Compressive sampling of speech signals using
Bessel basis
Senthil Murugan S∗ , Manoj C
Sathiesh Kumaar L, Magesh S
Electronics and Communication Engineering
Amrita Vishwa Vidyapeetham
Coimbatore, India
∗ amrita.senthil@gmail.com
Abstract—Compressive sampling is an evolving approach for
sampling signals at rates much less than Nyquist rate. For Com-
pressive sampling of signal, suitable basis function is required
which gives sparsity on transformation. This paper proposes com-
pressive sampling of speech signals by using Bessel function as
basis function for transformation of signal from time domain. By
using Bessel function as basis, it is shown in this paper that speech
signal has been sampled at a rate less than the Nyquist rate.
The Signal-to-Distortion ratio (SDR) of reconstructed speech
signals using Bessel basis and Sinusoidal basis are compared
for compressive sampling rate of 8000Hz, 6400Hz and 3200Hz.
Results shows that the proposed Bessel basis performance is far
superior to Sinusoidal basis in terms of signal reconstruction.
Index Terms—Sampling, Compressive sensing (CS), Bessel
function, Sampling of speech signal
I. I NTRODUCTION
The compression of speech signals by signal processing
techniques after sampling the analog signal at twice the
maximum frequency -Nyquist-Shannon sampling theorem-is
generally accepted and found in many literature. Though the
amount of data after compression is reduced, the data to be
sampled are generally large in these methods. To avoid this
resource(sensing) wastage, the work on compressive sampling
have been progressing in recent years. Compressive sam-
pling[1] takes only small amount of samples directly from
the sampling process for reconstruction of speech signal.
For compressive sampling of signal, the signal have to
be sparse-expanded with suitable basis functions. Generally,
sinusoids are used as basis function for expansion of speech
signals in speech recognition,speaker identification and veri-
fication,etc. But, the sinusoidal expansions i.e. Fourier trans-
forms doesn’t give sparse coefficients on expansion. Though
there are many works on Compressive sampling of audio
signals [14], [17], [18], the papers on speech signals [13],
[14] are rare because of the above mentioned problem. In [13],
compressive sampling is done only for sparsely excited speech
using Matching Pursuit method.
In our paper, we propose Zeroth order Bessel function,
which are decaying in nature, as the basis function for
compressive sampling of speech signals, as it captures the
characteristics of speech better than sinusoids [9] and produces
sparse coefficients. Our proposed basis function works not
Rakesh Peter†
Computational Engg. and Networking
Amrita Vishwa Vidyapeetham
Coimbatore, India
† rakesh.peter@gmail.com
only for sparsely excited speech but also for voiced speech.
We use ‘Basis pursuit’ framework [3] for CS recovery.
The paper is organised in the following order. In Section II,
we give the mathematical model for compressive sampling.
Section III describes the characteristics of Bessel function
and Fourier Bessel transform for speech signal. Section IV
proposes the compressive sampling of speech signal using
Bessel as basis function. The experiment details and results are
dealt in Section V. Finally, the conclusion is given in Section
VI followed by overview of future work in Section VII.
II. M ATHEMATICAL MODEL FOR C OMPRESSIVE SAMPLING
A. Compressive sampling
Compressive sampling relies on the fact that many nat-
ural signals are sparse under suitable basis function for
sampling and reconstruction. Consider a discrete time sig-
nal x of N samples as vector in vector space ℜN . Let
Ψ = [ψ1 |ψ2 |ψ3 . . . ψN ] be N x N basis matrix where
ψ1 , ψ2 , . . . ψN are basis vectors of space ℜN .
x = Ψs (1)
where s = [s1 s2 . . . sN ] is a vector of length N containing
projection coefficients si = hx, ψi i.
The vector x is said to be K-sparse in Ψ domain if K
non-zero coefficients of s are enough to reconstruct x with
minimal distortion. We observe sampling redundancy in signal
acquistion which will be removed by compressive sampling by
taking only M samples as follows:
Let y be measurement vector of length M given by y=Φx
where Φ is an M x N measurment matrix and K < M << N .
By equation (1), y is written as
y = Θx where Θ = ΦΨ (2)
The selection of Φ have to be done carefully in a way
that Φ and Ψ are incoherent which is the essential condition
(Restricted Isometry Property) for reconstructing x from y [4].
In papers [1], [2], [3], it is shown that Gaussian matrix of size
M x N satisfy isometry property for all most any Ψ with high
probability if M ≥ cKlog(N/K) for small constant c.
B. Reconstruction Algorithm Bessel Basis
1
N
The reconstruction of signal x in space ℜ is equivalent to
reconstruction of sparse vector s in Ψ domain by equation (1).
Unfortunately, finding exact s in ℜN which satisfy Θs′ = y is
not that trivial because of the following reason: Θs′ = y for
all s′ = s + r where vector r is in the Null space N (Θ) of Θ. 0.5
However, approximate solution can be obtained using l1 norm

