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3/8/2017 SamplingTheoremBandpassorIntermediateorUnderSamplingGaussianWaves

Published July 28, 2011 by Mathuranathan

Sampling Theorem Bandpass or


Intermediate or Under Sampling
(4 votes, average: 4.50 out of 5)

Prerequisite: Sampling theorem baseband


sampling
Intermediate Sampling or Under-Sampling

Asignal is a bandpass signal if we can fit all its frequency content inside
abandwidth Fb . Bandwidth is simply the difference between the lowest
and the highest frequency present in the signal.

In order for a faithful reproduction and reconstruction of a bandpass


analog signal with bandwidth Fb , the signal should be sampled at a
Sampling frequency (Fs ) that is greater than or equal to twice the
maximum bandwidth of the signal.

Fs 2Fb

Consider a bandpass signal extending from 150Hz to 200Hz. The


bandwidth of this signal is Fb = 200H z 150H z = 50H z. In order to
faithfully represent the above signal in the digital domain the sampling
frequency must be Fs >= 100H z (2 F b) . Note that the sampling
frequency 100Hz is far below the maximum content of the signal (which is
200Hz). That is why the bandpass sampling is also called under-sampling.
As long as the sampling frequency is greater than or equal to twice the

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3/8/2017 SamplingTheoremBandpassorIntermediateorUnderSamplingGaussianWaves

bandwidth of the signal, the reconstruction back to analog domain will be


error free.

Going back to the aliasing zone figure, if the signal of interest is in the zone
other than zone 1, it is called a bandpass signal and the sampling operation
is called Intermediate Sampling or Harmonic Sampling or Under
Sampling or Bandpass Sampling.

Folding Frequencies and Aliasing Zones

Note that zone 1 is a mirror image of zone 2 (with frequency reversal).


Similarly zone 3 is a mirror image of zone 4 etc.., Also, any signal in zone 1
will be reflected in zone 2 with frequency reversal which inturn will be
copied in zone 3 and so on.

Lets say the signal of interest lies in zone 2. This will be copied in all the
other zones. Zone 1 also contains the sampled signal with frequency
reversal which can be correct by reversing the order of FFT in digital
domain.

No matter in which zone the signal of interest lies, zone 1 always contains
the signal after sampling operation is performed. If the signal of interest
lies in any of the even zones, zone 1 contains the sampled signal with
frequency reversal. If the signal of interest lies in any of the odd zones,
zone 1 contains the sampled signal without frequency reversal.

Example:

Consider an AM signal centered at carrier frequency 1MHz, with two


components offset by 10KHz 0.99 MHz and 1.01 MHz. So the AM signal
contains three frequency components at 0.99 MHz, 1 MHz and 1.01 MHz.

Our desire is to sample the AM signal. As with the usual sampling theorem
(baseband), we know that if we sample the signal at twice the maximum

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3/8/2017 SamplingTheoremBandpassorIntermediateorUnderSamplingGaussianWaves

frequency i.e Fs>=2*1.01MHz=2.02 MHz there should be no problem in


representing the analog signal in digital domain.

By the bandpass sampling theorem, we do not need to use a sampler


running at Fs>=2.02 MHz. Faster sampler implies more cost. By applying
the bandpass sampling theorem, we can use a slower sampler and reduce
the cost of the system. The bandwidth of the signal is 1.01MHz-0.99 MHz =
20 KHz. So, just sampling at Fs >= 2 20K H z = 40K H z will convert
the signal to digital domain properly and we can also avoid using an
expensive high rate sampler (if Fs >= 2.02M H z used according to
baseband sampling theorem).

Lets set the sampling frequency to be Fs = 120K H z (which is 3 times


higher than the minimum required sampling rate of 40KHz or
oversampling rate =3).

Now we can easily find the position of the spectral components in the
sampled output by using the aliasing zone figure as given above. Since
,
Fs = 120K H z Fs /2 will be 60KHz. So the zone 1 will be from 0 to 60
KHz, zone 2 -> 60-120KHz and so on. The three spectral components at
0.99MHz, 1MHz and 1.01 MHz will fall at zone 17 ( how ? 0.99 MHz/60 KHz
= 16.5 , 1MHz/60KHz = 16.67 and 1.01MHz/60KHz = 16.83 , all figures
approximating to 17). By the aliasing zone figure, zone 16 contains a copy
of zone 17, zone 15 contains a copy of zone 16, zone 14 contains a copy of
zone 15 and so on Finally zone 1 contains the copy of zone 2 (Frequency
reversal also exist at even zones). In effect, zone 1 contains a copy of zone
17. Since the original spectral components are at zone 17, which is an odd
zone, zone 1 contains the copy of spectral components at zone 17 without
frequency reversal.

