• The Journal of Korean Institute of Communications and Information Sciences
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• v.39A no.4
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• pp.172-179
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• 2014

HEVC (High Efficiency Video Coding) fine-tuned many existing coding tools and adopted also many new coding techniques. As a result, HEVC has accomplished about 2 times of compression efficiency enhancement compared to the existing video coding standard of H.264/AVC. One of the newly adopted tools in HEVC is the transform skip scheme which performs quantization without transform. This technique improves coding efficiency especially with computer-generated images. However, the unavailability of global or local properties of general video signals demands encoder to decide whether performing transform or not for each TU (Transform Unit). The necessity of computing rate-distortion costs for this decision is one reason to increase encoder complexity. In this paper, a fast transform skip mode decision method is proposed, which is based on the fast decision of rate-distortion cost calculation for transform skip mode, by considering frequency characteristics of residual signal. The proposed method can reduce $4{\times}4$ TU encoding time by about 27.1% with only about 0.03% consequential decrement in BDBR.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.2
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• pp.30-38
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• 2002

In this paper, we propose a new method for image coding which efficiently use the relationship between the properties of spatial image and its wavelet transform. Firstly, an original image is decomposed into several layers by the wavelet transform, and simultaneously decomposed into 2$^n$$\times$2$^n$blocks. Each block is classified into two regions according to their standard deviation, i.e., low activity region(LAR) and high activity region(HAR). The region with low frequency in spatial domain does not only appears as zero regions in wavelet frequency domain like HL, LH, and HH but also gives little influence to the quality of reconstructed image. The other side, the high frequency regions are related to significant coefficients which gives much influence to image reconstruction. In this paper, we propose a image coding method to obtain high compression rate at low bit rate by these properties. The LAR region is encoded by LAR coding method which is proposed in this paper, the HAR by a technique similar to bitplane coding in hierarchical tree. Simulation results show that th,$\boxUl$ proposed coding method has better performance than EZW and SPIHT schemes in terms of image quality and transmitted bit rates, can be successfully applied to the application areas that require of progressive transmission.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.3
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• pp.213-224
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• 1992

An encoding technique based on region splitting and vector quantization is proposed for the lower level Laplacian pyramid images. The lower level Laplacian pyramid images have lower variance than higher levels but a great influence on compression ration due to large spatial area. And so from data compression viewpoint, we subdivide them with variance thresholding into two regions such as one called : flat region” and the other “edge region”, and encode the flat region with its mean value and the edge region as vector quantization method. The edge region can be reproduced faithfully and significant improvement on compression ratio can be accomplished with a little degradation of PSNR in spite of the effect of large flat region since the codebook used is generated from the edge region only on from the entire image including the flat region. It can be verified by computer simulation results that proposed method is more efficient in compression ratio and processing time than the conventional encoding technique of vector quantization.

• Journal of KIISE:Software and Applications
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• v.33 no.4
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• pp.411-417
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• 2006

Motion Estimation (ME) is an important part of video coding process and it takes the largest amount of computation in video compression. Half-pixel and quarter-pixel motion estimation can improve the video compression rate at the cost of higher computational complexity In this paper, we suggest a new efficient low-complexity algorithm for half-pixel and quarter pixel motion estimation. It is based on the experimental results that the sum of absolute differences(SAD) shows parabolic shape and thus can be approximated by using interpolation techniques. The sub-pixel motion vector is searched from the minimum SAD integer-pixel motion vector. The sub-pixel search direction is determined toward the neighboring pixel with the lowest SAD among 8 neighbors. Experimental results show that more than 20% reduction in computation time can be achieved without affecting the quality of video.

