• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.28-36
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• 1995

In this paper, a double loop control model using leaky delay LMS algorithm are proposed for active noise control. The proposed double loop control model estimates the loudspeaker characteristic and the error path transfer function with on-line using only gain and acoustic time delay to reduce computation burden. The control of error signal through double loop control scheme makes the more robust cntrol system. The input signal of filter to estimate acoustic time delay is used difference between input signal of input microphone and adaptive filter output. And also, in nonstationary environments, the leaky delay LMS algorithm is employed to counteract parameter drift of delay LMS algorithm. For practical noise signal, the proposed double loop control model reduces noise level about 12.9 dB.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.425-433
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• 2003

Acoustic Echo Canceller (AEC) is a signal processing system for removing unwanted echo signals in teleconference and hands-free communication. Least mean square (LMS) algorithm is one of the adaptive echo cancellation algorithms and it has been most attractive because of its simplicity and robustness. However, the convergence properties of the LMS algorithm degrade with highly correlated input signals such as speech. For this reason, transform-domain adaptive filtering algorithm was introduced to decorrelate the colored input samples by using the orthogonal transform matrix such as DCT, DFT and then LMS adaptive filtering process is applied. In this paper, we propose a MLT domain adaptive echo canceller base on the MLT (Modulated lapped Transform) orthogonal transform matrix. The proposed algorithm achieves high decorrelation efficiency and fast convergence speed via modulated lapped transform of size 2NXN instead of NXN unitary transform such as DCT, DFT, Hadamad and it is applied to the acoustical echo cancellation system. Form the computer simulation with both synthesis and real speech, the proposed MLT domain adaptive echo canceller shows approximately twice faster convergence speed and 20∼30 ㏈ ERLE improvements over the DCT frequency domain acoustic echo cancellation system.

• Journal of IKEEE
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• v.14 no.1
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• pp.33-39
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• 2010

In this paper, a 5Gb/s receiver with an adaptive equalizer for serial link interfaces is proposed. For effective gain control, a least-mean-square (LMS) algorithm was implemented with two internal signals of slicers instead of output node of an equalizing filter. The scheme does not affect on a bandwidth of the equalizing filter. It also can be implemented without passive filter and it saves chip area and power consumption since two internal signals of slicers have a similar DC magnitude. The proposed adaptive equalizer can compensate up to 25dB and operate in various environments, which are 15m shield-twisted pair (STP) cable for DisplayPort and FR-4 traces for backplane. This work is implemented in $0.18-{\mu}m$ 1-poly 4-metal CMOS technology and occupies $200{\times}300{\mu}m^2$. Measurement results show only 6mW small power consumption and 2Gbps operating range with fabricated chip. The equalizer is expected to satisfy up to 5Gbps operating range if stable varactor(RF) is supported by foundry process.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.22 no.4
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• pp.453-461
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• 2011

In order to use more efficiently limited frequency resources at the broadcasting band and to eliminate blanket area of the terrestrial broadcasting and to improve broadcasting quality. The importance of repeaters has increasing continuously. However, in case of T-DMB digital on channel repeater in OFDM systems, some of the signal radiated feedback again at the receiver antenna. So it generates feedback signal interference in repeater system. Also phase noise increases ICI(Inter Carrier Interference). It affects seriously the frequency domain equalizer. In this paper, we remove the feedback signal interference by LMS with correlation. Also we propose an effective equalizer algorithm that can remove ICI caused by phase noise and the power amplifier's back-off. In this simulation results, this system is satisfied the performance of BER=$10^{-4}$ at less than SNR=14 dB after compensation of phase noise.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.14 no.4
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• pp.247-256
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• 2021