Since there is no frequency reversal, in zone 1 the three components will


be at 30KHz, 40KHz and 50KHz (You can easily figure this out ).

This operation has downconverted our signal from zone 17 to zone 1


without distorting the signal components. The downconverted signal can
be further processed by using a filter to select the baseband
downconverted components. Following figure illustrates the concept of
bandpass sampling.

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Bandpass Sampling

Consider the same AM signal with three components at 0.99MHz, 1MHz


and 1.01 MHz. Now we also have an unwanted fourth component at 1.2
MHz along with the incoming signal. If we sample the signal at 120 KHz, it
will cause aliasing (because the bandwidth of the entire signal is 1.2-0.99 =
0.21 MHz = 210 KHz and the sampling frequency of 120KHz is below twice
the bandwidth). In order to avoid anti-aliasing and to discard the
unwanted component at 1.2 MHz, an anti-aliasing bandpass filter has to be
used to select those desired component before performing the sampling
operation at 120KHz. This is also called pre-filtering. The following figure
illustrates this concept.

Bandpass sampling with pre-filtering

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3/8/2017 SamplingTheoremBandpassorIntermediateorUnderSamplingGaussianWaves

See also:

[1] Oversampling, ADC DAC Conversion,pulse shaping and Matched Filter


[2] Sampling Theorem Basics and Baseband Sampling

Recommended Books:

External Resources:

[1] Video Lectures on Signals and Systems, Filter Design and Transforms
[2] A Guide on Frequency Analysis and Manipulation using MATLAB
[3] A short presentation on Quantization
[4] Useful Matlab Functions and Scripts for Audio Signals and Systems
[5] Mathworks Designing a Sigma-Delta ADC from Behavioral Model to
Verilog and VHDL
[6] Anti-aliasing and quantization of audio signals and digital images using
Matlab

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Mathuranathan
Mathuranathan Viswanathan - Founder and Author @
gaussianwaves.com which has garnered worldwide
readership. He is a masters in communication
engineering and has 9 years of technical expertise in
channel modeling and has worked in various
technologies ranging from read channel design for hard
drives, GSM/EDGE/GPRS, OFDM, MIMO, 3GPP PHY layer
and DSL. He also specializes in tutoring on various
subjects like signal processing, random process, digital
communication etc.., LinkedIn Profile

L AT E ST A RT I C L E S S I G N A L P RO C E S S I N G

A N A L O G TO D I G I TA L C O N V E R S I O N BA N D PA S S SA M P L I N G D I G I TA L TO
ANALOG CONVERSION I N T E R M E D I AT E SA M P L I N G OV E R SA M P L I N G
SA M P L I N G T H E O R E M S I G N A L P RO C E S S I N G U N D E R SA M P L I N G

10Comments Gaussianwaves
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Jointhediscussion

RAJESH8monthsago
thankyou,nicelyexplainedcanyoutellmehowthesethreecomponent30KHz,40KHz
and50KHzappearedfrom0.99MHz,1MHzand1.01MHz?
Reply Share

Mathuranathan Mod >RAJESH 7monthsago

ForFs=120KHz,Fs/2willbe60KHz.Thenthezone1willbefrom0to60KHz.
http://www.gaussianwaves.com/2011/07/samplingtheorembandpassorintermediateorundersampling2/ 6/12
3/8/2017 SamplingTheoremBandpassorIntermediateorUnderSamplingGaussianWaves

Asexplainedinthepost,theactualfrequencies,0.99M,1M,1.01Mfallswillfallin
zone17
Sincethebandwidthofeachzoneis60KHz,theratiosofthesefrequenciesto
theBWare
[0.99M/60K,1M/60K,1.01M/60K]=[16.5,16.67,16.83]

Thus,
thefrequency0.99Mis0.5awayfromthebeginningofzone17
thefrequency1Mis0.67awayfromthebeginningofzone17
thefrequency1.01Mis0.83awayfromthebeginningofzone17

Translatingthistozone1frequencies(exactvaluescalculatedbymultiplying
theratiosandthemaxfrequencyinzone1)
0.5*60KHz=30KHz,
0.67*60KHz=40.2KHz
0.83*60KHz=49.8KHz

Iapproximatedthemas30KHz,40KHzand50KHz
Reply Share

alihassan>Mathuranathan6monthsago
dearMrMathuranathanIwantyoutohelpinmywifeprojecthowcan
contactyoupleaseitissourgent.
Reply Share

