• The Journal of the Korea institute of electronic communication sciences
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• v.14 no.6
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• pp.1249-1256
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• 2019

In this paper, we propose a new algorithm to remove acoustic feedback and noise in hearing aids. Instead of using the conventional FIR structure, this algorithm is a deep learning algorithm using neural network adaptive prediction filter to improve the feedback and noise reduction performance. The feedback canceller first removes the feedback signal from the microphone signal and then removes the noise using the Wiener filter technique. Noise elimination is to estimate the speech from the speech signal containing noise using the linear prediction model according to the periodicity of the speech signal. In order to ensure stable convergence of two adaptive systems in a loop, coefficient updates of the feedback canceller and noise canceller are separated and converged using the residual error signal generated after the cancellation. In order to verify the performance of the feedback and noise canceller proposed in this study, a simulation program was written and simulated. Experimental results show that the proposed deep learning algorithm improves the signal to feedback ratio(: SFR) of about 10 dB in the feedback canceller and the signal to noise ratio enhancement(: SNRE) of about 3 dB in the noise canceller than the conventional FIR structure.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• The Journal of the Acoustical Society of Korea
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• v.12 no.5
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• pp.54-63
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• 1993

적응 능동 잡음제거를 위한 새로운 방식을 기술했으며, 잡음원이 존재하는 환경에서 물리적인 잡음 제거를 하는데 목표를 두었다. 기존의 시스템 식별 방식과 달리 적응 예측기를 사용하였으며 신호를 검출하는데 쓰이는 마이크를 하나로 구성하였다. 잡음 신호 자체를 예측함으로써 적극적인 대처를 할 수 있으며 적응 알고리즘은 한번만 사용되고 적응 제어형 시스템 식별 방식에서 필요로 하는 피드백 항을 제거시킴으로써 시스템을 간략화시켰다. 컴퓨터 모의 실험 결과를 통하여 제안한 방식이 주기성을 갖거나 혹은 대역 제한된 잡음에 대해서 매우 유용함을 보였다.

• Journal of Korea Multimedia Society
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• v.13 no.7
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• pp.973-982
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• 2010

In this paper, we proposed the single-channel adaptive noise canceller for the enhancement of heart sound (HS) in the auscultation signal. In case of either normal or emergency, a HS diagnosis is difficult due to the various signal source in the chest. Therefore, the HS enhancement is necessary. The conventional active noise canceller(ANC) has two channel, main signal and reference signal. For signal channel, the reference signal in ANC was generated by the proposed HS analyser and BS-Gate based on the characteristic of HS. This reference signal is suitable to the ANC condition. Experimental data were acquisited from MP36, SS30L in BIOPAC Inc., By the experiment, we confirmed that the proposed single-channel ANC was efficient for HS enhancement. And by the comparison with active linear enhancement, it was validate that the proposed ANC is not affected by the variation of a heartbeat.

• Journal of Korea Multimedia Society
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• v.2 no.1
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• pp.99-108
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• 1999

In this paper, a new robust adaptive algorithm is proposed to improve the performance of AEC without computational burden. The proposed adaptive algorithm is based on NLMS algorithm, and its step-size is varied with the reference input signal power and the desired signal power. Its step-size is normalized by the sum of the powers of the reference input signal and the desired signal. When the near-end speaker's speech and noise are applied into the microphone, the step-size becomes small and the misalignment of coefficients are reduced. The convergence speed is comparable to NLMS algorithm at AEC application because the echo signals are attenuated about 10∼20 dBSPL. The characteristics of this algorithm is also analyzed and compared with conventional ones in this paper.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10A
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• pp.1006-1012
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• 2008

This paper presents a Duvall-structure-based adaptive beamforming method which efficiently cancels coherent and incoherent interferences. The proposed method exploits several correlation vectors to increase the dimension of the weight vector, compared to the existing method which uses a single correlation vector only. The increased dimension of the weight vector leads to an improvement in the signal-to-interference plus noise ratio (SINR) performance. Moreover, the proposed method can suppress more interferences than the existing one. Simulation shows that the former is superior to the latter in terms of the steady-state and transient responses.

• Journal of KSNVE
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• v.7 no.3
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• pp.427-437
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• 1997

Mean active intensity based active control for the cancellation of radiated noise out of the duct exit is studied. The active intensity control strategy is drerived based on the relation of the exterior sound field out of the duct termination and interior sound field of the duct. One of the characteristics of this control strategy is that the control performance can be maintained regardless of the sensor loction, compared with the conventional local pressure control methods at either interior downstream or exterior field positions. It is also suggested that the digital filtering for the active intensity control can be achieved by time-domain filtered-x LMP (Lest-Mean-Product) adaptive algorithm. Experiments for an open-ended duct are performed to compare the active intensity control performance with conventional pressure control one. Active control experiment of local sound pressure is conducted by widely used filtered-x LMS adaptive Algorithm and active intensity control implementaion uses the derived filter d-x LMP algorithm. It is shown that the exterior sound fileds was much better observable by sensing of the active intensity than by just sound pressure. It is also demonstrated that the global control performance of external field by acoustic intensity is superior to the conventional sound pressure control performance.

• 한국정보통신설비학회:학술대회논문집
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• 2005.08a
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• pp.397-405
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• 2005

This paper presents a headphone system which has adopted two algorithm to increase sound clearness and to separate signal from noisy environment. In the field of adaptive signal processing, LMS algorithm which is a kind of steepest decent method, can be implemented with more simple calculation, so that we use it to eliminate unwanted noise elements for the proposed system. Futhermore we generate early echo using some delays, then mix it in signal. This process can increase the clearness of signal. In this paper, we prove that the proposed system can be implemented in real time. The proposed system is satisfied to subject assessment test base on MOS(Mean Opinion Score) of ITU-T.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.7B
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• pp.505-516
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• 2012

According to distribute of resource of macro cell and reduce distance between transmitter and receiver, Femto cell system is promising to provide costeffective strategy for high data traffic and high spectral efficient services in future wireless cellular system environment. However, the co-channel operation with existing Macro networks occurs some severe interference between Macro and Femto cells. Hence, the interference cancellation or management schemes are imperative between Macro and Femto cells in order to avoid the decrease of total cell throughput. First, we briefly investigate the conventional resource allocation and interference cancellation scheme between Macro and Femto cells. So we found that cell throughput and frequency reuse ware decreased Then, we propose an adaptive resource allocation scheme based on the distribution of Femtocell traffic in order to increase the cell throughput and also maximize the spectral efficiency over the FFR (Fractional Frequency Reuse) based conventional resource allocation schemes. Simulation Results show that the proposed scheme attains a bit similar SINR (Signal to Interference Noise Ratio) distribution but achieves much higher total cell throughput performance distribution over the conventional resource allocation schemes for FFR and future IEEE 802.16m based Femtocell network environment.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.3
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• pp.13-18
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• 2011

Sphere Decoding (SD) is a decoding technique able to achieve the Maximum Likelihood (ML) performance in fading environments; nevertheless, the main disadvantage of this technique is its high complexity, especially in poor channel conditions. In this paper, we present an adaptive hybrid algorithm which reduces the conventional Sphere Decoder's complexity and keeps the ML performance. The system called Adaptive OSIC-SD modifies its operation based on Signal to Noise Ratio (SNR) information and achieves an optimal performance in terms of Bit Error Rate (BER) and complexity. Through simulations, we probe that the proposed system maintains almost the same bit error rate performance of the conventional SD, and exhibits a lower, quasi-constant complexity.