2016 Progress In Electromagnetic Research Symposium (PIERS), Shanghai, China, 8–11 August

A Blind Source Separation Algorithm of Non-stationary Signals

Based on Local Polynomial Fourier Transform
Wei Nie1 , Yibing Li1 , and Dandan Liu2
1
College of Information and Communication Engineering
Harbin Engineering University, Harbin 150001, China
2
College of Electrical and Control Engineering
Heilongjiang University of Science and Technology, Harbin 150001, China

Abstract— The separation of non-stationary signals is a hot topic. The blind source separa-
tion algorithms based on time-frequency analysis can not only describe the time-domain feature
and the frequency-domain feature of signals but also extract the time-varying feature of non-
stationary signals. Blind source separation owns better performance in solving the problem of
non-stationary signals separation. In traditional methods based on linear time-frequency anal-
ysis, the time-frequency resolution is not high. Meanwhile, in traditional methods based on
bilinear time-frequency, there is the problem of the cross-term interference. According to these
problems, this paper proposes a blind source separation algorithm to separate the source signals
based on local polynomial Fourier transform that adjusts the time-frequency resolution and the
cross-term interference. First, the observed signals that are received by the senors are processed
through whitening. Then, the whitened signals are transformed to get time-frequency distribu-
tion matrices through local polynomial Fourier transform. What’s more, a strategy of choosing
time-frequency distribution matrices is proposed to detect the matrices that own the underly-
ing diagonal or off-diagonal structure and remove other matrices. The thresholds of traditional
strategies in other papers are not needed in this paper. Therefore, the strategy of this paper
can avoid the blindness of choosing the threshold. Finally, a joint diagonalization of a combined
set of time-frequency distribution matrices is done to estimate the mixing matrix and the source
signals. The simulation results indicate that the proposed algorithm of this paper can extract
the non-stationary source signals effectively and own better performance than other comparison
algorithms.

1. INTRODUCTION
Blind source separation (BSS) can recover the source signals from the observed signals with few prior
information. Because BSS owns good separation performance, it receives more and more attention.
The BSS algorithms have been widely applied in digital image [1], wireless communication [2],
bio-medical engineering [3], speech processing [4], array signal processing [5], mechanical fault
diagnosis [6], etc.. In these fields, there is a type of special signal, namely the non-stationary
signal. Traditional non-stationary signals include speech signals, bio-medical signals, radar signals
and so on. These non-stationary signals don’t exist as a single form in actual life. They consist of
several signals and all kinds of interferences. The situation in which several non-stationary signals
are mixed is common in reality. For these special signals, the BSS algorithms based on time-
frequency distribution can not only describe the time-domain feature and the frequency-domain
feature of signals but also extract the time-varying feature of non-stationary signals. Therefore,
these algorithms own better performance than traditional algorithms. The BSS algorithm based
on time-frequency distribution was first proposed in [7]. This algorithm utilizes the time-frequency
difference of different source signals to get the time-frequency matrices and estimates the source
signals with joint diagonalization. Traditional BSS techniques such as independent component
analysis need to satisfy the assumption that only a Gaussian source signal exists in all sources.
However, the assumption is unnecessary for the algorithm in [7]. However, the algorithm adopts
the bilinear time-frequency distribution and then produces some cross-term interferences that affect
the performance of the algorithm. Meanwhile, linear time-frequency distribution does not own
enough time-frequency resolution and affects the separation of sources. Many scholars propose their
methods to solve this problem and discuss the choice strategy of the auto-term, the cross-term and
time-frequency matrices. A novel method of choosing time-frequency matrices was proposed in [8].
This method utilizes the non-orthogonal joint diagonalization method to extract the source signals.
Authors in [9] used multiple hypothesis testing to choose time-frequency points and separated
non-stationary signals. Peng [10] proposed a method to choose the auto-term and applied it to

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underdetermined blind source separation. A novel time-frequency distribution that owns good time-
frequency resolution and restrains the interferences was proposed in [11]. Authors in [12] utilized
improved time-frequency WVD and the special method of choosing time-frequency matrices to
separate sources. The time-frequency method is applied to separate underwater signals in [13]. This
paper introduces the local polynomial Fourier transform (LPFT) whose time-frequency distribution
owns the advantages of linear time-frequency transform and bilinear time-frequency transform.
Meanwhile, we propose a novel strategy of choosing time-frequency matrices to lay the foundation
for joint diagonalization.
The rest of this paper is organized as follows. In Section 2, we introduce the basic BSS model
of non-stationary signals. Our algorithm is described in Section 3. Section 4 shows the simulation
results and analysis, and the conclusion is made in Section 5.

