• Title/Summary/Keyword: signal compression method

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Noise additived image encoding By EZW algorithm (EZW를 이용한 잡음 영상의 부호화)

  • 김형준;김재필;김향진;김영애;임재윤
    • Proceedings of the IEEK Conference
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    • 2000.06d
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    • pp.27-30
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    • 2000
  • In this paper, we propose new method for denoising in processing the image compression. Usually, to compress the noise image, we must have the denoising step before encoding. But this method has a embedded character, so need not an additional noise eliminator. In SAQ step, an embedded signal is quantized more detail and the other side is suppressed. Comparing with the conventional method, we can get the enhanced image quality.

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An Implementation of FM-extended Stereo System via Application of Quadrature Modulation and Companding Method (직교 변조 및 압신 방식을 이용한 확장형 FM 스테레오 방식의 구현)

  • Heo, Dong-Kyu;Kim, Kee-Keun;Kim, Ju-Koang;Ryu, Heung-Gyoon
    • The Journal of the Acoustical Society of Korea
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    • v.10 no.2
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    • pp.44-51
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    • 1991
  • In this paper, we have studied the FM-extended broadcasting which makes the enhancement of stereo signal-to-noise ration by utilizing the compression technique and quadrature moduation method of standard subcarrier signal in transmitter station, and corresponding demodulation method and expansion technique in receiver system. The proposed system is completely compatible with the conventional FM stereo system. We have confirmed the validity of the proposed method by implementation of transmitter/receiver system including quadrature modulation/demodulation and companding system.

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Methods of Knock Signal Analysis in a S.I. Engine (4 기통 스파크 점화 기관의 노킹 신호 해석 방법)

  • Kim, K.W.;Chun, K.M.
    • Transactions of the Korean Society of Automotive Engineers
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    • v.1 no.3
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    • pp.12-21
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    • 1993
  • In recent years, high efficiency, high performance, and low pollutant emmision engines have been developed. Knock phenomenon has drawn interests because it became an hinderance to engine power and efficiency increase through higher compression ratio. Knock phenomenon is an abnormal combustion originated from autoignition of unburned gas in the end-gas region during the later stage of combustion process and accompanied a high pitched metallic noise. And this phenomenon is characterized by knock occurrence percentage, knock occurrence angle and knock intensity. A four cylinder spark ignition engine is used in our experiment, and its combustion chamber pressure is measured at various engine speeds, ignition timing. The data are analyzed by numerous methods in order to select the optimum methods and to achieve better understanding of knock characteristics. Methods using band-pass filter, third derivative and step method are shown to be the most suitable, while methods using frequency analysis are shown to be unsuitable. Because step method only uses signals above threshold value during knocking condition, pressure signal analyses with this method show good signal-to-noise ratio.

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Design of FPGA-based Signal Processing of EWRG for Localized Heavy Rainfall Observation (국지성 호우 관측을 위한 FPGA 기반의 전파강수계 신호처리 설계)

  • Choi, Jeong-Ho;Lee, Bae-Kyu;Park, Hyeong-Sam;Park, Jeong-Min;Lim, Sang-Hun
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.24 no.9
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    • pp.1215-1223
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    • 2020
  • Recently, the number of natural disasters caused by inclement weather conditions such as localized heavy rainfall, Typhoon, etc. is increasing in Korea, which requires relevant prevention and water management measures. Rain gauges installed on the ground have strengths in continuously·directly measures ground precipitation but cannot provide accurate information on spatial precipitation distribution in the areas without the rain gauges. The present research has designed and developed an electromagnetic-based multi-purpose precipitation gauge(EWRG, Electromagnetic Wave Rain Gauge) that can measure rainfall at the real time, by overcoming spatial representativeness. In this paper, we propose an FPGA-based signal processing design method for EWRG. The signal processing of the EWRG was largely designed by calculating the ADC and DDC of the LFM waveform, pulse compression, correlation coefficient and estimating the precipitation parameter. In this study, the LFM waveform and pulse compressed signal were theoretically analyzed.

Robust Digital Image Watermarking Based on MTF of HVS (인간 시각의 MTF에 기반한 견고한 디지털 영상 워터마킹)

  • 홍수기;조상현;최흥문
    • Proceedings of the IEEK Conference
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    • 2000.06d
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    • pp.114-117
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    • 2000
  • In this paper, we proposed robust digital image watermarking based on modulation transfer function (MTF) of human visual system (HVS). Using the proposed method, robust watermarking is possible both in common image processing operations such as cropping and lossy compression and in geometrical transforms such as rotation, scaling, and translation, because it can embed watermark and template signal maximally using MTF of HVS. Experimental results show that the proposed watermarking method is more robust to several common image processing operations and geometrical transforms.

