• Journal of Broadcast Engineering
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• v.12 no.6
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• pp.585-595
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• 2007

Distributed video coding is a new video coding paradigm based on Slepian-Wolf and Wyner-Ziv's information theory Distributed video coding whose decoder exploits side information transfers its computational burden from encoder to decoder, so that encoding with light computational power can be realized. RD performance is superior than that of standard video coding without motion compensation process but still has a gap with that of coding with motion compensation process. This parer introduces basic theory of distributed video coding and its structure and then shows RD performances of DVCs whose coding style is different from each other and of a DVC as a spatial scalable video coder.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.12 no.1
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• pp.37-42
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• 2019

In this paper, we propose a syndrome decoding scheme that can correct two errors for single error correcting Hamming codes within a code length. The decoding scheme proposed in this paper has the advantage of significantly improving the error rate performance compared to the decoder complexity by correcting multiple errors without substantially increasing the decoding complexity. It is suitable for applications in which the energy use of encoder/decoder is extremely limited and the low error rate performance is required, such as IoT communications and molecular communications. In order to verify the improvement of the error rate performance of the Hamming code with the proposed decoding scheme, we performed simulation on Hamming codes with short code length in the AWGN and BPSK modulation environments. As a result, compared with the conventional decoding method, the proposed decoding scheme showed performance improvement of about 1.1 ~ 1.2[dB] regardless of the code length of the Hamming code.

• The Journal of the Korea Contents Association
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• v.12 no.8
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• pp.20-28
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• 2012

DVC(Distributed Video Coding) technique is a new paradigm, which is based on the Slepian-Wolf and Wyner-Ziv theorems. DVC offers not only flexible partitioning of the complexity between the encoder and decoder, but also robustness to channel errors due to intrinsic joint source-channel coding. Many conventional research works have been focused on the light video encoder and its rate-distortion performance improvement. However, in this paper, we propose a new DVC codec which is effectively applicable for error-prone environment. The proposed method adopts a quantiser without dead-zone and symmetric Gray code around zero value. Through computer simulations, the proposed method is evaluated by the bit errors position as well as the number of burst bit errors. Additionally, it is shown that the maximum and minimum transmission rate for the given application can be linearly determined by the number of bit errors.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7B
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• pp.1393-1399
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• 1999

In this paper, a VLSI design of the convolutional codec chip of code rate r=l/3, and constraint length K=9 is presented, which is able to correct errors of the received data when transmitted data is corrupted in channels. The circuit design mainly aimed for simple implementation. In the decoder, Viterbi algorithm with 3-bit soft-decision is employed. For information sequence updating and storage, the register exchange method is employed, where the register length is 5$\times$K(45 stages). The codec chip is designed using VHDL language and Design Analyzer and VHDL Simulator of Synopsys are used for simulation and synthesis. The chip is composed of ENCODER block, ALIGN block, BMC block, ACS block, SEL_MIN block and REG_EXCH block. The operation of the codec chip is verified though the logic simulations, where several error conditions are assumed. As a result of the timing simulation after synthesis, the decoding speed of 325.5Kbps is achieved, and 6,894 gates is used.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.63-68
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• 2001

This paper describes the details of ITU-T SGIS G.729A speech coder implementation using ARM9 Thumb/sup R/ processor core and various techniques used in the optimization process. ITU-T G.729 speech coder is the standard of the toll quality 8 kbit/s speech coding. The input to the speech encoder is assumed to be a 16 bits PCM signal at a sampling rate of 8000 samples per second. G.729A is reduced complexity version of the G.729 coder. This version is bit stream interoperable with the full version. The implemented coder requires 34.8 MIPS for the encoder and 8.1 MIPS for the decoder, 36.5 kBytes of program ROM and 6.3 kBytes of data RAM, respectively. The implemented coder is tested against the set of 9 test vectors provided by ITU-T for bit exact implementation.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.7 no.4
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• pp.567-574
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• 2006

