• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.238-247
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• 2005

In this Paper. we Presents subband design techniques of a compensating filter and adaptive feedback canceller for the digital hearing aid. The sensorineural hearing loss has a hearing threshold that shows a nonlinear characteristic in frequency domain. and its compensation suffers from an echo that produced by an undesired time varying feedback path. Therefore. the digital hearing aid requires the compensator that can adjust gains nonlinearly in frequency bands and eliminate the echo rapidly In the Proposed digital hearing aid. the compensating filter is designed by the adaptive system identification method in subband structure, and the adaptive feedback canceller is designed by the subband affine projection algorithm. The designed compensation filter can control the nonlinear gain in each subband respectively, therefore precise compensation is possible. And the feedback canceller using the subband adaptive filter achieves fast convergence rate. The Performances of the Proposed method are verified by computer simulations as comparing with the behaviors of the previous trials.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.56 no.12
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• pp.2251-2256
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• 2007

In this paper, the wavelet transform based lossless image compression algorithm is proposed. The proposed algorithm transforms the input image using 9/7 ICFB and S+P filter, and eliminates the spacious correlation of the subband coefficients, applying the context modeling predictive technique based on the multi-resolution structure and the feedback of the prediction error. The prediction context exploits the subordination and direction property of the different level subband in the vertical, horizontal, and diagonal subband coefficients. The simulation result of the high frequency images such as PEPPERS, BOAT, and AIRPLANE shows that the proposed algorithm efficiently predicts the edge area of each multi-resolution subband.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.42 no.4
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• pp.41-46
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• 2005

Acoustic problems in the environment have gained attention due to the noise and vibration contact often in industry as well as life on modem society. Noise is one of safety element in industry, this proves damaging to humans from both a physical and a psychological aspect and so drop working efficiency. Various noises are happened from machines of manufacturing process in PCB industry. This paper present a new approach for subband feedback Active Noise Control (ANC) using Mel scale for headset system in PCB industry. The proposed Mel scale subband algorithm had a performance advantage over the subband algorithm in the noise attenuation and convergence time.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.2
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• pp.386-388
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• 2017

In this paper, we propose a novel protocol design for multi-subband feedback-based multicast services in LTE systems, which reduces feedback overheads with guaranteeing QoS. Through the experimental results, it is manifested that the proposed protocol is able to reduce more feedback overheads in comparison with the conventional scheme.

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.85-88
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• 2002

In this paper, the proposed watermark algorithm is based on energy distribution of the subband coefficients in the frequency domain and edge of the original image in the spacial domain. Out of these information, the KeyMap which decides the embedded position of watermark is produced. And then the binary watermark is embedded into the wavelet coefficient of LL3 subband using KeyMap and LFSR(Linear Feedback Shift Register).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.5C
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• pp.706-715
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• 2004

In this paper, to decide the watermark embedding positions and embed the watermark we use the subband tee structure which is presented in the wavelet domain and the edge information in the spatial domain. The significant frequency region is estimated by the subband searching from the higher frequency subband to the lower frequency subband. LH1 subband which has the higher frequency in tree structure of the wavelet domain is divided into 4${\times}$4 submatrices, and the threshold which is used in the watermark embedding is obtained by the blockmatrix which is consists by the average of 4${\times}$4 submatrices. Also the watermark embedding position, Keymap is generated by the blockmatrix for the energy distribution in the frequency domain and the edge information in the spatial domain. The watermark is embedded into the wavelet coefficients using the Keymap and the random sequence generated by LFSR(Linear feedback shift register). Finally after the inverse wavelet transform the watermark embedded image is obtained. the proposed watermarking algorithm showed PSNR over 2㏈ and had the higher results from 2％ to 8％ in the comparison with the previous research for the attack such as the JPEG compression and the general image processing just like blurring, sharpening and gaussian noise.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.14 no.1
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• pp.94-101
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• 2000

In this paper, a multi-channel active noise control(ANC) system using subband hybrid control techniques is proposed. Subband techniques could reduce computational burden and improve the performance of ANC systems by dividing several frequency subband and adjusting adaptive filter coefficients. So it can effectively cancel noises at wanted frequency range and use lower order adaptive filter than the existing algorithms. The adjoint LMS algorithm, which prefilter the error signals instead of the divided reference signals in frequency band, is also used for adaptive filter algorithms to reduce the computational burden of the subband adaptive systems. To improve performance of the ANC system, a weighted hybrid control technique, which has weightily properties of feedforward control systems and feedback control systems, is applied. This algorithm shows higher stability and good noise attenuation property in broad band ANC systems. Computer simulations were performed to show the effectiveness of the proposed algorithm.

• ETRI Journal
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• v.33 no.6
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• pp.831-840
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• 2011

This paper focuses on codebook-based precoding for space-division multiple access/orthogonal frequency-division multiple access (SDMA-OFDMA) systems aiming to guarantee high throughput for their users as well as to mitigate interference to fixed satellite service (FSS). A systematic design of SDMA codebook for subband-based OFDMA is proposed, which forms multiple orthogonal beams with common spatial null in the direction of a victim FSS earth station (ES). The design enables both transmitter and receiver to independently construct identical codebook by sharing only on the direction angle of an FSS ES, which takes fewer overhead bits than Gram-Schmidt process, a general method satisfying our design criterion. A system-level throughput evaluation shows that the proposed precoding provides superior performance over existing spectrum sharing method, that is, subband deactivation. The spectrum sharing analysis shows that the proposed precoding, even with an estimation error of the direction angles of an FSS ES, causes lower interference than existing precoding, knockdown precoding.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.5
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• pp.538-542
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• 2018

Echo generated in digital hearing aid vent give rise to user's discomfort. For cancelling feedback echo in vent, it is required to estimate vent impulse response exactly. The vent impulse response has time varying and sparse characteristics. The IPNLMS has been known a useful adaptive algorithm to estimate vent impulse response with these characteristics. In this paper, subband sparse adaptive filter which applying IPNLMS to subband hearing aid structure is proposed to cancel echo of vent by estimating sparse vent impulse response. In the propose method, the decomposition of input signal to subband can pre-whiten each subband signal, so adaptive filter convergence speed can be improved. And the poly phase component decomposition of adaptive filter increases sparsity of each components, and the better echo cancellation can be possible without additional computation. To derive coefficients update equation of the adaptive filter, by defining the cost function based weight NLMS is defined, and the coefficient update equation of each subband is derived. For verifying performances of the adaptive filter, convergence speed, and steady state error by white signal input, and echo cancelling results by real speech input are evaluated by comparing conventional adaptive filters.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.645-651
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• 2003

The system in which a microphone and a loudspeaker are simultaneously used can cause an echo. The echo is caused by feedback between the output of the loudspeaker and the input of the microphone. The acoustic echo canceller is a device to cancel the echo in a communication system. Its general procedure for cancellation is first estimating the plant response of the feedback path and then eliminating the feedback signal from the input signal. In this paper, full-band and sub-band approaches are compared by using some simulation examples.