Int J Speech Technol
DOI 10.1007/s10772-017-9443-z
Development of two-input adaptive noise canceller for wideband
and narrowband noise signals
Jafar Ramadhan Mohammed1   
Received: 15 November 2016 / Accepted: 20 July 2017
© Springer Science+Business Media, LLC 2017
Abstract  In this paper, the development of the single-
stage two-input adaptive noise canceller that proposed by
Widrow et  al. is investigated. It is shown that the use of
multi-stage cancellers instead of a single stage improves
the performance of the conventional two-input adaptive
noise canceller in a real life high-noise environment. In
such environment, the background noise may contain both
the wideband and narrowband (sinusoidal) noise compo-
nents. In this contribution, the wideband noise and sinu-
soidal noise signals can be significantly suppressed using
a new multi-stage adaptive noise cancellation scheme
based on adaptive line enhancer (ALE) and Least Mean
Square (LMS) filter. The proposed scheme is comprised
of two stages. The first stage uses ALE filters, which are
used to cancel the sinusoidal noise from the primary and
reference input signals, whereas the wideband noise is can-
celled using LMS adaptive filter in the second stage. The
good performance of the proposed scheme has been veri-
fied via real-time implementation on the Texas Instruments
TMS320C6713DSK.
Keywords  Adaptive noise canceller · Adaptive line
enhancer · Wideband noise · Narrowband noise
1 Introduction
The widespread use of cellular phones has significantly
increased the use of communication devices in high-noise
* Jafar Ramadhan Mohammed
jafarram@yahoo.com
1
College of Electronic Engineering, Ninevah University,
Mosul 41001, Iraq
environments. Intense background noise, however, often
corrupts speech and degrades the performance of many
communication systems. Thus a large variety of noise
reduction techniques have been proposed for reducing
background noise in noisy environment, see (Narendra and
Han 2012; Moir and Harris 2012; Niedźwiecki and Meller
2012; Roshahliza 2017; Jayakumar et  al. 2016) and refer-
ences therein. The recent noise reduction techniques prefer
multi-microphone systems (Bitzer et  al. 2001; Moham-
med 2009; Touazi and Debyeche 2017), over single-
microphone systems (Boll 1979; Sasaoka et  al. 2005) due
to the obvious advantages provided by the former over the
later. A multi-microphone system can be considered as a
directional microphone with a zero response (null) in the
noise direction. However, the microphone array involves
increased cost in the form of more microphones, D/A con-
verters, memory, signal processing power, etc. On the other
hand, spectral subtraction method (Boll 1979) is known to
use single-microphone system. However, in the spectral
subtraction method, the musical tones arise from residual
noise and it requires the speech/non-speech section detector
under the noisy environments.
The widely used adaptive noise canceller (Widrow et  al.
1975) consists a single stage with two microphones namely,
the primary microphone which is placed close to the signal
source to pick up the desired signal and the reference micro-
phone which is placed close to the noise source to sense
only the noise. Also it employs an adaptive filter with the
LMS algorithm to cancel the noise component embedded in
the primary signal. The adaptive filter has the task of mod-
eling the impulse response path between noise source and
primary microphone. In real-time environment, the back-
ground noise typically comes from various sources such as
ventilating fan, audio equipment, engines etc. Therefore, the
background noise maybe consists of wideband noise as well
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as the sinusoidal noise. A block diagram of the conventional
ANC scheme is illustrated in Fig. 1, where ­H1(z) represents
the transfer function between the wideband noise source and
the primary microphone, and H ­ 2(z) represents the transfer
function between the sinusoidal noise and the primary micro-
phone. The conventional ANC scheme has two inputs: the
primary input x(n) = s(n) + 𝜂0 (n) + 𝜁0 (n) and the reference
input v1 (n) = 𝜂1 (n) + 𝜁1 (n). The reference input contains
sinusoidal noise 𝜂1 (n) and wideband noise 𝜁1 (n) while the
primary input consists of speech s(n) plus sinusoidal noise
𝜂0 (n) and wideband noise 𝜁0 (n). The objective of the adap-
tive filter W(z) is to use the reference inputs 𝜁1 (n) + 𝜂1 (n)
to estimate the noise signals 𝜁o (n) + 𝜂o (n). The filter output
y(n), which is an estimate of noises 𝜁o (n) + 𝜂o (n), is then sub-
tracted from the primary microphone signal, producing the
desired speech signal plus reduced noises. Xiao and Wang
(2011) showed that the conventional ANC system may per-
form very poorly if its primary and reference noise signals
contain both wideband and narrowband components simul-
taneously. They also showed that if the length of the adaptive
filter W(z) is set longer than the paths ­H1(z), ­H2(z) and fre-
quencies of the sinusoids involved are not very low, the con-
ventional ANC scheme usually indicates good performance.
However, if a single sinusoid with very low frequency and
relatively large amplitude is present, and the adaptation is not
slow, the conventional ANC-only system may lose its power.
Its steady-state residual noise signal will sometimes show
spike-like peaks that sound like fireworks (Xiao and Wang
2011). Moreover, when the length of the adaptive controller
W(z) is shorter than the paths ­H1(z), ­H2(z), the conventional
ANC performance will deteriorate too, because the adaptive
