INTERNATIONAL JOURNAL OF LATEST TECHNOLOGY IN ENGINEERING,  
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ISSN 2278-2540 | DOI: 10.51583/IJLTEMAS | Volume XV, Issue I, January 2026  
Noise Reduction and Signal Estimation For 5g Anntenna Using Least  
Mean Square (LMS) Algorithm and Kalman Filter  
Bayem Donatus I, Alumona Theopilus  
Department of Electronic and Computer Engineering, Faculty of Engineering, Nnamdi Azikiwe  
University Awka  
Received: 05 January 2026; Accepted: 13 January 2026; Published: 24 January 2026  
ABSTRACT  
The Kalman filter code used in the active control system is described in detail in this thesis. In order to react to  
variations in the primary noise and 5G environment, traditional active noise management techniques typically  
use an adaptive filter, such as the filtered reference least mean square (FxLMS) algorithm. However, the weak  
convergence features of the FxLMS algorithm typically affect how well dynamic noise is reduced. This research  
suggests utilizing the Kalman filter in the active noise control (ANC) system to enhance the efficacy of noise  
reduction for dynamic noise. The Kalman filter is used effectively by the ANC application using a new dynamic  
ANC model. The numerical simulation shows that the proposed Kalman filter works better than the FxLMS  
approach in terms of convergence performance for handling dynamic noise. This suggests that the transition of  
the control filter has a higher degree of confidence than the observation function. When the effects of various  
mu and K values were examined, it was discovered that the LMS algorithms had a slow rate of convergence for  
mu = 0.01 and high starting error signals for both echo and noise cancellation that subsequently decreased.  
Although there was some improvement in the noise and echo-canceled signals, the residual noise and echo  
persisted. Error signals dropped more quickly and the convergence rate was better with mu = 0.05 than with mu  
= 0.01; this suggests more efficient cancellation. There was less lingering echo and noise in the resulting clearer  
signals, with echo and noise cancelled. Hybrid LMS algorithms exhibited rapid convergence at μ = 0.1, with  
error signals declining sharply, indicating effective cancellation. With little lingering interference, the noise-  
cancelled and echo-cancelled signals were noticeably clearer. Although the convergence was quite quick with  
mu = 0.5, there was a higher chance of instability, particularly for the LMS method. It is often advised to use a  
hybrid algorithm with a mu value of between 0.05 and 0.1. Achieving the ideal balance between convergence  
speed and stability requires proper mu tuning, which guarantees efficient cancellation without creating  
instability.  
INTRODUCTION  
The fifth-generation (5G) technology, which promises previously unheard-of speeds, capacity, and connection,  
is the result of wireless communication's growth. Managing and reducing noise, which can seriously impair  
signal quality and system performance, is one of the major issues facing 5G networks. For 5G antennas to  
guarantee high data speeds and dependable connectivity, noise reduction is consequently crucial (Araújo and  
Almeida, 2019). Massive machine-type communication, ultra-reliable low latency communication, and  
improved mobile broadband are the goals of 5G technology. To achieve these objectives, advanced signal  
processing methods are needed to manage the network's growing density and complexity. Noise is a major  
problem that can originate from a number of sources, such as ambient conditions, interference from other  
devices, and thermal noise. Any undesired signal that obstructs the intended communication signal is called  
noise. Noise can cause errors and lower service quality in wireless communication by distorting the sent signal  
(Uwaechia and Mahyuddin, 2019). To preserve the integrity of the data being communicated and to maximize  
the performance of communication systems, effective noise reduction techniques are crucial. With its promise  
of previously unheard-of data speeds, extremely low latency, and extensive connectivity, 5G technology  
represents a major turning point in the development of wireless communication. Applications ranging from  
augmented reality and Internet of Things (IoT) gadgets to driverless cars and smart cities will be supported by  
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5G networks. Advanced antenna systems, such as large Multiple Input Multiple Output (MIMO) and  
beamforming technologies, which are essential to 5G infrastructure, are required to achieve these capabilities.  
REVIEW OF RELATED LITERATURE  
Srinivas et al. (2022) suggest a semi-blind estimator that uses one-bit Massive MIMO to mitigate the impact of  
a polluted pilot, which could significantly impair system performance. To enhance the CSI estimation, the semi-  
blind approach makes use of pilots and a few data symbols. Spectral efficiency and estimation accuracy are  
increased when data symbols are used in the channel estimation process. When compared to the pilot-based  
estimators, the suggested iteration-based semi blind one-bit massive MIMO algorithm performed better in terms  
of BER and MSE. Moreover, although utilizing a small number of pilot symbols, the spectral efficiency has been  
enhanced.  
