• Journal of Broadcast Engineering
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• v.1 no.2
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• pp.85-95
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• 1996

The purpose of this paper Is to propose a dubbed signal time-synchroniztion technique based on the SOLA(Synchronized Over-Lap and Add) method which has been widely used to modify the time scale of speech signal. In broadcasting audio recording environments, the high degree of background noise requires dubbing process. Since the time difference between the original and the dubbed signal ranges about 200mili seconds, process is required to make the dubbed signal synchronize to the corresponding image. The proposed method finds he starting point of the dubbing signal using the short-time energy of the two signals. Thereafter, LPC cepstrum analysis and DTW(Dynamic Time Warping) process are applied to synchronize phoneme positions of the two signals. After determining the matched point by the minimum mean square error between orignal and dubbed LPC cepstrums, the SOLA method is applied to the dubbed signal, to maintain the consistency of the corresponding phase. Effectiveness of proposed method is verified by comparing the waveforms and the spectrograms of the original and the time synchronized dubbing signal.

• Korean Journal of Computational Design and Engineering
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• v.16 no.4
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• pp.249-259
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• 2011

In this paper we present our work on the parameterized construction of virtual drivers' reach motion to seat belt, by using motion capture data. A user can generate a new reach motion by controlling a number of parameters. We approach the problem by using multiple sets of example reach motions and learning the relation between the labeling parameters and the motion data. The work is composed of three tasks. First, we construct a motion database using multiple sets of labeled motion clips obtained by using a motion capture device. This involves removing the redundancy of each motion clip by using PCA (Principal Component Analysis), and establishing temporal correspondence among different motion clips by automatic segmentation and piecewise time warping of each clip. Next, we compute motion blending functions by learning the relation between labeling parameters (age, hip base point (HBP), and height) and the motion parameters as represented by a set of PC coefficients. During runtime, on-line motion synthesis is accomplished by evaluating the motion blending function from the user-supplied control parameters.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2002.05a
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• pp.1-5
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• 2002

The purpose of this paper is developing a speech recognition module in a wheelchair for the sake of convenience. of the disability. For this system, we used TMS320C32 as a main processor; eliminated noise by applying Winer filler while considering characteristics of noise environment in pre-processing stage, and; extracted 12 feature patterns per france using LPC&Cepstrum. Then, we implemented the hybrid form combining DTW (Dynamic Time Warping), which is generally used for isolated words in the conventional algorithms, in the recognition Part, and NN (Neural network) to prevent any error of recognition. In this research, we achieved a recognition rate of more than 96% on isolated words when DTW and Hybrid forms were individually experimented in noise environment

• Journal of the Korean Wood Science and Technology
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• v.4 no.1
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• pp.48-53
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• 1976

This Study was carried out to investigate the course of air-drying and drying defects of red pine (Pinus densiflora S. et Z.) and Italian poplar (Populus eurameriana I-476) boards 1,2 and 3cm thick in the flat pile. The results are as follows. 1. Air-drying curves for red pine and Italian poplar boards were same as figure 1 and 2. These moisture contents were lower in July and August during seasoning periods. 2. Air drying time of red pine board required to dry 15 percent moisture content in one week for 1 cm board, five weeks for 2 cm board and six weeks for 3cm board respectively. In case of Italian poplar boards, in one week for 1 cm board, four weeks for 2 cm and five weeks for 3 cm board. The thickness of board influenced the time for air drying. 3. Drying defects such as checking, warping and staining happened badly in pine than in Italian poplar boards. Especially, checking was severe in thicker board and warping in thinner board.

• The Journal of the Acoustical Society of Korea
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• v.13 no.3
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• pp.25-32
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• 1994

This paper presents the Korean digit recognition method under noise environment using the spectral mapping training based on static supervised adaptation algorithm. In the presented recognition method, as a result of spectral mapping from one space of noisy speech spectrum to another space of speech spectrum without noise, spectral distortion of noisy speech is improved, and the recognition rate is higher than that of the conventional method using VQ (vector quatization) and DTW(dynamic time warping) without noise processing, and even when SNR level is 0dB, the recognition rate is 10 times of that using the conventional method. It has been confirmed that the spectral mapping training has an ability to improve the recognition performance for speech in noise environment.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.258-260
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• 2011