controller cannot realize a good approximation of the impulse
response paths between noise sources and primary micro-
phone, and the existence of sinusoids make the system lack of
power to take care of both the path and the sinusoids simul-
taneously. Winberg et al. (1999) used a hybrid ANC headset
Fig. 1  The conventional adap-
tive noise cancellation scheme
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Int J Speech Technol
which combines both feedforward and feedback ANC tech-
niques. The adaptive feedforward controller is based on a dig-
ital system, while the feedback system is based on an analog
system.
On the other hand, the adaptive line enhancer (ALE) tech-
nique has been successfully applied to a wide range of appli-
cations such as speech coding and denoising periodic signals
from white noise (Levin et al. 2011). By combining the ALE
and conventional ANC scheme in an effective manner, the
purpose of cancellation of both the wideband and sinusoidal
noise is accomplished. The idea is to first introduce an adap-
tive line enhancer to remove the sinusoids from the primary
and reference signals to let the conventional ANC do what it
is good at, and then place a conventional ANC scheme cas-
cade to the ALE to cancel the wideband noise. To the best
of my knowledge, the proposed ANC scheme has almost not
been investigated and discussed so far. The proposed scheme
uses two stages of adaptive filters where the first stage consist
of two ALE filters placed in parallel to estimate and cancel
the sinusoidal noise included in the primary input and refer-
ence input signals. The second stage consists of conventional
adaptive noise cancellation scheme which estimates and can-
cels the wideband noise present in the noisy output speech
signal from the first stage and will provide the required
speech enhancement.
2 The proposed scheme
Figure 2 shows the new proposed adaptive noise cancellation
scheme. The primary input and reference input signals of the
proposed scheme are given as follows:
x(n) = s(n) + v0 (n)=s(n) + 𝜁0 (n) + 𝜂0 (n) (1)
v1 (n) = 𝜁1 (n) + 𝜂1 (n) (2)
Int J Speech Technol
Fig. 2  The proposed adaptive noise cancellation scheme
where x(n) is primary input signal, s(n) is clean speech,
v0 (n) and v1 (n) represents the noise signals received by pri-
mary microphone and reference microphone respectively,
𝜁1 (n) and 𝜂1 (n) represent the wideband noise and the sinu-
soidal noise, respectively.
The proposed adaptive noise cancellation scheme shown
in Fig. 2 consists of two cascade stages. The first stage con-
sists of two ALE filters which are named ALE1 and ALE2
placed in parallel and used for reducing the sinusoidal noise
included in the primary input and reference input signals.
This connection gives the advantage of adaptation conver-
gence at same time for both ALE filters (ALE1 and ALE2)
if we choose same value of the step size and filter length for
both ALE filters. In this setup the purpose of the signal path
that contains the delay element in the ALE1 and ALE2, is to
generate a reference noise signal that may be used to estimate
the sinusoidal noise signal 𝜂̂0 (n) and 𝜂̂1 (n) in the ALE1 and
ALE2 respectively. First, the sinusoidal noise is estimated by
both ALE filters (ALE1 and ALE2). The decorrelation fac-
tors or time delays d1 and d2 , which are carefully selected
to be long enough to decorrelate the sinusoidal noise and
short enough to maintain its effect on the speech and wide-
band noise negligible, so that the outputs are just replicas of
s(n) + 𝜁0 (n) for ALE1 output and of 𝜁1 (n) for ALE2 output.
The second stage in the proposed scheme is used for
wideband noise cancellation. The 𝜁1 (n) is represent the ref-
erence input signal to the LMS adaptive filter in the second
stage, according to the adaptive noise filtering principles.
The system output ŝ (n) is the enhanced speech signal.
The details of the adaptive line enhancer (ALE) used
to reduce sinusoidal noise in the proposed scheme are
explained with the help of the block diagram given in
Fig. 2. The ALE is in fact a degenerate form of the adap-
tive noise canceller in that its reference signal, instead of
being derived separately, consists of a delayed version of
the primary signal. The delay, denoted by d1 as shown in
the Fig.  2, is called the decorrelation delay of the ALE,
measured in units of the sampling period. Let the sinusoi-
dal noise be the stationary periodic signal, for example,
ventilating fan and engine noise. On the other hand, the
phoneme of speech changes, so the speech signal is the
non-stationary signal (Hayes 1996).
The delayed signal of x(n) as shown in Fig. 2, is repre-
sented as
x(n − d1 ) = s(n − d1 ) + 𝜂0 (n − d1 ) + 𝜁0 (n − d1 ) (3)
where d1 is time delay. Since characteristics of speech
change, the autocorrelation of speech fades as d1 increases.
On the other hand, the delayed sinusoidal noise 𝜂0 (n − d1 )
is correlated with 𝜂0 (n). In addition, the delayed wideband
noise 𝜁0 (n − d1 ) has low correlation with 𝜁0 (n). Thus, if two
signals, 𝜂0 (n) and 𝜂0 (n − d1 ), are correlated, then 𝜂0 (n) may
be estimated by 𝜂̂0 (n) from 𝜂0 (n − d1 ) (Pearson 2003).
Estimating 𝜂̂0 (n) depends on the strategy of how the cost
function is to be minimized, be it either least mean squares
or recursive least squares (Pearson 2003). For this paper,
the cost function will be minimized based on least mean
squares. The mean squared error of the ALE1 as shown in
Fig. 2, is defined as
E[e2 (n)] = E[(s(n) + 𝜁0 (n) + (𝜂0 (n) − 𝜂̂0 (n)))2 ]
= E[(s(n) + 𝜁0 (n))]2 + 2E[(s(n) + 𝜁0 (n))(𝜂0 (n) − 𝜂̂0 (n))]
+ E[(𝜂0 (n) − 𝜂̂0 (n))]2 (4)
13

