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Dive into the research topics where Muhammad Tahir Akhtar is active.

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Featured researches published by Muhammad Tahir Akhtar.


IEEE Transactions on Audio, Speech, and Language Processing | 2006

A new variable step size LMS algorithm-based method for improved online secondary path modeling in active noise control systems

Muhammad Tahir Akhtar; Masahide Abe; Masayuki Kawamata

This paper proposes a new method for online secondary path modeling in active noise control systems. The existing methods for active noise control systems with online secondary path modeling consist of three adaptive filters. The main feature of the proposed method is that it uses only two adaptive filters. In the proposed method, the modified-FxLMS (MFxLMS) algorithm is used in adapting the noise control filter and a new variable step size (VSS) least mean square (LMS) algorithm is proposed for adaptation of the secondary path modeling filter. This VSS LMS algorithm is different from the normalized-LMS (NLMS) algorithm, where the step size is varied in accordance with the power of the reference signal. Here, on the other hand, the step size is varied in accordance with the power of the disturbance signal in the desired response of the modeling filter. The basic idea of the proposed VSS algorithm stems from the fact that the disturbance signal in the desired response of the modeling filter is decreasing in nature, (ideally) converging to zero. Hence, a small step size is used initially and later its value is increased accordingly. The disturbance signal, however, is not available directly, and we propose an indirect method to track its variations. Computer simulations show that the proposed method gives better performance than the existing methods. This improved performance is achieved at the cost of a slightly increased computational complexity.


Signal Processing | 2012

Employing spatially constrained ICA and wavelet denoising, for automatic removal of artifacts from multichannel EEG data

Muhammad Tahir Akhtar; Wataru Mitsuhashi; Christopher J. James

Detecting artifacts produced in electroencephalographic (EEG) data by muscle activity, eye blinks and electrical noise, etc., is an important problem in EEG signal processing research. These artifacts must be corrected before further analysis because it renders subsequent analysis very error-prone. One solution is to reject the data segment if artifact is present during the observation interval, however, the rejected data segment could contain important information masked by the artifact. The independent component analysis (ICA) can be an effective and applicable method for EEG denoising. The goal of this paper is to propose a framework, based on ICA and wavelet denoising (WD), to improve the pre-processing of EEG signals. In particular we employ concept of the spatially constrained ICA (SCICA) to extract artifact-only independent components (ICs) from the given EEG data, use WD to remove any cerebral activity from the extracted-artifacts ICs, and finally project back the artifacts to be subtracted from EEG signals to get clean EEG data. The main advantage of the proposed approach is faster computation, as it is not necessary to identify all ICs. Computer experiments are carried out, which demonstrate effectiveness of the proposed approach in removing focal artifacts that can be well separated by SCICA.


international symposium on circuits and systems | 2005

A method for online secondary path modeling in active noise control systems

Muhammad Tahir Akhtar; Masahide Abe; Masayuki Kawamata

The paper proposes a new method for online secondary path modeling in active noise control (ANC) systems. The proposed method is a modified version of the basic method, proposed by L.J. Eriksson and M.C. Allie (1989). The objective is to realize improved performance at a reasonable computational cost. The control filter is adapted by using the same error signal as used in the adaptation of the modeling filter. Furthermore, a variable step size (VSS) LMS algorithm is used to adapt the modeling filter. The basic idea of VSS stems from the fact that the disturbance signal in the desired response of the modeling filter is decreasing in nature, (ideally) converging to zero. Hence, a small step size is used initially and later its value is increased accordingly. The computational complexity of the proposed method is comparable to the Eriksson and Allie method, and it gives the best performance among existing methods. Computer simulations are presented to show the effectiveness of the proposed method.


IEEE Transactions on Audio, Speech, and Language Processing | 2007

On Active Noise Control Systems With Online Acoustic Feedback Path Modeling

Muhammad Tahir Akhtar; Masahide Abe; Masayuki Kawamata

The presence of strong acoustic feedback degrades the convergence speed of the active noise control (ANC) filter, and in the worst case the ANC system may become unstable. A fixed feedback neutralization filter, obtained offline, can be used to neutralize the acoustic feedback. The feedback path, however, may be time varying, and we may need continual adjustments during online operation of the ANC system. This paper proposes a new method for online modeling of the acoustic feedback path in ANC systems. The proposed method uses three adaptive filters; a noise control filter, a feedback path modeling (FBPM) filter, and an adaptive noise cancelation (ADNC) filter. The objective of ADNC filter is to remove the disturbance from the desired response of FBPM filter. In comparison with the existing method, which works only for predictable noise sources, the proposed method can work, as well, with the broadband noise sources. The computer simulations are carried out for narrowband (predictable) (case I) and broadband (random) noise sources (case II). It is demonstrated that the proposed method performs better than the existing method in both cases


international symposium on circuits and systems | 2012

Recursive independent component analysis for online blind source separation

Muhammad Tahir Akhtar; Tzyy-Ping Jung; Scott Makeig; Gert Cauwenberghs

This study proposes and evaluates a recursive algorithm for incremental estimation of independent components from on-line data. The algorithm offers the convergence properties of batch independent component analysis (ICA) with incremental updates of a form similar to natural gradient (NG) on-line information maximization (Infomax). We employ recursive procedure to arrive at steady state solution given by NG Infomax. Furthermore, we propose a novel procedure to compute corrective updates on the basis of previous estimates. Implementation of this algorithm incurs linear complexity in data size, input dimensions, and number of estimated independent components. Significant gains in convergence rate over on-line natural gradient ICA are demonstrated.


