• Proceedings of the Korea Institutes of Information Security and Cryptology Conference
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• 2002.11a
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• pp.417-420
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• 2002

본 논문에서는, 지문인증 시스템에 있어서 방향성분의 통계적 방법을 이용한 reference point 검출 알고리즘과 이 reference point를 중심으로 하여 기존의 filterbank의 단점을 개선한 feature 추출 알고리즘을 소개한다. 제안한 reference point검출 알고리즘은 기존의 Poincare방법의 단점인 세밀한 전처리 과정과 arch type의 지문에 대해 장점을 가지며, 개선한 filterbank방법은 기존의 filterbank방법보다 reference point의 검출 위치에 덜 민감하며, 좀 더 높은 인증률을 보였다.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.178-181
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• 2000

In this paper, we present the implementation of filterbank for MPEG-2 Advanced Audio Coding (AAC) decoder with VHDL. The filterbank of AAC employs a technique called time-domain aliasing cancellation (TDAC). In order to make the algorithm more efficiently, we decompose and reorganize the filterbank algorithm lot the high speed decoding process and lower computational cost. And we make this filterbank algorithm to be used with other modules of AAC decoder in parallel processing.

• Journal of Institute of Control, Robotics and Systems
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• v.20 no.8
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• pp.882-889
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• 2014

This paper deals with a filterbank based on the IMM (Interacting Multiple Model) that combines data from a sensor and uses them selectively depending on a level of maneuver. Furthermore, within the maneuver interval, the existing IMM filter has disadvantages such as unnecessary target estimation errors caused by using a constant velocity model and an increase of computation load because of a fixed structure. On the other hand, the proposed IMM filterbank overcomes these disadvantages by using three model groups and designs a filterbank to cope with a wideband-maneuvering target. The performances of the IMM filterbank was evaluated through comparison with the existing IMM via computer simulations. The results show good performances for a wideband-maneuvering target.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.40 no.6
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• pp.95-105
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• 2003

A novel reference point detection method is proposed by exploiting tile gradient probabilistic model that captures the curvature information of fingerprint. The detection of reference point is accomplished through searching and locating the points of occurrence of the most evenly distributed gradient in a probabilistic sense. The uniformly distributed gradient texture represents either the core point itself or those of similar points that can be used to establish the rigid reference from which to map the features for recognition. Key benefits are reductions in preprocessing and consistency of locating the same points as the reference points even when processing arch type fingerprints. Moreover, the new feature extraction method is proposed by improving the existing feature extraction using filterbank method. Experimental results indicate the superiority of tile proposed scheme in terms of computational time in feature extraction and verification rate in various noisy environments. In particular, the proposed gradient probabilistic model achieved 49% improvement under ambient noise, 39.2% under brightness noise and 15.7% under a salt and pepper noise environment, respectively, in FAR for the arch type fingerprints. Moreover, a reduction of 0.07sec in reference point detection time of the GPM is shown possible compared to using the leading the poincare index method and a reduction of 0.06sec in code extraction time of the new filterbank mettled is shown possible compared to using the leading the existing filterbank method.

• Proceedings of the Korean Information Science Society Conference
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• 2011.06a
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• pp.255-258
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• 2011

An application of Gammatone auditory filterbank for emotion recognition from speech is described in this paper. Gammatone filterbank is a bank of Gammatone filters which are used as a preprocessing stage before applying feature extraction methods to get the most relevant features for emotion recognition from speech. In the feature extraction step, the energy value of output signal of each filter is computed and combined with other of all filters to produce a feature vector for the learning step. A feature vector is estimated in a short time period of input speech signal to take the advantage of dependence on time domain. Finally, in the learning step, Hidden Markov Model (HMM) is used to create a model for each emotion class and recognize a particular input emotional speech. In the experiment, feature extraction based on Gammatone filterbank (GTF) shows the better outcomes in comparison with features based on Mel-Frequency Cepstral Coefficient (MFCC) which is a well-known feature extraction for speech recognition as well as emotion recognition from speech.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.3
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• pp.119-128
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• 2017

In this paper, the wideband direction finding algorithm and system design method for short duration signal such as frequency hopping or burst signal is presented. The polyphase filterbank that it is possible for the near perfect reconstruction was used as a pre-processing and in each subband power measurement was performed to determine whether the presence of a signal and finally general direction finding algorithm was performed. In addition, various experiments was performed using Matlab Simulink and collected data from wideband receiver to verification of the proposed algorithm.

• Journal of Korea Society of Industrial Information Systems
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• v.6 no.4
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• pp.81-88
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• 2001

In this paper, a Korean digit recognition system based on a multilayer Perceptron is implemented. We also investigate the performance of widely used speech features, such as the Mel-scale filterbank, MFCC, LPCC, and PLP coefficients, by applying them as input of the proposed recognition system. In order to build a robust speech system, the experiments for demonstrating its recognition performance for the clean data as well as corrupt data are carried out. In experiments of recognizing 20 Korean digit, we found that the Mel-scale filterbank coefficients performs best in terms of recognition accuracy for the speech dependent and speech independent database even though noise is considerably added.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.245-248
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• 1998

In this paper, a new variable wavelet packet decomposition audio coder, based on the time varying characteristic of the audio signals, is proposed and presents a technique to incorporate psychoacoustic models into an adaptive wave let packet scheme. The proposed filterbank improves the defect of the polyphase filterbank that could not properly represent the critical band and the defect of QMF-tree filter that need high complexity to implement. The filterbank consists of varying number of subband from 4 to 26 bands and use Daubechies 6-order wave let. The codec yields excellent quality at total bit rates of about 128kbps for monophonic CD-quality signals with an sampling frequency of 44.1kHz and reduces complexity by 19% for various bit-rates and sources with encoding and decoding process.

• MALSORI
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• no.66
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• pp.73-86
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• 2008

Beamforming is one of the spatial filtering techniques which extract only desired signals from noisy environments using microphone arrays. Fixed beamforming is a simple concept and easy to implement. However, it does not show good performance in real noisy conditions. As an adaptive beamforming, Frost algorithm can be a good candidate. It uses the concept of the linearly constrained minimum variance (LCMV) algorithm. The difference between the Frost and the LCMV algorithm is the error correction scheme which is very effective feature in the aspect of performance. In this paper, as quadrature mirror filtering (QMF)-based filterbank is utilized as the pre-processing of the Frost beamformning, the filter length and the learning rate of each band is optimized to improve the performance. The performance is measured by the signal-to-noise ratio (SNR) and the Bark's scale spectral distortion (BSD).

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1717-1720
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• 2002

Objective quality measurement schemes that in- corporate properties of the human auditory system. The basilar membrane(BM) acts as a spectrum analyzer, spatially decomposing the signal into frequency components. Each filterbank is an implementation of the ERB, gam-machirp function. This filterbank is level-dependent asymmetric compensation filters. And for the validation of the auditory model, we calculate the CPD. Quality measurement is obtained from the result.