Since speech signal, sinusoidal noise and wideband noise are
uncorrelated, E[s(n)𝜂0 (n)] = 0 and E[𝜁0 (n)𝜁0 (n − d2 )] = 0,
then 2E[(s(n) + 𝜁0 (n))(𝜂0 (n) − 𝜂̂0 (n))] = 0. The mean
squared error becomes
E[e2 (n)] = E[(s(n) + 𝜁0 (n))]2 + E[(𝜂0 (n) − 𝜂̂0 (n))]2 (5)
Minimizing E[e (n)] is equivalent to minimizing
2
E[(𝜂0 (n) − 𝜂̂0 (n))]2 . Therefore, this minimization will cause
𝜂̂0 (n) to be the minimum mean-square estimate of 𝜂0 (n)
(Pearson 2003).
The estimated output of ALE1 filter 𝜂̂0 (n) which is
shown in Fig. 2 is given by
L
∑
𝜂̂0 (n) = wk (n)x(n − d1 − k)
k=1
where wk (n) is the impulse response of the ALE1 filter and
L is filter length. The ALE1 filter can estimate only the cur-
rent signal that correlated with the delayed signal.
Applying the same mathematical analysis to part 2
of Fig.  2 above, for the reference input signal v1 (n). The
delayed signal of v1 (n) is represented as
v1 (n − d2 ) = 𝜂1 (n − d2 ) + 𝜁1 (n − d2 ) (7)
where d2 is time delay. Since the delayed sinusoidal noise
𝜂1 (n − d2 ) is correlated with 𝜂1 (n). In addition, the wide-
band noise 𝜁1 (n − d2 )has low correlation with 𝜁1 (n), the
output of ALE2 filter 𝜂̂1 (n) is given by,
L
∑ eter used in this simulation.
𝜂̂1 (n) = wk (n)v1 (n − d2 − k)
k=1
Even though the details of conventional adaptive noise
canceller using LMS algorithm could be found elsewhere
(Pearson 2003), the author would like to include a brief
overview of this filter for ready reference. A conventional
ANC is shown in Figs. 1 and 2 (stage 2). The output signal
y(n) is formed as the weighted sum of a set of input sig-
nal sample 𝜁1 (n), 𝜁1 (n − 1), ......., 𝜁1 (n − L + 1). Mathemati-
cally, the output y(n) is equal to the inner product of the
input vector 𝛇1 (n) and the weight vector 𝐰(n),
y(n) = 𝛇1 𝐓 (n)𝐰(n)
where 𝐰(n) = [w(n), w(n − 1), ....................., w(n − L + 1)],
(10)
is the weight vector of the adaptive filter. During the adap-
tation process, the weights are adjusted according to the
LMS algorithm (Pearson 2003). The primary input signal
contains the speech s(n), the wideband noise 𝜁0 (n) as well
as residual sinusoidal noise from output of ALE1 filter. The
reference input signal contains the wideband noise 𝜁1 (n) as
well as the residual sinusoidal noise from output of ALE2.
The wideband noise 𝜁1 (n) is correlated with noise signal
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Int J Speech Technol
𝜁0 (n) in the primary input signal. A general expression of
the output can be obtained as follows,
e(n) = d(n) − y(n) = d(n) − 𝛇1 𝐓 (n)𝐰(n) (11)
The LMS algorithm updates the filter coefficients accord-
ing to (Pearson 2003)
𝐰(n + 1) = 𝐰(n) + 𝜇e(n)𝛇1 (n) (12)
where 𝜇 is the step size which controls the convergence
speed and the stability of the adaptive filter.
3 Simulation results
(6)
To demonstrate the effectiveness of the proposed scheme,
extensive simulation and real-time implementation are
performed. All sound data prepared in this simulation
was sampled at 11,025  Hz with 8bit resolution. The
speech signal spoken by the male talker was used. As the
noise signals, the Gaussian white noise signal generation
and the sinusoidal signal generation in Matlab is used.
In the simulation, the power ratio of sinusoidal noise to
wideband noise was set to 10  dB. The two paths, H ­ 1(z)
and ­H2(z) are simulated by two FIR filters each of length
32 taps. The adaptive algorithm for updating filter coef-
ficients for all adaptive filters is chosen as is done in LMS
algorithm (Pearson 2003). Table 1 indicates each param-
(8) In order to show the noise cancellation performance
of the proposed and conventional ANC schemes in the
presence of wideband and sinusoidal noises, these noises
are filtered with ­H1(z) and ­H2(z) and then fed into inputs
of both schemes. Note that only in this case the speech
signal is considered to be inactive. The output (error)
signal by the proposed scheme is shown in Fig.  3a. The
improvement in the noise reduction performance pro-
vided by the proposed scheme over that of the conven-
tional scheme is evident when this performance in Fig. 3a
is compared with that of the conventional scheme given
(9)
Table 1  Parameters used in the proposed scheme
Stage 1 ALE1 Decorrelation delay 1 Sample
No. of weights 128
Step size 0.001
ALE2 Decorrelation delay 1 Sample
No. of weights 128
Step size 0.001
Stage 2 LMS algorithm No. of weights 32
Step size 0.1
Int J Speech Technol