Amplitude
minimization given by

bs = arg min||s′ ||1 , (3)

0
with constraint Θs′ = y
This convex optimization problem can be solved using
linear program algorithm called as Basis pursuit [3]. Many
reconstruction algorithms are also proposed in [5] and [6].
As we are concentrating mostly on finding basis function for
−0.5
CS of speech signal rather than optimization techniques, we 0 10 20 30 40 50 60 70 80 90 100
x
adopted basis pursuit algorithm for reconstruction of speech chch
signal.
Fig. 1. Bessel function of First kind.
III. F OURIER B ESSEL S ERIES
A. Bessel function
Speech signal in time domain
Bessel functions are the solutions for the differential equa- 0.1

tion
′′ ′
x2 y + xy + (x2 − n2 )y = 0, n > 0 (4) 0

which is called Bessel’s differential equation. The solution for −0.1

0 50 100 150 200 250 300 350
this equation is given by
Fourier Bessel transform of speech signal
0.1
y = C1 Jn (x) + C2 Yn (x) (5)
where Jn (x) is called as Bessel function of the first kind of 0
order n and Yn (x) is called a Bessel function of the second
kind of order n. −0.1
0 50 100 150 200 250 300 350
Bessel function of first kind is given by Fourier transform of speech signal
3
X∞
(−1)r (x/2)n+2r )
Jn (x) = ( (6) 2
r=0
r!Γ(n + r + 1)
1
Figure 1 clearly shows that bessel function of first kind
0
resembles damped sinusoids i.e. amplitude of bessel function 0 50 100 150 200 250 300 350
of first kind decays as ‘x’ increases. It is observed that
all physical systems including speech signal have naturally chch
decaying impulse response. When the signal is projected on Fig. 2. Fourier Bessel transform and Fourier transform of speech signal.
basis which have resemblance as that of signal itself, the
projected coefficients will be spase. Hence, the bessel function
which has the characteristics of speech signal and sastisfying where J0 (λm t/a) are the Zero-order Bessel functions, λm
orthogonal property [7] can be used as basis function for for m=1,2,3,. . . are the ascending order positive roots of
expanding speech signal. J0 (λ) = 0, Q is the order of the Bessel expansion.
The coefficient Cm called as Fourier Bessel coefficients
B. Fourier Bessel expansion for speech signal
(FB) are computed as
Consider a signal s(t) defined over arbitary time interval Z a
(0,a). This signal can be expressed in terms of bessel function 2
Cm = 2 2 ts(t)J0 (λm t/a)dt (8)
as a J1 (λm ) 0

Q where J1 (λm ) are the First-order Bessel function.

