Automatic Classification of Analogue Modulation Signals by

Statistical Parameters

Shintaro Taira* and Eiichiro Murakami

Second Research Center, Technical Research and Development Institute, Japan Defense Agency
1-2-24, Ikejiri, Setagaya-ku, Tokyo 154-8511, JAPAN

Abstra&Automatic classification of modulation signals the mean of the envelope. Nandi et al. [2] used the
plays an imporkmt role in communication applicationssuch spectrum symmetry around the ctier frequency, the
as an intelligent demodulator interference identiikation maximum value of the power spectral density of the
and monitoring, so many investigations have been carried instantaneous amplitude and the standard deviation
out in the past. In this paper, new automatic classification of instantaneous phase for the classification. Aisbet
procedure of analogue modulation signals including phase [3] proposed the statistical parameter of the envelope
continuous FSK signals is proposed. As for the unbiased by Gaussian noise. But there are some
~.. .
ation between flequency modulation signals and problems about those investigations. [1] supposed that
amplitude modulation signals and the classification among the modulating signal is the Gaussian noise in the
amplitude modulation signals, the statistical parameters of derivation of the classflcation parameter. But if the
signal envelope are used. For the classification among modulating signal is not Gaussian noise, the value of
frequency modulation signals, the compactness of parameter may change from the anticipated one. And
instantaneous fi’equency distribution is used. Good the discrimination between analogue modulation
claasiikation possibility have been ascertained by the signals and digital modulation signals is also
computer simulations when SNR > 10dB. And the necessary prior to the class~lcation of analogue
. . .
ckcrmnnation between aualogue modulation signals and modulation signals in the field use.
digital modulation signals based on the block pmceseing is In this paper, new automatic classification
alm discussed” procedure of analogue modulation signals, whose
modulating signals have an auto-regressive
1. INTRODU~ON characteristic (simulated speech signal), is discussed
Recently with the diversification of communication by the computer simulations. And the discrimination
styles, an universal demodulator which has the between analogue modulation signals and digital
intelligent function becomes needed. As one of the modulation signals based on the block processing is
functions which is necessm to such an universal also shown. In Section 2, the discrimination between
demodulator, there is an automatic classification of analogue modulation signals and digital modulation
modulation signals. Namely considering the situation signals is discussed. In Section 3, the Classflcation
where the transmitter changes the modulation types procedure of analogue modulation signals is shown,
adaptively in response im the environment, the and conclusions are presented in Section 4.
demodulator must recognize the opposite modulation
types in real time and select the suitable II. BLOCK PROCESSING
demodulation method. Estimation result of the symbol rate is available for
Automatic classification of modulation signals has the discrimination between analogue modulation
other applications besides the universal demodulator, signals and digital modulation signals, because
for example, interference identification, monitming analogue modulation signals do not have symbol rate
and so on. So many investigations have been carried originally and its estimation result will become an
out. [1-8] arbitrary value. In this study, the block demodulation
The investigations about the classification of method based on the block processing [9,10] is used
analogue modulation signals are as follows. Chan et al. for the estimation of symbol rate.
[1] developed the classification algorithm based on the Fig. 1 shows a block diagram of the block
ratio of the variance of the envelope to the square of demodulation method. Block demodulation method

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 follows.
Signal amplitudes are normalized by the maximum
MO]
values of each analysis blocks.
htmtion Noise component: additive white Gaussian noise
I 17.39 sampling points per symbol period (sampling
timing is asynchronous to symbol timing).
4096 sampling points per analysis block
Carrier offset and initial phase error : 3% of the true
value and rc/4
Cosine roll-off factor for band-limitation: 1.0 (only for
1
digital modulation signals)
Iler!tion al $wifid Numbu 10 times practices are carried out to evaluate the
classification possibility for each modulation signal
Fig.1. Block demodulation method. and SNR.
AM, FM, PM, DSB and SSB signals are expressed as
consists of three parts as follows, the quasi follows respectively.
synchronous orthogonal demodulator, the estimator of
s(t)= [1+ mx(t)lcos(2@ct) (1)
symbol points and the estimator of carrier
information.
Received signal is multiplied by a ftxed reference
(
s(t) = cos 2#Ct + Kf ~:mx(t)dr) (2)

carrier adjusted around the true carrier fi-equency s(t) = cos(2@ct + Kpx(d) (3)
( quasi synchronous orthogonal demodulator), so the
extracted in-phase(I) and quadrature-phase(Q) s(t) = x(0cos(2#ct) (4)
components have the camier frequency offset and
initial phase error. But the influence of those error s(t) = x(t)cos(2#Ct)p y(t) sin(2#Ct) (5)
components is eliminated by the calculation of square where ~ is the carrier frequency, x(O is the modulating
sum of I components and Q components., and only signal, m is the amplitude depth of AM signal, Kf is
symbol rate components remain. Square sum the frequency deviation coefficient of FM signal, and
components of each analysis block are processed by K, is the phase deviation coefficient of PM signal.
FFT algorithm to extract the symbol components. The And the frequency modulation index of FM signal,
maximum peak component and largest one among the D ,is defined as follows.
adjacent components in the frequency spectrum are
Kf [x(t)]maX
extracted as the symbol rate components, because the D= (6)
symbol timing is asynchronous to the sampling point. 2~x