(a) (b)
0.25 Primary Signal 0.25 Primary Signal
Output Signal Output Signal
0.2 0.2
0.15 0.15
0.1 0.1
0.05 0.05
Amplitude

Amplitude
0 0
-0.05 -0.05
-0.1 -0.1
-0.15 -0.15
-0.2 -0.2
-0.25 -0.25

0.5 1 1.5 2 2.5 3 3.5 4 0.5 1 1.5 2 2.5 3 3.5 4

Time index 4 Time index 4
x 10 x 10

Fig. 3  Output (error) signal of the a proposed scheme. b Conventional scheme

(a) Output of the Conventional Scheme (b) Output of the Conventional Scheme

Primary Signal Primary Signal

0.25 0.25
Output Signal Output Signal
0.2 0.2

0.15 0.15

0.1 0.1

0.05 0.05
Amplitude

Amplitude

0 0

-0.05 -0.05

-0.1 -0.1

-0.15 -0.15

-0.2 -0.2

-0.25 -0.25

0.5 1 1.5 2 2.5 3 3.5 4 0.5 1 1.5 2 2.5 3 3.5 4

Time index 4 Time index x 10
4
x 10

(c) Spectrogram of Output of The Conventional Scheme

0
5000

-10

4000 -20

-30
Frequency

3000
-40

-50
2000
-60

1000 -70

-80
0
0.5 1 1.5 2 2.5 3 3.5
Time

Fig. 4  Output (error) signal of the conventional scheme a without sinusoidal noise. b With sinusoidal noise. c Spectrogram of (b)

13
 Int J Speech Technol

in Fig.  3b. Note that the filter length used with the con- Spectrogram of Primary Signal

ventional scheme is chosen to be 32 taps and the step

size of the LMS algorithms is chosen to be 0.1. For com- 5000

parison, these parameters of the conventional ANC are

chosen to be equal to those of the second stage of the 4000
proposed scheme. Obviously the conventional scheme
is incapable of dealing with both noises simultaneously.