X
s(t) = Cm J0 (λm t/a), 0 < t < a (7) The Fourier Bessel expansion for speech signal have already
m=1 been explored in many areas of speech processing like speech
coding [9], speech recognition, speaker verification and iden- breaking down convex problem to LPP is that the s, Θ and
tification systems [10], [11], [12]. The paper [9] dicuss about y should have real-value entries [20]. One of the remarkable
compression of speech signal using FB expansion. But the property of Fourier Bessel expansion is that bessel coefficient
application of the same technique on compressive sampling is in Bessel domain are all real values [11]. Also while designing
first in this work. Measurement matrix, we make sure it contains real values.
So, Convex optimization problem can be solved using Linear
IV. C OMPRESSIVE SAMPLING OF SPEECH USING B ESSEL programs. This is another advantage of going to Bessel basis
BASIS
as sinusoidal basis does not have these features.
A. Design of Bessel basis matrix
As already mentioned in the paper, Compressive sampling V. E XPERIMENTAL RESULTS
of any signal in vector space ℜN starts with formulation of
N x N basis matrix Ψ containing N column vectors which A. Experiments
are the basis vectors for ℜN . The Bessel basis vector ψi is The voiced speech is taken for testing the proposed method
given by   of compressive sampling of speech. We sampled the signal
J0 ( λNi ) at 16000Hz for simulation in MATLAB. Since speech signal
 J0 ( 2λi )  is non-stationary in nature, it is split into frames x of length
 N 
ψi =  ..  (9) 20ms which corresponds to 320 samples.
 . 
J0 ( NNλi ) 1) Compressive sampling of speech for M = 160: The ma-
trix Θ of size 160 X 320 is designed as mentioned previously
where λi is the ith positive root of J0 (λ) = 0. The Bessel basis and applied on x to get compressive sampled vector y of
matrix Ψ = [ψ1 |ψ2 | . . . ψN ] is designed and given below size 160 resulting in compressive sampling rate of 8000Hz.
  For reconstruction of speech vector x, the sparse vector bs
J0 ( λN1 ) J0 ( λN2 ) . . . J0 ( λNN )
 J0 ( 2λ1 ) J0 ( 2λ2 ) . . . J0 ( 2λN )  is estimated by l1 minimization with equality constraints
 N N N  (also called as basis pursuit). We used l1 -MAGIC MATLAB
Ψ= .. .. .. ..  (10)
 . . . .  package [15] for reconstruction of s via Convex programming.
J0 ( NNλ1 ) J0 ( NNλ2 ) . . . J0 ( NNλN ) The Bessel basis matrix Ψ of size 320 x 320 is calculated
following the steps given in previous section and it is applied
Now, if the discretized speech signal is projected on the
on the estimated sparse vector bs of size N to get reconstructed
Inverse Bessel matrix Ψ−1 ,the vector s containing sparse
speech vector bx. The SDR value of reconstructed speech signal
bessel coefficients is obtained.
using Bessel basis is found to be an average of 13dB. We
B. Design of measurement matrix also found SDR of reconstructed speech using Sinusoid basis
to be an average of 7dB. This clearly shows the efficieny
Measurement matrix forumulation should be carried out
of the our proposed system with the existing model. The
with care, such that dimensionality reduction from ℜN to
original speech signal and its reconstruction using compressive
ℜM should not affect signal reconstruction. In our work, we
sampling techniques for both sinusoidal and bessel basis are
designed the matrix Θ of size M x N instead of Measurement
plotted.
matrix by random selection of M rows of Bessel matrix Ψ and
multiplying it with Gaussian distributed random weights. Now, 2) Compressive sampling of speech for M=64: Compres-
the random measured vector y or compressive samples will be sive sampling for M=64 corresponds to sampling rate of
obtained as given below: 3200Hz. Here, only the matrix Θ have to be desinged for
size of 64 x 320. The SNR value for Bessel and Sinusoidal
y = Θs (11) basis are almost same (average of 2.5dB) for this sampling
rate. Anyhow, the quality of reconstructed speech using Bessel
C. Optimization technique basis is far superior to Sinusiodal basis which can be seen
The papers[3], [5] gives various methods for signal recovery from graphs of the reconstructed speech signal. We also
by convex programming from highly incomplete information. experimented for other compressive sampling rates of 6400Hz,
We are adopting basis pursuit (also called l1 minimization with 9600Hz, 12800Hz and the SDR values are compared for both
equality constraints) for our work. The framework of basis basis functions.
pursuit for speech recovery is stated as
B. Results
min ||s||1 subject to
PΘs = y
where ||s||1 := |si | The plots containing original speech signal (for 20ms), re-
i constructed speech signal using Bessel basis and reconstructed
There are two ways in which convex problem can be solved speech signal using Sinusoidal basis are given in this section
with less complexity: (1) By recasting convex problem in for different CS rates. It can be seen from the plot that only
to Linear Programming Problem (LPP) (2) By recasting it some intial values of reconstructed speech varies much from
into Second Order Cone Problem (SOCP). The condition for original speech signal.
 Original speech signal Original speech signal
0.1 0.1

