• The Journal of the Korea Contents Association
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• v.9 no.11
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• pp.10-17
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• 2009

Wyner-Ziv video coding is a new video compression paradigm based on distributed source coding theory of Slepian-Wolf and Wyner-Ziv. Wyner-Ziv coding enables light-encoder/heavy-decoder structure by shifting complex modules including motion estimation/compensation task to the decoder. Instead of performing the complicated motion estimation process in the encoder, the Wyner-Ziv decoder performs the motion estimation for the generation of side information in order to make the predicted signal of the Wyner-Ziv frame. The efficiency of side information generation deeply affects the overall coding performance, since the bit-rates of the Wyner-Ziv coding is directly dependent on side information. In this paper, an improved side information generation method using field coding is proposed. In the proposed method, top fields are coded with the existing SI generation method and bottom fields are coded with new SI generation method using the information of the top fields. Simulation results show that the proposed method improves the quality of the side information and rate-distortion performance compared to the conventional method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.11
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• pp.2412-2418
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• 2009

As an astonishing progress of FTTH infrastructure, the new technologies have been widely studied to use the tantalizing benefits of high bandwidth in fiber optic cable. In this paper, a new VONU is presented to perform all necessary optical functions. It can converts digital and analog CATV signals and satellite-based signal transmitted via one fiber optic cable to electrical signals (electric lights). However, most previous VONU systems have the problems such as interference between difference services, signal distortion, and noise increasing rate. These problems cause the quality deterioration in broadcasting. Therefore, we suggest the new VONU system to solve all problems listed above. In addition, we show that how our system performs well by measuring the real data with implemented system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7A
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• pp.529-536
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• 2003

In this Paper, a new DCT HLMS DFE(Discrete Cosine Transform Hierarchical Least Mean Square Decision Feedback Equalizer) algorithm is proposed to improve the convergence speed and MSE(Mean Square Error) performance of a receive channel equalizer in ATSC(Advanced Television System Committee) 8VSB(Vestigial Side Band) digital terrestrial TV system. The proposed algorithm reduces the eigenvalue range of input data autocorrelation by transforming LMS (Least Mean Square) DFE into the subfilter of hierarchical structure. Moreover, the use of DCT and power estimation algorithm makes it possible to reduce the eigenvalue deviation of input data which results from distortion and delay of the receive signal in the miulti-path environment. Simulation results show that proposed DCT HLMS DFE has SNR improvement of approximately 3.8dB, 5dB and 2dB as compared to LMS DFE when the equalized symbol error rate is 0.2 in ATTC defined digital terrestrial TV broadcasting channels A, B and F, respectively.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.6C
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• pp.585-593
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• 2007

For the improvement of coding efficiency, the H.264 standard uses new coding tools such as VBS, 1/4-pel accurate ME, multiple references, intra prediction, loop filter, etc. Using these coding tools, H.264 has achieved significant improvements from rate-distortion point of view compared to existing standards. However, the encoder complexity is greatly increased due to these coding tools. We focus on the complexity reduction method of intra-mode decision. Our algorithm first restricts selective prediction modes of Intra4x4 using a simple preprocessing. The prediction modes of Intra4x4 are used for restricting those of the other inter-modes. Simulation results show that the proposed method outperforms other conventional methods and save about 82% of total encoding time.

• The KIPS Transactions:PartB
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• v.19B no.1
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• pp.9-18
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• 2012

Many audio watermarking researches which have been adapted to authenticate contents can not recover the original media after watermark removal. Therefore, reversible watermarking can be regarded as an effective method to ensure the integrity of audio data in the applications requiring high-confidential audio contents. Reversible watermarking inserts watermark into digital media in such a way that perceptual transparency is preserved, which enables the restoration of the original media from the watermarked one without any loss of media quality. This paper presents a new interval-based audio integrity authentication algorithm which can detect malicious tampering. To provide complete reversibility, we used differential histogram-based reversible watermarking. To authenticate audio in parts, not the entire audio at once, the proposed algorithm processes audio by dividing into intervals and the confirmation of the authentication is carried out in each interval. Through experiments using multiple kinds of test data, we prove that the presented algorithm provides over 99% authenticating rate, complete reversibility, and higher perceptual quality, while maintaining the induced-distortion low.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.612-621
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• 2003

As the use of MP3 audio compression increased, the demand for the insertion of additive data about copyright or information on music contents has been groved and the related research has been progressed actively. When an additive data is inserted into MP3 bitstream, it should not to happen any distortion of music quality or the change of file size, due to the modification of MP3 bitstream structure. In our study, to make these conditions satisfied, we inserted some additive data to bitstream by modifying some bits of linbits among the quantized integer coefficients having big values. At this time, we consider the characteristics of linbits and their distributions. As a result of subjective sound quality test through MOS test, we confirmed that the quality of MOS 4.6 can be achieved at the data insertion rate of 60 bytes/sec. Using the proposed method, it is possible to effectively insert an additive data such as copyright information or information about media itself, so that various applications like audio database management can be realized.