Frequency
3000
The adaptive filter W(z) in the conventional scheme does
not have ability to model both paths H ­ 1(z) and H­ 2(z) (see
Fig. 1) and it only takes care of the stronger noise. Thus, 2000

the conventional scheme loses its power for such appli-

cations, as it was originally designed to cancel wideband 1000
noise.
With the proposed scheme, the signal of the reference 0
input v1 (n) in (2) is not used to directly feed the adaptive Primary Signal
filter W(z), as its sinusoidal components may destroy the 0.8

ability of adaptive filter to reduce the wideband noise in the

0.6
primary input. Instead, the signal of the reference input is
pre-filtered by an adaptive line enhancer. The adaptive line 0.4
enhancer cancels the sinusoidal noise components resid-
ing in the reference input such that adaptive filter W(z) in Amplitude
0.2

the second stage could work effectively for wideband noise 0

cancellation.
In order to further analyze the performance deteriora- -0.2

tion of the conventional ANC scheme in the presence of

mixture of wideband and sinusoidal noises, we consider
the case were the filter length is longer than the impulse
response paths H ­ 1(z) and ­H2(z). Here in this scenario, two
cases are simulated; (a) reference microphone contains
wideband noise only, and (b) reference microphone con-
tains both wideband and sinusoidal noises. For comparison
fairness, the reference signal power of (a) is adjusted to be
the same as that of (b). Figure 4 shows the results for the
conventional ANC with filter length = 256 taps and the step
size = 0.1 such that the system indicates relatively fast con-
vergence. This figure clearly shows the poor performance
of the conventional ANC scheme if its primary and refer-
ence noise signals contain both wideband and sinusoidal
noises simultaneously even though the filter length of the
adaptive filter W(z) is set longer than the paths H ­ 1(z) and
­H2(z) (see Fig. 4b, c). On the other hand, as it anticipated,
the conventional scheme performs very well if its primary
and reference noise signals contain only wideband noise
(see Fig. 4a).
The spectrogram and the noisy signal of the primary
input are shown in Fig. 5. Note that, because the wideband
and sinusoidal noises, the quality of the original speech is
significantly degraded. Figure  6 shows the outputs of the
ALE1 and ALE2 in the first stage of the proposed scheme.
Note that the reduction of the sinusoidal noise is done to a
high degree. The enhanced speech signal by the proposed
adaptive noise cancellation scheme is shown in Fig. 7. The
improvement in the performance provided by the proposed
-0.4
-0.6
-0.8
0 0.5 1 1.5 2 2.5 3 3.5 4
Time index 4
x 10
Fig. 5  Spectrogram and signal of the primary input
scheme over that of the conventional scheme is evident
when this performance in Fig.  7 is compared with that of
the conventional scheme given in Fig. 8.
4 Real‑time implementation of the proposed
scheme
To examine the real-time behavior, the proposed
scheme is implemented by using Texas Instruments
TMS320C6713DSK. This DSK board contains a 32-bit
stereo codec for input and output (I/O) support and a uni-
versal synchronous bus (USB) cable that connects the DSK
board to a PC. It also includes Code Composer Studio
(CCS), which provides an integrated development envi-
ronment (IDE), bringing together the C compiler, assem-
bler, linker, and debugger. The onboard codec AIC23 uses
a sigma–delta technology that provides ADC and DAC.
It connects to a 12-MHz system clock. Variable sampling

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Int J Speech Technol

Spectrogram of ALE 1 Output Spectrogram of ALE 2 Output

5000 5000

4000 4000

Frequency
Frequency

3000 3000

2000 2000

1000 1000

0 0

Output of ALE1 Output of ALE2

0.8 0.8

0.6 0.6

0.4 0.4

0.2 0.2
Amplitude
Amplitude

0 0

-0.2 -0.2

-0.4 -0.4

-0.6 -0.6

-0.8 -0.8
0.5 1 1.5 2 2.5 3 3.5 4 0.5 1 1.5 2 2.5 3 3.5 4
Time index 4 Time index 4
x 10 x 10