이 논문에서는 쿼리-바이-싱잉/허밍 (Query-by-singing/humming, QbSH) 시스템 또는 커버 노래 인식 (cover song identification) 시스템에서 사용 가능한 멜로디 유사도를 제안한다. QbSH 또는 커버 노래 인식은 디지털 음악의 사용이 보편화되면서 음악 검색의 방법으로 많은 연구가 진행되어 오고 있다. 멜로디 유사도는 이런 시스템을 구현하는데 필수적인 요소이며, 두 개의 음악에서 멜로디가 추출되었다고 가정하고, 추출된 멜로디 사이의 유사한 정도를 수치로 표현한다. QbSh 시스템이나 커버 노래 인식 시스템은 멜로디 유사도에 기반하여 입력 노래와 유사한 노래를 데이터베이스에서 검색하는 작업을 수행한다. 이 논문에서 제안하는 멜로디 유사도 방식은 기존의 많이 연구되던 동적 시간 왜곡 (dynamic time warping, DTW) 방법과 크로마 표현 방법 (chroma representation)을 사용하였다. DTW방법은 비대칭적으로 사용하고 미디 노트 영역에서 표현된 멜로디 특징은 0이상 12 미만의 크로마 레벨로 표현하였다. 기존의 방법에서는 정수값을 많이 사용하였으나 이 논문에서는 실수값을 사용한다. DTW 에 사용하는 거리 함수를 기존에 사용하던 차이의 절대값 대신 꺾인 함수 형태를 사용함으로써 성능을 높였다. QbSH 시스템에서의 실험을 통해서 성능을 검증하였다. 본 논문에서는 10-12초 길이의 1000번의 쿼리(Query)에 대해서 28시간 정도의 데이터베이스에서 실험한 결과, 순위 역의 평균 (Mean reciprocal rank, MRR) 값이 0.713을 보였다.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.6
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• pp.559-566
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• 2008

In this paper, we propose a range subsequence matching under dynamic time warping (DTW) distance. We exploit Dual Match, which divides data sequences into disjoint windows and the query sequence into sliding windows. However, Dual Match is known to work under Euclidean distance. We argue that Euclidean distance is a fragile distance, and thus, DTW should be supported by Dual Match. For this purpose, we derive a new important theorem showing the correctness of our approach and provide a detailed algorithm using the theorem. Extensive experimental results show that our range subsequence matching performs much better than the sequential scan algorithm.

• Journal of KIISE:Software and Applications
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• v.36 no.12
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• pp.1009-1015
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• 2009

Handwriting verification is a technique of distinguishing the same person's handwriting specimen from imitations with any two or more texts using one's handwriting individuality. This paper suggests an effective verification method for the handwritten signature or text on the off-line environment using pattern recognition technology. The core processes of the method which has been researched in this paper are extraction of letter area, extraction of features employing structural characteristics of handwritten text, feature analysis employing DTW(Dynamic Time Warping) algorithm and PCA(Principal Component Analysis). The experimental results show a superior performance of the suggested method.

• Phonetics and Speech Sciences
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• v.7 no.1
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• pp.3-10
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• 2015

In this paper, different frequency scales in cepstral feature extraction are evaluated for the text-independent speaker recognition. To this end, mel-frequency cepstral coefficients (MFCCs), linear frequency cepstral coefficients (LFCCs), and bilinear warped frequency cepstral coefficients (BWFCCs) are applied to the speaker recognition experiment. In addition, the spectro-temporal features extracted by the cepstral-time matrix (CTM) are examined as an alternative to the delta and delta-delta features. Experiments on the NIST speaker recognition evaluation (SRE) 2004 task are carried out using the Gaussian mixture model-universal background model (GMM-UBM) method and the joint factor analysis (JFA) method, both based on the ALIZE 3.0 toolkit. Experimental results using both the methods show that BWFCC with appropriate warping factor yields better performance than MFCC and LFCC. It is also shown that the feature set including the spectro-temporal information based on the CTM outperforms the conventional feature set including the delta and delta-delta features.

• Structural Engineering and Mechanics
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• v.28 no.4
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• pp.387-410
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• 2008

The fatigue damage accumulation rates of horizontally curved thin walled box-girder bridge have been estimated from vehicle-induced dynamic stress history using rain flow cycle counting method in the time domain approach. The curved box-girder bridge has been numerically modeled using computationally efficient thin walled box-beam finite elements, which take into account the important structural actions like torsional warping, distortion and distortional warping in addition to the conventional displacement and rotational degrees of freedom. Vehicle model includes heave-pitch-roll degrees of freedom with longitudinal and transverse input to the wheels. The bridge deck unevenness, which is taken as inputs to the vehicle wheels, has been assumed to be a realization of homogeneous random process specified by a power spectral density (PSD) function. The linear damage accumulation theory has been applied to calculate fatigue life. The fatigue life estimated by cycle counting method in time domain has been compared with those found by estimating the PSD of response in frequency domain. The frequency domain method uses an analytical expression involving spectral moment characteristics of stress process. The effects of some of the important parameters on fatigue life of the curved box bridge have been studied.