The Enhanced Channel Estimation for MIMO¬OFDM in 5G NR was examined by Taheri et al. in 2021. They  
believed that Two potential technologies to achieve high data rate transmission capabilities, excellent spectral  
efficiency, and greater robustness against multi-path fading are Orthogonal Frequency Division Multiplexing  
(OFDM) and Multiple Input Multiple Output (MIMO). An essential component of the MIMO OFDM system is  
the channel estimating approach, which is utilized to recover the originally sent signal by reducing the impact of  
the channel. Training-based, semi-blind, and blind channel estimation algorithms are the three primary categories  
into which the channel estimation techniques fall.  
The study's primary goal was to choose and put into practice an appropriate channel estimation algorithm with  
a low degree of computational complexity that produces good performance. This study examined and carried  
out channel estimation in the MIMO¬OFDM system using block type pilot symbol layout. The Low Rank  
Approximation (LRA¬LMMSE) and Minimum Mean Square Error (MMSE) algorithms are the channel  
estimation techniques that have been examined. The performance of algorithms' Bit Error Rate (BER) and Block  
Error Rate (BLER) as well as their level of computational complexity are the primary subjects of this study.  
MATLAB simulations based on BER and BLER vs Single to Noise Rate (SNR) are used to assess the efficacy  
of the developed techniques. The assessment takes into account each algorithm's level of computational  
complexity. According to the final results of several channel models, LRA¬LMMSE performs better than Linear  
Minimum Mean Square Error (LMMSE) in terms of complexity degree, whereas MMSE surpasses Least Square  
(LS)  
in  
terms of BER and BLER.  
Multiple Input Multiple Output-Orthogonal Frequency Division Multiplexing (MIMO-OFDM) is a well-known  
contemporary wireless broadband technology because of its spectral efficiency, high data transmission rate, and  
resilience to multipath fading (Manasa & Venugopal, 2023). This method offers a wide range of coverage and  
reliable communication. Two significant issues for MIMO-OFDM systems are the accurate recovery of Channel  
State Information (CSI) and the synchronization between the transmitter and receiver. Channel state information  
is recovered using a variety of estimate techniques, including blind, pilot-aided, and semi-blind channel  
estimating. However, those systems operate poorly due to a number of shortcomings. Therefore, the basic  
introduction of the Channel Estimation (CE) process in the MIMO-OFDM system is described in this study.  
This survey's primary objective is to investigate the analysis and classification of simulation tools and channel  
estimation techniques in various contributions. It also highlights the performance study using various  
performance measures from various contributors.  
Mathematical Formulation  
Least Mean Square (LMS) Algorithm  
An adaptive filter, the LMS method modifies its filter coefficients to reduce the mean square error (MSE)  
between the output signal and the intended signal. The following stages can be used to express the LMS  
algorithm, which is derived from Wiener filter theory:  
Initialization: Initialize the filter weights w(0)w(0)w(0) to zero or small random values.  
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(3.1)  
Filter Output: Compute the filter output y(n)y(n)y(n) by convolving the input signal x(n)x(n)x(n) with the filter  
weights w(n)w(n)w(n):  
(3.2)  
Error Signal: Determine the difference between the intended signal and the error signal, e(n)e(n)e(n).  
d(n)d(n)d(n) and the filter output y(n)y(n)y(n):  
(3.3)  
Weight Update: Update the filter weights using the LMS update rule:  
(3.4)  
where μ\muμ is the learning rate, or step size, that regulates the algorithm's stability and rate of convergence.  
Until the weights converge to their ideal values and the mean square error is minimized, the iterative process is  
continued.  
Kalman Filter  
One of the best recursive algorithms for estimating the state of dynamic systems is the Kalman filter. By reducing  
the mean square error, it offers the most accurate approximation of the state of the system. Prediction and  
updating are the two primary processes via which the Kalman filter functions.  
State Space Model: Define the state space model of the system:  
(3.5)  
(3.6)  
where uku_kuk is the control input, wkw_kwk and vkv_kvk are the process and measurement noise, respectively,  
and xkx_kxk is the state vector at time kkk. It is assumed that the noise is Gaussian with zero mean and  
covariance matrices QQQ and RRR.  