Fig. 6  Spectrograms and signals at the outputs of ALE1 and ALE2

rates from 8 to 96  kHz can be set readily. There are four
connectors on the board provide input and output: MIC IN
for microphone input, LINE IN for line input, LINE OUT
for line output, and HEADPHONE for a headphone out-
put (multiplexed with line output). The DSK operates at
225 MHz. More information about this board can be found
in (Texas Instruments 17).
In our first experiment, we implement the proposed
adaptive noise canceller for the cancellation of sinusoi-
dal noise in the presence of wideband random noise.
Note that in this experiment as well as the next one, the
speech signal is considered to be inactive. The power of
the sinusoidal noise is set to be higher than that of the
wideband noise. The sinusoidal noise and wideband
signals are generated by using an external function gen-
erator and Goldwave respectively. This Goldwave is a
software tool that can be used to generate the wideband
random noise. An adapter with two connectors at one
end for each input signal and one connector at the other
end, which connects to the DSK is required to implement
these experiments in this section. The wideband signal
output is connected into left channel of the adaptor and a
sinusoidal noise with a frequency of 2 kHz is connected
into right channel of the adaptor. Then, the adaptor out-
put is connected to the input line in the DSK. The out-
put line of the DSK is connected to the digital storage
oscilloscope (type RIGOL DS5202CA). Figure 9a shows
the input signal (waveform plot on the top of Fig. 9a) and
its spectrum (FFT plot on the bottom of Fig.  9a) before
adaptation process of the proposed ANC system. Fig-
ure 9b shows the output (error) signal (waveform plot on
the top of Fig. 9b) and its spectrum (FFT plot on the bot-
tom of Fig. 9b) after the adaptation process converged to
the wideband signal. It can be seen from FFT plots of the
input and output signals that the undesirable 2 kHz sinu-
soidal noise is completely cancelled.

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 Int J Speech Technol

5000 5000

4000 4000
Frequency

Frequency
3000 3000

2000 2000

1000 1000

0 0

0.8 0.8

0.6 0.6

0.4 0.4

0.2 0.2
Amplitude

Amplitude
0 0

-0.2 -0.2

-0.4 -0.4

-0.6 -0.6

-0.8 -0.8
0.5 1 1.5 2 2.5 3 3.5 4 0.5 1 1.5 2 2.5 3 3.5 4
Time index 4 Time index 4
x 10 x 10

Fig. 7  Spectrogram (top) and signal at the output (bottom) of the Fig. 8  Spectrogram (top) and signal at the output (bottom) of the
proposed scheme conventional scheme

In our second experiment, we investigate the effect of
multiple sinusoidal noises with different frequencies on
the performance of the proposed scheme. In this case, two
undesired sinusoidal noise signals with frequencies 2 and
1 kHz and one wideband signal were considered. Figure 10a
(top) shows the waveform of the input signal, while the bot-
tom plot in the same figure shows the spectrum of the input
signal. Figure  10b (top) shows the waveform of the out-
put (error) signal, while the bottom plot in the same figure
shows the spectrum of the output signal. It can be seen from
FFT plots of the input and output signals that the undesira-
ble 1 and 2 kHz sinusoidal noises are completely cancelled.
The sinusoidal noise no longer exists in the residual noise of
the proposed ANC scheme that implement on the DSK.
In our last experiment, we port the proposed adaptive
noise canceller to the DSK to examine real time behavior.
An adaptor with two connectors at one end for each input
signal and one connector at the other end, which connects
to the DSK, is also required to implement this experiment.
The desired speech signal is fed through the left chan-
nel of the adaptor and the undesirable signals (sinusoidal
plus wideband) through the right channel of the adaptor.
Figure 11a shows the noisy speech signal at the input and
Fig.  11b is the snapshot of output captured in real time
by using digital storage oscilloscope. It can be seen from
Fig.  11b that the proposed ANC scheme is very effective
in reducing both wideband and sinusoidal noises in the pri-
mary signal. The only sacrifice we have to make is a mod-
erate increase of computational cost due to the addition of
two adaptive line enhancers.
5 Real‑world applications of the proposed scheme
The proposed scheme can be applied successfully in dif-
ferent kinds of environments. Some of them are elaborated
below:

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Int J Speech Technol

Fig. 9  Plots illustrating the

adaptation process obtained
with a digital storage oscil-
loscope; a input signals (2 kHz
sinusoidal and wideband
signals) before adaptation; b
cancellation of sinusoidal noise
and reduction of wideband
noise after adaptation

Fig. 10  Plots illustrating the

adaptation process obtained
with a digital storage oscil-
loscope; a input signals (multi
sinusoidal with frequencies
1, and 2 kHz plus wideband
signal) before adaptation; b
cancellation of both sinusoidal
noises and reduction of wide-
band noise after adaptation

Fig. 11  Plots illustrating
the real-time behavior of the
proposed scheme obtained
with a digital storage oscillo-
scope; a snapshot of the noisy
speech signal; b snapshot of the
enhanced speech signal at the
output of the proposed scheme

5.1 Cell phones
In order to ensure clear voice connectivity, the recent model
of iPhones is deploying the concept of using two input
microphones to cancel out the noise from speech signal
using adaptive noise cancellation (Narendra and Han 2012;
Sugiyama et  al. 2011). The proposed scheme can be used
here to ensure effective speech communication especially
when using cell phone during journey by car or train in
which the desired speech signal encounters various types of

13

noise signals such as the car-engine noise and wind noise.
In this case, the noise signal to be targeted has both wide-
band and sinusoidal components.
5.2 ECG analysis
The electrocardiogram (ECG) is a graphical representation
of hearts functionality and is an important tool used for
diagnosis of cardiac abnormalities. In the clinical environ-
ment during acquisition, the ECG signal encounters various
types of artifacts. The predominant artifacts present in the
ECG includes: Baseline Wander, Power-Line Interference,
Muscle Artifacts, and Motion Artifacts. These artifacts
strongly affects the ST segment, degrades the signal qual-
ity, frequency resolution, produces large amplitude signals
in ECG that can resemble PQRST waveforms, and masks
tiny features that are important for clinical monitoring,
diagnosis. Cancellation of these artifacts in ECG signals is
an important task for better diagnosis (Zia Ur Rahman et al.
2012; Mohammed 2014). The extraction of high-resolu-
tion ECG signals from recordings which are contaminated
with both wideband and sinusoidal noise components is an
important issue to investigate. The proposed scheme can
also be deployed for solving this problem. The sinusoidal
noise as well as wideband noise signals can be effectively
removed by using ALE and LMS filter combination to
yield better results.
5.3 Active noise control
In many real-life active noise control applications to rotat-
ing machines such as factory cutters, fans, motors etc., the
noise to be targeted has both wideband and sinusoidal sig-
nals. Namely, one or more sinusoids usually exist in both
the reference and primary inputs. For such applications,
the conventional narrowband active noise control (Kuo
and Morgan 1999) takes care of the sinusoidal components
only, leaving the wideband component untouched (Xiao
and Wang 2011). On the other hand, the broadband active
noise control (Kuo and Morgan 1999) may also lose its
power for such applications, as it was originally designed
to suppress wideband noise (Xiao and Wang 2011). There-
fore, cancellation of both noise signals simultaneously is an
important task for these applications.
5.4 Propeller induced cabin noise during flight
During flight, the engine noise present in the cockpit heav-
ily corrupts the communication because it consists of mul-
tiple harmonics that occupy the same frequency band as
that of the pilot‘s speech (Winberg et al. 1999). Note that
the noise in this case may be comprised of both sinusoi-
dal and wideband noise signals. The conventional adaptive
13
Int J Speech Technol
noise cancellation scheme when applied will reduce the
wideband noise components only. To recover the pilot‘s
speech effectively, the proposed scheme may be applied to
remove both wideband and sinusoidal noises.
6 Conclusions
In this paper, an efficient and simplified adaptive noise
cancellation scheme has been proposed to reduce both the
wideband and the sinusoidal noises in a noisy environment.
The proposed scheme consists of two cascade stages. The
first stage consists of two adaptive line enhancers placed
