• 방송공학회논문지
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• 제15권1호
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• pp.133-143
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• 2010

본 논문은 영상을 효율적으로 부호화하면서도 자연스러운 압축 영상을 만들어내는 쿼드트리 영상 압축 기법을 제안한다. 제안하는 압축 기법은 유의미한 에지 선을 보존하기 위해서 에지-기반 쿼드트리 분할을 적용하고, 잎 노드 블록 사이의 높은 상관성을 활용하기 위해서 예측부호화 기법을 이용한다. 제안하는 압축 기법이 JPEG에 비해서 부호 효율이 27% 이상 개선됨을 $256\times256$ 휘도영상에 대한 전산실험을 통하여 검증하였다. 제안하는 압축 기법은 JPEG과 같은 고정 블록 부호화 기법에서 나타나곤 하는 압축 영상에서의 파문 현상을 발생시키지 않아서, 보다 자연스러운 압축 영상을 제공할 수 있다.

• 대한전자공학회논문지
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• 제27권4호
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• pp.491-501
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• 1990

Recently, four speech coding techniques, namely, SBC-APCM(sub-band coding adaptive PCM), RPE-LPC(regualr pulse excitation linear predictive codec), MPE-LTP(multi-pulse excited long-term prediction) and CELP (code-excited linear prediction) are proposed for digital mobile radio applications. However, a performance comparison of these coders in the Rayleigh fading environment has not been made yet. In this paper, the performances of the four spech coders in the random bit error and burst error environment are investigated. For the channel coding of SBC-APCM, RPE-LPC and MPE-LTP, the sensitivity of output bit stream is measured and a bit selective forward error correction is provided acording to the measured bit sensitivity. And for an attempt to improve the performance of CELP, an optimum quantizer is applied for transmitting scalar quantities in CELP. However, an improvement over the conventional approach is found to be negligible. For the channel coding of CELP, Reed-Solomon code, Golay code, convolutional code of rate 1/2 shows the best performance. Finally, from the simulation results, it is concluded that CELP is the best candidate for digital mobile radio and is followed by MPE-LTP, SBC-APCM and RPE-LPC.

• 한국지리정보학회지
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• 제3권1호
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• pp.1-11
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• 2000

본 논문에서는 인공위성으로부터 얻어진 다중스펙트럼영상의 부호화 방법을 다룬다. 위성영상의 공간 및 스펙트럼 해상도가 급속도로 향상되면서 처리해야 할 다중스펙트럼 영상의 데이터량은 엄청나게 증가하였다. 이에 따라 위성영상을 활용하기 위해서는 효율적으로 부호화하는 기술이 필요하다. 본 논문에서는 벡터양자화에 근거한 예측부호화, 영상의 quadtree 분할, 그리고 예측오차의 압축을 위한 DCT를 복합적으로 적용한 부호화 기법을 제시한다. 벡터양자화를 통해 대역영상간의 공간적인 특징이 동일하다는 점을 이용한 예측을 하고, 영상분할을 통해 영상의 공간적인 정보량에 따라 적응적으로 비트를 할당하며, DCT를 통해 예측오차의 공간적응적인 부호화를 수행한다. Landsat TM 영상을 대상으로 수행한 실험을 통해 제안 알고리듬의 위성영상 압축기법으로서의 타당성을 보였다.

• 한국음향학회지
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• 제23권4호
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• pp.333-339
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• 2004

본 논문에서는 벡터 양자화기와 피라미드 벡터 양자화기를 직렬로 결합하여 16차 벡터 소스에 대한 vector quantizer-pyramid vector quantizer (VQ-PVQ)를 개발하였으며, 예측 구조와 세이프티-넷 (safety-net) 개념을 결합시켜 광대역 음성 부호화기용 LPC 계수 양자화 기를 설계하였다. 본 양자화기의 성능은 AMR-WB(ITRT-T G.722.2)의 LPC양자화기 성능과 비교하였는데, 스펙트럼 왜곡 및 메모리 요구량에서 상당한 이득을 얻었다.

• 전자공학회논문지B
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• 제32B권1호
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• pp.96-102
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• 1995

This paper presents the VHSIC Hardware Description Language(VHDL) implementation of the Fixed Point Covariance Lattice(FLAT) algorithm for an Linear Predictive Coding(LPC) analysis and its related algorithms, such as the forth order high pass Infinite Impulse Response(IIR) filter, covariance matrix calculation, and Spectral Smoothing Technique(SST) in the Vector Sum Exited Linear Predictive(VSELP) speech coder that has been Selected as the standard speech coder for the North America and Japanese digital cellular. Existing Digital Signal Processor(DSP) chips used in digital cellular phones are derived from general purpose DSP chips, and thus, these DSP chips may not be optimal and effective architectures are to be designed for the above mentioned algorithms. Then we implemented the VHDL code based on the C code, Finally, we verified that VHDL results are the same as C code results for real speech data. The implemented VHDL code can be used for performing logic synthesis and for designing an LPC Application Specific Integrated Circuit(ASOC) chip and DsP chips. We first developed the C language code to investigate the correctness of algorithms and to compare C code results with VHDL code results block by block.

