• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.12
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• pp.1274-1285
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• 1991

This paper presents a statistical convergence analysis of the normalized least mean square(NLMS)algorithm that employs a single-pole lowpass filter, In this algorithm the lowpass filter is used to adjust its output towards the estimated value of the input signal power recursively. The estimated input signal power so obtained at each time is then used to normalize the convergence parameter. Under the assumption that the primary and reference inputs to the adaptive filter are zero mean wide sense stationary, and Gaussian random processes, and further making use of the independence assumption. we derive expressions that characterize the mean and maen squared behavior of the filter coefficients as well as the mean squared estimation error. Conditions for the mean and mean squared convergence are explored. Comparisons are also made between the performance of the NLMS algorithm and that of the popular least mean square(LMS) algorithm Finally, experimental results that show very good agreement between the analytical and emprincal results are presented.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.7
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• pp.1650-1658
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• 1998

This paper describes how to remove echoes effectively using speech parameter information provided form speech coder. More specially, the proposed adaptive echo canceller utilizes the excitation signal or linearly predicted error signal instead of output speech signal of vocoder as the input signal for adaptation algorithm. The normalized least mean ssquare(NLMS) algorithm is used for the adaptive echo canceller. The proposed algorithm showed a fast convergece charactersitcis in the sinulatio compared to the conventional method. Specially, the proposed echo canceller utilizing the excitation signal of speech coder showed about four times fast convergence speed over the echo canceller utilizing the output speech signal of the speech coder for the adaptation input.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.4
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• pp.18-26
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• 1996

In this paper we propose a new dual structured tap selection algorithm for voice echo canceller in digital cellular communication system, investigating adaptive filtering algorithms for echo cancellation in long distance telephony or mobile communication system. The proposed algorithm has a two-stage processing structure that after a dispersive region in an impulse response of an echo path is found out, the tap coefficients of a short length filter are adjusted adaptively for the region, because the impuse response has a very little portion of the dispersion. Simulation results show that the proposed algorithm with 256 taps gives a performance of convergence speed superior to both full-tap normalized least mean with 256 taps gives a performance of convergence speed superior to both full-tap normalized least mean square (NLMS) and a scrub taps waiting in a queue (STWQ) algorithms by about eighty per cent, also to a tap selection algorithm by about twenty per cent. And the resutls diplay that if the more tap coefficients are used due to a long delayed dispersive zone, the proposed algorithm produces the better performance.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.5 no.1
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• pp.25-36
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• 2006

This paper discusses the implementation and performance evaluation of a DSP-based digital beamformer using the Texas Instrument TMSC6711 DSP processor for smart antenna applications. Two adaptive beamforming algorithms which served as the brain for the beamformer, the Normalized Least-Mean-Square (NLMS) and the Constant Modulus Algorithms (CMA) were embedded into the processor and evaluated. Result shows that the NLMS-based digital beamformer outperforms the CMA-based digital beamformer: 1)For NLMS algorithm, the antenna steers to the direction of the desired user even at low iteration value and the suppression level towards the interferer increases as the number of iteration increase. For CMA algorithm, the beam radiation pattern slowly steers to the desired user as the number of iteration increased, but at arate slower than NLMS algorithm and the sidelobe level is shown to increases as the number of iteration increase. 2) The NLMS algorithm has faster convergence than CMA algorithm and the error convergence for CMA algorithm sometimes is subject to misadjustment.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.413-416
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• 2002

In this paper, a speech enhancement system using microphone array with adaptive Post-Processing is implemented in real-lime with TMS320C6201 DSP. It consists of delay-and-sum beamformer and adaptive post-processing filters with NLMS (Normalized Least Mean Square) algorithm. THS1206 ADC is used for collection of 4-channel microphone signals. Sizes of program memory, data ROM and data RAM of the implemented system are 15,744, 748 and 47,540 bytes, respectively. Finally 21.839${\times}$106 clocks per second is required for real-time operation.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.10-17
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• 2010

In a passive sonar, the improvement of detection performance by using noise cancellation is usually a important problem. In this paper, we have analyzed the own-ship noise cancellation in the two operation modes which are used in the HMS system. In the operator mode, an adaptive line enhancer(ALE) is applied to improve the tonal detection by using broadband noise cancellation and the normalized least mean square(NLMS) algorithm is applied to the design of an adaptive filter. The reference input that is correlated with a primary input can be used to remove the noise incident on the observation directionin the automatic mode. Computer simulations with real sea that data show that the proposed adaptive noise canceller has good performance in passive detection under HMS operation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.2A
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• pp.91-99
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• 2011

Several repeater systems are used to enhance the cell coverage to location such as shadow and rural areas in mobile systems. But the general RF repeater solutions are not suitable for high power outdoor environment because it has the weakness such as self oscillation problem With adoption of a adaptive digital filter technology, feedback interference cancellation repeater prevents oscillation by detecting and canceling the unwanted feedback signal between transmission and receiver antenna. In this paper, dual NLMS based interference cancellation method is proposed and the step size adaptation can be implemented by the estimation of the feedback channel Doppler frequency characteristics. The performance of the proposed algorithm is quantified via analysis and simulation for the static and multipath fading feedback channels.

• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.559-567
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• 2020

This paper proposes a noise cancelling algorithm to remove own-ship noise for a towed array sonar. Extra beamforming is performed using partial channels of the acoustic array to get a reference beam signal robust to the noise bearing. Frequency domain Adaptive Noise Cancelling (ANC) is applied based on Normalized Least Mean Square (NLMS) algorithm using the reference beam. The bearing of own-ship noise is estimated from the coherence between the reference beam and input beam signals. Own-ship noise level is calculated using a beampattern of the noise with estimated steering angle, which prevents loss of a target signal by determining whether to update a filter so that removed signal level does not exceed the estimated noise level. Simulation results show the proposed algorithm maintains its performance when the own-ship gets out off its bearing 40 % more than the conventional algorithm's limit and detects the target even when the frequency of the target signal is same with the frequency of the own-ship signal.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.22 no.4
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• pp.430-439
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• 2011

In this paper, we design a digital predistortion(DPD) for linearity enhancement of RF power amplifier operating in 900 MHz band. We verify improvement of linearity by comparing the proposed DPD using tanh as a nonlinear indexing function and the DPD using linear indexing function based on signal amplitude. The digital predistortion is realized by look-up table(LUT) method, and the Saleh model is employed for power amplifier modeling, then a commercial power amplifier module is used for measurement. The LUT has 256 tables, and the NLMS(Normalized Least Mean Square) algorithm was utilized for an adaptive algorithm for estimation. As a result, we improve the ACLR(Adjacent Channel Leakage Ratio) by around 15 dB.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.23 no.4
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• pp.484-490
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• 2012

In this paper, we design a digitally pre-distorted RF power amplifier operating in 900 MHz band. The linearity of RF power amplifier is improved by employing the digital pre-distortion(DPD) based on affine projection(AP) algorithm, where the look-up table(LUT) method is used with non-linear indexing. The proposed DPD with AP algorithm is compared with that with normalized least mean square(NLMS) algorithm, applied to the RF power amplifier. A commercial power amplifier module is used for verification of the proposed algorithm which shows improvement of adjacent channel leakage ratio(ACLR) by about 21 dB.