• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.4 no.2
• /
• pp.407-416
• /
• 2000

Adaptive digital filtering and noise cancelling technique using a tree structured filter bank are presented to reduce a undesirable aliasing due to the decimation of filtered output and improve the performance in terms of mean-square error and the convergence speed using a aliasing canceller. A signal is split into two subband by analysis filter bank and decimated by decimator and reconstructed by interpolation technique and synthesis filter bank. A variable step-size LMS algorithm is used to improve the convergence speed in case of existing the measurement noise in desired input of filter. It is shown by computer simulation that the proposed subband structure in this paper is superior to conventional subband filter structure in terms of mean-square error and convergence speed.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2014.10a
• /
• pp.150-155
• /
• 2014

본 논문에서는, 원거리 음성인식을 위한 서브밴드 필터링 기반의 빠르고 효율적인 에코제거 시스템을 제안한다. 제안하는 에코제거 시스템은 우선 채널간 유사도 (correlation) 가 높을 경우 적응필터가 오작동하는 것을 방지하기 위해 spatial decorrelation 을 적용하게 된다. 그리고 tree 형태를 가지는 IIR filterbank 기반의 subband 구조를 채택함으로써, 적은 차수로도 효과적인 analysis, synthesis 필터링을 수행할 수 있도록 한다. 이 과정에서 불가피하게 발생하는 서브 밴드간 spectral aliasing은 notch filter를 적용해 해결할 수 있다. 또한 적응 필터로는 improved proportionate normalized least-mean-square (IP-NLMS) 알고리즘을 사용해 수렴속도 및 에코제거 성능에서 우수함을 확인하였다. 마지막으로 decision-directed estimation 기반의 residual echo suppressor를 적용해 잔여 에코를 제거하게 된다. 본 논문에서는 각 단계를 구성하게 된 이론적인 배경을 소개하고, 실제 에코가 존재하는 환경에서 ERLE, 원거리 음성 인식률, computational complexity를 통해 제안하는 에코제거 시스템의 효과를 입증하도록 한다.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.11 no.1
• /
• pp.18-25
• /
• 2018

In this paper, we present a low-power implementation of the multi-channel hearing aid system using a general-purpose DSP chip. The system includes an acoustic amplification algorithm based on Wide Dynamic Range Compression (WDRC), an adaptive howling canceller, and a single-channel noise reduction algorithm. To achieve a low-power implementation, each algorithm is re-constructed in forms of integer program, and the integer program is converted to the assembly program using BelaSigna(R) 250 instructions. Through experiments using the implementation system, the performance of each processing algorithm was confirmed in real-time. Also, the clock of the implementation system was measured, and it was confirmed that the entire signal processing blocks can be performed in real time at about 7.02MHz system clock.

• The Journal of the Acoustical Society of Korea
• /
• v.18 no.5
• /
• pp.35-43
• /
• 1999

In this paper, a new robust adaptive algorithm and a post-processing method are proposed to improve the performance of AEC without computational burden. 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. To reduce the residual echoes, a new post-processing method, which is co-operated with the proposed noise-robust adaptive algorithm, is proposed in this paper. The method is based on the correlation of the desired signal and the estimation error signal. The residual echoes are attenuated as proportional to the correlation normalized with the power of desired signals. The normalized correlation plays a role as Wiener filter for residual echoes. In the double-talk situation, the estimation error signals, that are residual echoes, dominantly include the near-end speaker's speech and the normalized correlation closes to 1. Therefore, the near-end speaker's speech can be transmitted without being attenuated. When the desired signals consists of only the acoustic echoes, the residual echoes are mostly attenuated and canceled by the proposed post-processor. The computation of AEC using the proposed post-processor is comparable to NLMS algorithm.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.11 s.353
• /
• pp.141-148
• /
• 2006

This paper deals with the performance improvement by eliminating loop interference signal and inserting adaptive equalizer for phase compensation in OFDMA based Wibro relay station. The Wibro relay station is used for the extension of communication service area and for throughput improvement of base station. The loop interference is important factor of performance determination of relay station when transmitter and receiver is very closely located. In order to design interference canceller, we generated base-band OFDMA signal and then transmitted the signal along with pilot tones alined with two different combinations for training mode. And then, we generated received fading signal due to the loop interference added noise to the received signal. In the receiver, the transmitted signal is recovered by elimination of the interference signal with channel estimate and compensating phase by adaptive equalizer. The performance improvement was verified by computer simulation which show channel estimation, constellation of signal and BER characteristics according to the variation of SNR ratio.