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Prediction Step: Estimate the error covariance Pkk−1P_{k|k-1}Pkk−1 and the next state  
x^kk−1\hat{x}_{k|k-1}x^kk−1:  
(3.7)  
(3.8)  
Update Step: Compute the Kalman gain KkK_kKk:  
(3.9)  
State Estimate and Error Covariance:  
(3.10)  
(3.11)  
To provide the best state estimation, the Kalman filter iteratively improves both the state estimate and the error  
covariance.  
(3.12)  
Kalman Prediction: Utilizing the Kalman filter equations, forecast the subsequent state and error covariance:  
(3.13)  
(3.14)  
Update the state estimate and error covariance with the latest measurement:  
(3.15)  
(3.16)  
(3.17)  
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By iteratively updating the filter weights with the LMS algorithm and fine-tuning the state estimate with the  
Kalman filter, the hybrid technique improves signal estimation and noise reduction performance.  
System Model Simulation  
The project's implementation was carried out using a MATLAB simulation framework. The downlink physical  
shared channel layer is where the system is simulated. Throughout the implementation process, MATLAB's  
signal processing and communication toolboxes were utilized. The communication toolbox provides test and  
measurement tools, such as bit error rate, waveform generator, CRC encoding, and decoding operations, to  
examine and verify the channel system that has been put into place. The signal processing toolbox made it easier  
to build the channel estimators by doing a number of tasks, including calculating the covariance, measuring the  
SNR, and autocorrelation of the matrix. Two statistical techniques for figuring out the relationship between two  
random variables in two distinct matrices are covariance and autocorrelation.  
The methodologies differ in that the autocorrelation measures the strength of the linear relationship between  
variables, whereas the covariance measures the linear relationship between two variables. As the number of  
antennas grows, the channel estimation's complexity rises linearly. Consequently, only 4x4that is, four antenna  
ports on the transmitter side and four antenna ports on the reception endis supported by the system.  
Implementing the system in the frequency domain is simpler than doing so in the time domain. Four transmitters  
(Tx), a channel, and four receivers (Rx) make up the majority of the system. Different outcomes of channel  
estimation were examined using a variety of channel model types.  
Channel model cases are:  
To add noise that replicates the noise in an actual communication environment, the communication  
toolbox offers an AWGN channel model function.  
One reception antenna is correlated by the channel for each transmission phase in the MIMO simple  
channel model. The channel correlation coefficient, which is multiplied by the data being communicated  
during the transmission, is shown in the following dcequation.  
(3.18)  
(3.19)  
MIMO sparse channel model, in which one or two receiver antennas are correlated by the channel for each  
transmission phase. The channel correlation coefficient is shown in the following equation:  
(3.20)  
(3.21)  
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In addition to their channel correlation, the basic MIMO and sparse MIMO channels also include AWGN.  
Figure 3.1: Block diagram of System model  
In order to assess the hybrid LMS-Kalman filter's performance, we constructed a simulation environment using  
the following settings:  
1. System Model:  
Carrier frequency: 28 GHz  
Bandwidth: 100 MHz  
Number of antennas: 64 (8x8 MIMO configuration)  
Channel model: Rayleigh fading  
2. Noise Model:  
Additive White Gaussian Noise (AWGN)  
Signal-to-Noise Ratio (SNR): Varied from 0 to 30 dB  
3. Performance Metrics:  
Signal-to-Interference-plus-Noise Ratio (SINR)  
Mean Square Error (MSE)  
Convergence speed  
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CRC Implementation  
The CRC block for error detection was implemented to start the channel coding process. The CRC Consultative  
Committee for International Telephony (CCIT) generator object is defined as gen by the crc.generator on the  
transmitter side. The 16-bit object gen defines a number of characteristics and generates the checksum. The  
transferred data bits are divided by a polynomial in many steps to determine the checksum. The 16-bit checksum  
(gen) is appended to the end of the sent data by the function generate. A 16-bit CRCCCIT detector object det is  
constructed on the receiver side by crc.detector. A number of properties are defined by the object det.  
To recreate a checksum on the recipient's end, the polynomial division is repeated. By comparing the object det  
regenerated checksum with the object gen checksum attached to the transmitted data, the function detect finds  
the problem in the received data. The function gives the number of subframe (block) faults as well as the output  
data with the checksum removed. The outdata and indata bits are compared by the function isequal. Data was  
not impacted or altered during transmission if the outcome is equal, and vice versa. The rate of the number bit  
errors is returned by the TotalBLER function.  
