• Journal of the Institute of Convergence Signal Processing
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• v.15 no.3
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• pp.91-96
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• 2014

For Sound source direction and separation method, Frequency Domain Binaural Model(FDBM) shows low computational cost and high performance for sound source separation. This method performs sound source orientation and separation by obtaining the Interaural Phase Difference(IPD) and Interaural Level Difference(ILD) in frequency domain. But the problem of reflection occurs in practical environment. To reduce this reflection, a method to simulate the sound localization of a direct sound, to detect the initial arriving sound, to check the direction of the sound, and to separate the sound is presented. Simulation results show that the direction is estimated to lie close within 10% from the sound source and, in the presence of the reflection, the level of the separation of the sound source is improved by higher Coherence and PESQ(Perceptual Evaluation of Speech Quality) and by lower directional damping than those of the existing FDBM. In case of no reflection, the degree of separation was low.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.925-928
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• 2008

With basis of the development of information communication techniques in recent year, the digital processing techniquy also has been growed fast. The digital processing technique have used signals - speech and image processing- for processing of transmission and analysis. After we get and save the signals. Effective signal processing techniques have varied filters and typical digital filters are FIR filter and IIR filter. The FIR digital filter is more secure because phase response characteristics have linear phase. But, FIR digital filters have a problem to product the Gibbs phenomenon generating around a discontinuous point. A propose of filer is to remove the problem. Therefore, in this paper I was proposed a method using FIR digital filter applied a modified window function and the method was compared with conventional methods.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2005.11a
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• pp.250-255
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• 2005

In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and reduces the computational complexity. The proposed filter is applied to wavelet based adaptive algorithm. Actually we compared the performance of the proposed algorithm with other algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, the frequency domain algorithm is prefer than the existent time domain. we analyzed the Wavelet algorithm, short-length fast running FIR algorithm, fast-short-length fast running FIR algorithm and proposed algorithm.

• 한국정보통신설비학회:학술대회논문집
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• 2005.08a
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• pp.397-405
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• 2005

This paper presents a headphone system which has adopted two algorithm to increase sound clearness and to separate signal from noisy environment. In the field of adaptive signal processing, LMS algorithm which is a kind of steepest decent method, can be implemented with more simple calculation, so that we use it to eliminate unwanted noise elements for the proposed system. Futhermore we generate early echo using some delays, then mix it in signal. This process can increase the clearness of signal. In this paper, we prove that the proposed system can be implemented in real time. The proposed system is satisfied to subject assessment test base on MOS(Mean Opinion Score) of ITU-T.

• The Journal of the Acoustical Society of Korea
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• v.13 no.6
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• pp.91-97
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• 1994

In speech processing, the waveform codings are concerned with simply preserving the waveform of signal through a redundancy reduction process. In the case of speech synthesis, the waveform codings with high quality are mainly used to the synthesis by analysis. Because the parameters of this coding are not classified as both excitation and vocal tract, it is difficult to apply the waveform coding to the synthesis by rule. Thus, in order to apply the waveform coding to synthesis by rule, it is necessary to alter the pitches. In this paper, we proposed a new pitch alteration method that can change the pitch period in waveform coding by dividing the speech signals into the vocal tract and excitation parameters. This method is a time-frequency domain method preserving the phase component of the waveform in time domain and the magnitude component in frequency domain. Thus, it is possible that the waveform coding is carried out the synthesis by rule in speech processing. In case of using the algorithm, we can obtain spectrum distortion with $2.94\%$. That is, the spectrum distortion is decreased more $5.06\%$ than that of the pitch alteration method in time domain.

• Proceedings of the Korean Society of Marine Engineers Conference
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• 2005.11a
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• pp.57-58
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• 2005

Recently the telematics system is used widely. Users want to high quality communications. Since the primary advantage of using an array is to enhance a desired signal and reject jamming interferences, array signal processing is essential to satisfy unmet demand of user. In general, beamforming is a spatial filtering operation performed on the data received by an array of sensors. So we propose the beamformer design that use FPGA for real time processing. And we use zero-padding interpolation for high resolution data.