in parallel. The second stage consists of an adaptive noise
canceller. First the sinusoidal noise is reduced from the
primary input and reference input signals by using two
adaptive line enhancers. Next the adaptive noise cancel-
ler is used to reduce the wideband noise and to provide the
enhanced speech signal at the output of the system. Simu-
lation and experimental results show that the proposed
scheme has the potential for reducing both the wideband
noise and the sinusoidal noise while maintaining the high
quality of enhanced speech.
References
Bitzer, J., Simmer, K. U., & Kammeyer, K. D. (2001). Multi-micro-
phone noise reduction techniques as front-end devices for speech
recognition. Speech Communication, 34(1–2), 3–12.
Boll, S. F. (1979). Suppression of acoustic noise in speech using spec-
tral subtraction. IEEE Transactions Acoust, Speech, Signal Pro-
cessing, ASSP-27(2), 113–120.
Hayes, M. (1996). Statistical Digital Signal Processing and Mod-
eling. New York: Wiley.
Haykin, S. (2003). Adaptive filter theory (4th ed.). New Delhi: Pear-
son Education (Singapore) Ltd.,.
Jayakumar, E. P., Shifas, P. V. M., Sathidevi, P. S. (2016). Integrated
acoustic echo and noise suppression in modulation domain.
International Journal of Speech Technology, 19(3), 611–621.
Kuo, S. M., & Morgan, D. R. (1999). Active noise control: A tutorial
review. Proceedings of the IEEE, 87(6), 943–973.
Levin, J., Perez-Arancibia, N. O., & Ioannou, P. A. (May 2011).
Adaptive notch filter using real-time parameter estimation. IEEE
Transactions Control Systems Technology, 19(3), 673–681.
Mohammed, J. R. (2009). An efficient method for combining adap-
tive echo and noise canceller in hands-free systems. Interna-
tional Journal of Adaptive Control and Signal Processing, 23,
278–292.
Mohammed, J. R. (2014). Low complexity adaptive noise canceller
for mobile phones based remote health monitoring. International
Journal of Electrical and Computer Engineering (IJECE), 4(3),
422–432.
Moir, T. J. & Harris, J. I. (2012). Decorrelation of multiple non-sta-
tionary sources using a multivariable crosstalk-resistant adaptive
noise canceller. International Journal of Adaptive Control and
Signal Processing 27(5), 349–367.
Int J Speech Technol
Narendra, K. S. & Han, Z. (2012). A new approach to adaptive con-
trol using multiple models,” International Journal of Adaptive
Control and Signal Processing, 26, 778–799.
Niedźwiecki, M. & Meller, M. (2012). Hybrid SONIC: joint feedfor-
ward–feedback narrowband interference canceler. International
Journal of Adaptive Control and Signal Processing, 27(12),
1048–1064.
Ramli R. M., Noor A. O. A., Samad, S. A. (2017). Noise cancella-
tion using selectable adaptive algorithm for speech in variable
noise environment. International Journal of Speech Technology,
online, 1–8.
Sasaoka, N., Sumi, K., Itoh, Y. & Fujii, K. (2005). A new noise reduc-
tion system based on ALE and noise reconstruction filter. Pro-
ceedings of the IEEE International Symposium Circuits Systems
(ISCAS), pp. 272–275.
Sugiyama, A., Miyahara, R., & Kato, M. (2011). An adaptive noise
canceller with adaptive delay compensation for a distant noise
source. Proceedings of the IEEE International Conference on
Acoustics, Speech, and Signal Processing (ICASSP), Vol. 5, pp.
265–268, USA.
Touazi, A., Debyeche, M. (2017). An experimental framework for
Arabic digits speech recognition in noisy environments. Interna-
tional Journal of Speech Technology, 20(2), 205–224.
TMS320C6713 Floating-Point Digital Signal Processor, SPRS186,
Texas Instruments, Dallas, TX.
Widrow, B., Glover, J. R. Jr., McCool, J. M., Kaunitz, J., Williams,
C. S., Hearn, R. H., Zeidler, J. R., Dong, E. Jr., & Goodlin R. C.
(1975). Adaptive noise cancelling: Principles and applications.
Proceedings of the IEEE, 63(12), 1692–1716.
Winberg, M., Johansson, S., Lago, T., & Claesson, I. (1999). A new
passive/active hybrid for a helicopter application. International
Journal of Acoustics and Vibration, 4(2), 51–58.
Xiao, Y. & Wang, J. (2011). A new feedforward hybrid active noise
control system. IEEE Signal Processing Letters, 18(no. 10),
591–594.
Zia Ur Rahman, M., Shaik, R. A., & Rama Koti Reddy, D. V. (2012).
Efficient and simplified adaptive noise cancelers for ECG sensor
based remote health monitoring. IEEE Sensors Journal, 12(3),
566–573.
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