• Title/Summary/Keyword: Codec method

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Similar Movie Retrieval using Low Peak Feature and Image Color (Low Peak Feature와 영상 Color를 이용한 유사 동영상 검색)

  • Chung, Myoung-Beom;Ko, Il-Ju
    • Journal of the Korea Society of Computer and Information
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    • v.14 no.8
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    • pp.51-58
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    • 2009
  • In this paper. we propose search algorithm using Low Peak Feature of audio and image color value by which similar movies can be identified. Combing through entire video files for the purpose of recognizing and retrieving matching movies requires much time and memory space. Moreover, these methods still share a critical problem of erroneously recognizing as being different matching videos that have been altered only in resolution or converted merely with a different codec. Thus we present here a similar-video-retrieval method that relies on analysis of audio patterns, whose peak features are not greatly affected by changes in the resolution or codec used and image color values. which are used for similarity comparison. The method showed a 97.7% search success rate, given a set of 2,000 video files whose audio-bit-rate had been altered or were purposefully written in a different codec.

A Bus Data Compression Method on a Phase-Based On-Chip Bus

  • Lee, Jae-Sung
    • JSTS:Journal of Semiconductor Technology and Science
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    • v.12 no.2
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    • pp.117-126
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    • 2012
  • This paper provides a method for compression transmission of on-chip bus data. As the data traffic on on-chip buses is rapidly increasing with enlarged video resolutions, many video processor chips suffer from a lack of bus bandwidth and their IP cores have to wait for a longer time to get a bus grant. In multimedia data such as images and video, the adjacent data signals very often have little or no difference between them. Taking advantage of this point, this paper develops a simple bus data compression method to improve the chip performance and presents its hardware implementation. The method is applied to a Video Codec - 1 (VC-1) decoder chip and reduces the processing time of one macro-block by 13.6% and 10.3% for SD and HD videos, respectively

Development of A Lossless Video Coding System Using Motion Compensation

  • Naoki Ono;Satoshi Ishbashi;Naoki Kobayashi
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 1998.06b
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    • pp.177-182
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    • 1998
  • In this paper, we propose a lossless motion video coding method using motion compensation. For realtime coding and transmission, we developed a lossless video codec based on the proposed method. It was confirmed the codec functions normally in an experiment involving coding and transmitting through an actual ATM network. Furthermore, we proposed a lossless coding method with bit-plane expansion under a constant bitrate. Two approaches, a closed bit-plane approach and a merged bit-plane approach, are considered and characteristics of their compression efficiency are estimated. Simulation results show that the proposed method is suitable for lossless video coding with bitrate control.

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Implementation of JBIG2 CODEC with Effective Document Segmentation (문서의 효율적 영역 분할과 JBIG2 CODEC의 구현)

  • 백옥규;김현민;고형화
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.27 no.6A
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    • pp.575-583
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    • 2002
  • JBIG2 is an International Standard fur compression of Bi-level images and documents. JBIG2 supports three encoding modes for high compression according to region features of documents. One of which is generic region coding for bitmap coding. The basic bitmap coder is either MMR or arithmetic coding. Pattern matching coding method is used for text region, and halftone pattern coding is used for halftone region. In this paper, a document is segmented into line-art, halftone and text region for JBIG2 encoding and JBIG2 CODEC is implemented. For efficient region segmentation of documents, region segmentation method using wavelet coefficient is applied with existing boundary extraction technique. In case of facsimile test image(IEEE-167a), there is improvement in compression ratio of about 2% and enhancement of subjective quality. Also, we propose arbitrary shape halftone region coding, which improves subjective quality in talc neighboring text of halftone region.

Study on Performance Improvement of Video in the H.264 Codec (H.264 코덱에서 동영상 성능개선 연구)

  • Bong, Jeong-Sik;Jeon, Joon-Hyeon
    • Proceedings of the KIEE Conference
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    • 2005.10b
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    • pp.532-535
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    • 2005
  • These days, many image processing techniques have been studied for effective image compression. Among those, 2D image filtering is widely used for 2D image processing. The 2D image filtering can be implemented by performing ID linear filtering separately in the direction of horizontal and vertical. Efficiency of image compression depends on what filtering method is used. Generally, circular convolution is widely used in the 2D image filtering for image processing. However it doesn't consider correlations at the region of image boundary, therefore filtering can not be performed effectively. To solve this problem. I proposed new convolution technique using Symmetric-Mirroring convolution, satisfying the 'alias-free' and 'error-free' requirement in the reconstructed image. This method could provide more effective performance than former compression methods. Because it used very high correlative data when performed at the boundary region. In this paper, pre-processing filtering in H.264 codec was adopted to analyze efficiency of proposed filtering technique, and the simulator developed by Matlab language was used to examine the performance of the proposed method.