Channel Estimation Implementation  
The transmitted pilot DMRS symbols (reference signal) provided by dmrsSym are utilized in the channel  
estimation process to estimate the channel coefficient specified by H, whereas dmrsDe specifies the DMRS  
symbols built at the receiver side. To recover the signal that was initially delivered, the receiver decodes the  
received signal using channel estimation. The LS and LMMSE algorithms are the channel estimation methods  
that are currently in use in the system; LS is used in four transmission phases. It is generally recognized that the  
pilot symbols channel estimations operate well. One disadvantage, though, is that delivering the pilot and data  
symbols simultaneously increases the transmission overhead.  
The channel estimation algorithms MMSE and LRALMMSE are selected for implementation because they  
estimate the channel coefficient using training symbols.  
Figure 3.2: MMSE channel estimation  
Algorithm Implementation  
LMS Algorithm  
In order to reduce the mean square error (MSE) between the intended and actual output, the LMS algorithm  
adaptively modifies the filter coefficients using a stochastic gradient-based methodology. The following is the  
fundamental LMS update equation:  
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where:  
w(n)w(n)w(n) is the weight vector at iteration nnn.  
μ\muμ is the step size (learning rate).  
e(n)e(n)e(n) is the error signal, defined as e(n)=d(n)−y(n)e(n) = d(n) - y(n)e(n)=d(n)−y(n).  
d(n)d(n)d(n) is the desired signal.  
y(n)y(n)y(n) is the output signal of the adaptive filter, y(n)=wT(n)x(n)y(n) =  
w^T(n)x(n)y(n)=wT(n)x(n).  
x(n)x(n)x(n) is the input signal vector.  
The step size μ\muμ, which is crucial for stability and quick convergence, determines how well the LMS  
algorithm converges. The following procedures are used to implement the LMS algorithm:  
Initialization:  
(3.22)  
Initialize the filter weights to zero or small random values.  
Filter Output:  
(3.23)  
Convolution of the input signal with the filter weights yields the filter output.  
Error Signal:  
(3.24)  
The difference between the intended signal and the filter output is the error signal.  
(3.25)  
Kalman Filter  
The following procedures are used to implement the Kalman filter:  
State Space Model: Describe the system's state space model:  
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(3.26)  
(3.27)  
Prediction Step:  
(3.28)  
(3.29)  
(3.30)  
(3.31)  
(3.32)  
Hybrid LMS-Kalman Filter Implementation  
The LMS and Kalman filter processes are integrated to create the hybrid LMS-Kalman filter. The procedure  
entails:  
Set the LMS filter weights and Kalman filter parameters to their initial values.  
Utilizing the LMS method, update the filter weights.  
Use the Kalman filter to forecast the next state and error covariance.  
Apply the most recent measurement to the state estimate and error covariance.  
Performance Evaluation  
The following measures are used to assess the hybrid LMS-Kalman filter's performance:  
Signal-to-Interference-plus-Noise Ratio (SINR):  
(3.33)  
Where PsP_sPs, PiP_iPi, and PnP_nPn are the signal, interference, and noise powers, respectively.  
Mean Square Error (MSE):  
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(3.34)  
where y(n)y(n)y(n) is the filter output, d(n)d(n)d(n) is the desired signal, and NNN is the number of samples.  
Convergence Speed: How much iteration is necessary to get the algorithm to a stable state? Setting up the  
state function is the first step in using the Kalman filter methodology.  