• Journal of Korea Multimedia Society
• /
• v.3 no.4
• /
• pp.347-355
• /
• 2000

This paper presents an adaptive AEC(acoustic echo canceller) based on the wavelet transform using M-channel subband QMF filter banks. The proposed algorithm improves the performance of AEC with a realtime process by a low complexity of wavelet transform filter banks, a subband processing and a orthogonality of wavelet subband filter. Adaptive filter coefficients of each subband are updated using LMS algorithm with a low complexity and a easy realization for a realtime processing and a reduction of hardware cost. For a input signal, a white Gaussian noise and a real speech signal with a environment noises are used for a performance estimation of the proposed algorithm. As a result of computer simulation, the proposed AEC has a low asymptotic error, a low computation complexity and a robust performance.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
• /
• v.3 no.3
• /
• pp.14-21
• /
• 2010

For the ubiquitous healthcare environment, real-time measurement of biomedical signals and accuracy of the measured biomedical information are very important. In addition, it is important to develop a healthcare device with low power In this paper, the synchronized pulse in a heartbeat was detected from the radial artery using the piezo film sensor, in order to eliminate inconvenience to wear a pulse detection finger probe. We can get a best output after applying the adaptive noise canceller using two piezo film sensor signals, pulse signal having motion artifacts and motion artifacts reference signal. To detect heartbeat, we use maximum point detection method from pulse removed motion artifacts.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.4A
• /
• pp.330-335
• /
• 2002

In the direct sequence code division multiple access system, the problem of multiple access interference due to multiple access is generated. A interference cancelling receiver is used to solve this problem. The conventional interference cancelling receiver is structure of successive interference canceller using antenna array. In this structure, the difference of between method I and method II depends on updating weight vector. In this paper, the adaptive blind CMA array interference cancelling receiver using cost function of constant modulus algorithms is proposed to update weight vector at conventional structure. The simulation compared the proposed interference cancelling receiver with two conventional interference cancelling receivers by signal to interference ratio and bit error rate curve under additive white Gaussian noise environment. The simulation results show that the proposed receiver has about the gain of SIR of 1.5[dB] more than method I which is conventional receiver at SIR curve, and about the gain of SIR of 0.5(dB) more than method II. In BER curve, the proposed IC receiver about the gain of SNR of 2[dB] more than method I and about the gain of SNR of 0.5[dB] more than method If, Thus, the proposed interference cancelling receiver has the higher performance than conventional interference cancelling receivers.

• Journal of Korea Society of Industrial Information Systems
• /
• v.5 no.2
• /
• pp.32-38
• /
• 2000

One of the most popular algorithm in adaptive signal processing is the least mean square(LMS) algorithm. The majority of these papers examine the LMS algorithm with a constant step size. The choice of the step size reflects a tradeoff between misadjustment and the speed of adaptation. Subsequent works have discussed the issue of optimization of the step size or methods of varying the step size to improve performance. However there is as yet no detailed analysis of a variable step size algorithm that is capable of giving both the adaptation speed and the convergence. In this paper we propose a new variable step size algorithm where the step size adjustment is controlled by the gradient of error square. The proposed algorithm is performed in the Walsh-Hadamard domain in real-valued orthogonal transform because of fast convergence. The simulation results using the new algorithm for noise canceller system is described. They are compared to the results obtained by other algorithms. It is shown that the proposed algorithm produces good results compared with conventional algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.28 no.10C
• /
• pp.967-983
• /
• 2003

In wireless spread-spectrum communication systems utilizing PN (pseudo noise) sequences, a variety of noise sources from the channel affect the data reception performance. Among them, in this paper we are concerned with the narrowband interference that may arise from the use of the spectral bands overlapped by the existing narrowband users or the intentional jammers as in military communication. The effect of this interference can be reduced to some extent at the receiver with the PN demodulation by processing gain. It is known, however, that when the interferers are strong, the reduction cannot be sufficient and thereby requiring the extra use of narrowband interference cancellers (NIC's) at the receivers. A class of adaptive NIC's are studied here based on different two cost functions. One is the chip mean-squared error (MSE) computed prior to the PN demodulation and used in the conventional cancellers. Since thses conventional cancellers should be operated at the chip rate, the computational requirements are enormous. The other is the symbol MSE computed after the PN demodulation in which case the weights of the NIC's can be updated at a lot lower symbol rate. To compare the performance of these NIC's, we derive a common measure of performance, i.e., the symbol MSE after the PN demodulation. The analytical results are verified by computer simulation. As a result, it is shown that the cancellation capability of the symbol-rate NIC's are similar or better than the conventional one while the computational complexity can be reduced a lot.