2. THE BASIC MODEL

The linear instantaneous mixed BSS model of non-stationary signals can be denoted as

x(t) = As(t) + n(t) (1)

where A is a n × m mixing matrix, x(t) = [x1 (t), x2 (t), . . . , xn (t)]T is the observation signals vector,
s(t) = [s1 (t), s2 (t), . . . , sm (t)]T is the source signals vector, n(t) is the noise, t is a time point. Before
the algorithm, the preprocessing steps that include removing mean and whitening are adopted. The
equation of removing mean is written as

x̃(t) = x(t) − E[x(t)] (2)

In this paper, the subspace-based method is utilized to whiten the observation signals to get the
whitened signals
z(t) = Wx̃(t) (3)
where W is the whitening matrix.

3. THE PROPOSED ALGORITHM

The linear time-frequency transform such as short time Fourier transform (STFT) can’t balance
time resolution and frequency resolution because of the effect of the window function. Meanwhile,
the bilinear time-frequency transform will result in the cross-term. Therefore, a time-frequency
transform that can adjust the time-frequency resolution and the cross-term interference is needed.
As a generalized form of STFT, LPFT [14, 15] owns wider description ability of non-stationary and
stronger noise suppression ability. Therefore, LPFT is adopted in this paper.
The discrete form of LPFT can be denoted as

∑
N/2−1
x(n + m)h(m)e−j[2πkm/N +k1 m
2
/2+...+kM −1 mM /M !]
LPFT(n,k, k1 , . . . ,kM −1 ) = (4)
m=−N/2

where x is a non-stationary signal, N is the sample number in the frequency band, ki is the
parameter of discrete LPFT, M denotes the highest order of local polynomial approximation, h(m)
is the window function. The spectrum of LPFT is the square of the LPFT module and can be
written as
LPPx (t, w, w̄) = |LPFTx (t, w, w̄)|2 = LPFTx (t, w, w̄)LPFT∗x (t, w, w̄) (5)
The above formula is the auto-spectrum of LPFT. In order to research the interaction of several
signals, the cross-spectrum of LPFT can be denoted as

CLPPx1 x2 (t, w; w̄) = LP F Tx1 (t, w; w̄)LP F Tx∗2 (t, w; w̄) (6)

where x1 and x2 are two non-stationary signals. When the sources are mixed as the linear instan-
taneous form, the local polynomial Fourier transform is adopted for whitened signals. For each
whitened signal, the LPFT spectrum can be denoted as LPPzi . For each two whitened signals, the
LPFT spectrum can be denoted as CLPPzk zl where k, l = 1, . . . , n and k ̸= l. The time-frequency

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matrix Rz (t, w) can be constructed based on the time-frequency distribution of observation signals.
For each time-frequency point, the following formula can be obtained
 
LPPz1 CLPPz1 z2 . . . CLPPz1 zm
 CLPPz2 z1 LPPz2 . . . CLPPz2 zm 
Rz (t, w) = 
 .
.. .
.. .. .. 
 (7)
. .
CLPPzm z1 CLPPzm z2 ... LPPzm

For the source signals, the similar time-frequency matrix Rs (t, w) can also be constructed.
Similar to the LPFT spectrum of observation signals, the auto-spectrum and the cross-spectrum
of source signals LPFT are LPPsi (i = 1, 2, . . . m) and CLPPsk sl (k, l = 1, . . . , M and k ̸= l),
respectively. If there are two source signals, there are four types of time-frequency points.
Type 1: Time-frequency points where there are neither source cross-terms nor source auto-terms.
The time-frequency matrix is [0, 0; 0, 0].
Type 2: Time-frequency points that correspond to source auto-terms only. The time-frequency
matrix is [LPPs1 (ti , wj ), 0; 0, 0] or [0, 0; 0, LPPs2 (ti , wj )].
Type 3: Time-frequency points that correspond to source cross-terms only. The time-frequency
matrix is [0, CLPPs1 s2 (ti , wj ); CLPPs2 s1 (ti , wj ), 0].
Type 4: Time-frequency T-F points that correspond to both source cross-terms and auto-terms.
The time-frequency matrix is [LPPs1 (ti , wj ), CLPPs1 s2 (ti , wj ); CLPPs2 s1 (ti , wj ), LPPs2 s2 (ti , wj )].
For these time-frequency points, we must take corresponding strategies to choose time-frequency
points and time-frequency matrices. First, the following rule of choosing time-frequency matrices
is chosen to remove the points with low energy

trace(Rz (t, w)) > ε (8)

where ε is a threshold. The time-frequency points that satisfy the above condition is retained.
Meanwhile, in order to choose more proper time-frequency points, the following rule is designed