Figure 3.3: Simulation Flowchart  
Result Analysis  
The details of the observations for each set of mu values are described below:  
Mu = 0.01 0.05  
So, that more noise will not be generated  
For K= 2:N  
K= 3:N  
K= 4:N  
K= 5:N  
Where N = number of time steps  
In Kalman gain with updated Noise Variance;  
Where Legends Time State  
Estimated State  
-
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In measurement of Noise Variance Estimation;  
Where Legends True Noise Variance  
-
Estimated Noise Variance  
State Estimation  
1
0
True State  
Estimated State  
-1  
0
0
0
10  
10  
10  
20  
20  
20  
30  
30  
40  
50  
60  
70  
70  
80  
90  
100  
100  
100  
Measurements  
2
0
-2  
40  
50  
60  
80  
90  
Measurement Noise Variance Estimation  
0.5  
Estimated Noise Variance  
True Noise Variance  
0
30  
40  
50  
60  
70  
80  
90  
Time step  
Figure 4.1 showing when the value of K is 2 with step size of noise variance at 0.01  
State Estimation  
2
True State  
Estimated State  
0
-2  
0
0
0
10  
10  
10  
20  
20  
20  
30  
30  
40  
50  
60  
70  
70  
80  
90  
100  
100  
100  
Measurements  
2
0
-2  
40  
50  
60  
80  
90  
Measurement Noise Variance Estimation  
0.5  
Estimated Noise Variance  
True Noise Variance  
0
30  
40  
50  
60  
70  
80  
90  
Time step  
Figure 4.2 showing when the value of K is 3 with step size of noise variance at 0.01  
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State Estimation  
2
0
True State  
Estimated State  
-2  
0
0
0
10  
10  
10  
20  
20  
20  
30  
30  
40  
50  
60  
70  
70  
80  
90  
100  
100  
100  
Measurements  
2
0
-2  
40  
50  
60  
80  
90  
Measurement Noise Variance Estimation  
0.5  
Estimated Noise Variance  
True Noise Variance  
0
30  
40  
50  
60  
70  
80  
90  
Time step  
Figure 4.3 showing when the value of K is 4 with step size of noise variance at 0.01  
State Estimation  
1
True State  
Estimated State  
0
-1  
0
0
0
10  
10  
10  
20  
20  
20  
30  
30  
40  
50  
60  
70  
70  
80  
90  
100  
100  
100  
Measurements  
2
0
-2  
40  
50  
60  
80  
90  
Measurement Noise Variance Estimation  
0.5  
Estimated Noise Variance  
True Noise Variance  
0
30  
40  
50  
60  
70  
80  
90  
Time step  
Figure 4.4 showing when the value of K is 5 with step size of noise variance at 0.01  
From figure 4.1 to figure 4.4 -: Based on the innovation squared error, the LMS algorithm modifies the  
measurement noise variance estimate \(r_k\). This hybrid strategy enhances state estimation and noise variance  
tracking in non-stationary noise settings, as the Kalman filter updates the state estimate using this adaptive noise  
variance  
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Kalman Filter Approach for Active Noise Control  
Figure 4.5: Block diagram of the modified ANC structure based on the FxLMS algorithm  
Figure 4.5 shows the improved FxLMS algorithm, another well-known method for adaptive active noise  
reduction. By recovering the disturbance from the error signal using the secondary path estimate, the internal  
model technique makes it possible to control noise using the conventional least mean square (LMS) algorithm.  
This modified setup can also be used to apply the Kalman filter methodology, which involves replacing the LMS  
model with the Kalman filter to finish updating the control filter. Setting up the state function is the first step in  
using the Kalman filter methodology. Using Appendis 2, the next paragraphs will clarify the ANC's state function  
definition.  
Code configuration of loading the primary and secondary path  
The KF.mat file, which applies the Kalman filter method for a single-channel active noise control (ANC)  
application, is briefly introduced in this section. Also, a comparative examination of the FxLMS algorithm is  
carried out. While the FxLMS method uses the traditional feed-forward ANC structure, the Kalman filter  
technique uses the modified feed-forward active noise control (ANC) structure.  
The primary path and secondary path are loaded in this section of the code from the Mat files, PriPath_3200.mat  
and SecPath_200_6000.mat. All these paths are synthesized from the band-pass filters, whose impulse responses  
are illustrated in Figure 4.  
load (PriPath_ 3200 . mat’);  
load (SecPath_200_6000 .mat’) ;  
subplot (2 ,1 ,1)  
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Figure 4:6a The impulse response of the primary path and the secondary path  
Simulation system configuration  
The active noise control (ANC) system's sampling rate is set to 16,000 Hz, and the simulation lasts for 0.25  
seconds. As seen in Figure 5, the main noise in this ANC system is a chirp signal, whose frequency progressively  
fluctuates between 20 Hz and 1600 Hz in order to replicate the dynamic noise.  
Parameter  
Definition  
Sampling rate  
Primary noise  
Parameter  
Definition  
fs  
y
T
N
Simulation duration  
Simulation taps  
Table 1: Simulation input parameters value description and definition  
% sampling rate 16 kHz  
; % Simulation duration (seconds ).  
;% Time variable .  
fs =  
= 0.25  
= 0:1/ fs:T  
=
fw  
500  
=
T,1600  
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Figure 4: 6b The waveform of the reference signal that is a chirp signal ranging from  
load (PriPath_ 3200 . mat’);  
load (SecPath_ 200_6000 .mat’) ;  
figure  
;
subplot (2 ,1 ,1)  
plot( PriPath );  
title(Primary Path );  
grid on  
;
20 o 1600 Hz.  
