• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.2C
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• pp.184-190
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• 2006

In this paper, we propose the blind multiuser detector which is robust against the effects of near-far and multiuser interference. The proposed detector is composed of the partial IC(interference canceller) and the blind MOE(minimum output energy) multiuser detector. The partial IC partially eliminates interference components from the received signal then the output of partial IC is fed into the input of multiuser detector. Simulation results show that the proposed detector has the robust property but the performance of conventional MOE multiuser detector is rapidly degraded in case of existing both near far and multiuser. Thus, the proposed partial IC BMUD(blind multiuser detection) technique has better performance than the conventional MOE.

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.4
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• pp.295-303
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• 2008

In general, transform domain adaptive filters show faster convergence speed than the time domain adaptive filters, but the amount of calculation increases dramatically as the filter order increases. This problem can be solved by making use of the subband structure in transform domain adaptive filters. In this paper, to increase the convergence speed on the generalized subband decomposition FIR adaptive filters, a structure of the adaptive filter with subfilter of dyadic sparsity factor in wavelet transform domain is designed. And, in this adaptive filter, the equivalent input in transform domain is derived and, by using the input, the convergence properties for the LMS algorithm is analyzed and evaluated. By using this sub band adaptive filter, the inverse system modeling and the periodic noise canceller were designed, and, by computer simulation, the convergence speeds of the systems on LMS algorithm were compared with that of the subband adaptive filter using DFT(discrete Fourier transform).

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.43-49
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• 1997

Generally, echo cancelers of a TV conference system or a audio conference system are to carry out a real time processing in the case of the closed room having long reverberation time because the system requires much time to modify filter coefficients to environmental changes. Therefore this paper proposes a new subband adaptive filtering method using polyphase filter banks of MPEG(Moving Picture Experts Group) audio system to solve the problems. This method divides signal spectra of input and output into several frequency bands, and each band is adaptively filtered by using ES-NLMS (Exponential Step-Normalized Least Mean Square) algorithm. The optimal number of subband is determined by computational simulations. According to the results of simulation, ERLE of the subband model is 2dB smaller than general full band, calculation rate's of the subband model is decreased about 88%.

• Journal of Korea Multimedia Society
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• v.3 no.4
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• pp.347-355
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• 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 Korean Institute of Communications and Information Sciences
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• v.25 no.1A
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• pp.156-166
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• 2000

In discrete multi-tone (DMT)-based very high bit-rate digital subscriber line (VDSL) systems, the ingressed RFI (Radio Frequency Interference) accompanied by transmitted signal at the receiver is known to cause the spectralleakage by the finite-point FFT, resulting in significant performance degradation.0 this paper, we propose a RFIcancellation technique which can compensate the ingressed RFI efficiently, especially for a high data-rate VDSLsystem. The proposed technique compensates the performance degradation of e VDSL system due to RFI byusing a time-domain RFI canceller whose coefficients are obtained from the estimated center frequency of RFI inthe frequency domain under the assumption that the ingressed RFI is a narrow-band signal compared to VDSLsampling frequency. The proposed technique requires no training symbol and convergence period, and worksproperly even when spectral shape of the ingressed RFI is unknown or arbitrary. Feasibility of the proposedtechnique is demonstrated via a computer simulation by comparing its performance with the performance of theprevious RFI cancellation technique.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.6
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• pp.95-101
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• 2009

Among the adaptive noise cancellers (ANC), the least mean square (LMS) algorithm has probably become the most popular algorithm because of its robustness, good tracking properties, and simplicity of implementation. However, it has non-uniform convergence and a trade-off between the rate of convergence and excess mean square error (EMSE). To overcome these shortcomings, a number of variable step size least mean square (VSSLMS) algorithms have been researched for years. These LMS algorithms use a complex variable step method approach for rapid convergence but need high computational complexity. A variable step approach can impair the simplicity and robustness of the LMS algorithm. The proposed instantaneous gain control (IGC) algorithm uses the instantaneous gain value of the original signal and the noise signal. As a result, the IGC algorithm can reduce computational complexity and maintain better performance.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.9 no.2
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• pp.205-212
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• 2016

In this paper, we implement the development platform of multichannel digital hearing aid algorithm using Simulink provided by Matlab. The digital hearing aids are considered medical devices designed to compensate for hearing loss, they need to be correctly selected, to help a person who has difficulty in hearing. The development platform that implemented in this paper, includes WOLA filterbank for analysis/synthesis of input signal, Wide dynamic range compression for hearing loss compensation and adaptive filter for feedback cancellation. Using the development platform, algorithm parameters for each block can be set depending on the hearing aid user. Thus it is possible to test the algorithm before the machine language. As a result, the time for algorithm development can be saved and performance and computational complexity can be optimized.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.1
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• pp.18-25
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• 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
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• v.20 no.7
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• pp.37-46
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• 2001

This paper proposes a new noise suppression method using the Wavelet transform analysis. The noise suppressor using the Wavelet transform shows the more effective advantages in a babble noise than one using the short-time Fourier transform. We designed a new channel structure based on spectral subtraction of Wavelet transform coefficients and used the Wavelet mask pattern with more higher time resolution in high frequency. It showed a good adaptation capability for babble noise with a non-stationary property. To evaluate the performance of proposed noise canceller, the informal subjective listening tests (Mos tests) were performed in background noise environments (car noise, street noise, babble noise) of mobile communication. The proposed noise suppression algorithm showed about MOS 0.2 performance improvements than the suppression algorithm of EVRC in informal listening tests. The noise reduction by the proposed method was shown in spectrogram of speech signal.

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