• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.584-587
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• 2008

In this paper, we propose a real time video codec adapter for enabling video communications with terminals having a codec which is different from each other. When multimedia services are playing with an office service phone such as a video phone or software phone which has video capability, each terminal is not being considered to have optimized video or voice codec. So when a video phone with only one type of video codec is used in the video streaming service which requires another type of codec, the streaming service is not successful without codec transformation. The real time codec adapter in this paper provides a real time code transformation which enables communication services such as video conferencing between terminals which have different codec.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.481-487
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• 2004

In this paper. the structure of bandwidth scalable wideband speech codec and its high-band codec are proposed. In the high-band codec. the signal is divided into frequency bands. and each band is quantized in DCT domain. The DCT coefficients are splitted into magnitude and sign, and each is quantized independently by a specialized method based on its characteristics. In addition. the quantized gain parameter in the low-band codec is utilized in the high-band codec for an enhanced performance. The bandwidth scalable wideband speech codec using G.729E for low-band and the proposed codec for high-band is developed, and it is confirmed that the proposed codec has better subjective performance than 24kbps G.722.1.

• The Journal of the Acoustical Society of Korea
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• v.26 no.3
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• pp.102-108
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• 2007

A high-Performance scalable speech codec generally requires a very low-rate first layer and a fine granule second layer, and this codec can be implemented with the harmonic codec and the MDCT-based transform codec for each layer. In this structure, however. each codec requires independent frequency transform and the time delay of each codec is accumulated. resulting in long time delay for the overall codec. In this paper, new MDCT structure in the second layer is Proposed. where MDCT is forced to share the look-ahead region of the first layer in order to prevent the time delay accumulation and the resulting functional error of MDCT is analyzed and removed after IMDCT The Proposed delayless MDCT requires no additional bits and Provides the equivalent coding performance with the reduced time delay, yielding a meaningful enhancement of the overall codec.

• Proceedings of the Korean Information Science Society Conference
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• 1999.10c
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• pp.632-634
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• 1999

기존의 화상전화 시스템 및 화상회의 시스템은 H/W로 구현된 H.263 CODEC을 사용한 경우는 있었지만, 컴퓨터의 성능변수와 직접적인 관계가 있는 S/W CODEC을 사용한 경우는 드물었다. 여기에 착안하여 본 연구는 PSTN 망에서의 화상회의 및 화상전화에 응용할 수 있는 실시간 H.263 S/W CODEC을 구현하였다. H.263 S/W CODEC을 MuX(Multimedia I/O Server)의 DLM으로 구현함으로써 현재 LAN 기반으로 동작하는 MuX를 PSTN망으로 확장하는데 있어 기반 기술로 사용할 수 있으며, 대역폭이 낮은 LAN 환경에서도 안정적인 화상회의가 가능하도록 하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.2
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• pp.369-374
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• 2018

The Line Code is a code used for data transmission and is a coding method used to prevent loss of data between transmitter and receiver. There are various kinds of line code such as B3ZS, HDB3, B8ZS, and HDB3 code is used in the tactical communication networks of Korea. In this paper, we implement the HDB3 Codec that is used in tactical communication networks in order to eliminate the risk factors for component discontinuance and to shorten development period and ensure reliability. Also it is confirmed that it can be implement by minimizing the amount of logic usage so that it can be used unrestrictedly in systems using FPGA and the implemented HDB3 Codec is simulated to confirm that it is equivalent to the performance of HDB3 Codec IC.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.395-401
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• 2005

In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.

• Broadcasting and Media Magazine
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• v.19 no.1
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• pp.48-55
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• 2014

One of the recent noticeable evolutions in mobile communication systems is that wideband-codec is deployed rapidly in VoLTE (Voice over Long Term Evolution) service or HD voice. This paper is concerned with next generation HD voice or VoLTE service that is coined to describe high quality communication with super-wideband voice codec. 3GPP EVS (Enhanced Voice Service) Codec is being standardized to develop the super-wideband voice codec. This paper deals with the codec design constraints, performance requirements, the status of standardization, and finally perspective on VoLTE service in future.

• Journal of Korea Society of Digital Industry and Information Management
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• v.10 no.4
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• pp.117-125
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• 2014

In this paper, a novel hardware architecture of the LT codec is presented where non-BP based decoding algorithm is applied. Novel LT codec architecture is designed with an efficient degree distribution unit using Verilog HDL. To perform permutation operation, different initial valued or time shifted counters have been used to get pretty well permutations and an effect of randomness. The codec will take 128 bits as input and produce 256 encoded output bits. The simulation results show expected performances as the implemented distribution and the original distribution are pretty same. The proposed LT codec takes 257.5 cycle counts and $2.575{\mu}s$ for encoding and decoding instead of 5,204,861 minimum cycle counts and 4.43s of the design mentioned in the previous works where iterative soft BP decoding was used in ASIC and ASIP implementation of the LT codec.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2003.06a
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• pp.197-200
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• 2003

본 논문에서는 IEEE 802.11a 무선 LAN 용 CODEC 회로를 설계하고, VHDL 코딩과 FPGA에 의한 회로설계 검증에 관해 기술한다. IEEE 802.lla WLAN CODEC의 구조는 크게 데이터 보호를 위한 스크램블러/디스크램블러, 채널 에러에 대한 정보보호를 위한 Convolutional 부호기와 Viterbi 복호기로 구성된 채널 코덱, 그리고 연집에러를 랜덤 에러로 변화시키는 인터리버/디인터리버로 구성된다. 본 논문에서는, 이와 같은 CODEC의 각 부분을 하드웨어로 구현하기 위한 새로운 회로구성을 제안하고, 그 성능을 VHDL 코딩에 의한 시뮬레이션과 FPGA에 의한 하드웨어 검증 결과를 제시한다.

• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.442-444
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• 2003

본 논문에서는 IEEE 802.113 무선 LAN 용 CODEC 회로를 설계하고, VHDL 코딩 과 FPGA에 의한 회로설계 검증에 관해 기술한다. IEEE 802.11a WLAN CODEC의 구조는 크게 데이터 보호를 위한 스크램블러/디스크램블러, 채널 에러에 대한 정보보호를 위한 Convolutional 부호기와 Viterbi 복호기로 구성된 채널 코덱, 그리고 연집에러를 랜덤 에러로 변화시키는 인터리버/디인터 리버로 구성된다. 본 논문에서는, 이와 같은 CODEC의 각 부분을 하드웨어로 구현하기 위한 새로운 회로구성을 제안하고, 그 성능을 VHDL 코딩에 의한 시뮬레이션과 FPGA에 의한 하드웨어 검증 결과를 제시한다.