Extracted symbol components are processed by where ~is the maximum frequency of the modulating
Inverse-FFT, and the points where arctangent signal.
between the real components and imaginary The Modulating signal of digital modulation signals
components of the IFFT outputs equals to z (symbol is generated by the FSR sequences expressed by
points) are decided by the 1inear i nt emo 1at i on. I+&+#3=0. For the modulating signal of analogue
Symbol rate is calculated by averaging the intervals modulation signals, the first-order auto-regressive
between those symbol points . After the estimation of model given by the following equation is used.
symbol rate, carrier offset and initial phase error are
x(t) = px(t – 1)+ n(t) (7)
estimated by the least square method and carrier
frequency is adjusted. Those calculations is omitted in
this study. where n@ is an uniform random number whose
The conditions for the computer simulations are as average is O. The value of o is decided so that -3dB

0-7803-5538-5/99/$10.00 (c) 1999 IEEE

band width of x(O is 4kHz. Sampling frequenq is s(t) = A(t) sin(h(t))+ n(t) (lo)
supposed to be about 278kHz in this simulation, so o
= A(t)cos(h(t))+
s’(t) n’(t) (11)
is chosen to be 0.914.
The estimation error of the symbol rate is defined as where n ‘(t) is the Hilbert transform of noise
the ratio of the standard deviation to the mean for component. The signal envelope &(@ is the square
10 analysis blocks. Fig.2 shows the results at SNR= sum of both components.
20dB and 50dB. For n/4 shift QPSK and 16QAM, the A2(t)+2A(t@)sid/i(t))+n’(t)c&(t)))
fit)=.?(t)+s’’(t)=
estimation errors are almost O because correct
+ ~2(~)+
d2(t)= A2(t)+ 2A(t)&)+ C2(t) (12)
estimation results are obtained for the digital
modulation signals. For the analogue modulation where
signals, the estimation errors are large and take an a(t) = n(t) sin(h(t))+ n’(t) cos(h(t)). (13)
arbitrmy values. Also for phase-continuous FSK
c2(t)= n2(t)+ ??’2(t). (14)
signals which is the gathering of the tone signals, the
correct estimation of the symbol rate is impossible. So The expected value of signal envelope is as follows.
FSK signals are also included in the object of the E(flt))= E(A2(t))+ 2.E(A(t)a(t))+ E(c2(t)) (15)
classification.
li’(A(ia(@) becomes O because signal is not correlated
~ .CLASSIFICA’I’IONOF AN.ALOGUE MODULATION with noise. So (15) becomes as follows.
SIGNAU3
E(&t))= E(A2(t))+ E(c2(t))= E(A2(t))+ 2& (16)
The discrimination between frequency modulation
signals (FSK, FM, PM) and amplitude modulation where Oz is the time averaged power of Gaussian
signals (AM, DSB, SSB) is carried out by the noise. The expected value of square of signal envelope
statistical parameter of the signal envelope. The is as follows.
derivation of the parameter is as follows. E@t))=<A’(t))+4E(A’(tMt)} 4~A’(tW))
The waveform of input signal is defined as follows.
+ 2~A2(tk2(t))+ 4~A(t~tk2(@+ lif4(t)) (17)
The following relations can be used because signal is
where A(fl is the instantaneous amplitude, ~ is the not correlated with noise.
carrier frequency, 0(t) is the modulation phase, 8 is E(A3(t)a(t))= O (18)
the initial carrier phase and n(fl is the Gaussian noise. E(A(t)a(t)c2(t))= O (19)
where
E(A2(t)a2(t))= o%(A2(t)) (20)
h(t)= 2njfct + @(t)+ e (9)
Signal component s(@ and its Hilbert transform s’(ij E(A2(t)c2(t))= 20-’E(A2(t)) (21)
are as follows.
E(c’(t))= E(n’(t))+ 2E(n2(t)n’2 (t))+ E(n’4 (t))
~ 0.9

z 0.8 =304 +204 + 304 =8G4 (22)

ii 0.7
u +Nvl
o 0.6 + FM
In (22), the relation 17(iz4(@)=l@”@))=3 ~’ is used.
5:0.5 --A- DSB Using (18)-(22), the expected value of square of
~<
~ a 0.4 ‘-*- 7T/4shii
z +160AM
signal envelope is as follows.
0.3
o + FSK2
$ 0.2 4 El E(f2(t))= E(A4(t))+ 4c?E(A2(t))+ 4& E(A2(t))+ 8C74
~ 01
1-
In
w 0 = E(A4(t))+ 8c?E(A2(t))+ 804 (23)
5CdB 20dB
S/NdS) Combining (16) and (23), the parameter K which is
not influenced by noise is defined as follows.
Fig.2. Estimation error of symbol rate.

