• Title/Summary/Keyword: Coding Parameter

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Survey on Nucleotide Encoding Techniques and SVM Kernel Design for Human Splice Site Prediction

  • Bari, A.T.M. Golam;Reaz, Mst. Rokeya;Choi, Ho-Jin;Jeong, Byeong-Soo
    • Interdisciplinary Bio Central
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    • v.4 no.4
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    • pp.14.1-14.6
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    • 2012
  • Splice site prediction in DNA sequence is a basic search problem for finding exon/intron and intron/exon boundaries. Removing introns and then joining the exons together forms the mRNA sequence. These sequences are the input of the translation process. It is a necessary step in the central dogma of molecular biology. The main task of splice site prediction is to find out the exact GT and AG ended sequences. Then it identifies the true and false GT and AG ended sequences among those candidate sequences. In this paper, we survey research works on splice site prediction based on support vector machine (SVM). The basic difference between these research works is nucleotide encoding technique and SVM kernel selection. Some methods encode the DNA sequence in a sparse way whereas others encode in a probabilistic manner. The encoded sequences serve as input of SVM. The task of SVM is to classify them using its learning model. The accuracy of classification largely depends on the proper kernel selection for sequence data as well as a selection of kernel parameter. We observe each encoding technique and classify them according to their similarity. Then we discuss about kernel and their parameter selection. Our survey paper provides a basic understanding of encoding approaches and proper kernel selection of SVM for splice site prediction.

PSNR-based Initial QP Determination for Low Bit Rate Video Coding

  • Park, Sang-Hyun
    • Journal of information and communication convergence engineering
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    • v.10 no.3
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    • pp.315-320
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    • 2012
  • In H.264/AVC, the first frame of a group of pictures (GOP) is encoded in intra mode which generates a large number of bits. The number of bits for the I-frame affects the qualities of the following frames of a GOP since they are encoded using the bits remaining among the bits allocated to the GOP. In addition, the first frame is used for the inter mode encoding of the following frames. Thus, the initial quantization parameter (QP) affects the following frames as well as the first frame. In this paper, an adaptive peak signal to noise ratio (PSNR)-based initial QP determination algorithm is presented. In the proposed algorithm, a novel linear model is established based on the observation of the relation between the initial QPs and PSNRs of frames. Using the linear model and PSNR results of the encoded GOPs, the proposed algorithm accurately estimates the optimal initial QP which maximizes the PSNR of the current GOP. It is shown by experimental results that the proposed algorithm predicts the optimal initial QP accurately and thus achieves better PSNR performance than that of the existing algorithm.

Design and Implementation of Simple Text-to-Speech System using Phoneme Units (음소단위를 이용한 소규모 문자-음성 변환 시스템의 설계 및 구현)

  • Park, Ae-Hee;Yang, Jin-Woo;Kim, Soon-Hyob
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.3
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    • pp.49-60
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    • 1995
  • This paper is a study on the design and implementation of the Korean Text-to-Speech system which is used for a small and simple system. In this paper, a parameter synthesis method is chosen for speech syntheiss method, we use PARCOR(PARtial autoCORrelation) coefficient which is one of the LPC analysis. And we use phoneme for synthesis unit which is the basic unit for speech synthesis. We use PARCOR, pitch, amplitude as synthesis parameter of voice, we use residual signal, PARCOR coefficients as synthesis parameter of unvoice. In this paper, we could obtain the 60% intelligibility by using the residual signal as excitation signal of unvoiced sound. The result of synthesis experiment, synthesis of a word unit is available. The controlling of phoneme duration is necessary for synthesizing of a sentence unit. For setting up the synthesis system, PC 486, a 70[Hz]-4.5[KHz] band pass filter for speech input/output, amplifier, and TMS320C30 DSP board was used.

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Equal Bit Rate Control for Low Bit-rate Coder based on Frame Statistics (저 전송률 부호화기를 위한 프레임 특성에 근간한 균등 비트 할당 기법)

  • Seo Dong-Wan;Choe Yoon-Sik
    • Journal of the Institute of Convergence Signal Processing
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    • v.6 no.4
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    • pp.176-181
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    • 2005
  • This paper presents an equal bit rate control algorithm utilizing the statistical change between the previous frame and the current frame. The previous studies on the model-based rate control have focused on the models of bit rate and distortion in types of coders, in terms of the quantization parameter. The proposed algorithm improves the typical model-based rate control by updating a model parameter instead of modeling a better model of the rate and distortion. The proposed algorithm updates this model parameter by recognizing the change in statistics between the previous frame and the current frame. We implement the proposed algorithm in MPEG-4 coders and verify its performance while comparing it to the TMN8's approach (up to 0.6dB of improvement).

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Frame-Layer H.264 Rate Control for Scene-Change Video at Low Bit Rate (저 비트율 장면 전환 영상에 대한 향상된 H.264 프레임 단위 데이터율 제어 알고리즘)

  • Lee, Chang-Hyun;Jung, Yun-Ho;Kim, Jae-Seok
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.44 no.11
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    • pp.127-136
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    • 2007
  • An abrupt scene-change frame is one that is hardly correlated with the previous frames. In that case, because an intra-coded frame has less distortion than an inter-coded one, almost all macroblocks are encoded in intra mode. This breaks up the rate control flow and increases the number of bits used. Since the reference software for H.264 takes no special action for a scene-change frame, several studies have been conducted to solve the problem using the quadratic R-D model. However, since this model is more suitable for inter frames, the existing schemes are unsuitable for computing the QP of the scene-change intra frame. In this paper, an improved rate control scheme accounting for the characteristics of intra coding is proposed for scene-change frames. The proposed scheme was validated using 16 test sequences. The results showed that the proposed scheme performed better than the existing H.264 rate control schemes. The PSNR was improved by an average of 0.4-0.6 dB and a maximum of 1.1-1.6 dB. The PSNR fluctuation was also in proved by an average of 18.6 %.

