• Title/Summary/Keyword: Pulse code modulation

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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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An Improved Digital Filter Design for the DSD Encoder with Multi-rate PCM Input (다중 표본화율의 PCM 입력을 위한 개선된 DSD 인코더용 디지털 필털 설계)

  • Moon, Dong-Wook;Kim, Lark-Kyo
    • Proceedings of the KIEE Conference
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    • 2005.10b
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    • pp.358-360
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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, uses 1 bit representation with a sampling frequency of 2.8224MHz (64${\times}$44.1kHz). For multi-rate PCM (Pulse Code Modulation) input such as 8${\sim}$192kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital mter structure composed of sample-rate converter and interpolaton filter for the DSD encoder with multi-rate (8${\sim}$192kHz) PCM input, without a external sample-rate converter.

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The Real-Time Implementation of G.726 ADPCM on OAK DSP Core based CSD17C00A (OAK DSP Core 기반 CSD17C00A에서의 G.726 ADPCM의 실시간 구현)

  • Hong SeongHoon;Shim MinKyu;Sung YooNa;Ha JungHo
    • Proceedings of the Acoustical Society of Korea Conference
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    • spring
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    • pp.52-55
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    • 1999
  • 다중 전송율(16, 24, 32, 40kbps)을 제공하는 G.726 부호화기는 ADPCM (Adaptive Differential Pulse Code Modulation) 부호화법을 사용한다. 본논문에서는 G.726 ADPCM 알고리즘을 C&S Technology에서 개발한 음성 신호 처리를 위한 범용 DSP인 CSD17C00A 칩을 이용하여 실시간 응용이 가능하도록 구현하였다. G.726에 대한 양방향 평가는 Codec Loopback test을 통해 수행되었으며, W-T에서 제공한 테스트 절차에 따라 평가되었다. 본 논문에서 구현된 G.726 부호화기는 평균 11 MIPS의 계산 속도를 갖고, 프로그램 메모리 크기는 2.8K Words이고, 데이터 메모리 크기는 550 Words 를 필요로 하였다.

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Residual DPCM in HEVC Transform Skip Mode for Screen Content Coding

  • Han, Chan-Hee;Lee, Si-Woong;Choi, Haechul
    • IEIE Transactions on Smart Processing and Computing
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    • v.5 no.5
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    • pp.323-326
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    • 2016
  • High Efficiency Video Coding (HEVC) adopts intra transform skip mode, in which a residual block is directly quantized in the pixel domain without transforming the block into the frequency domain. Intra transform skip mode provides a significant coding gain for screen content. However, when intra-prediction errors are not transformed, the errors are often correlated along the intra-prediction direction. This paper introduces a residual differential pulse code modulation (DPCM) method for the intra-predicted and transform-skipped blocks to remove redundancy. The proposed method performs pixel-by-pixel residual prediction along the intra-prediction direction to reduce the dynamic range of intra-prediction errors. Experimental results show that the transform skip mode's Bjøntegaard delta rate (BD-rate) is improved by 12.8% for vertically intra-predicted blocks. Overall, the proposed method shows an average 1.2% reduction in BD-rate, relative to HEVC, with negligible computational complexity.

Development of PCM Data Recorder for Telemetry System (원격측정용 PCM 데이터 저장장치 개발)

  • Koh, Kwang-Ryul;Lee, Sang-Bum;Lee, Hyun-Kyu;Kim, Whan-Woo
    • Journal of the Korea Institute of Military Science and Technology
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    • v.14 no.4
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    • pp.607-614
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    • 2011
  • This paper describes the development of pulse code modulation(PCM) data recorder with design, implementation and environmental test. PCM serial data that diverged from telemetry encoder output is used as the input and is reformed to parallel signal through FPGA processing. Controllers construct the packet by the sector and record it into non-volatile memory. Compact flash(CF) memory for data storage media, USB interface for data downloading, and a software for operating status diagnosis and file format conversion are used.