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 o
K= =J5(A4(d)- 2(~A2(t))]
E(I&(d}2@d)~ (24) ---------------A
——__—___——
---------- -----
~ -0.2
2
$
The classification results of analogue modulation > -0.4 ~m=O.4
n
u -W–m=O.6
signals by K are shown in Fig. 3. The amplitude
~ -0.8 --*-- m=O.8
r“-
modulation depth of AM signal is 1.0, the frequency E
z

modulation index of FM signal is 9.0, and the phase = –0,8

deviation coefficient of PM signal is x. It can be

-1
known that the values of amplitude modulation 10dS 20dB 50dB
S/N
signals are around O and the values of frequency
modulation signals are around –1.0 respectively, so Fig.5. Influence of amplitude modulation depth(AM).
the discrimination by the threshold value is possible.
The instantaneous frequency of FSK signals takes
And there is not the influence due to the noise
only the predetermined values, so the frequency
component.
distribution in the spectrum is discrete. On the other
The results of FSK signals are shown in Fig.4. It
hand, the ones of FM and PM signals are continuous.
can be known that the values of FSK signals are also
So the parameter which expresses the compactness of
around -1.0 and not influenced by the noise
the frequency distribution [11] is available for the
component.
classification among frequency modulation signals.
Fig.5 shows the results when the amplitude
The instantaneous frequency at each sampling point
modulation depth of AM signal changes. The values
t, f(i, is calculated as follows.
somewhat change by the modulation depth, but the
discrimination against the frequency modulation
(25)
signals is possible.

0.2 where d@ is the instantaneous phase obtained by

-----
Uo 1i-------- Hilbert transform of signal, and T, is the sampling
3
—Ah!

El
J -0.2
< –-w--FM
interval. From f($, the normalized centered
>
n
-0.4 .--h--- PM instantaneous frequency, fit) ,is calculated as follows.
; -0.6 -.%. -DSB
< _ USB
~ -0.8
*. . .. . .. . . . . . . . . . . . . .. ---- LSB
z *-. . . . . . . . . . . . . . . . . . . .~
3 -1 . .

T-
-1.2
10dB 20dB 50dB .f~(t)= f-n(t)- ma (27)
s/N
where f.a is the maximum instantaneous frequency
Fig.3. Results of analogue modulation signals. in each analysis block, and ma is the average of
instantaneous frequency for each analysis block. The
o –- . . .. . . . .. ..
compactness of the instantaneous frequency, C , is
-0.2 defined as follows.
w
3
J
2
–0.4
~ FSK2
~= ‘m’)) (28)
~ –0,6 – * – FSK4
In
< --*--MSK
El
(E(f;(@~
Z -0.8
z
3 The classification result by the compactness, C’ ,is
shown in Fig.6. The frequency modulation index of
-1.2 FM signal is 5.0, and the phase deviation coefficient of
10dB 20dB 50dB

s/N
PM signal is z/2. The values of FSK signals are
small in comparison to FM and PM signals because
Fig.4. Results of FSK signals. the instantaneous frequency takes the predetermined

0-7803-5538-5/99/$10.00 (c) 1999 IEEE

ones and the compactness is low. It can be known that
5 I
the diserimination between FSK4 and the other FSK + .--. .—. . . . . . . . - ..-. .—. . -...-..-6
5 4.5
signals is also possible at SNR = 20dB. And the Li= d
03
discrimination between FM signal and PM signal is cLlQ3.5’ ‘-- .-. <--;
~K
——+——
FSti
---
w+ 3 * ------ ._ +._. _ . + - +3-– FSK4
also possible if the threshold value is chosen to be 2!4?
.- A--- MSK
1:
around 4.0. –-x--FM
--”&------------
The classification among amplitude modulation ....................
--o-. PM
4
signals is carried out by the ratio of the variance of the
# 0.5
envelope tQ the square of the mean of the envelope
‘o
referred to [1]. The key parameter for the 10dB 20dB 50dB
classification, R ,is defined as follows. S/N

Fig.6 Classification among frequency modulation signals.

