• Title/Summary/Keyword: CODEC

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Design of the LSF Parameter Quantizer for the Wideband Speech Codec (광대역 음성 부호화기용 선 스펙트럼 주파수 계수 양자화기 설계)

  • 지상현;강상원;윤병식
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.4
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    • pp.29-34
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    • 2001
  • In this paper, we designed an LSF coefficient quantizer of the wideband speech codec that can produce high quality speech service. For the efficient LSF coefficient quantizer, the interframe correlation was used. Also we separately quantized the LSF coefficients with high and low interframe correlation. Predictive pyramid vector quantizer (PVQ) was used for quantizing the LSF coefficients with high interframe correlation, and PVQ was used for quantizing the LSF coefficients with low interframe correlation. Experiments show that the proposed UF quantizer can quantize LSF information in 40 bits/frame, with an average spectral distortion (SD) of 1 dB and less than 3.87% frames having SD greater than 2 dB.

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Efficient Motion Estimation Algorithm and Circuit Architecture for H.264 Video CODEC (H.264 비디오 코덱을 위한 효율적인 움직임 추정 알고리즘과 회로 구조)

  • Lee, Seon-Young;Cho, Kyeong-Soon
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.47 no.12
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    • pp.48-54
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    • 2010
  • This paper presents a high-performance architecture of integer-pel motion estimation circuit for H.264 video CODEC. Full search algorithm guarantees the best results by examining all candidate blocks. However, the full search algorithm requires a huge amount of computation and data. Many fast search algorithms have been proposed to reduce the computational efforts. The disadvantage of these algorithms is that data access from or to memory is very irregular and data reuse is difficult. In this paper, we propose an efficient integer-pixel motion estimation algorithm and the circuit architecture to improve the processing speed and reduce the external memory bandwidth. The proposed circuit supports seven kinds of variable block sizes and generates 41 motion vectors. We described the proposed high-performance motion estimation circuit at R1L and verified its operation on FPGA board. The circuit synthesized by using l30nm CMOS standard cell library processes 139.8 1080HD ($1,920{\times}1,088$) image frames per second and supports up to H.264 level 5.1.

The Design and Performance Analysis of Effficient VoIP Service Scheme for High Speed Packet Switching based on IMT-2000 (IMT-2000 기반 고속 패킷 교환 방식에서의 효율적인 VoIP 서비스 지원 방안 설계와 성능 분석)

  • Lee, Tae-Ro;Lee, Sung-Won;Han, Chi-Geun;Ryoo, In-Tae
    • The Transactions of the Korea Information Processing Society
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    • v.7 no.8
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    • pp.2463-2472
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    • 2000
  • In this paper, we consider pits and falls of V oIP service scheme over the air link environment. It results that V oIP over packet switching is a more attractive approach in several points. We point out the ffilIior requirements for successful V oIP service over the air. Also, we propose V oIP CP concept for efficient wireless channel utilization. Additionally, we analyze and evaluate the performance. According to the results, It shows that the long cycle VolP vocoder CODEC such as ITU-T G.723 is better than short cycle V oIP vocoder CODEC. In this case, the increase of the simultaneous user of system is almost 60% larger than conventional circuit switching.

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Implementation of G.726 ADPCM Dual Rate Speech Codec of 16Kbps and 40Kbps (16Kbps와 40Kbps의 Dual Rate G.726 ADPCM 음성 codec구현)

  • Kim Jae-Oh;Han Kyong-Ho
    • Journal of IKEEE
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    • v.2 no.2 s.3
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    • pp.233-238
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    • 1998
  • In this paper, the implementation of dual rate ADPCM using G.726 16Kbps and 40Kbps speech codec algorithm is handled. For small signals, the low rate 16Kbps coding algorithm shows almost the same SNR as the high rate 40Kbps coding algorithm , while the high rate 40Kbps coding algorithm shows the higher SNR than the low rate 16Kbps coding algorithm fur large signal. To obtain the good trade-off between the data rate and synthesized speech quality, we applied low rate 16Kbps for the small signal and high rate 40Kbps for the large signal. Various threshold values determining the rate are applied for good trade-off between data rate and speech quality. The simulation result shows the good speech quality at a low rate comparing with 16Kbps & 40Kbps.

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A Study on the Interframe Image Coding Using Motion Compensated and Classified Vector Quantizer (Ⅱ : Hardware Implementation) (이동 보상과 분류 벡터 양자화기를 이용한 영상 부호화에 관한 연구 (Ⅱ: 하드웨어 실현))

  • Jeon, Joong-Nam;Shin, Tae-Min;Choi, Sung-Nam;Park, Kyu-Tae
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.27 no.3
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    • pp.21-30
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    • 1990
  • This paper describes a hardware implementation of the interframe monochrome video CODEC using a MC-CVQ(Motion Compensated and Classified Vector Quantization) algorithm. The specifications of this CODEC are (1) the resolution of image is $128{\times}128$ pixels, and (2) the transmission rates are about 10frames/sec at the 64Kbps channel. In order to design the CODEC under these conditions, it is implemented by a multiprocessor system composed of MC unit, CVQ nuit and decoder unit, which are controlled by microprogramming technique. And the 3~stage pipelined ALU(Arithmetic and Logic Unit) is adopted to calculate the minimum error distance in the MC unit and CVQ nuit. The realized system shows that the transmission rates are 6-15 frames/sec according to the relative motion of the video signal.

