• Title/Summary/Keyword: Pulse code modulation

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Power Spectral Density of Antipodal Ultra Wideband Signal (Antipodal 초광대역(UWB) 신호의 전력 스펙트럼 밀도 분석)

  • Kim, Jong Han;Lee, Jung Suk;Kim, Yoo Chang;Kim, Won Hoo;Kim, Jung Sun
    • Journal of Advanced Navigation Technology
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    • v.5 no.1
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    • pp.54-61
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    • 2001
  • In conventional Ultra Wide Band(UWB) system, it uses Pulse Positioning Modulation Method to modulate data signal. In this paper, however, we derive power spectral density characteristic of time hopped antipodal signal using stochastic process. UWB signal employes Gaussian monopulse and Rayleigh monopulse which pulse width is 0.5 nsec and interval is 5 nsec. But comb line which produces unintentionally could be evidently reduced by the time hopped code, so this code be used to channelize for multiple access and minimize to different communication system.

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Double Binary Turbo Coded Data Transmission of STBC UWB Systems for U-Healthcare Applications

  • Kim, Yoon-Hyun;Kim, Eun-Cheol;Kim, Jin-Young
    • International journal of advanced smart convergence
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    • v.1 no.1
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    • pp.27-33
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    • 2012
  • In this paper, we analyze and simulate performance of space time block coded (STBC) binary pulse amplitude modulation-direct sequence (BPAM-DS) ultra-wideband (UWB) systems with double binary turbo code in indoor environments for various ubiquitous healthcare (u-healthcare) applications. Indoor wireless channel is modeled as a modified Saleh and Valenzuela (SV) model proposed as a UWB indoor channel model by the IEEE 802.15.SG3a in July 2003. In the STBC encoding process, an Alamouti algorithm for real-valued signals is employed because UWB signals have the type of real signal constellation. It is assumed that the transmitter has knowledge about channel state information. From simulation results, it is shown that the STBC scheme does not have an influence on improving bit error probability performance of the BPAM-DS UWB systems. It is also confirmed that the results of this paper can be applicable for u-healthcare applications.

Bitwise Decomposition Algorithm for Gray Coded M-PSK Signals (Gray 부호화된 M-PSK 신호의 비트 정보 분할 알고리듬)

  • Kim Ki-Seol;Hyun Kwang-Min;Park Sang-Kyu
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.8A
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    • pp.784-789
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    • 2006
  • In this paper, we propose a bitwise information decomposition algorithm for an M-PSK signal based on the Max-Log-MAP algorithm. In order to obtain the algorithm, we use a coordinate transformation from M-PSK to M-PAM signal space. Using the proposed algorithm, we analyze the performance of a Turbo iterative decoding method. The proposed algorithm can be applicable not only for a communication with PSK and iterative decoding method but for adaptive modulation and coding system.

Power-efficiency Analysis of the MIMO-VLC System considering Dimming Control (조광제어를 고려한 MIMO-VLC 시스템의 전력 효율 분석)

  • Kim, Yong-Won;Lee, Byung-Jin;Lee, Byung-Hoon;Lee, Min-Jung;Kim, Kyung-Seok
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.18 no.6
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    • pp.169-180
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    • 2018
  • White light-emitting diodes (LEDs) are more economical than fluorescent lights, and provide high brightness, a high lifetime expectancy, and greater durability. As LEDs are closely connected with people's daily lives, dimming control of LED is an important component in providing energy savings and improving quality of life. In visible light communications systems using these LEDs, multiple input multiple output (MIMO) technology has attracted a lot of attention, in that it can attain the channel capacity in proportion to the number of antennas. This paper analyzes the power performance of three kinds of modulation in visible light communications (VLC) systems applied space-time block code (STBC) techniques. The modulation schemes are return-to-zero on-off keying (RZ-OOK), variable pulse position modulation (VPPM), and overlapping pulse position modulation (OPPM), and dimming control was applied. The power requirements and power consumption were used as metrics to compare the power efficiency in $2{\times}2$ STBC-VLC environments under the three kinds of modulation. We confirm that dimming control affects the communications performance of each modulation scheme. VPPM showed greater consumption among the three modulations, and OPPM showed energy savings comparable to VPPM.

A Study on EUROFIX Reed Solomon Code Design Using Finite Galois Field Fourier Transformation (유한체 푸리에 변환을 이용한 EUROFIX RS Code 설계에 관한 연구)

  • Kim, Min-Jee;Kim, Min-Jung;Chung, Se-Mo;Cho, Hyung-Rae
    • Journal of Navigation and Port Research
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    • v.28 no.1
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    • pp.23-29
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    • 2004
  • This paper deals with Reed-Solomon Coding for EUROFIX system EUROFIX is an integrated navigation and communication system, which combines Differential GNSS and Loran-C EUROFIX transmits DGNSS(Differential Global Navigation Satellite Systems) (data by pulse position modulation of Loran-C pulses. Loran-C system is regarded as a satellite backup system in recent. And now, it is important to detect and correct much errors in communication systems. Error corrections or correction algorithm is actively studied nowadays because of this. In this paper, we study and design encoder and decoder of Reed Solomon Code using Finite Galois Field Fourier Transformation for error corrections in EUROFIX data transmission. Through extensive simulation, the designed Reed Solomon code is shown to be effective for error correction in EUROFIX data transmission.