0 0

−0.1 −0.1
0 50 100 150 200 250 300 350 0 50 100 150 200 250 300 350
Recovered signal taking only 160 samples − Bessel basis Recovered signal taking only 64 samples − Bessel basis
0.1 0.1

0 0

−0.1 −0.1
0 50 100 150 200 250 300 350 0 50 100 150 200 250 300 350
Recovered signal taking only 160 samples − Sinusoidal basis Recovered signal taking only 64 samples − Sinusoidal basis
0.1 0.1

0 0

−0.1 −0.1
0 50 100 150 200 250 300 350 0 50 100 150 200 250 300 350

chch chch
Fig. 3. Reconstructing speech signal using 160 samples. Fig. 5. Reconstructing speech signal using 64 samples.

Original speech signal

0.1

−0.1
0 50 100 150 200 250 300 350
Recovered signal taking only 128 samples − Bessel basis
0.1

−0.1
0 50 100 150 200 250 300 350
Recovered signal taking only 128 samples − Sinusoidal basis
0.1

0
−0.1
0 50 100 150 200
chch
Fig. 4. Reconstructing speech signal using 128 samples.
The SDR values for reconstructed speech for various M/N
ratios are also provided for selecting Compressive sampling
rate according to desired quality.
VI. S UMMARY AND C ONCLUSION
In this paper, Compressive sampling of speech using Bessel
function as basis is proposed. It is shown that recovered
speech signal using Bessel basis has outperformed Sinusoidal
basis (Fourier transform). It is important to notice that our
proposed algorithm is the first of its kind to perform well
on speech audio.The computational complexity involved in
solving convex optimization problem for sparse vector is also
reduced in our proposed method due to real-valued Fourier
Bessel coefficients. The quality of reconstructed speech can
be enhanced by improving the optimization techniques.
chch
250 300 350 Fig. 6. SDR values of reconstructed speech signal for various M/N ratio
VII. F UTURE WORK
Our proposed method is being applied for tracking people
in Wireless Sensor Networks. Using microphone sensors and
the person’s speech signal as identity, person tracking systems
can be developed. Similar approach in [18] requires the use of
audio signal source attached to a person for tracking. In our
future work, we are planning to use USRP based Software
Radio Basestation [21] to receive the compressed samples
of speech signal from wireless microphone sensor nodes and
compute the Bessel sparse coefficients by Convex optimization
technique in the base station PC. The estimated Bessel sparse
coefficients will be given to speaker identification system [10],
[12] without transferring to time domain, opposed to the case
in [16]. The Compressive sampling of speech signal has a
lot of application in security systems using Wireless Sensor
Network and our proposed system will help further research
in such areas.
 R EFERENCES
[1] Emamnuel J Candés, “Compressive sampling,” Proceedings of the In-
ternational Congress of Mathematicians,vol. 3,pp. 1433-1452, Madrid,
Spain, 2006.
[2] Massimo Fornasier,Holger Rauhut, “Compressive Sensing,” Handbook of
Mathematical Methods in Imaging (O. Scherzer Ed.), Springer, 2011.
