• Title/Summary/Keyword: PCM(Pulse Code Modulation)

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Maximum Likelihood Based Doppler Estimation and Target Detection with Pulse Code Modulated Waveform (ML 기법을 이용한 PCM 파형에서의 표적 탐지 및 도플러 추정)

  • Yang, Eunjung;Lee, Heeyoung;Song, Junho
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.25 no.12
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    • pp.1275-1283
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    • 2014
  • Characteristics of PCM(Pulse Code Modulation) waveform are suitable for target tracking. Especially in terms of dwell time, it is desirable to detect and track a moving target with the single PCM waveform for a MFR(Multi-Function Radar) which carries out multiple tasks. General PCM waveform processing includes Doppler filter bank caused by the characteristics of ambiguity function, to detect target and estimate Doppler frequency, which induces hardware burden and computational complexity. We propose a ML(Maximum Likelihood) based Doppler estimator for a PCM waveform, which is the closed form suboptimal solution and computationally efficient to estimate Doppler frequency and detect a moving target.

Pulse Code Signal Recognition using Integra-Normalizer (인테그라-노말라이저를 이용한 펄스코드 신호인식)

  • Kim, Seong-Su
    • The Transactions of the Korean Institute of Electrical Engineers D
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    • v.49 no.8
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    • pp.491-494
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    • 2000
  • A scheme is proposed for measuring similarities between the binary pulse signals in the pulse-code modulation using the Integra-Normalizer. The Integra-Normalizer provides a better interpretation of the relationship between the pulse signals by removing redundant codes, which maps all possible observed signals to one of the codes to be received with relative similarities between each pair of compared signals. The proposed method provides better error tolerance than L2 metric, such as Hamming distance, since the distances between pulse signals are measured not useful for the time-delay detection in the pulse-code modulation.

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PCM Encoder Structure for Real-time Updating of Telemetry System Parameters (원격 측정 시스템 파라미터 실시간 업데이트 PCM 엔코더 구조)

  • Park, Yu-Kwang;Yoon, Won-Ju
    • Journal of Advanced Navigation Technology
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    • v.23 no.5
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    • pp.452-459
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    • 2019
  • In this paper, we describe a PCM encoder structure that can update the telemetry system parameters in real time. In the PCM encoder, an analog signal control unit for FPGA, flash memory, and sensor data acquisition was constructed. UART communication, analog signal control, flash memory control, and frame generation are possible through logic inside FPGA of PCM encoder. UART communication allows the PC to transmit parameter data to the PCM encoder, and flash memory is controlled to update the parameter of the telemetry system in real time and finally the frame is formed. Simulation and verification were performed to confirm whether the parameter data is updated in real time, and the proposed structure was used to construct a telemetry system with enhanced flexibility and convenience.

On Predictive Coding of Speech Signals (음성신호의 예측부호화에 관하여)

  • 은종관
    • The Magazine of the IEIE
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    • v.12 no.5
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    • pp.23-35
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    • 1985
  • 본 논문은 디지털 음성통신에서 사용되는 예측부호화(predictive coding) 방식에 관하여 기술하고 있다. 특히 전송속도가 16∼48kbit/s 대역에서 많이 사용하고 있는 adaptive differential pulse code modulation(ADPCM)과 adaptive delta modulation(ADM)에 관하여 중점적으로 토의한다. 또한 variable-rate ADPCM과 ADM에 관해서 기술하고, 이들 시스템의 noisy channel에서의 효과 및 성능개선방법, 그리고 PCM과의 transcoding에서의 문제점 등을 통의한다. ADPCM은 최근 CCITT에서의 표준화 결과로 앞으로 PCM과 함께 많이 쓰여질 전망이며, ADM은 시스템이 간단하고 또한 channel error에 강한 이유로 특수통신에 많이 쓰여질 것이다.

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On the Comparison of MTF in Sub-Band Coding Technique Employing the Human Visual System (인간의 시각특성을 고려한 Sub-Band 부호화에서 MRF 비교에 관한 연구)

  • 김용관;박섭형;이상욱
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.27 no.5
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    • pp.784-792
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    • 1990
  • In this paper, attempts have been made to compare the subjective performance of sub-band coding (SBC) techniques incorporating four representative modulation transfer functions (MTF's) of th human visual system(HVS): Sakrison, Nill, Ngan, and Rao. In SBC, the frequency band of input signal is split into 16 equal sub-bands. In addition, 28 sub-band splitting schemd which splits more sub-bands in low spatial frequency is considered to compare the 4 MTF's effectively. In encoding of each sub-band, the weight of each sub-band obtained from the MTF of HVS is applied to bit allocation process which minimize the weighted mean square error (WMSE). The differential pulse code modulation(DPCM) coder is used to encode the lowest sub-band and the pulse code modulation(PCM) coder is used for the rest of sub-bands. It is found that the images incorporating the MTF of Rao yields the best results in subjective criteria, followed by Ngan, Nill, Sakrison, and the images not employing the HVS.

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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.

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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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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Design of Voice processing module Using RTP in VoIP system (SIP기반의 VoIP시스템에서 RTP를 이용한 Voice 처리 모듈의 개발)

  • 윤원동;백은경;박일규;최양희
    • Proceedings of the Korean Information Science Society Conference
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    • 2001.04a
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    • pp.292-294
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    • 2001
  • VoIP(Voice over IP) system은 현재 크게 2가지 형태로 진행되어가고 있다. 첫 번째는 H.323을 이용한 방법이고, 두 번째는 SIP(Session Initiation Protocol)를 이용한 방법이다. H.323은 실제 데이터를 전송하기전 호처리에 많은 signaling이 이루어지는 관계로 SIP보다 많은 RTT(Round Trip Time)를 소모하게 된다. 따라서 매우 복잡하고, LAN환경을 바탕으로 만들어서 확장성면에서도 여러 문제점을 가지고 있다. 그래서 본 논문은 호처리는 SIP를 이용하고, 실제 음성전송은 RTP(Real-Time Transport Protocol)와 RTCP(RTP Control Protocol)를 이용하는 시스템 구현을 제시한다. RTP는 실시간 특성을 가지는 데이터에 대해서 종단간 전송 서비스를 제공해주는 프로토콜로, 어떠한 인코딩에도 적합한 프레임워크를 제공한다. 그런데, RTP는 완전한 하나의 프로토콜이 되기 위해서는 RTP와 페이로드 포맷이 함께 제공되어야 하므로, 구현시스템은 음성신호를 PCM(Pulse Code Modulation), ADPCM(Adaptive Differential PCM)등의 여러 압축기술을 이용하여 파일을 생성하여 실시간으로 RTP와 RTCP를 이용하여 전송하는 방법을 제시한다.