This paper focuses on lossy medical image compression methods for medical images that operate on three-dimensional(3-D) irreversible integer wavelet transform. We offer an application of unbalanced tree structure algorithm to medical images, using a 3-D unbalanced wavelet decomposition and a 3-D unbalanced spatial dependence tree. The wavelet decomposition is accomplished with integer wavelet filters implemented with the lifting method. We have tested our encoder on volumetric medical images using different integer filters and 16 coding unit size. The coding unit sizes of 16 slices save considerable dynamic memory(RAM) and coding delay from full sequence coding units used in previous works. If we allow the formation of trees of different lengths, then we can accomodate more transaxial scales than three. Then the encoder and decoder can then keep track of the length of the tree in which each pixel resides through the sequence of decompositions. Results show that, even with these small coding units, our algorithm with I(5,3)filter performs as well and better in lossy coding than previous coding systems using 3-D integer unbalanced wavelet transforms on volumetric medical images.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.1
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• pp.46-56
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• 1991

In this paper, 16word X 8bit Content Addressable and Reentrant Memory(CARM) is described. This device has 4 operation modes(read, write, match, reentrant). The read and write operation of CARM is like that of static RAM, CARM has the reentrant mode operation where the on chip garbage collection is accomplished conditionally. Thus function can be used for high speed matching unit of dynamic data flow computer. And CARM also can encode matching address sequentially according to therir priority. CARM consists of 8 blocks(CAM cell, Sequential Address Encoder(S.A.E). Reentrant operation. Read/Write control circuit, Data/Mask Register, Sense Amplifier, Encoder. Decoder). Designed DARM can be used in data flow computer, pattern, inspection, table look-up, image processing. The simulation is performed using the QUICKSIM logic simulator and Pspice circuit simulator. Having hierarchical structure, the layout was done using the 3{\;}\mu\textrm{m} n well CMOS technology of the ETRI design rule.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1C
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• pp.92-101
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• 2004

This paper proposes a algorithm to extract the variety region (object) from video for the real-time encoding of MPEG-4 based. The previous object segmentation methods cannot used the videophone or videoconference required by real-time processing. It is difficult to transfer a video to real-time because it increased complexity for the operation of each pixel on the spatial segmentation and temporal segmentation method proposed by MPEG-4 Working Group. But algorithm proposed for this thesis not operates a pixel unit but operates a macro block unit. Thus this enables real-time transfer. But this algorithm cannot extract several object for a image using proposed algorithm as previous algorithm. On system constructed by encoder and decoder. A proposed algorithm inserted for encoder as pre-process.

• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.37-44
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• 2000

In this paper, we described the real-time implementation effort of MPEG-4 audio HVXC (Harmonic Vector eXcitation Coding) algorithm for very low bitrates, which has target applications from mobile communications to Internet telephony, on current high performance floating point TMS320C6701 DSP. We adopted a hardware structure for real-time operation. In order for software optimization, we used C- and assembly-language level optimizations for time-critical functional codes. Utilizing the internal program memory of the DSP as the program cache, the internal data memory overlap technique and DMA functionality, we could get a goal of realtime operation of HVXC codec both at 2 kbit/s and at 4 kbit/s. For an encoder at 2 kbit/s, the optimization ratio to original code is about 96 %. Finally, we got the subjective quality of MOS 2.45 at 2 kbit/s from an informal quality test.

• Journal of Korea Multimedia Society
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• v.15 no.8
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• pp.1017-1030
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• 2012

Most watermarking schemes for video contents protection have been studied to increase watermark's robustness and invisibility against such compressions and many kinds of signal processing after embedding copyright information to the original contents. This paper proposes an active watermarking that infect watermark to contents in the video decoding process using embedded infectious watermark and control signals from a video encoder side. To achieve this algorithm, we design a kernel based watermarking in video encoder side that is possible to recover the original contents and watermark in watermark detection procedure perfectly. And then, by reversible de-watermarking in video decoder side, we design the active watermark infection method using detected watermark and control signal. This means that our system can provide secure re-distributions of video contents without any quality degration and watermark bit error against transcoding or re-encoding processing. By experimental results, we confirmed that the embedded watermark was infected by video contents and codec perfectly without any declines of compression ratio and video quality.