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Signal Enhancement of a Variable Rate Vocoder with a Hybrid domain SNR Estimator

  • Park, Hyung Woo
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.13 no.2
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    • pp.962-977
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    • 2019
  • The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.

A Novel Method for Bitrate Control within Macroblocks Using Kalman and FIR Filters

  • Seok, Jin-Wuk;Yoon, Ki-Song;Kim, Bum-Ho;Lee, Jeong-Woo
    • ETRI Journal
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    • v.33 no.4
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    • pp.641-644
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    • 2011
  • In this letter, we propose a novel bitrate control, using both Kalman and FIR filters, based on a Hamiltonian analysis with respect to the amount of bits from each macroblock, in an encoding of a general video codec such as H.264/AVC. Since the proposed bitrate control is based on the simple computation of an optimal control method based on the Hamiltonian analysis, it is not necessary to use additional computation, such as a DCT or quantization, to estimate the bits for bitrate control. As a result, the proposed algorithm can be applied to single-pass encoding and can provide sufficient encoding speed with respect to various applications, even those requiring real-time control.

A Study of Multi-Channel Video Transfer System with EBCOT (EBCOT를 이용한 다 채널 영상 전송 시스템에 대한 연구)

  • 추연학;김영민
    • Journal of Korea Multimedia Society
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    • v.4 no.1
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    • pp.75-81
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    • 2001
  • A EBCOT(Embedded Block Coding with Optimized Truncation) is image compression codec using in JPEG2000, currently the new standard for still image coding. this paper proposes multi-channel video transfer system with EBCOT using a single codec to transfer video to difference band-width channel. This parer testify that compression rate of EBCOT higher than ordinary VLC using RLC and Huffman codec and apply EBCOT to JPEG structure. this structure increases parallelism and error resilience using black coding method. finally it looks into difficult to apply MPEG structure to multi channel video transfer system, and proposes solution using EBCOT.

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A Performance Assessment of Real-time Multichannel Audio Codec

  • Kim, Sunghan;Jang, Daeyoung;Hong, Jinwoo
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.3E
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    • pp.56-61
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    • 1997
  • In this paper, we describe a real-time implementation of a multi-channel auido codec system that is based on the MPEG-1 audio algorithm. The major feature of this system is that it has a flexible multi-DSP system that can be adapted for various applications with using up to four TMS320C40 DSPs. The purpose of this paper is to present the problems of the system and is to describe the optimized methods to solve the problems in the view of hardware and software. Our audio codec is composed of an encoder an a decoder system and the bit rate of bitstream is up to 384 kbps. Fast input/output interfaces, DSP overloads, and inter-DSP communications methods with high speed are considered in multi-DSP H/W. Also, to run real-time in S/W, optimizing methods of algorithm are considered. After implementation of system, the subjective assessment method, and 'triple stimulus/hidden reference/double blind' that recommended by ITU-R TG10/3 is adopted for the quality of our system. All test items except one are awarded difference grades(diffgrade) better than 1-. Form the results, multi-channel audio system can be used for HDTV service.

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A Design for Extension Codec based on Legacy Codec (레거시 코덱 기반 확장 코덱 설계)

  • Young, Su Heo;Bang, Gun;Park, Gwang Hoon
    • Journal of Broadcast Engineering
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    • v.20 no.4
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    • pp.509-520
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    • 2015
  • A design for the merge mode of three dimensional High Efficiency Video Coding (3D-HEVC) is proposed in this paper. The proposed design can reduce the implementation complexity by removing the duplicated modules of the HEVC. For the extension codec, the implementation complexity is as crucial as coding efficiency, meaning if possible, extension codec needs to be easily implemented through by reusing the design of the legacy codec as-is. However, the existing merging process of 3D-HEVC had been built-in integrated in the inside of the HEVC merging process. Thus the duplicated merging process of HEVC had to be fully re-implemented in the 3D-HEVC. Consequently the implementation complexity of the extension codec was very high. The proposed 3D-HEVC merge mode is divided into following two stages; the process to reuse the HEVC modules without any modification; and the reprocessing process for newly added and modified merging modules in 3D-HEVC. By applying the proposed method, the re-implemented HEVC modules, which accounted for 51.4% of 3D-HEVC merge mode confirmed through the operational analysis of algorithm, can be eliminated, while maintaining the same coding efficiency and computational complexity.