• The Journal of the Acoustical Society of Korea
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• v.11 no.4
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• pp.36-44
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• 1992

To reduce the redundancy within samples that resulted from uniform sampling method, nonuniform sampling or nonredundant-sample coding methods can be considered. But it is well-known that when conventional nonuniform sampling methods are applied directly to speech signal, the amount of data required is comparable to or more than that required by uniform sampling method like PCM. To overcome this problem, we consider properties of speech signal in the sense of perception, and suggest a nonuniform sampling method at the maxima and minima of speech wave. To analyze the performance of the suggested method, compression ratio is considered. We show that compression ratio can be improved by silence detection, which can't be implemented by conventional methods based on uniform sampling. As experimental results, compression ratios of 1.54 without silence detection and 2.88 with silence detection for 8kHz 8-bit PCM signals are obtained.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.7C
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• pp.658-665
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• 2005

Re-assignment scheme of index in index image is called reindexing. It has been well known that index image can be reindexed without losslessness. In this paper, we introduces an adaptive rank based reindexing scheme using co-occurrence frequency between neighboring pixels. Original index image can be converted into rank image by the proposed scheme. Using the proposed scheme, a better compression efficiency can be expected because most of the reindexed values(rank) get distributed with a smaller variance than the original index image. Experinental results show that the proposed scheme achieves a much better compression performance over GIF, arithmetic coding, Zeng's algorithm and RIAC scheme.

• Journal of Broadcast Engineering
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• v.27 no.3
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• pp.318-331
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• 2022

With the recent development of hardware computing devices and software based frameworks, machine tasks using deep learning networks are expected to be utilized in various industrial fields and personal IoT devices. However, in order to overcome the limitations of high cost device for utilizing the deep learning network and that the user may not receive the results requested when only the machine task results are transmitted from the server, Collaborative Intelligence (CI) proposed the transmission of feature maps as a solution. In this paper, an efficient compression method for feature maps with vast data sizes to support the CI paradigm was analyzed and presented through experiments. This method increases redundancy by applying feature map reordering to improve compression efficiency in traditional video codecs, and proposes a feature map method that improves compression efficiency and maintains the performance of machine tasks by simultaneously utilizing image compression format and video compression format. As a result of the experiment, the proposed method shows 14.29% gain in BD-rate of BPP and mAP compared to the feature compression anchor of MPEG-VCM.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.17 no.7
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• pp.49-53
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• 2016

Recently, the use of signal compression methods to improve the efficiency of wireless networks have increased. In particular, the MPC system was used in the pitch extraction method and the excitation source of voiced and unvoiced to reduce the bit rate. In general, the MPC system using an excitation source of voiced and unvoiced would result in a distortion of the synthesis speech waveform in the case of voiced and unvoiced consonants in a frame. This is caused by normalization of the synthesis speech waveform in the process of restoring the multi-pulses of the representation segment. This paper presents an ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform. The experiments were performed with 16 sentences of male and female voices. The voice signal was A/D converted to 10kHz 12bit. In addition, the ACFBD-MPC system was realized and the SNR of the ACFBD-MPC estimated in the coding condition of 8kbps. As a result, the SNR of ACFBD-MPC was 13.6dB for the female voice and 14.2dB for the male voice. The ACFBD-MPC improved the male and female voice by 1 dB and 0.9 dB, respectively, compared to the traditional MPC. This method is expected to be used for cellular telephones and smartphones using the excitation source with a low bit rate.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.3
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• pp.37-43
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• 2009

The new still image compression standard JPEG2000 is consisted of DWT and EBCOT. In this paper, proposed and designed new algorithm in efficient EBCOT. BPC based on the contort. Proposed BPC Algorithm is forecasted coding pass using Sigstage, column, mpass value. BAC design apply 4-pipeline stage. EBCOT designed using Verilog HDL. Verification and Synthesis using Xillinx FPGA technology.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.64-70
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• 1997

This paper is aimed to propose a new wavelet audio signal coding scheme which reduces the complexity of well-known MPEG(Moving Picture Expert Group)-Audio. The filters of MPEG0audio apply subband technique on the 16-bits PCM audio to aquire bitstream of subband sample using dynamic bit allocation. If we use the wavelet coefficients instead of subband samples and 6 bands which is less than 32 bands of MPEG-audio, the complexity can be reduced. A new audio signal compression algorithm in this paper is based on wavelet transform and the proposed algorithm is compared with MPEG-audio. At the bitrate of 256kbps, the proposed algorithm maintains the CD(Compact-disc) quality. We were able to reduce the about 40% of complexity at encoder and about 70% at decoder.