• Journal of the Korea Society of Computer and Information
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• v.15 no.10
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• pp.49-57
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• 2010

In this thesis, We studied on hand-written character recognition, that characters entered into a digital input device and remove noise and separating character elements using preprocessing. And processed character images has done thinning and 3-level wavelet transform for making normalized image and reducing image data. The structural method among the numerical Hangul recognition methods are suitable for recognition of printed or hand-written characters because it is usefull method deal with distortion. so that method are applied to separating elements and analysing texture. The results show that recognition by analysing texture is easily distinguished with respect to consonants. But hand-written characters are tend to decreasing successful recognition rate for the difficulty of extraction process of the starting point, of interconnection of each elements, of mis-recognition from vanishing at the thinning process, and complexity of character combinations. Some characters associated with the separation process is more complicated and sometime impossible to separating elements. However, analysis texture of the proposed character recognition with the exception of the complex handwritten is aware of the character.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.37 no.4
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• pp.35-45
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• 2000

The vertex-based boundary encoding scheme is widely used in object-based video coding area and computer graphics due to its scalability with natural looking approximation. Existing single framebased vertex encoding algorithm is not efficient for temporally correlated video sequences because it does not remove temporal redundancy. In the proposed method, a vertex point is selected from not only the boundary points of the current frame but also the vertex points of the previous frame to remove temporal redundancy of shape information in video sequences. The problem of selecting optimal vertex points is modeled as finding shortest path in the directed acyclic graph with weight The boundary is approximated by a polygon which can be encoded with the smallest number of bits for maximum distortion. The temporal redundancy between two successive frames is efficiently removed with the proposed scheme, resulting in lower bit-rate than the conventional algorithms.

• JSTS:Journal of Semiconductor Technology and Science
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• v.17 no.1
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• pp.129-140
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• 2017

In this paper, a $4{\times}32$-channel neural recording system capable of acquiring neural signals is introduced. Four 32-channel neural recording ICs, complex programmable logic devices (CPLDs), a micro controller unit (MCU) with USB interface, and a PC are used. Each neural recording IC, implemented in $0.18{\mu}m$ CMOS technology, includes 32 channels of analog front-ends (AFEs), a 32-to-1 analog multiplexer, and an analog-to-digital converter (ADC). The mid-band gain of the AFE is adjustable in four steps, and have a tunable bandwidth. The AFE has a mid-band gain of 54.5 dB to 65.7 dB and a bandwidth of 35.3 Hz to 5.8 kHz. The high-pass cutoff frequency of the AFE varies from 18.6 Hz to 154.7 Hz. The input-referred noise (IRN) of the AFE is $10.2{\mu}V_{rms}$. A high-resolution, low-power ADC with a high conversion speed achieves a signal-to-noise and distortion ratio (SNDR) of 50.63 dB and a spurious-free dynamic range (SFDR) of 63.88 dB, at a sampling-rate of 2.5 MS/s. The effectiveness of our neural recording system is validated in in-vivo recording of the primary somatosensory cortex of a rat.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.5
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• pp.69-75
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• 2010

Advanced T-DMB(Terrestial DMB )system which is a new portable mobile broadcasting system has been developed to increase data rate up to double of conventional T-DMB in same bandwidth while maintaining backward compatibility, using hierarchical modulation method. The Advanced T-DMB system realize high qualification of conventional T-DMB system by adding BPSK signal or QPSK signal as enhanced layer to existing DQPSK signal. The enhanced layer signal should be small enough to maintain backward compatibility and to minimize the coverage loss of existing T-DMB service area. But this causes the enhanced layer signal of Advanced T-DMB susceptible to fading effect in transmission channel. In this paper we applied the duo-binary turbo code which has powerful error correction capability to the enhanced layer signal for compensating channel distortion. And the computer simulation results about the performance of the duo-binary turbo code in Advanced T-DMB system are presented along with analysis comments.