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In-service Real-time and Continuous Objective Video Quality Assessment for DTV Broadcasting

  • Han, Chan-Ho
    • Journal of Korea Multimedia Society
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    • v.16 no.1
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    • pp.50-55
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    • 2013
  • This article presents a simple and reasonable in-service, real-time, and continuous single-ended objective video quality assessment model for DTV broadcasting using a multiburst signal at the bottom of the transient effect area, similar to Vertical Interval Test Signals. The issue of in-service video-quality monitoring in DTV broadcasting is addressed, and an effective method of quality monitoring is presented. The proposed method is also implemented and tested in a range of situations using a simulated HDTV broadcasting network.

Speech Transition Detection and approximate-synthesis Method for Speech Signal Compression and Recovery (음성신호 압축 및 복원을 위한 음성 천이구간 검출과 근사합성 방식)

  • Lee, Kwang-Seok;Kim, Bong-Gi;Kang, Seong-Soo;Kim, Hyun-Deok
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2008.05a
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    • pp.763-767
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    • 2008
  • In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.

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A study on optimal Image Data Multiresolution Representation and Compression Through Wavelet Transform (Wavelet 변환을 이용한 최적 영상 데이터 다해상도 표현 및 압축에 관한 연구)

  • Kang, Gyung-Mo;Jeoung, Ki-Sam;Lee, Myoung-Ho
    • Proceedings of the KOSOMBE Conference
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    • v.1994 no.12
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    • pp.31-38
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    • 1994
  • This paper proposed signal decomposition and multiresolution representation through wavelet transform using wavelet orthonormal basis. And it suggested most appropriate filter for scaling function in multiresoltion representation and compared two compression method, arithmetic coding and Huffman coding. Results are as follows 1. Daub18 coefficient is most appropriate in computing time, energy compaction, image quality. 2. In case of image browsing that should be small in size and good for recognition, it is reasonable to decompose to 3 scale using pyramidal algorithm. 3. For the case of progressive transmittion where requires most grateful image reconstruction from least number of sampls or reconstruction at any target rate, I embedded the data in order of significance after scaling to 5 step. 4. Medical images such as information loss is fatal have to be compressed by lossless method. As a result from compressing 5 scaled data through arithmetic coding and Huffman coding, I obtained that arithmetic coding is better than huffman coding in processing time and compression ratio. And in case of arithmetic coding I could compress to 38% to original image data.

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Microwave Signal Spectrum Broadening System Based on Time Compression

  • Kong, Menglong;Tan, Zhongwei;Niu, Hui;Li, Hongbo;Gao, Hongpei
    • Current Optics and Photonics
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    • v.4 no.4
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    • pp.310-316
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    • 2020
  • We propose and experimentally demonstrate an all-optical radio frequency (RF) spectrum broadening system based on time compression. By utilizing the procedure of dispersion compensation values, the frequency domain is broadened by compressing the linear chirp optical pulse which has been multiplexed by the radio frequency. A detailed mathematical description elucidates that the time compression is a very preferred scheme for spectrum broadening. We also report experimental results to prove this method, magnification factor at 2.7, 8 and 11 have been tested with different dispersion values of fiber, the experimental results agree well with the theoretical results. The proposed system is flexible and the magnification factor is determined by the dispersion values, the proposed scheme is a linear system. In addition, the influence of key parameters, for instance optical bandwidth and the sideband suppression ratio (SSR), are discussed. Magnification factor 11 of the proposed system is demonstrated.

DCT and DWT Based Robust Audio Watermarking Scheme for Copyright Protection

  • Deb, Kaushik;Rahman, Md. Ashikur;Sultana, Kazi Zakia;Sarker, Md. Iqbal Hasan;Chong, Ui-Pil
    • Journal of the Institute of Convergence Signal Processing
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    • v.15 no.1
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    • pp.1-8
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    • 2014
  • Digital watermarking techniques are attracting attention as a proper solution to protect copyright for multimedia data. This paper proposes a new audio watermarking method based on Discrete Cosine Transformation (DCT) and Discrete Wavelet Transformation (DWT) for copyright protection. In our proposed watermarking method, the original audio is transformed into DCT domain and divided into two parts. Synchronization code is applied on the signal in first part and 2 levels DWT domain is applied on the signal in second part. The absolute value of DWT coefficient is divided into arbitrary number of segments and calculates the energy of each segment and middle peak. Watermarks are then embedded into each middle peak. Watermarks are extracted by performing the inverse operation of watermark embedding process. Experimental results show that the hidden watermark data is robust to re-sampling, low-pass filtering, re-quantization, MP3 compression, cropping, echo addition, delay, and pitch shifting, amplitude change. Performance analysis of the proposed scheme shows low error probability rates.