R=% (29)
P TABLE 1
Decision rule by R [1]

where U2 is the variance of the envelope, and ,u is Range of R Decision result

the mean of the envelope. The envelope is obtained as 0.396ZR FM,PM

the square sum of signal component and its Hilbert
transform component. Y.T. than et al. have shown 0.897 >~0.396 AM

that R becomes the function of the carrier to noise

1.105 Z~0.897 SSB
ratio and the modulating signal to noise ratio and all
analogue modulation signals can be classified by this D1.105 DSB
parameter. The decision rule presented in [1] is shown
in Table 1 But in [1], it is assumed that the
modulating signal is the complete Gaussian noise, so 2, I
1.8
if the modulating signal is not Gaussian, for example
1.6

r-
voice signal, the value of 2? may vary horn the
--+--AM
decision rule shown in Table 1. -+s - DSB

We assumed that the modulating signal is the first- --*--USB

- -X - LSB
order auto-regressive voice, and the simulation results
0.4
for amplitude modulation signals are shown in Fig.7. h
0,2 —
The amplitude modulation depth of AM signal is 0.2. o-
1

From this figure, it can be known that the R of DSB 10dB 20dB 50dB

S/N
signal and SSB signal is within the range regulated
by Table 1 but the ones of AM signal are smaller than Fig7. Classification result among amplitude modulation signals
the one of Table 1 and within the range of FM and
PM signals. So AM signal whose modulating signal is amplitude modulation depth, the discrimination from
voice cannot be discriminated from frequency DSB signal and SSB signal by threshold value is still
modulation signals by R. But if the discrimination by possible.
the aforementioned parameter K is practiced and
iiequency modulation signals are discriminated IV. CONCLUSIONS
beforehand, this parameter is available for the The new methods for the discrimination between
classification among amplitude modulation signals. analogue modulation signals and digital modulation
Fig.8 shows the results when the amplitude signals and the classification among analogue
modulation depth of AM signal changes. Though the modulation signals including phase-continuous FSK
value of 1?becomes larger with the increment of the signals have been proposed. Symbol rate is estimatid

0-7803-5538-5/99/$10.00 (c) 1999 IEEE

 ~FERENCES
[1] Y.T.Chan ,L.G.Gadbois, and P. Yansouni, “Identi6cation of the
* “%. -.
modulation type of a signal’’,IEEE International Conference on

m
o . .
“: 0.4 .. . --+-- “.4 Acoustic, Speech and Signal Processing,pp.838-841 (1985).
LY i L ----
.. . .. . . -%. -._. -.-. _._. _. -4s--0.6 [2] A.K.Nandi and E.E.Azzouzj’’Automatic analogue modulation
.. . . (
!& . .. A--- 0.8
“-. % . . .. . . recognition’’, Signal Processing, 46, pp.211-222(1995).
<
--.- . . .. .
. . . . . . . . . . . . . . . . . . . ..+.
-.)+--1 [3] Janet AISBETT, “Automatic modulation recognition using time
L

—- +_____ domain parameters”, Signal Processing, 13, pp.323-328 (1987).

0.1
“~
1 [4] Samir S.Soliman and Shue-Zen Hsue~Signal classification using
statistical moments’’,IEEE Trans. Commun.,vol.40,No.5,pp.908-
10C6 2ikB 5od3 916(May. 1992).
M’@) [5] Yawpo Yang and Samir.S.Soliman~An improved moment-based
algorithm for signal classification’’, Signal Processing,43,pp.231 -244
Fig.8. Influence of amplitude modulation depth(~.
(1995).
[6] Masafumi Hagiwara and Masao Nakagawa, “Automatic
by the block demodulation method, and analogue
estimation of an input signal type”, IEEE GLOBECOM’87, pp.254-
modulation signals can be discriminated from digital
258 (1987).
modulation signals by the estimation error of symbol
[’7jE.E.Azr.ouz and AX.Nandi, “Automatic identification of digital
rate. In this paper, it is assumed that modulating
modulation types’’, SignaJ Processing,47,pp.55-69 (1995).
signal is a first-order auto-regressive voice and not
[8] AK.Nandi and E.E.Azzouz,’’Modulation recognition using
Gaussian, so the classification of all analogue
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[9] Yasushi.Higashida, Masafami Hagiwara and Masao
Chan et al is impossible. But if the statistical
Nakagawa, “Timing extraction and carrier estimation for PSK signal
parameter K defined in Section3 is used in
block demodulation method”, IEICE Trans.B ,Japan,vol.J71-B,No.4,
combination with the R, all analogue modulation
pp.540-546(Apr.1988).
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[10] Yutaka Miyake, Masafami Hagiwara and Masao
In future, the influence of the statistical
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Recognition of Communication Signale, KluwerAcademic Publishers,
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