• Journal of the Institute of Convergence Signal Processing
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• v.13 no.1
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• pp.19-25
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• 2012

An adaptive system is a well-known method for removing noise from noise-corrupted speech. In this paper, we perform a least mean square (LMS) based on wavelet adaptive algorithm. It establishes the faster convergence rate of as compared to time domain because of eigenvalue distribution width. And this paper provides the basic tool required for the FIR algorithm whose algorithm reduces the arithmetic complexity. We consider a new fast short-length running FIR structure in discrete wavelet adaptive algorithm. We compare FIR algorithm and short-length fast running FIR algorithm (SFIR) to the proposed fast short-length running FIR algorithm(FSFIR) for arithmetic complexities.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.277-280
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• 2001

본 연구에서는 연속음성인식 시스템의 성능개선을 위한 기초 연구로서 음소결정트리 상태분할과 한국어 음성학적 지식을 이용하여 문맥의존 음향모델의 작성방법을 검토하고. 한국어 연속음성인식에 적용을 소개한다. 음소결정트리 상태분할 알고리즘은 각 노드에서 한국어 음성학적 지식으로 구성된 음소 질의어 집합에 따라 2진 트리로 SSS(Successive State Splitting) 알고리즘에 의해 상태분할 하는 방법으로서 상태분할 후 각 상태를 네트워크로 연결한 구조를 HM-Net(Hidden Markow Network)이라 하며 문맥의존 음향모델로 표현된다. 작성한 문맥의존 음향모델의 유효성을 확인하기 위해 본 연구실의 항공편 예약 문장(YNU200)에 대해 연속음성인식 실험을 수행하였다. 인식실험 결과, 문맥의존 음향모델에 대한 화자독립 연속음성인식률이 기존의 단일 HMM 모델보다 평균적으로 1-pass의 경우 9.9%, 2-pass의 경우 4.1% 향상된 인식률을 보였다. 따라서 문맥의존 음향모델을 작성하는데 음소결정트리 상태분할과 한국어 음성학적 지식이 유효함을 확인하였다.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2006.06a
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• pp.17-20
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• 2006

본 논문에서는 잡음이 첨가된 연속음성에서의 자동 음절분할을 위해 기존에 사용되고 있는 특징 파라미터인 단구간 에너지 이외에 잡음에 강인한 특성을 가지고 있는 새로운 특징인 스펙트럼 밀도비교척도와 의사역행렬을 이용한 선형판별함수를 제안한다. 기존에 사용되는 단구간 에너지는 잡음이 없는 환경에서는 좋은 성능을 나타내지만 잡음환경에서는 그렇지 못하다. 반면에 논문에서 제안한 척도들은 반대의 성능을 가지므로 주변잡음의 크기에 따라 각각의 파라미터를 적절한 가중치로 조합하는 음절구간 결정함수와 유한상태 머신을 추가로 사용면 무 잡음 환경뿐만 아니라, 잡음이 첨가된 연속음성에서도 일정수준 이상의 음절구간을 분리해 낼 수 있다.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.177-180
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• 2000

최근에 음성인식의 상용화가 급격히 추진되고 있다. 그러나 음성인식 응용제품의 부족과 음성인식 시스템의 성능문제로 인하여 일반인의 이용은 그다지 많지 않다. 본 논문에서는 연구실에서 만든 가변 어휘 음성인식기를 이용하여 음성인식 끝말 이어가기 게임을 구현하였다. 가변어휘 음성 인식기는 VCCV(Vowel＋consonant＋Consonant＋vowel) 기반의 화자독립으로 구현하였다. 끝말 이어가기 게임을 위해서 약 500만 어절이 포함된 문장에서 추출한 단어의 일부를 이용하여 사전을 구축하였고, 같은 음절로 시작하는 단어가 많은 경우에는 그 수를 제안하였다. 본 연구에서 구현한 음성인식 끝말 이어가기 게임은 제한된 단어사전을 이용하도록 하였으나 음성인식기의 성능향상과 완전한 사전구축이 이루어지면 음성인식을 이용한 언어 학습기나 게임 등의 개발과 이용의 활성화에 크게 기여할 것이라 생각된다.