• Title/Summary/Keyword: Digital codec

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Digital Voice Ground Wave Range Analysis of HF Radios that Applied MELPe CODEC Using GRWAVE Program (MELPe 코덱이 적용된 HF 무전기의 GRWAVE 분석 툴을 이용한 디지털 음성 지상파 통달거리 분석)

  • Heo, Jin;Lee, Sangjin;Lee, Kangchun;Seo, Sungwon;Kim, Jungsup;Han, Sungwoo
    • Journal of the Korea Institute of Military Science and Technology
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    • v.20 no.3
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    • pp.431-440
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    • 2017
  • HF communications are used as a last means of long distance communications without any relay node in NLOS (Non Line-Of-Sight) environment. Conventional analog voice communication in the HF band is vulnerable to security as well as severe background noise. To overcome these shortcoming, digital voice was introduced into HF radios in the early 1980s. In this paper, we analyze avaliable digital voice communication ground wave range of HF radios that applied MELPe CODEC and MIL-STD-188-110B physical layer standard using GRWAVE program. And we evaluate usefulness of digital voice communication in HF band.

An MP@HL Codec and its Performance

  • Kanda, Kikufumi;Nishida, Yukihiro;Nakasu, Eisuke;Ohtsuka, Yoshimichi;Okawa, Yuji
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 1998.06b
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    • pp.10-15
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    • 1998
  • In Japan, digital HDTV services via broadcasting satellite are planned to start in 2000. NHK has developed an HDTV codec conforming to the MPEG-2 $MPHL$ and estimated the coded picture quality with diverse image sequences. This paper shows the codec specifications and presents the result of the picture quality assessment.

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A Design of ADPCM CODEC Core for Digital Voice and Image Processing SOC (디지털 음성 및 영상 처리용 SOC를 위한 ADPCM CODEC 코어의 설계)

  • 정중완;홍석일;한희일;조경순
    • Proceedings of the IEEK Conference
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    • 2001.06b
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    • pp.333-336
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    • 2001
  • This paper describes the design and implementation results of 40, 32, 24 and 16kbps ADPCM encoder and decoder circuit, based on the protocol CCITT G.726. We verified the ADPCM algorithm using C language and designed the RTL circuit with Verilog HDL. The circuit has been simulated by Verilog-XL, synthesized by Design Compiler and verified using Xilinx FPGA. Since the synthesized circuit includes a small number of gates, it is expected to be used as a core module in the digital voice and image processing SOC.

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Implementation and evaluation of stereo audio codec using perceptual coding (지각 부호화를 이용한 스테레요 오디오 코덱의 구현 및 음질 평가)

  • 차경환;장대영;홍진우;김천덕
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.33B no.4
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    • pp.156-163
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    • 1996
  • In this paper, we described the implementation and the sound quality assessment of a real-time stereo audio codec using TMS320C40 DSP (digital signal processing) chip for low bitrte and high quality audio. We implemented hardware and software in order to overcome a real-time processing problem of audio compression algorithm that can be produced by largely recursive computing and complexity of the process. We have studied five types of distortion that can be produced by perceptual coding and the codec was evaluated by eight test musics that are selected in SQAM (sound quality assessment material) 422-2-4-2 produced by EBU (european broadcast union). The subjective listening tests were carried out on the codec quality and preformance by double blind method in a listening room with eleven listeners. As a result, 5 grade-impairment scale was scored under minus one and the codec quality was evaluated to be perceptible, but not annoying.

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Real-Time Implementation of AMR Speech Codec Using TMS320VC5510 DSP (TMS320VC5510 DSP를 이용한 AMR 음성부호화기의 실시간 구현)

  • Kim, Jun;Bae, Keun-Sung
    • MALSORI
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    • no.65
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    • pp.143-152
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    • 2008
  • This paper focuses on the real time implementation of an adaptive multi-rate (AMR) speech codec, that is a standard speech codec of IMT-2000, using the TMS320VC5510. The series of TMS320VC55x is a 16-bit fixed-point digital signal processor (DSP) having low power consumption for the use of mobile communications by Texas Instruments (TI) corporation. After we analyze the AMR algorithm and source code as well as the structure and I/O of 7MS320VC55x, we carry out optimizing the programs for real time implementation. The implemented AMR speech codec uses 55.2 kbyte for the program memory and 98.3 kbyte for the data memory, and it requires 709,878 clocks, i.e. about 3.5 ms, for processing a frame of 20 ms speech signal.