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Turbo Coded OFDM for Digital Audio Broadcasting System (디지털 오디오 방송을 위한 터보 부호화된 OFDM)

  • Kim, Han-Jong
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.38 no.11
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    • pp.19-29
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    • 2001
  • The Pan-European Digital Audio Broadcasting(DAH) system's performance is characterized and improved with the aid of turbo codec. From the fact that the first bit among the four coded bits at the RCPC coding defined in the Eureka 147 DAD system is not. punctured and always transmitted, this paper proposes a new turbo coded DAB system model that replaces the existing RCPC codec by a turbo codec without modifying the puncturing procedure and puncturing vectors defined in the standard DAB system for compatibility. The performance of a new system is compared to that of the conventional system under the Rician fading channel and the Rayleigh fading channel in conjunction with DAD transmission mode I and III suitable for the terrestrial single frequency network and satellite broadcasting.

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Density Scalability of Video Based Point Cloud Compression by Using SHVC Codec (SHVC 비디오 기반 포인트 클라우드 밀도 스케일러빌리티 방안)

  • Hwang, Yonghae;Kim, Junsik;Kim, Kyuheon
    • Journal of Broadcast Engineering
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    • v.25 no.5
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    • pp.709-722
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    • 2020
  • Point Cloud which is a cluster of numerous points can express 3D object beyond the 2D plane. Each point contains 3D coordinate and color data basically, reflectance or etc. additionally. Point Cloud demand research and development much higher effective compression technology. Video-based Point Cloud Compression (V-PCC) technology in development and standardization based on the established video codec. Despite its high effective compression technology, point cloud service will be limited by terminal spec and network conditions. 2D video had the same problems. To remedy this kind of problem, 2D video is using Scalable High efficiency Video Coding (SHVC), Dynamic Adaptive Streaming over HTTP (DASH) or diverse technology. This paper proposed a density scalability method using SHVC codec in V-PCC.

The Subjective Assessment Testing of Basic and Transmission Video Quality for Digital Broadcasting Satellite (디지털 위성 방송 기본 화질과 전송 화질의 주관적 평가 시험)

  • 박대철;김용선;유태선;전병민
    • Journal of Broadcast Engineering
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    • v.2 no.1
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    • pp.24-35
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    • 1997
  • Basic and transmission video quality testing was performed in real-time mode using hardware video codec based on MPEG-2 MP@ML standard including subsequent transmitting and receiving satellite simulator unit. The double-stimulus Impairment scale method and the double-stimulus continous quality scale method based on CCIR 500-5 were used as an evaluation method. The whole digital broadcasting satellite system consisting of MPEG-2 codec, system mux/demux, channel codec, channel, modem, antenna, etc. was put into the overall video quality testbed and the basic and transmission error quality assessment was performed at various bitrates and BER for an integrated system performance evaluation. In transmission error video quality testing, transmission error video quality maintained on average above 3.9 point on the 5-point scale. The low-bit rate quality such as film mode(@2Mbps) highly depended on the statitical characteristics of video source and maintained on average around 2.7 point.

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Security Enhancing of Authentication Protocol for Hash Based RFID Tag (해쉬 기반 RFID 태그를 위한 인증 프로토콜의 보안성 향상)

  • Jeon, Jin-Oh;Kang, Min-Sup
    • Journal of Internet Computing and Services
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    • v.11 no.4
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    • pp.23-32
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    • 2010
  • In this paper, we first propose the security enhancing of authentication protocol for Hash based RFID tag, and then a digital Codec for RFID tag is designed based on the proposed authentication protocol. The protocol is based on a three-way challenge response authentication protocol between the tags and a back-end server. In order to realize a secure cryptographic authentication mechanism, we modify three types of the protocol packets which defined in the ISO/IEC 18000-3 standard. Thus active attacks such as the Man-in-the-middle and Replay attacks can be easily protected. In order to verify effectiveness of the proposed protocol, a digital Codec for RFID tag is designed using Verilog HDL, and also synthesized using Synopsys Design Compiler with Hynix $0.25\;{\mu}m$ standard-cell library. Through security analysis and comparison result, we will show that the proposed scheme has better performance in user data confidentiality, tag anonymity, Man-in-the-middle attack prevention, replay attack, forgery resistance and location tracking.

Implementation of a 4-Channerl ADPCM CODEC Using a DSP (DSP를 사용한 4채널용 ADPCM CODEC의 실시간 구현에 관한 연구)

  • Lee, Ui-Taek;Lee, Gang-Seok;Lee, Sang-Uk
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.22 no.5
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    • pp.29-38
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    • 1985
  • In this paper we have designed and implemented in real time a simple, efficient and flexible AOPCM cosec using a high speed digital processor, NEC 7720. For ADPCM system, we have used an instantaneous adaptive quantizer and a first-order fixed predictor. The software for NEC 7720 has been developed and it was found that the NEC 7720 was capable of performing the entire ADPCAt algorithm for 4 channels in real time as optimizing the program. Computer simulation has born made to investigate a computational accuracr of NEC 7720 and to de-termine necessary parameters for a ADPCM codec. Real telephone speech, RC-shaped Gaussian noise and 1004 Hz tone signal were used for simulation. In simulation, the parameters werc optimized from the computed SNR and the informal listening test. The developed software was tested in real time operation using a hardware emulator for NEC 7720. It took a maximum 23.25$\mu$s to encode one sample and 113.5$\mu$s, including all the necessary 1/0 operations, to encode 4 channels. In the case of decoding process, it took 24.75$\mu$s to decode one sample and 119.5$\mu$s to decode 4 channels.

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