• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2002.11a
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• pp.81-84
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• 2002

본 논문에서는 간략화 된 형상학적 다항식 변환(Morphological Polynomial Transform)과 형상학적 보간법(Morphological Interpolation)을 이용하는 비선형 예측 방법을 제안한다. 형상학적 다항식 변환은 형상학적 연산을 통해 데이터를 구조함수들의 계수들로 표현하는 변환이며, 형상학적 보간법은 형상학적 다항식 변환에 의한 계수들을 이용하여 보간하는 방법이다. 형상학적 다항식 변환을 간략화 하여 정수 연산만으로 적용할 수 있도록 개선하였으며, 보다 영상에 적합한 형상학적 보간법에 기반 한 예측 방법을 사용한다. 제안하는 예측 방법과 허프만 부호화를 사용하여 적은 비트로 영상을 손실 없이 저장할 수 있음을 실험으로 검증한다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.2 s.302
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• pp.131-142
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• 2005

The low rate speech coders under 4 kbit/s are based on sinusoidal transform coding (STC) or multiband excitation (MBE). Since the harmonic coders are not efficient to reconstruct the transient segments of speech signals such as onsets, offsets, non-periodic signals, etc, the coders do not provide a natural speech quality. This paper proposes method of a efficient transient model :d a multi-mode low rate coder at 2.4 kbit/s that uses harmonic model for the voiced speech, stochastic model for the unvoiced speech and a model using aperiodic pulse location tracking (APPT) for the transient segments, respectively. The APPT utilizes the harmonic model. The proposed method uses different models depending on the characteristics of LPC residual signals. In addition, it can combine synthesized excitation in CELP coding at time domain with that in harmonic coding at frequency domain efficiently. The proposed coder shows a better speech quality than 2.4 kbit/s version of the mixed excitation linear prediction (MELP) coder that is a U.S. Federal Standard for speech coder.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.2
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• pp.186-192
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• 2011

In this paper, we propose a new procedure for improvement of the linear predictive coding. To reduce the error power incurred by the coding, we interchanged the order of the two procedures of windowing on the signal and linear prediction. This scheme corresponds to LPC extraction with windowed autocorrelation. The proposed method requires more calculational time because it necessitates matrix inversion on more parameters than the conventional technique where an efficient Levinson-Durbin recursive procedure is applicable with smaller parameters. Experimental test over various speech phonemes showed, however, that our procedure yields about 5 % less power distortion compared to the conventional technique. Consequently, the proposed method in this paper is thought to be preferable to the conventional technique as far as the fidelity is concerned. In a separate study of speaker-dependent speech recognition test for 50 isolated words pronounced by 40 people, our approach yielded better performance too.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8C
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• pp.822-829
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• 2009

This paper proposes a new method of improving the quality of High Efficiency Advanced Audio Coding (HE-AAC). HE-AAC encodes input source by allocating bits for each scalefactor bands appropriately according to human ear's psychoacoustic property. As a result, insufficient bits are assigned to the bands which have relatively low energy. This imbalance between different energy bands can cause decreasing of sound quality like musical noise. In the proposed system, a Linear Prediction (LP) module is combined with HE-AAC as a pre-processor to improve sound quality by even bits distribution. To apply accurate human being's psychoacoustic property, the psychoacoustic model uses Fast Fourier Transform (FFT) spectrum of original input signal to make masking threshold. In its implementation, masking threshold of psychoacoustic model is normalized using the LP spectral envelope in prior to quantization of the LP residual. Experimental result shows that, the proposed algorithm allocates bits appropriately for insufficient bits condition and improves the performance of HE-AAC.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.246-249
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• 1995

4.16kbps의 전송률을 갖는 음성 부호화기 구조에 관하여 기술한다. 제안된 음성 부호화기는 개방 회로 피치 검출기와 이로부터 생성된 pulse train을 코드북으로 갖는 CELP 부호화기이다. Pulse-Train codebook은 분석 프레임별로 부호화 및 복호화 양단에서 생성되며 음성의 피치 및 포만트 정보를 내포하고 있다. 구현된 PT-CELP는 random codebook 방식의 CELP에 비해 적은 크기로 codebook을 만들 수 있으며 음성의 특징을 충분히 반영하므로 합성된 음성의 음질을 향상시킬 수 있다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.109-111
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• 2004

