List of Articles LMS

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  1 - A New Switched-beam Setup for Adaptive Antenna Array Beamforming
  Shahriar Shirvani Moghaddam Farida Akbari
  10.7508/jist.2016.01.001
  In this paper, a new spatio-temporal based approach is proposed which improves the speed and performance of temporal-based algorithms, conventional Least Mean Square (LMS), Normalized LMS (NLMS) and Variable Step-size LMS (VSLMS), by using the switched beam technique. I More

  In this paper, a new spatio-temporal based approach is proposed which improves the speed and performance of temporal-based algorithms, conventional Least Mean Square (LMS), Normalized LMS (NLMS) and Variable Step-size LMS (VSLMS), by using the switched beam technique. In the proposed algorithm, first, DOA of the signal source is estimated by MUltiple SIgnal Classification (MUSIC) algorithm. In the second step, depending on the desired user's location, the closest beam of the switched beam system is selected and its predetermined weights are chosen as the initial values for the weight vector. Finally, LMS/NLMS/VSLMS algorithm is applied to initial weights and final weights are calculated. Simulation results show improved convergence and tracking speed and also a higher efficiency in data transmission through increasing the Signal to Interference plus Noise Ratio (SINR) as well as decreasing the Bit Error Rate (BER) and Mean Square Error (MSE), in a joint state. Moreover, Error Vector Magnitude (EVM) as a measure for distortion introduced by the proposed adaptive scheme on the received signal is evaluated for all LMS-based proposed algorithms which are approximately the same as that for conventional ones. In order to investigate the tracking capability of the proposed method, the system is assumed to be time varying and the desired signal location is considered once in the centre of the initial beam and once in the edge of the fixed beam. As depicted in simulation results, the proposed DOA-based methods offer beamforming with higher performance in both cases of the initial beam, centre as the best case and edge as the worst case, with respect to conventional ones. The MSE diagrams for this time varying system show an ideal response for DOA-based methods in the best case. Also, in the worst case, initial height of MSE is reduced and consequently the required iteration to converge is less than the conventional LMS/NLMS/VSLMS Manuscript profile
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  2 - Achieving Better Performance of S-MMA Algorithm in the OFDM Modulation
  Saeed Ghazi-Maghrebi Babak Haji Bagher Naeeni Mojtaba Lotfizad
  10.7508/jist.2013.01.002
  20.1001.1.23221437.2013.1.1.2.7
  Effective algorithms in modern digital communication systems provide a fundamental basis for increasing the efficiency of the application networks which are in many cases neither optimized nor very close to their practical limits. Equalizations are one of the preferred More

  Effective algorithms in modern digital communication systems provide a fundamental basis for increasing the efficiency of the application networks which are in many cases neither optimized nor very close to their practical limits. Equalizations are one of the preferred methods for increasing the efficiency of application systems such as orthogonal frequency division multiplexing (OFDM). In this paper, we study the possibility of improving the OFDM modulation employing sliced multi-modulus algorithm (S-MMA) equalization. We compare applying the least mean square (LMS), multi modulus algorithm (MMA) and S-MMA equalizations to the per tone equalization in the OFDM modulation. The paper contribution lies in using the S-MMA technique, for weight adaptation, to decreasing the BER in the OFDM multicarrier modulation. For more efficiency, it is assumed that the channel impulse response is longer than the cyclic prefix (CP) length and as a result, the system will be more efficient but at the expense of the high intersymbol interference (ISI) impairment existing. Both analysis and simulations demonstrate better performance of the S-MMA compared to LMS and MMA algorithms, in standard channels with additive white Gaussian noise (AWGN) and ISI impairment simultanously. Therefore, the S-MMA equalization is a good choice for high speed and real-time applications such as OFDM based systems. Manuscript profile
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  3 - Acoustic Noise Cancellation Using an Adaptive Algorithm Based on Correntropy Criterion and Zero Norm Regularization
  Mojtaba Hajiabadi
  10.7508/jist.2015.03.003
  The least mean square (LMS) adaptive algorithm is widely used in acoustic noise cancellation (ANC) scenario. In a noise cancellation scenario, speech signals usually have high amplitude and sudden variations that are modeled by impulsive noises. When the additive noise More

  The least mean square (LMS) adaptive algorithm is widely used in acoustic noise cancellation (ANC) scenario. In a noise cancellation scenario, speech signals usually have high amplitude and sudden variations that are modeled by impulsive noises. When the additive noise process is nonGaussian or impulsive, LMS algorithm has a very poor performance. On the other hand, it is well-known that the acoustic channels usually have sparse impulse responses. When the impulse response of system changes from a non-sparse to a highly sparse one, conventional algorithms like the LMS based adaptive filters can not make use of the priori knowledge of system sparsity and thus, fail to improve their performance both in terms of transient and steady state. Impulsive noise and sparsity are two important features in the ANC scenario that have paid special attention, recently. Due to the poor performance of the LMS algorithm in the presence of impulsive noise and sparse systems, this paper presents a novel adaptive algorithm that can overcomes these two features. In order to eliminate impulsive disturbances from speech signal, the information theoretic criterion, that is named correntropy, is used in the proposed cost function and the zero norm is also employed to deal with the sparsity feature of the acoustic channel impulse response. Simulation results indicate the superiority of the proposed algorithm in presence of impulsive noise along with sparse acoustic channel. Manuscript profile
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  4 - An Acoustic Echo Canceller using Moving Window to Track Energy Variations of Double-Talk-Detector
  Mouldi  MAKDIR Mourad BENZIANE Mohamed  BOUAMAR
  10.61186/jist.40125.12.46.105
  As a fundamental device in acoustic echo cancellation (AEC) systems, the echo canceller based on adaptive filters relies on the adaptive approximation of the echo-path. However, the adaptive filter must face the risk of divergence during the double-talk periods when th More

  As a fundamental device in acoustic echo cancellation (AEC) systems, the echo canceller based on adaptive filters relies on the adaptive approximation of the echo-path. However, the adaptive filter must face the risk of divergence during the double-talk periods when the near-end is present. To solve this problem, the double-talk-detector (DTD) is often used to detect the double-talk periods and prevent the echo canceller from being disturbed by the other end of the speaker’s signal. In this paper, we propose a DTD based on a new method that can detect quickly and track accurately double-talk periods. It is based on the sum of energies of the estimated echo and the microphone signals which is continuously compared to the error energy. A window that moves with time and tracks energy variations of the different input signals of the DTD represents a fundamental feature of the proposed method compared to several other methods based on correlation. The goal is to outperform conventional normalized cross-correlation (NCC) methods which are well-known in terms of small steady-state misalignment and stability of decision variable. In this work, the normalized least mean squares (NLMS) algorithm is used to update the filter coefficients along speech signals which are taken from the NOIZEUS database. Efficiency of the proposed method is particularly compared to the conventional Geigel algorithm and normalized cross-correlation method (NCC) that depends on the cross-correlation between the microphone signal and the error signal of AEC. Performance evaluation is confirmed by computer simulation. Manuscript profile