• Journal of information and communication convergence engineering
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• 제10권2호
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• pp.175-180
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• 2012

This paper presents a blind watermark detection scheme for an additive watermark embedding model. The proposed estimation-correlation-based watermark detector first estimates the embedded watermark by exploiting non-Gaussian of the real-world audio signal and the mutual independence between the host-signal and the embedded watermark and then a correlation-based detector is used to determine the presence or the absence of the watermark. For watermark estimation, blind source separation (BSS) based on independent component analysis (ICA) is used. Low watermark-to-signal ratio (WSR) is one of the limitations of blind detection with the additive embedding model. The proposed detector uses two-stage processing to improve the WSR at the blind detector; the first stage removes the audio spectrum from the watermarked audio signal using linear predictive (LP) filtering and the second stage uses the resulting residue from the LP filtering stage to estimate the embedded watermark using BSS based on ICA. Simulation results show that the proposed detector performs significantly better than existing estimation-correlationbased detection schemes.

• 전자공학회논문지B
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• 제31B권10호
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• pp.130-139
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• 1994

Video compression algorithms are based on removing spatial and/or temproal redundancy inherent in image sequences by predictive(DPCM) encoding, transform encoding, or a combination of predictive and transform encoding. In this paper, each 8$\times$8 DCT coefficient of DFD(displaced frame difference) is adaptively quantized by one of the four quantizers depending on total distortion level, which is determined by characteristics of HVS(human visual system) and buffer status. Therefore, the number of possible quantizer selection vectors(patterns) is 4$^{64}$. If this vectors are coded, toomany bits are required. Thus, the quantizer selection vectors are limited to 2048 for Y and 512 for each U, V by the proposed method using SWAD(sum of weighted absolute difference) for discriminating vectors. The computer simulation results, using the codebook vectors which are made by the proposed method, show that the subjective and objective image quality (PSNR) are goor with the limited bit allocation. (17Mbps)

• 한국정보통신학회논문지
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• 제15권12호
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• pp.2521-2526
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• 2011

본 논문에서는 퍼셉트론 신경회로망과 선형예측부호화 켑스트럼 계수를 사용한 화자인식 알고리즘을 제안한다. 제안하는 화자인식 알고리즘은 입력받은 음성신호에 대해서 유성음 구간을 추출한다. 추출된 유성음 구간에 대하여 선형예측 분석에 의하여 화자의 특성을 가지고 있는 선형예측부호화 켑스트럼 계수를 구한다. 구해진 선형예측부호화 켑스트럼 계수를 분류하기 위하여 이 켑스트럼 계수를 퍼셉트론 신경회로망의 입력으로 사용하여 네트워크의 학습을 수행한다. 본 실험에서는 선형예측부호화 켑스트럼 계수와 신경회로망을 사용하여 본 화자인식 알고리즘이 유효하다는 것을 인식률을 통하여 확인한다.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송∙미디어공학회 2019년도 추계학술대회
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• pp.22-23
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• 2019

The emerging Versatile Video Coding (VVC) standard introduces Bi-prediction with CU-level Weights (BCW) to enhance the bi-predictive prediction. The syntax element of BCW index is adaptively coded according to the value of NoBackwardPredFlag which indicates if there is no future picture in the display order among the reference pictures, and it can violate the flexibility of codec and cause the dependency issue. This paper proposes BCW clean-up design that allows all weights can be parsed without any condition. The experimental results show negligible BD-rate losses while resolving the issues.

• 전자공학회논문지B
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• 제28B권11호
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• pp.866-875
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• 1991

In this paper, multi pulse excited-linear predictive coding (MPE-LPC), where the correlation eliminated residual signal is modeled by a few pules, is shown to be effective for the compression of electrocardiogram (ECG) data, and a more efficient scheme for a faithful reconstruction of ECG is proposed. The reconstruction charateristic of QRS's and P.T waves is improved using the adaptive pulse allocation (APA), and the compression ratio (CR) can be changed by controlling the mumber of modeling pulses. The performance of the proposed method was evaluated using 10 normal and 10 abnormal ECG data. The proposed method had a better performance than the variable threshold amplitude zone time epoch coding (AZTEC) algorithm and the scan-along polygonal approximation (SAPA) algorithm with the same CR. With the CR in kthe range of 8:1 to 14:1, we could compress ECG data efficiently.