 C(t, w) = det(Rz (t, w))
∥GradC (t, w)∥2 ≤ εGrad (9)

HC (t, w) > 0

where det(Rz (t, w)) denotes the value of the determinant, GradC (t, w) is the gradient equation of
C(t, w), HC (t, w) is the Hessian matrix of C(t, w), εGrad is a threshold. K time-frequency matrices
can be obtained after the above processes. The joint diagonalization method is utilized to get the
unitary matrix, and then estimate the source signals and the mixing matrix.

ŝ = UH Wx(t) (10)
#
 = W U (11)

where ŝ is the estimated source signals and  is the estimated mixing matrix.
The steps of the proposed algorithm are as follows
Step 1. Whiten the observation signals.
Step 2. Make the LPFT transform to get the time-frequency matrices.
Step 3. Adopt the novel strategy of choosing the time-frequency matrices to get more proper
time-frequency matrices.
Step 4. Utilize the method of joint diagonalization to estimate the mixing matrix and the source
signals.

4. SIMULATION RESULTS AND ANALYSIS

The sources consist of four LFM signals. The slopes of the frequency modulation are both 5 Hz/s.
The initial frequency is 5 Hz and 10 Hz. The sampling rate is 128 Hz/s. The number of sampling
points is 256. The time-domain plot of the source signals is shown in Figure 1. If the mixing
matrix is [0.266, 0.7848; 0.966, 0.5736], the mixed signals are shown in Figure 2. The estimated
source signals with the proposed algorithm are shown in Figure 3.
From Figure 3, we can know that this algorithm is effective to recover the source signals.

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2016 Progress In Electromagnetic Research Symposium (PIERS), Shanghai, China, 8–11 August

Source 1 Observation 1
1 2

0.5 1

0 0

-0.5 -1

-1 -2
0 50 100 150 200 250 0 50 100 150 200 250

Source 2 Observation 2
1 2

0.5 1

0 0

-0.5 -1

-1 -2
0 50 100 150 200 250 0 50 100 150 200 250

Figure 1. The source signals. Figure 2. The mixed signals.

Estimated source 1
2

-1

-2
0 50 100 150 200 250

Estimated source 2
2

-1

-2
0 50 100 150 200 250

Figure 3. The estimated source signals.

In order to demonstrate the performance of the proposed algorithm and compare with other
algorithms, the normalized mean square error (NMSE) of the estimated mixing matrix is chosen
as the performance index and described as
(∑ )
2
(â
mn∑ mn − amn )
NMSE = 10log10 2
(12)
a
mn mn

where âmn is the (m, n)th element of the estimated mixing matrix and amn is the (m, n)th element
of the initial mixing matrix. The NMSE(dB) of different algorithms are shown in Table 1.

Table 1. The NMSE (dB) of different algorithms.

the algortihm in [12] the algorithm in [13] our algorithm

−51.5092 −65.1648 −92.5462

Another performance index is source to interference ratio (SIR). It can be written as

{ }
E[|si (t)|2 ]
SIRi = 10log10 (13)
E[|yi (t) − si (t)|2 ]
where SIRi is the SIR of the ith source signal, si (t) is the ith source signal, yi (t) is the ith estimated
source signal. The average SIR (dB) of different algorithms are shown in Table 2.
From Table 1 and Table 2, it is known that our algorithm has better performance than the
comparison algorithms.

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Table 2. The average SIR (dB) of different algorithms.

The algortihm in [13] Our algorithm

57.6939 84.6294

5. CONCLUSION
This paper proposes a BSS algorithm of non-stationary signals based on LPFT that can adjust
the time-frequency resolution and the cross-term interference. First, the observation signals are
processed by the whitening step and the LPFT to get time-frequency matrices. Then, a novel
strategy of choosing time-frequency matrices is proposed to get more proper time-frequency ma-
trices. Finally, a method of joint diagonalization is utilized to estimate the mixing matrix and the
source signals. Simulation results show that this algorithm can separate the non-stationary signals
effectively and owns better performance than other algorithms.
ACKNOWLEDGMENT
This paper is funded by the International Exchange Program of Harbin Engineering University for
innovation-oriented Talents Cultivation, the National Natural Science Foundation of China (Grant
No. 51374099, No. 51509049), the Natural Science Foundation of Hei-longjiang Province (Grant
No. F201345).
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