• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.12
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• pp.2521-2526
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• 2011

This paper proposes a speaker recognition algorithm using a perceptron neural network and LPC (Linear Predictive Coding) cepstrum coefficients. The proposed algorithm first detects the voiced sections at each frame. Then, the LPC cepstrum coefficients which have speaker characteristics are obtained by the linear predictive analysis for the detected voiced sections. To classify the obtained LPC cepstrum coefficients, a neural network is trained using the LPC cepstrum coefficients. In this experiment, the performance of the proposed algorithm was evaluated using the speech recognition rates based on the LPC cepstrum coefficients and the neural network.

• The Journal of the Acoustical Society of Korea
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• v.23 no.4
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• pp.333-339
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• 2004

This Paper proposes the vector quantizer-pyramid vector quantizer(VQ-PVQ) structure. Also both predictive structure and safety-net concept are combined into the VQ-PVQ to quantize the IPC parameter of wideband speech codec. The Performance is compared to the LPC vector quantizer used in the AMR-WB(ITU-T G.722.2). demonstrating reduction in both spectral distortion and encoding memory.

• Journal of Korea Multimedia Society
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• v.11 no.11
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• pp.1480-1490
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• 2008

Design of a linear predictor and matching of an entropy coder is the art of lossless audio coding. In this paper, we use the covariance method and the Choleskey decomposition for calculating linear prediction coefficients instead of the autocorreation method and the Levinson-Durbin recursion. These results are compared to the polynomial predictor. Both of them, the predictor which has small prediction error is selected. For the entropy coding, we use the Golomb-Rice coder using the block-based parameter estimation method and the sequential adaptation method with LOCO-land RLGR. The proposed predictor and the block-based parameter estimation have $2.2879%{\sim}0.3413%$ improved compression ratios compared to FLAC lossless audio coder which use the autocorrelation method and the Levinson-Durbin recursion. The proposed predictor and the LOCO-I adaptation method could improved by $2.2879%{\sim}0.3413%$. But the proposed predictor and the RLGR adaptation method got better results with specific signals.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.5
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• pp.458-467
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• 1991

In this paper, the cause of linear prediction error is analysed and the theoretical basis for nomalizing the format bandwidth to 0is given and its validity is verified. The formant and bandwidth in relation to the position of the poles of AR filter are measured for an alaysis of the relation between the pole position and the formant bandwidth. By changing the glottis reflection coefficient to 1. the pole position and the formant bandwidth. By changing the glottis reflection coefficient to 1. the effect of the glottis is eliminated and as the result a new linear preiction coefficients are obtained by normalizing the formant bandwidth of the signal to 0. since these coefficients are symmetrical, the standard deviation is larger than the coefficients with fixed glottis reflection coefficient. The bit rate for speech coding can be reduced by a factor of 2 without any loss of information. Through computer simulation, recognition rate of 96.7% is botained by using the proposed algorithm in recognizing 5 Korean vowels in noisy environment.

• 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.

• 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 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.10C
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• pp.1484-1489
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• 2004

Given a video frame in terms of its 8${\times}$8 block-DCT coefncients, we wish to obtain a downsized or upsized version of this Dame also in terms of 8${\times}$8 block DCT coefficients. The DCT being a linear unitary transform is distributive over matrix multiplication. This fact has been used for downsampling video frames in the DCT domains in Dugad's, Mukherjee's, and Park's methods. The downsampling and upsampling schemes combined together preserve all the low-frequency DCT coefficients of the original image. This implies tremendous savings for coding the difference between the original frame (unsampled image) and its prediction (the upsampled image).This is desirable for many applications based on scalable encoding of video. In this paper, we extend the earlier works to various DCT sizes, when we downsample and then upsample of an image by a factor of two. Through experiment, we could improve the PSM values whenever we increase the DCT block size. However, because the complexity will be also increase, we can say there is a tradeoff. The experiment result would provide important data for developing fast algorithms of compressed-domain image/video resizing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.3A
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• pp.196-204
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• 2002

The Sinusoidal Transform Coding(STC) is a vocoding scheme based on a sinusoidal model of a speech signal. The low bit-rate speech coding based on sinusoidal model is a method that models and synthesizes speech with fundamental frequency and its harmonic elements, spectral envelope and phase in the frequency region. In this paper, we propose the 2.4kbps low-rate speech coding algorithm using the sinusoidal model of a speech signal. In the proposed coder, the pitch frequency is estimated by choosing the frequency that makes least mean squared error between synthetic speech with all spectrum peaks and speech synthesized with chosen frequency and its harmonics. The spectral envelope is estimated using SEEVOC(Spectral Envelope Estimation VOCoder) algorithm and the discrete all-pole model. The phase information is obtained using the time of pitch pulse occurrence, i.e., the onset time, as well as the phase of the vocal tract system. Experimental results show that the synthetic speech preserves both the formant and phase information of the original speech very well. The performance of the coder has been evaluated in terms of the MOS test based on informal listening tests, and it achieved over the MOS score of 3.1.