• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.476-485
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• 2003

This paper proposers a preprocessing approach to improving the quality of the music produced by the EVRC(enhanced variable rate codec) which is one of the CDMA(Code Division Multiple Access) voice codecs. Since the EVRC is optimized only for speech signals, it can deteriorate the quality of the music passed through it. One of the problems with the EVRC-coded music is time-clipping, which usually occurs when subsequent frames are encoded at Rate l/8. Since the EVRC determines the bit rate for an input frame based on the long-term prediction gain, we increase the long-term prediction gain in order for the most of the frames to be encoded at Rate 1 or Rate 1/2. Experimental results show that the approach works well on music signals and the number of time-clipped frames is considerably reduced.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.07a
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• pp.176-177
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• 2012

사람의 음성을 압축하는 방법으로 Code Excited Linear Prediction (CELP) 코더가 주로 사용되어 왔다. CELP 코더의 수신단에서는 양자화 된 여기신호를 LPC 필터로 합성하여 신호를 복원한다. LPC 합성필터의 영향으로 양자화 된 여기신호의 보로노이 셀 모양이 변형되는 문제점이 있기 때문에 이런 문제점을 해결하기 위해서 Karhunen-Loeve-Transform based Classify vector Quantization (KLT-CVQ) 코더가 제안되었다. 기존 KLT-CVQ 코더는 KLT 변환과 class 선택을 위해서 Eigen Value Decomposition (EVD)을 이용해서 eigen vector와 eigen value를 계산한다. 본 논문에서는 EVD 대신에 UTV Decomposition (UTVD)을 이용하여 KLT-CVQ의 계산량 문제점을 개선하는 방법을 제안한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.2
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• pp.413-418
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• 2009

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high qualify approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.50-53
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• 1991

벡터 양지화(VQ)는 신호 처리분야에서 정보의 압축을 위해 사용하는 아주 잘 알려진 방법이다. 벡터 양지화는 정보를 대량으로 줄이면서 그 효율을 떨어 뜨리지 않는 방향으로 발전해 왔다. VQ코드북의 크기가 커지면 하나의 코드워드를 찾기위한 시간이 증가하게 된다. 코드북의 빠른 검색을 위하여 다른 방법에 제안 되기도 했으나 최적 검색 방법이라고는 볼 수 없다. 본 고에서는 음성인식에 적용할 목적으로 기존의 방법으로 구성된 코드북의 구성을 변형 하지 않고 검색 속도를 증가 시킬 수 있는 방법을 기수랗고 그 효율에 대해서 설명한다.

• Proceedings of the KAIS Fall Conference
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• 2000.10a
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• pp.203-207
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• 2000

최근 신호 처리 분야에서 많은 연구가 되고 있는 웨이브렛 변환을 적용하고, DSP(Digital Signal Processor)인 TMS320C31을 사용하여 고속 처리 가능한 하드웨어를 구현하였다. 그리고, 컴퓨터하고 일정한 통신 대역을 유지하고 다른 장치에 영향을 주지 안기 위해서 ISA 버스를 사용하였다. 여기서는 웨이브렛 변환과 푸리에 변환의 차이 및 필터뱅크에 대해서 알아보고, DSP를 이용하여 웨이브렛 변환을 시키는 하드웨어를 구현했다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7C
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• pp.962-970
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• 2004

This paper presents an efficient implementation of the 2.4 kbps EHSX(Enhanced Harmonic Stochastic Excitation) speech coder on a TMS320C6701$^{TM}$ floating-point digital signal processor. The EHSX speech codec is based on a harmonic and CELP(Code Excited Linear Prediction) modeling of the excitation signal respectively according to the frame characteristic such as a voiced speech and an unvoiced speech. In this paper, we represent the optimization methods to reduce the complexity for real-time implementation. The complexity in the filtering of a CELP algorithm that is the main part for the EHSX algorithm complexity can be reduced by converting program using floating-point variable to program using fixed-point variable. We also present the efficient optimization methods including the code allocation considering a DSP architecture and the low complexity algorithm of harmonic/pitch search in encoder part. Finally, we obtained the subjective quality of MOS 3.28 from speech quality test using the PESQ(perceptual evaluation of speech quality), ITU-T Recommendation P.862 and could get a goal of realtime operation of the EHSX codec.c.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.5
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• pp.924-930
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• 2015

In this paper, we propose a novel approach of low computational complexity to improve the speech quality of the small acoustic equipment in noisy environment. The conventional gain control algorithm suppresses the noise of input signal, and then the part of wide dynamic range compression (WDRC) amplifies the undesired signal. The proposed algorithm controls the gain of hearing aids according to speech present probability by using the output of a voice activity detection (VAD). The performance of the proposed scheme is evaluated under various noise conditions by using objective measurement and yields superior results compared with the conventional algorithm.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.3
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• pp.24-31
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• 2001

The K-L expansion is a method for compressing dimensions of features and thus reduces computational cost in recognition process. Also This is well known that features can be extracted without much loss of information in the statistical pattern recognition. In this paper, the method that effectively applies K-L(Karhunen-Loeve) expansion to feature parameters of speech is proposed to improve the recognition accuracy of the Korean speech recognition system. The recognition performance of a novel feature parameters obtained by the proposed method(K-L coefficients) is compared with those of conventional Mel-cepstrum and regressive coefficients through speaker independent connected digits recognition experiments. Experimental results showed that average recognition rates using the K-L coefficients with regression coefficients obtained higher accuracy than conventional Mel-cepstrum with their regression coefficients.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.2
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• pp.25-30
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• 2003

Recently, development of digital technology is occupying a large part of multimedia information like character, voice, image, video, etc. Research about video indexing and retrieval progresses especially in research relative to video. In this paper, we propose new algorithm(Frame Feature Comparison) for MPEG video segmentation. Shot, Scene Change detection is basic and important works that segment it in MPEG video sequence. Generally, the segmentation algorithm that uses much has defect that occurs an error detection according to a flash of camera, movement of camera and fast movement of an object, because of comparing former frames with present frames. Therefore, we distinguish a scene change one more time using a scene change point detected in the conventional algorithm through comparing its mean value with abutted frames. In the result, we could detect more corrective scene change than the conventional algorithm.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.153-156
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• 2000

본 논문에서는 멀티미디어 서비스들 중에서 음성 또는 오디오 신호를 저속으로 압축할 때 사용되는 G.723.1 부호화기의 line spectral frequency(LSF) 계수 양자화 방식을 고속으로 처리하는 알고리즘을 제안하였다. 제안된 고속탐색 방법은 LSF 계수의 순서성질을 이용하여 코드북의 탐색 범위를 줄임으로써 계산량을 크게 감소시킨다. 제안된 고속탐색 방법을 predictive split VQ(PSVQ) 구조를 갖는 G.723.1 에 적용한 결과 spectral distortion(SD) 성능 감쇄 및 추가적인 메모리 증가 없이 최적 코드벡터를 찾기 위한 코드북 탐색 과정에서 코드북의 평균 탐색 범위가 $20.1\%$ 감소했으며, 이는 additions, subtractions, multiplies 및 comparisons 수가 각각 $19.1\%$, $20.1\%$, $19.4\%$ 및 $12.2\% 감소하는 결과를 얻었다.