Creating the disturbance and filtered reference  
The chirp signal is passed through the loaded primary and secondary routes to provide the disturbance and  
filtered reference utilized in the ANC system.  
Parameter  
Definition  
Parameter  
Y
Definition  
X
D
Reference signal vector  
Disturbance vector  
Filtered reference vector  
Primary noise  
PriPath  
SecPath  
Primary path vector  
Secondary path vector  
Rf  
Table 2: Simulation input parameters value description and definition  
=
% plot(X(end -100: end ))  
D
=
=
% plot(D(end -100: end ))  
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Dynamic noise cancellation by the single-channel FxLMS algorithm  
Here, the chirp disturbance is minimized using the single-channel FxLMS method. In the FxLMS method, the  
control filter's length consists of 80 taps, with a step size of 0.0005. The error signal detected by the ANC  
system's error sensor is seen in Figure 4.6. This figure demonstrates that the dynamic noise during the 0.25  
second cannot be completely attenuated by the FxLMS algorithm.  
Parameter Definition  
Parameter Definition  
X
D
L
Reference signal vector  
Disturbance vector  
Y
E
Control signal  
Error signal  
Step size  
Length of the control filtermuW  
Table 3: Simulation input parameters value description and definition  
= dsp  
=
Figure 4.6c the error signal of the single-channel ANC system based on the FxLMS algorithm.  
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Dynamic noise cancellation by the Kalman filter approach  
The Kalman filter is employed in the signal-channel ANC system to track the fluctuation of the chirp disturbance.  
The variance of the observed noise is initially set to 0.005, and the auto- correlation matrix of the state error is  
initially set to I, respectively. Figure 4. 7 The chirp disturbance was reduced in this section using the single-  
channel FxLMS technique. The step size is set to 0.0005, and the length control filter of the FxLMS algorithm  
has 80 taps. The error signal of the Kalman filter algorithm is displayed in Figure 4.6. Furthermore, Figure 4.8  
shows how the coefficients w5(n) and w60(n) change over time. The result demonstrates how well the Kalman  
filter reduces the chirp disruptions. As shown in Figure 4.9, the convergence behavior of the Kalman filter  
approach is noticeably better than that of the FxLMS method. Appendix 1 illustrates the fluctuation caused by  
the ANC system's error sensor. This figure demonstrates that the dynamic noise during the 0.25 second cannot  
be completely attenuated by the FxLMS algorithm.  
Parameter Definition  
Parameter Definition  
q
Variance of observe error  
P
Cross-correlation matrix of state error  
W
Xd  
Rf  
Control filter  
Ek  
Yt  
Error signal  
Anti-noise  
Input vector  
Reference signal vector  
Table 4: Simulation input parameters value description and definition  
Figure .4 .7: The error signal of the single-channel ANC system based on the Kalman filter.  
Figure 4. 8: The time history of the coefficients w5(n) and w60(n) in the control filter.  
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Figure4. 9: Comparison of the error signals in the FxLMS algorithm and the Kalman filter.  
CONCLUSION  
The conclusion was based on the outcomes of the simulation. This thesis offers a thorough examination of the  
active control system's Kalman filter code. The adaptive filter, such as the filtered reference least mean square  
(FxLMS) algorithm, is usually adjusted by traditional active noise management to accommodate variations in  
the primary noise and 5G environment. However, the noise reduction for dynamic noise is typically impacted  
by the FxLMS algorithm's slow convergence behavior. In order to enhance the noise reduction performance for  
dynamic noise, this work x-rayed employing the Kalman filter in the ANC system. The Kalman filter performs  
exceptionally well in the ANC application with a new dynamic ANC model. When it comes to handling dynamic  
noise, the numerical simulation showed that the suggested Kalman filter performs significantly better in terms  
of convergence than the FxLMS algorithm.  
It should be noted that the approach still accounts for the observed error even though it assumes a variance of 0  
for the state error. This suggests that the transition of the control filter has a higher degree of confidence than the  
observation function.  
It should be noted that the approach still accounts for the observed error even though it assumes a variance of 0  
for the state error. This suggests that the transition of the control filter has a higher degree of confidence than the  
observation function.  
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MANAGEMENT & APPLIED SCIENCE (IJLTEMAS)  
ISSN 2278-2540 | DOI: 10.51583/IJLTEMAS | Volume XV, Issue I, January 2026  
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