IEEE Transactions on Audio, Speech, and Language Processing | 2011

Improving Performance of Hybrid Active Noise Control Systems for Uncorrelated Narrowband Disturbances

Muhammad Tahir Akhtar; Wataru Mitsuhashi

In filtered-x LMS (FxLMS) single-channel feedforward active noise control (ANC) systems, a reference signal is available that is correlated with the primary disturbance at the error microphone. In some practical situations, there may also be a disturbance uncorrelated with the primary disturbance at the error microphone, for which a correlated reference signal is not available. This disturbance, being uncorrelated with the primary noise, cannot be controlled by the standard FxLMS algorithm, and increases the residual noise. In this paper we propose an improved hybrid ANC system that can simultaneously control both the correlated and uncorrelated noise signals. The proposed method comprises three adaptive filters: 1) the FxLMS-based ANC filter to cancel the primary noise; 2) a separate FxLMS-based ANC filter to cancel the uncorrelated disturbance; and 3) an LMS-based supporting adaptive filter to generate appropriate signals for the two ANC filters. Computer simulations demonstrate that the proposed method can effectively mitigate the correlated and uncorrelated primary disturbances. This improved performance is achieved at only a small increase in computational complexity.


IEEE Transactions on Speech and Audio Processing | 2005

A new structure for feedforward active noise control systems with improved online secondary path modeling

Muhammad Tahir Akhtar; Masahide Abe; Masayuki Kawamata

This paper proposes a new structure for feedforward active noise control (ANC) systems with online secondary path modeling. The proposed method: 1) uses the same error signal for updating the noise control process as used for the secondary path modeling process and 2) incorporates an adaptive filtering with averaging based filtered-reference algorithm in the noise control process. The computer simulations have been conducted with both narrowband and broadband noise signals. It is shown that in the proposed ANC system the residual noise signal and the secondary-path-modeling error can be reduced at a faster convergence rate than the existing methods. This improved performance is achieved at the expense of a slightly increased computational complexity.


IEICE Electronics Express | 2007

Noise power scheduling in active noise control systems with online secondary path modeling

Muhammad Tahir Akhtar; Masahide Abe; Masayuki Kawamata

In active noise control (ANC) systems, the online secondary path modeling (OSPM) methods that use additive random noise are often applied. The additive random noise, however, contributes to the residual noise, and thus deteriorates the noise control performance of ANC systems. This paper proposes a new OSPM method with power scheduling of additive random noise. Here the OSPM filter is adapted using a variable step size (VSS) LMS algorithm already proposed by authors. Furthermore, the additive-random-noise power is scheduled based on the convergence status of an ANC system. Computer simulations demonstrate the effectiveness of the proposed method.


Signal Processing | 2008

Online secondary path modeling in multichannel active noise control systems using variable step size

Muhammad Tahir Akhtar; Masahide Abe; Masayuki Kawamata; Akinori Nishihara

In single-channel feedforward active noise control (ANC) systems, additive random noise based methods are often applied to achieve secondary path modeling (SPM) during online operation. This paper investigates the issue of online SPM in multichannel ANC systems. It is shown that the application of existing methods for online SPM in multichannel ANC systems greatly increases the computational complexity. Here we extend our previous work on single-channel variable step-size online SPM to multichannel ANC systems. It is shown that the proposed method has reduced computational complexity as compared with other methods. Computer simulations are carried out that demonstrate the effectiveness of the proposed method.


IEEE Transactions on Audio, Speech, and Language Processing | 2013

Robust Auxiliary-Noise-Power Scheduling in Active Noise Control Systems With Online Secondary Path Modeling

Shakeel Ahmed; Muhammad Tahir Akhtar; Xi Zhang

This paper deals with the auxiliary noise-based methods for active noise control (ANC) systems with online secondary path modeling (SPM). The proposed method comprises two adaptive filters: the modified Filtered-X normalized least-mean-square algorithm-based ANC filter, and the normalized least-mean-square algorithm-based SPM filter excited by auxiliary noise. The auxiliary noise injected for online SPM, degrades the noise-reduction performance of the ANC system. A two-stage gain scheduling strategy is proposed to vary power of the auxiliary noise. In the first stage the gain is varied on the basis of power of the error signal of SPM filter, and in the second stage the gain is varied on the basis of the correlation estimate of the two adjacent samples of the error signal of SPM filter. The main idea is to inject large-power auxiliary noise at the start up or when a change in the acoustic paths is detected, and to reduce the power as the system converges. The proposed method achieves a fast convergence of the SPM filter and gives a robust performance in the presence of strong perturbation in acoustic paths. Furthermore, the proposed method improves the noise-reduction performance at steady-state even in the presence of an uncorrelated disturbance at the error microphone. Moreover, the improved performance is achieved at a lower computational cost as compared with a recent method proposed in [A. Carini, and S. Malatini, “Optimal variable step-size NLMS algorithms with auxiliary noise power scheduling for feedforward active noise control,”IEEE Trans. Audio, Speech Lang. Process., vol. 16, no. 8, pp. 1383-1395, Nov. 2008]. Extensive simulations are carried out to verify the effectiveness of the proposed method.

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Wataru Mitsuhashi

University of Electro-Communications

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Akinori Nishihara

Tokyo Institute of Technology

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Shakeel Ahmed

University of Electro-Communications

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Xi Zhang

University of Electro-Communications

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Muhammad Tufail

Pakistan Institute of Engineering and Applied Sciences

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Marko Kanadi

University of Electro-Communications

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Shakeel Ahmed

University of Electro-Communications

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