발성율은 일정한 시간동안 발성되는 음성신호 내에 몇 개의 음절이 포함되어 있는 지를 나타낸다. 발성율은 화자마다 다르고 각 음소들의 특징에 따라 변화할 수 있다. 발성율의 사전 측정이 이루어 진다면 음성부호화 측면에서도 중용한 정보로 사용될 수 있다. 기존의 음성부호화기는 발성율에 관계없이 고정적인 분석 구간을 정하여 전송률을 결정하고 있다. 따라서, 발성율을 미리 측정한다면, 발성율이 느린 부분과 빠른 부분에 각기 다른 부호화 방법을 적용하여 음질을 향상할 수도 있고 전송률을 가변적으로 적용할 수 도 있게 된다. 정확한 발성율을 측정하기 위해서는 음절의 변화를 추정하여야 한다. 음절의 변화를 추정하기 위한 방법으로 음성신호의 에너지 포락선 측정법과 LSP를 이용한 측정법이 각각 제안된 바 있으나, 본 논문에서는 위 두 가지 방법을 혼합한 방법을 사용하였다. 에너지 변동은 음성신호의 시간영역 처리방법으로 LSP 파라미터는 음성신호의 선형예측 분석에 의해 구해질 수 있다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2016.06a
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• pp.373-374
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• 2016

Residual Differnetial pulse-code Modulation (RDPCM) 기법은 비디오의 압축을 위한 시간 및 공간 예측 후 남은 잔여신호를 인접 화소를 이용하여 추가적인 중복정보를 제거하는 기법을 의미한다. 본 논문에서는 우선 잔차 신호의 예측을 위하여 인접 화소 사이 선형 가중 합으로 예측 모델을 세우고, 각 가중치를 $L_1$ 정규화를 포함하는 비용함수를 통해 추정함으로써 보다 효율적인 부호화 성능을 제공하는 알고리즘을 제안한다.

• The Journal of the Acoustical Society of Korea
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• v.25 no.6
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• pp.282-290
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• 2006

This paper proposes a new Digital Reverberator that models Analog Helical Coil Spring Reverberator for guitar amplifiers. While the conventional digital reverberators are proposed to provide better sound field mainly based on room acoustics, no algorithm or analysis of digital reverberators those model Helical Coil Spring Reverberator was proposed. Considering the fact that approximately $70{\sim}80$ percent of guitar amplifiers are still with Helical Coil Spring Reverberator, research was performed based not on Room Acoustics but on Helical Coil Spring Reverberator itself as an effector. After performing simulations with proposed algorithm, it was confirmed that the Digital Reverberator by proposed algorithm provides perceptually equivalent response to the conventional Analog Helical Coil Spring Reverberators.

• Journal of KIISE:Information Networking
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• v.35 no.3
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• pp.235-242
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• 2008

To improve video quality and coding efficiency, H.264/AVC adopted an adaptive rate control. But this method has a problem as it cannot predict an accurate quantization parameter(QP) for the first frame. The first QP is decided among four constant values by using encoder input parameters. It does not consider encoding bits, results in significant fluctuation of the image quality and decreases the average quality of the whole coded sequence. In this paper, we propose a new algorithm for the first frame QP decision in the H.264/AVC encoder. The QP is decided by the existing algorithm and the first frame is encoded. According to the encoded bits, the new initial QP is decided. We can predict optimal value because there is a linear relationship between encoded bits and the new initial QP. Next, we re-encode the first frame using the new initial QP. Experimental results show that the proposed algorithm not only achieves better quality than the state of the art algorithm, but also adopts a rate control forthe sequence that was impossible with the existing algorithm. By reducing fluctuation, subjective quality also improved.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9C
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• pp.1330-1336
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• 2004

This paper addresses an adaptive motion vector prediction algorithm to improve the performance of video encoder. The block-based motion vector is characterized by non-stationary local statistics so that the coefficients of LS (Least Squares) based linear motion can be optimized. However, it requires very expensive computational cost. The proposed algorithm using LS approach with spatially varying motion-directed property adaptively controls the coefficients of the motion predictor and reduces the computational cost as well as the motion prediction error. Experimental results show the capability of the proposed algorithm.