A LSF Quantizer for the Wideband Speech Using the Predictive VQ-Pyramid VQ (예측 VQ-Pyramid VQ를 이용한 광대역 음성용 LSF 양자학기 설계)

  • 이강은;이인성;강상원
    • 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.

Analysis of Quantization Parameter of Key Pictures in Distributed Video Coding (분산비디오 기술의 율 왜곡 성능 개선을 위한 키 픽처의 양자화 계수 분석)

  • Eun, Hyun;Shim, Hiuk Jae;Jeon, Byeungwoo
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2010.11a
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    • pp.239-241
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    • 2010
  • 분산 비디오 기술의 대표적인 기술 중 하나는 와이너 지브 부호화 기술이다. 와이너 지브 부호화 구조에서 보조정보는 인트라 복호화된 키 픽처들을 이용하여 생성한다. 키 픽처의 객관적 화질은 보조정보의 성능에 많은 영향을 끼치고, 잡음이 많은 보조정보를 복호화에 이용할 경우 부호화로부터 많은 패리티 비트를 요구하게 되어 율 왜곡 성능을 저하된다. 기존의 부호화 기술은 키 픽처 부호화 시 Quantization Matrix에 따라 미리 정의된 양자화 계수를 이용한다. 본 논문에서는 미리 정의된 양자화 계수 보다 낮은 계수 값을 사용하여 부호화 하는 방법을 제안한다. 제안방법은 키 픽처의 객관적 화질이 높아짐에 따라 보조정보의 화질을 향상시킨다. 잡음이 적은 보조정보는 와이너 지브 복호화 시 율 왜곡 성능을 향상시킨다. 실험결과는 기존 방법에 비해 최대 0.7dB에 이르는 성능향상을 보인다.

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An Accurate Estimation of Channel Loss Threshold Set for Optimal FEC Code Rate Decision (최적의 FEC 부호율 결정을 위한 정확한 채널손실 한계집합 추정기법)

  • Jung, Tae-Jun;Jeong, Yo-Won;Seo, Kwang-Deok
    • Journal of Broadcast Engineering
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    • v.19 no.2
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    • pp.268-271
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    • 2014
  • Conventional forward error correction (FEC) code rate decision schemes using analytical source coding distortion model and channel-induced distortion model are usually complex, and require the typical process of model parameter training which involves potentially high computational complexity and implementation cost. To avoid the complex modeling procedure, we propose a simple but accurate joint source-channel distortion model to estimate channel loss threshold set for optimal FEC code rate decision.

Spoken digit recognition Using the ZCR and PARCOR Coefficient (ZCR과 PARCOR 계수를 이용한 숫자음성 인식)

  • 김학윤
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1985.10a
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    • pp.75-78
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    • 1985
  • 본 연구는 시간 영역의 parament를 이용하여 한국어 숫자음(영, 일, 이, 삼, 사, 오, 육, 칠, 팔, 구)을 인식했다. 입력 음성 신호 X(n)의 Beginning Point와 Ending point를 ZCR(Zero-crossing Rate), Magnitude, Energy, Autocorrelation을 이용 Beginning point와 Ending point를 구하고 자음부의 인식은 위 계수들을 이용하여 행했다. 또, 유성음 부분에서는 PARCOR(Partial Autocorrelation), LPC(Linear Predictive Coding)를 이용 모음부와 유성자음을 인식하여 모음을 6개 부류(ㅏ, ㅑ, ㅗ, ㅜ, ㅠ, ㅣ)로 구분 인식했다. 이 방법에 의하면 입력 음성 신호 X(n)의 B.P(Beginning Point)와 E.P(Ending Point)를 쉽게 추출 가능하며 또한 각 Parameter를 이용하여 94.4%의 인식율을 얻었다.

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Deblocking Method Based on Mode Classification in the Low Bit-Rate Real-Time Video Coding (저비트율 실시간 비디오 압축에서의 모드구분에 기반한 블록킹 효과 제거 기법)

  • 이웅호;정동석
    • Proceedings of the IEEK Conference
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    • 2001.09a
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    • pp.723-726
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    • 2001
  • 일반적으로 블록기반의 동영상 압축방식은 블록킹효과를 필연적으로 수반한다. 특히 저비트율의 동영상에서는 블록킹 효과가 다른 어떤 영상의 왜곡보다 많이 발생한다. 본 논문에서는 이러한 블록킹효과를 효율적으로 인간 시각체계에 적합하게 실시간으로 제거하는 후처리 알고리즘을 제안한다. 우선 복원된 영상에서 인간의 시각체계와 동영상의 특성에 따라 3가지의 모드로 분리하여 QP(quantization Parameter)에 따라 임계치를 변화함으로써 각 모드의 필터링 범위를 가변시켰다. 이후에 각 모드에 알맞은 일차원 및 적응형 필터링을 적용한다. 적용된 모드별 필터링은 과도한 블러링 현상을 방지하고 영상내의 실제 에지성분읓 보호하면서 효과적으로 블록킹효과를 제거한다. 본 논문에서 제안하는 알고리즘을 실험 영상에 적용하였을 경우에 주관적 화질 및 객관적 화질인 PSNR로 0.5dB 정도 향상되었다.

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