.

M. Renfors, TUT/DCE

TLT-5806/IQ/56

21.9.2010

In a analysis-synthesis or transmultiplexer system, aliasing

and imaging are taking place at different processing

stages, but the effects cancel each other and perfect

reconstruction is achieved in ideal conditions.

However, in case of transmultiplexers, non-ideal,

frequency-selective transmission channel destroys the

orthogonality of the subcarrier signals, introducing

intercarrier interference (ICI, crosstalk) between adjacent

subchannels and intersymbol interference (ISI) within each

subchannel. These effects cannot be compensated

effectively by subband-wise processing. Thus, in the

critically sampled case, the channel equalizers after

analysis bank have to include connections between

adjacent subchannels.

Usually in the filter bank design, the roll-off is no more than

100 %, i.e., the overall subchannel bandwidth is twice the

channel spacing, or less. If the subchannel signals can be

obtained in 2x -oversampled form, then they are essentially

alias free (the alias components are attenuated at least by

the stopband attenuation of the channel filters). Then, in

principle, it becomes possible to do the channel

equalization for each subcarrier independently of the

others.

In the EMFB, there is a simple way to obtain 2x

oversampling for the subchannel signals: using the

complex subchannel signals instead of the real ones that

are sufficient in the critically sampled case.

DCE - Tampere University of Technology. All rights reserved.

M. Renfors, TUT/DCE

TLT-5806/IQ/57

21.9.2010

2x-Oversampled Analysis Bank and PerSubcarrier Equalization

CMFB

Analysis

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

bank is oversampled by 2, and after subchannel

equalizers, symbol-rate sequences are obtained by taking

the real part of the equalizer output (usually, the real part

operation can be efficiently combined with the equalizer).

M. Renfors, TUT/DCE

TLT-5806/IQ/58

21.9.2010

System

One good application of analysis-synthesis FB systems is

in frequency-domain channel equalization (FDE) in

communication systems.

FFT-IFFT -based FDEs have received a lot of attention

recently because, in wideband transmission, they result in

significantly

lower

complexity

than

time-domain

equalization. FFT-based FDEs use commonly a cyclic

prefix, in the same way as OFDM, to achieve simple and

robust system.

Using analysis-synthesis FB system in FDE, instead of

FFT-IFFT, has a number of benefits due to the good

frequency selectivity, as mentioned already in the

multicarrier context. Also here, in the critically sampled

case, subband-wise equalization results in poor

performance, and cross-connections between subbands

would be needed.

A 2x -oversampled EMFB-type analysis bank can be

realized in the same manner as in the transmultiplexer

case. This makes it feasible to do the channel equalization

subband-wise. Then the real parts of the subband

equalizer outputs are connected to a critically sampled

synthesis bank.

Naturally, analysis-synthesis systems with oversampled

subband processing could be a useful tool for various other

applications.

DCE - Tampere University of Technology. All rights reserved.