ArrozConCostra2yearsago
IwishIhadbeenexplainedthesetopicsintheeasyandintuitivewayyoudo.ThisFmax
vsBWthinghasalwaysbeeninmyheadandIhaveneverbeenabletofigureoutwhy
sometimesFswas2*Fmaxandsomeotherwas2*BW.Nowitallmakessense.
Thanks!
Reply Share

Mathuranathan Mod >ArrozConCostra 2yearsago

Thanksforyourwordsofencouragement.
Reply Share

MalikMehbob2yearsago
Whythefrequencyofcarrierwave
isalwayskeptatleast10timeshigherthenthehighestfrequencyofbaseband
signal?
Reply Share

Mathuranathan Mod >MalikMehbob 2yearsago

Thecarrierfrequencyshouldbegreaterthanthemessagefrequency,butnot
necessarily10times.
Thesignalqualitywillbegoodifthecarrierfrequencyisfargreaterthanthe
http://www.gaussianwaves.com/2011/07/samplingtheorembandpassorintermediateorundersampling2/ 7/12
3/8/2017 SamplingTheoremBandpassorIntermediateorUnderSamplingGaussianWaves
Thesignalqualitywillbegoodifthecarrierfrequencyisfargreaterthanthe
messagefrequency.Meaningthatthesignalquality(distinguishableatthe
receiver)for10xcarrierfrequencywillbebettercomparedtothesignalquality
with2xcarrierfrequency.
Reply Share

yasser3yearsago
aninterviewquestionaskedtomewasgivenasignalofBW500HZandFmaxis600
Hzwhatisminsamplingrate?1000or1200
Reply Share

Mathuranathan Mod >yasser 3yearsago

Hintforyoutosolve:SamplingTheoremspecifieshowtocomputethesampling
rateforgivenbandwidthNOTfrommaximumfrequency.

Whydoweconsiderbandwidthandnotmaximumfrequency?Practicalexample
below.

InPGSM900,thefrequencyrangeforuplinkis890.0915.0MHz.IfIgowith
maximumfrequency,thesamplingratewouldbe2*915MHz=1830MHz=1.83
GSPS.IsitpossibletohaveanAnalogtoDigitalConverter(ADC)withsucha
fastsampleratcheapercost?.Evenifwehavesuchahighspeedsampler,the
costwillbeprohibitive.Thesolutionistosamplethesignalwithrespecttogiven
bandwidth(ThankstoNyquistShannonTheorem).Thisgivestheminimum
samplingas2*(915890)=2*25Mhz=50MSPSonly.(GSPS=GigaSamples
PerSecond,MSPSMegaSamplesPerSecond)

Note:Theabovesampleisforillustrativepurposeonly.Foractualsamplingrate
employedinthedevices,pleaseconsultrespectivetechnologymanual.High
speedADCs(>1GSPS)areavailableinthemarket.Somehere:
http://www.ti.com/lsds/ti/d...
Reply Share

vutq3yearsago
oversamplinganalogsignal

Dearall

IhaveNsampleOFDMwhichireceivedfromDACsuchas

0.264050592243675160.064089949573707560.12561630116446681
0.258923396277778550.243541808380088740.11023471326677702
0.10510751730088040.1051075173008804

....

Iwouldliketooversamplingthesesamplesbymatlabcodeforsuitablewithmatrixsiein
optisystem(thesignalwillbetransmittooptisystemforsimulation...

http://www.gaussianwaves.com/2011/07/samplingtheorembandpassorintermediateorundersampling2/ 8/12
3/8/2017 SamplingTheoremBandpassorIntermediateorUnderSamplingGaussianWaves

BecausethesamplingrateDACisonly4GSPS,butthesamplingrateinoptisystemis
16GSPS.

Aftertransmitoverfiber,ireceivethesignalafterphotodectector,iwouldliketo
downsamplethesesamples,howcanidownsamplethemtoreceivetheorginalsample?

Manythanksforyoursupport
Reply Share

ALSOONGAUSSIANWAVES

HowtoInterpretFFTresultscomplex IntroductiontoMultipleAntenna
DFT,frequencybinsandFFTShift Systems
3commentsayearago 9comments3yearsago
MathuranathanThanksforyour mikelThankyouverymuchforyour
valuablefeedback.Iwilltrytoincorporate fastreply!
yoursuggestionstofurtherimprovethe
article.
ChirpSignalFrequencySweeping HowtointerpretFFTresultsobtaining
FFTandpowerspectraldensity magnitudeandphaseinformation
6comments3yearsago 2commentsayearago
MathuranathanThefrequencyofthe MathuranathanIdonothavethecode

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