[3] Emmanuel Candés, Justin Romberg, and Terence Tao, “Robust uncer-
tainity principles: Exact signal reconstruction from highly incomplete
frequency information,” IEEE Trans.on Information Theory, Vol. 52,no.
2, pp.489-509, February 2006.
[4] Richard Baraniuk, “Compressive sensing,” IEEE Signal Processing Mag-
azine, Vol. 24,no. 4, pp.118-121, July 2007.
[5] J. Haupt, R. Nowak, “Signal reconstruction from noisy random
projections,” IEEE Trans. Inform. Theory,” vol. 52, no.9, pp.4036-
4048,Sept.2006.
[6] J. Tropp, A.C. Gilbert, “Signal recovery from partial information
via orthogonal matching pursuit,” Apr. 2005 [Online]. Available:
http://www.personal.umich.edu/ jtropp/papers/TG06-Signal Recovery.pdf
[7] J. Schroeder, “Signal Processing via Fourier-Bessel series expansion,”
Digital Signal Processing 3, 112-124, 1993.
[8] K. Gopalan, T8. R. Anderson, “A Comparison of speaker identification
results using features based on Cepstrum and Fourier-Bessel expansion,”
IEEE Transactions on Speech and Audio processing, Vol. 7, no. 3, May
1999.
[9] K. Gopalan, “Speech coding using Fourier-Bessel expansion of speech
signals,” IECON’01: The 27th Annual Conference of the IEEE Industrial
Electronics Society.
[10] K. Gopalan, “Speaker identification using Bessel function expansion of
speech signals,” Final Rep. Summer Faculty Res. Program, Armstrong
Lab., AFOSR, Aug. 1993.
[11] K. Gopalan and T. R. Anderson, “Speech processing using Bessel func-
tions,” Proc. Symp. Intelligent Systems in Communications and Power,
Mayaguez, PR, Feb. 1994, pp. 255-259.
[12] K. Gopalan and T. R. Anderson, “Speaker Identification using Bessel
function representation and a back-propagation neural network,” Proc. of
the IEEE International Symposium on Industrial Electronics, 1995.
[13] T.V. Sreenivas, W. Batiaan Kleijn, “Compressive sensing for sparsely
excited speech signal,” Proc. IEEE Int. Conf. Acoust., Speech, Signal
Processing, 2009, pp.4125-4128.
[14] M. G. Christensen, J. ∅stergaard, and S. H. Jensen, “On compressed
sensing and its applications to speech and audio signals,” Asilomar Conf.,
2009.
[15] E. Candès and J. Romberg, “l1-magic: Recovery of sparse signals via
convex programming,” code package available at www.l1-magic.org
[16] Anthony Griffin, Elen Karamichali, “Speaker Identification using
sparsely excited speech signals and compressed sensing”, 18th European
Signal Processing Conference, August 23-27, 2010.
[17] Laura Balzano, Robert Nowak, “Compressed Sensing Audio Demon-
stration,” in website http://sunbeam.ece.wisc.edu/csaudio/
[18] Anthony Griffin, Panagiotis Tsakalides, “Compressed sensing of Audio
signals using multiple sensors”,16th European Signal Processing Confer-
ence, August 25-29, 2008.
[19] Chong Luo, Feng Wu, “Compressive Data Gathering for Large-Scale
Wireless Sensor Networks”,in Mobicom 2009.
[20] S.S. Chen, D.L. Dohono, “Atomic decomposition by basis pursuit,”
SIAM J.Sci. Comput., 20:33-61, 1999.
[21] Ettus Research LLC, 1043 North Shoreline Blvd., Suite 100, Mountain
View, CA 94043, www.ettus.com