In this paper, we propose a packet loss concealment (PLC) algorithm using pitch harmonic motion prediction and adaptive signal amplitude prediction and. The spectral motion prediction method divides the spectral motion of the previous usable frame into predetermined sub-bands to predict and restore the motion of the lost signal. In the proposed algorithm, the speech signal is classified into voiced and unvoiced sounds. In the case of voiced sounds, it is further divided into pitch harmonics using the pitch frequency to predict and restore the pitch harmonic motion of the lost frame, and for the unvoiced sound, the lost frame is restored using the spectral motion prediction method. When the continuous loss of speech frames occurs, a method of adjusting the gain using the least mean square (LMS) predictor is proposed. The performance of the proposed algorithm was evaluated through the objective evaluation method, PESQ (Perceptual Evaluation of Speech Quality) and was showed MOS 0.1 improvement over the conventional method.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2009.10a
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• pp.240-241
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• 2009

The method of the reduction of the duct noise can be classified by the method of passive control and the method of active control. However, the passive control method has a demerit to reduce the effect of noise reduction at low frequency (below 500Hz) range and to be limited by a space. Whereas, the active control method can overcome the demerit of passive control method. The algorithm of active control is mostly used the Least-Mean-Square (LMS) algorithm because the LMS algorithm can easily obtain the complex transfer function in real-time. Especially, When the Filtered-X LMS (FXLMS) algorithm is applied to an ANC system.

• IEIE Transactions on Smart Processing and Computing
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• v.3 no.5
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• pp.267-270
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• 2014

This paper presents the comparative performance of an adaptive FIR filter for a Delayed LMS algorithm. The delayed error signal was used to obtain a Delayed LMS algorithm to allow efficient pipelining for achieving a small critical path and area efficient implementation. This paper presents hardware efficient results (device utilization parameters) and power consumed. The FPGA families (Artix-7, Virtex-7, and Kintex-7) for a low voltage perspective are shown. The synthesis results showed that the artix-7 CMOS family achieves the lowest power consumption of 1.118 mW with 83.18 % device utilization. Different Precision strategies, such as the speed optimization and power optimization, were imposed to achieve these results. The algorithm was implemented using MATLAB (2013b) and synthesized on the Leonardo spectrum.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.64 no.4
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• pp.267-272
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• 2015

Active noise control (ANC) is a method of cancelling a noise signal in an acoustic cavity by generating an appropriate anti-noise signal via canceling loudspeakers. The continuous progress of ANC involves the development of improved adaptive signal processing algorithms, transducers, and DSP hardware. In this paper, the convergence behavior and the stability of the FxLMS algorithm in ANC systems with real-time implementation is proposed. Specially, The advanced DSP H/W with dual core(DSP+ARM) and API(application programming interface) S/W programming was developed to improve the real-time implementation performance under the FxLMS algorithms of input noise such as road noise environment. The experimental results are found to be in good agreement with the theoretical predictions.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1998.04a
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• pp.670-677
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• 1998

Many of the adaptive noise control systems utilize a form of the least mean square (LMS) algorithm. In the active control of noise, it is common practice to locate an error microphone far from the control source to avoid the near-field effects by evanescent waves. Such a distance between the control source and the error microphone makes a certain level of time-delay inevitable and, hence, may yield undesirable effects on the convergence properties of control algorithms such as filtered-x LMS. This paper discusses the dependence of the convergence rate on the acoustic error path in these popular algorithms and introduces new algorithms which increase the convergence region regardless of the time-delay in the acoustic error path. Performances of the new LMS algorithms are presented in comparison with those by the conventional algorithms based on computer stimulations and experiments.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.62 no.3
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• pp.111-116
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• 2013

Noise problem that occurs on the road is raising a lot of problems in the economic, social and environmental aspects. The objective of this paper is to propose ANC(active noise cancellation)-based road traffic noise reduction algorithm-model which can reduce noise by generating frequency opposed to noise sources to improve and complement the problem that existing physical form of a noise barrier. In this paper, we measured the noise characteristic from collection of two difference car noise also ANC simulation has been performed by using road traffic noises input. In order to compare the control performance, we performed noise reduction simulation of ANC by filtered-X LMS algorithm and delayed control signal injection. As a result of this simulation, we confirmed that convergence performance and noise decrease effect to the filtered-X LMS algorithm by inputting the road traffic noise.