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Copyright Protection using Encryption of DCT Coefficients and Motion Vector in Video Codec of Mobile Device (모바일 기기내의 비디오 코덱에서 DCT 계수와 움직임 벡터의 암호화를 이용한 저작권 보호)

  • Kwon, Goo Rak;Kim, Young Ro
    • Journal of Korea Society of Digital Industry and Information Management
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    • v.4 no.1
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    • pp.41-46
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    • 2008
  • With widespread use of the Internet and improvements in streaming media and compression technology, digital music, video, and image can be distributed instantaneously across the Internet to end-users. However, most conventional Digital Right Management are often not secure and fast enough to process the vast amount of data generated by the multimedia applications to meet the real-time constraints. In this paper, we propose the copyright protection using encryption of DCT coefficients and motion vector in MPEG-4 video codec of mobile device. This paper presents a new Digital Rights Management that modifies the Motion Vector of Macroblock for mobile device. Experimental results indicate that the proposed DRM can not only achieve very low cost of the encryption but also enable separable authentication to individual mobile devices such as Portable Multimedia Player and Personal Digital Assistants. The performance of the proposed methods have low complexity and low increase of bit rate in overhead.

A Study on the design of voice cryptograph system (음성암호시스템 설계에 관한 연구)

  • Choi, Tae-Sup;Ahn, In-Soo
    • Journal of the Institute of Electronics Engineers of Korea TE
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    • v.39 no.2
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    • pp.51-59
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    • 2002
  • In this paper, we studied the voice cryptograph system designed by the SEED algorithm for the safe transmission and receipt on the voice communication. Voice band signal converts to digital signal by the CODEC and DSP that applied the improved SEED algorithm encrypt the digital signal. The CODEC convert Encryption signal into analog voice signal. This voice signal is transmitted safely because of encryption signal even if someone wiretap. Receiver can hear the source voice, because the encryption signal decrypted using the SEED algorithm. In this paper, We designed the 32 round key instead of 16 round key in the SEED algorithm so that we improve the truncated differential probability from $2^{-143.1}$ to $2^{-286.6}$

Logic implementation of HDB3 Codec (HDB3 Codec의 로직 구현)

  • Eom, Joon;Kim, Young-kil
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2017.05a
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    • pp.397-399
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    • 2017
  • The HDB3 code, a type of line code, is a data coding method used for digital data transmission. It is used to remove the DC wander on the transmission line which occurs when the DC component data is transmitted continuously. The military tactical communication network uses HDB3 code for data transmission and develops equipment using commercial HDB3 Codec IC. Because it is operated for more than 10 years due to the characteristics of military equipment, if a failure occurs in the equipment, the equipment can not be repaired due to the discontinuance of the part, so that the entire equipment may not be used. In this paper, we implement the HDB3 Codec as a logic to solve this problem and verify that the performance is equivalent to that of commercial parts.

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Development of Digital Watermark Inserting and Detecting System for MPEG-2 Codec (MPEG-2 Codec에 대한 디지털 워터마크 삽입 및 검출 시스템 개발)

  • Hwang, Seon-Cheol
    • The Transactions of the Korean Institute of Electrical Engineers P
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    • v.59 no.4
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    • pp.390-394
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    • 2010
  • This paper describes a study on developing an insert and detect system of digital watermark to the MPEG-2 coder and decoder. Digital watermark inserting system substitutes a coefficient among the DCT coefficients for new one. This replacement performs in MPEG-2 PS format. Watermark detecting system analyzes the input data stream of MPEG-2 and detect the probability of occurrence of a specific code or watermarking code. Our inserting system had very high invisibility and the PSNR of the results was 130 dB around. And our detection rate of watermark was over 99% in digital format and over 85% in analog captured format according to the quality of contents.

Real-Time Implementation of a SBC Codec Using a NEC 7720 DSP (NEC 7720 DSP를 이용한 SBC codec의 실시간 구현)

  • Oh, Soo Hwan;Lee, Sang Uk
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.23 no.4
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    • pp.429-438
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    • 1986
  • In this paper we have designed and implemented a real-time, full-duplex SBC (sub-band coding) codec at 16kbps using a high speed digital signal processor, NEC 7720. The SBC codec employs a QMF(quadrature mirror filter) filter bank based on the tree structures of two-band analysis-synthesis pairs to partition speech signal into 4 octabe bands. Computer simulation has been done to investigate the effect of fixed-point computation of the NEC 7720. Three different performance measures, the conventional signal-to-noise ratio, the informal listening test, and an LPC(linear predictive coding)distance measure, have been used in this simulation. The necessary parameters have been optimized through the simulation. The developed hardware and software have been tested in real-time operation using a hardware emulator.

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