High Bit Rate Image Coder Using DPCM based on Sample-Adaptive Product Quantizer (표본 적응 프러덕트 양자기에 기초한 DPCM을 이용한 고 전송률 영상 압축)

  • 김동식;이상욱
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.24 no.12B
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    • pp.2382-2390
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    • 1999
  • In this paper, we employed a new quantization scheme called sample-adaptive product quantizer (SAPQ) to quantize image data based on the differential pulse code modulation (DPCM) coder, which has fixed length outputs and high bit rates. In order to improve the performance of traditional DPCM coders, the scalar quantizer should be replaced by the vector quantizer (VQ). As the bit rate increases, it will be nearly impossible to implement a conventional VQ or modified VQ, such as the tree-structured VQ, even if the modified VQ can significantly reduce the encoding complexity. SAPQ has a form of the feed-forward adaptive scalar quantizer having a short adaptation period. However, since SAPQ is a structurally constrained VQ, SAPQ can achieve VQ-level performance with a low encoding complexity. Since SAPQ has a scalar quantizer structure, by using the traditional scalar value predictors, we can easily apply SAPQ to DPCM coders. For synthetic data and real images, by employing SAPQ as the quantizer part of DPCM coders, we obtained a 2~3 dB improvement over the DPCM coders, which are based on the Lloyd-Max scalar quantizers, for data rates above 4 b/point.

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Linear Predictor Using Charge-Coupled Devices (CCD를 이용한 선형예측기)

  • 최태영;신철재
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.12 no.1
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    • pp.9-18
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    • 1987
  • An electro-optic system using linear photosensitive Charge Coupled Devices(CCDs) having dummy pixels has been proposed for realzation of linear prodictor in the differential pulse code modulation(DPCM). The system consists of three components as conventional system:input light source, spatial filter(mask) and CCD line scanning sensor. For the delay time due to the dummy pixels in CCD, modifying conventional mask, a new dispersive mask is proposed, of which every prediction coefficient is dispersed on the more than one element, the characteristics of the system using the proposed dispersive mask are analyzed theoretically and verified with experiment.

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A Fixed Rate Speech Coder Based on the Filter Bank Method and the Inflection Point Detection

  • Iem, Byeong-Gwan
    • International Journal of Fuzzy Logic and Intelligent Systems
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    • v.16 no.4
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    • pp.276-280
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    • 2016
  • A fixed rate speech coder based on the filter bank and the non-uniform sampling technique is proposed. The non-uniform sampling is achieved by the detection of inflection points (IPs). A speech block is band passed by the filter bank, and the subband signals are processed by the IP detector, and the detected IP patterns are compared with entries of the IP database. For each subband signal, the address of the closest member of the database and the energy of the IP pattern are transmitted through channel. In the receiver, the decoder recovers the subband signals using the received addresses and the energy information, and reconstructs the speech via the filter bank summation. As results, the coder shows fixed data rate contrary to the existing speech coders based on the non-uniform sampling. Through computer simulation, the usefulness of the proposed technique is confirmed. The signal-to-noise ratio (SNR) performance of the proposed method is comparable to that of the uniform sampled pulse code modulation (PCM) below 20 kbps data rate.

Estimation of Weight Coefficients of Residual DPCM based on L1 Regularization in HEVC Format Range Extension (HEVC 확장 표준 내 Residual DPCM 을 위한 L1 정규화 기반의 가중 계수 추정 기법)

  • Ryu, Su-Kyung;Kang, Je-Won
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2016.06a
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    • pp.373-374
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    • 2016
  • Residual Differnetial pulse-code Modulation (RDPCM) 기법은 비디오의 압축을 위한 시간 및 공간 예측 후 남은 잔여신호를 인접 화소를 이용하여 추가적인 중복정보를 제거하는 기법을 의미한다. 본 논문에서는 우선 잔차 신호의 예측을 위하여 인접 화소 사이 선형 가중 합으로 예측 모델을 세우고, 각 가중치를 $L_1$ 정규화를 포함하는 비용함수를 통해 추정함으로써 보다 효율적인 부호화 성능을 제공하는 알고리즘을 제안한다.

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Digital Filter Design for the DSD Encoder with Multi-rate PCM Input (PCM 입력의 DSD 인코더를 위한 디지털 필터 설계)

  • Moon, Dong-Wook;Kim, Lark-Kyo
    • Proceedings of the KIEE Conference
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    • 2005.05a
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    • pp.170-172
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    • 2005
  • The DSD(Direct Stream Digital) encoder, which is a standard for SACD(Super Audio Compact Disc) proposed by Sony and philips, use 1 bit representation with a sampling frequency of 2.8224 MHz (64 $\times$ 44.1 kHz). For multi-rate PCM (Pulse Code Modulation) input like as 48/96/192 kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital filter structure composed of sample-rate converter and interpolation filter for the DSD encoder with multi-rate (48/96/192 kHz) PCM input. without a external sample-rate converter.

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