A Study on a Tracking Method of Telemetry Signal using Self-Slaving (Self-Slaving을 이용한 원격측정 신호추적 기법 연구)

  • Lee, Sung-Pil
    • Journal of the Korea Institute of Military Science and Technology
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    • v.11 no.3
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    • pp.50-57
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    • 2008
  • A telemetry ground station has a highly directional, parabolic tracking antenna to receive a weak telemetry signal from a target at a distance. The tracking antenna with narrow beam-width normally uses the auto-track method for the target tracking. This paper presents several issues in the auto-track method and introduces a new tracking method using Self-Slaving technique. Self-Slaving means that the tracking antenna is slaved to not data measured by RADAR but GPS/INS informations received by the telemetry system for pointing. The Self-Slaving method shows good performance in comparison with auto-track method.

Image Information Compression by Adaptive Differential Pulse Code Modulation (적응 DPCM에 의한 영상정보 감축)

  • Cho, B.G.;Han, Y.O.;Oh, J.S.;Rhee, S.H.;Chang, I.H.;Rhee, Y.C.;Park, S.H.
    • Proceedings of the KIEE Conference
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    • 1988.07a
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    • pp.229-231
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    • 1988
  • In this paper, we design adaptive DPCM by the two dimensional linear prediction using the covariance method and propose adaptive algorithm which determines analysis frame size by the compassion of the local standard deviation and the global standard deviation of images.

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A Study on Implementation fo NTSC 90Mb/s digital TV CODEC (NTSC 90Mb/S 디지털 TV 코덱구성에 관한 연구)

  • Shin, W.B.;Kye, Y.C.;Rho, H.J.;Cheong, Y.;Lee, M.S.;Kim, J.K.
    • Proceedings of the Korean Institute of Communication Sciences Conference
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    • 1984.04a
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    • pp.9-12
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    • 1984
  • A digital TV CODEC has been implemented. This CODEC uses a pulse code modulation with a resolution of 9 bits per sample and generates digital ouput at the rate of 90.194Mb/s. In this system, we aadopted a horizontal blanking control scheme to meet the contraint of 90.194Mb/s transmission rete, conswequently ontaining data compression of 9.4%, our system's performances have been measured. Its SNR

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Quantization of Line Spectrum Pair Frequencies using Lattice Vector Quantizers (격자벡터양자화기를 이용한 음성신호의 LSP 주파수 양자화)

  • 강정원;정재호;정대권
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.21 no.10
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    • pp.2634-2644
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    • 1996
  • Two different low rate speech coders using one of four types of lattice vector quantizers(LVQ's) with fairly low complexity were investigated for an application to mobile communications. More specifically, two-stage vector quantizer-lattic vector quantizer(VQ-LVQ) systems and vector differenctial pulse code modulation(VDPCM)systems with lattice vector quantizers simulated to encode the line spectrum frequencies of various sentences at the rate 22 to 39 bits per 20 msec frame. The simulation results showed that the VDPCM system with the lattice VQ can save up to 10 bits/fram compared to the quantization scheme used in QCELP system. For the VQ-LVQ system, the spherical quasi-uniform LVQ below 36 bits/frame outperformed the other 3 types of LVQ's and the pyramidal quasi-uniform LVQ at 37 bits/frame outperformed the other 3 types of LVQ's with the spectral distortion 0.97.

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Variable Threshold Detection with Weighted BPSK/PCM Speech Signal Transmitted over Gaussian Channels (가우시안 채널에 있어 가중치를 부여한 BPSK/PCM 음성신호의 비트거물 한계치 변화에 의한 신호재생)

  • 안승춘;서정욱;이문호
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.24 no.5
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    • pp.733-739
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    • 1987
  • In this paper, variable threshold detection with weighted pulse code modulation-encoded signals transmitted over Gaussian channels has been investigated. Each bit in the \ulcornerlaw PCM word is weighted according to its significance in the transmitter. It the output falls into the erasure zone, the regenerated sample replaced by interpolation or prediction. To overall system signal to noise ratio for BPSK/PCM speech signals of this technique has been found. When the input signal level was -17 db, the gains in overall signal s/n compared to weighted PCM and variable threshold detection were 5 db and 3 db, respectively. Computer simulation was performed generating signals by computer. The simulation was in resonable agreement with our theoretical prediction.

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