• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.4 no.5
• /
• pp.963-969
• /
• 2000

In this paper, we define two input-output gains of linear periodic time-varying systems. One is the ratio of output with worst-case l2-norm over all inputs with unit 12-norm. It denotes G($\iota_2,\iota_2$.The other is the ratio of output with worst-case RMS value over all inputs with unit RMS value. It denotes G(RMS, RMS) .It is fact that these two gains are equivalent for linear time-invariant system. In this paper, we prove these two gains are also equivalent for linear periodic time-varying system. In addition, the relationship between two method of obtaining the generalized frequency responses for linear periodic time-varying system is derived. Finally, we apply the defined input-output gains to M-channel filter-bank which is multi-rate signal Processing system, used to speech coding. In the filter-bank, generally, aliasing distortion, magnitude distortion, and phase distortion are present. It is shown that these are kept small if the filter-bank is designed by a method that optimizes the gain G($\iota_2,\iota_2$ of an error system.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.2 no.3
• /
• pp.335-341
• /
• 1998

In this paper, we present a real time implementation of CS-ACELP(Conjugate Structure Algebraic Code Excited Linear Prediction) speech coder. ITU-T has standardized the CS-ACELP algorithm as G.729. Areal-time implementation of CS-ACELP speech coder algorithm is achieved using 16 bit fixed-point DSP chip. To implement in fixed-point DSP Chip, integer simulation of CS-ACELP algorithm is used. Furthermore. input/output function and communication function included in CS-ACELP speech coder is described. We develope CS-ACELP speech coder in DSP evaluation board and evaluate in IMT-2000 Test-bed.

• Proceedings of the Korea Contents Association Conference
• /
• 2018.05a
• /
• pp.189-190
• /
• 2018

Digital content requires the concept and structure that give us insights into the languages between computers and humans and how humans experience manifested among the flow of characters, images, and voice. Communicology, $Vil{\acute{e}}m$ Flusser's original study, allows us to reconsider and to reconstruct the boundary of human awareness. This paper intends to begin understanding digital content consisting of numerical codes by reviewing communicology. communicology helps to break up pre-existing categories and thinking about new standards. ith the help of information technology. Planning content can be actualized by classifying and reconstructing content that are input/output of senses. The standard of classification is 'boundary' and 'direction,' communication elements that cannot be broken down any further. There is no need to communicate if there is no boundary. The operation of communication is comprised of 'direction.' Considering humankind as the standard, the boundary that takes in stimulation from outside can be seen as senses. Direction can be expressed as input/output. Output assumes that technical pictures receive information. The coordinates for various pre-existing platforms and content and uncovered platforms can be set with a consistent standard. This allows us to escape from the standard of flat content that was activated by sight and rationality at the ideology of characters, to seek a three-dimensional standard that can be vitalized by various senses and irrationality, and to reconstruct the input/output of senses to show the possibility of planning a new platform.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.19 no.5
• /
• pp.920-927
• /
• 1994

A development of an efficient 14.0~14.5GHz 3 Watt SSPA is described in this paper, which is applicable to the very small aperture terminal(VSAT) for bidirectional data and voice signal transmission in low cost and with small size. The SSPA consists of two stages of low noise amplifiers using the low noise GaAs FETs. two stages of medium power amplifiers using the medium power GaAs FETs, and three stages of power amplifiers including a balanced amplifier using an internally matched power GaAs FET. The achieved with this seven stage amplifiers are 42dB signal power gain, 7dB noise figure, 35dBm output power at 1dB gain compression point and 2.0 and 1.5 input and output VSWR respectively.

• The Journal of the Acoustical Society of Korea
• /
• v.23 no.3
• /
• pp.262-267
• /
• 2004

AMR-WB (Adaptive Multi Rate Wideband) speech codec, the most recent voice codec standardized by 3GPP, has the wider audio bandwidth of 50∼7000 Hz and operates on nine speech coding bit rates between 6.60 and 23.85 kbit/s. This Paper presents the real-time implementation of AMR-WB speech codec by using a 16 bit fixed-point TeakLite DSP. The implemented AMR-WB codec requires the complexity of 52.2 MIPS at 23.85 kbit/s mode and also needs the program memory of 17.9 kwords, data RAM of 11.8 kwords, and data ROM of 10.1kwords. It was verified through passing the all test vectors provided by 3GPP with maintaining bit exactness. Stable operations on the real-time testing board were also proved without any distortions and delays for the audio in/out.

• Journal of Internet Computing and Services
• /
• v.6 no.1
• /
• pp.95-103
• /
• 2005

Recently researches about multi-modal system has been studied widely and actively, Such multi-modal systems are enable to increase possibility of HCI(Human-computer Interaction) realization, enable to provide information in various ways and also enable to be applicable in e-business application, If ideal multi-modal system can be realized in future, eventually user can maximize interactive usability between information instrument and men in hands-free and eyes-free, In this paper, a new multi-modal browsing system using Java RMI as communication interface, which integrated by HTML browser and voice browser is suggested and also English-English dictionary search application system is implemented as example.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 1999.11b
• /
• pp.261-266
• /
• 1999

본 논문에서는 인터넷 폰이나 디지털 이동통신에서와 같이 낮은 비트율이 요구되는 응용분야에서 사용될 수 있는 HVXC 부호화 및 복호화 알고리즘을 TMS320C6701 160MHz DSP를 사용하여 실시간 동작을 구현한 내용을 기술한다. 사용한 최적화 방법으로는 기본적으로 연산 시간이 많이 소요되는 함수 루틴에 대한 C 언어레벨의 최적화 및 어셈블리어 레벨의 최적화를 수행하였고, TMS320C6701 DSP 내부 프로그램 메모리를 프로그램 캐쉬로 사용하였다. 또한, 계산량이 많은 부분과 테이블 참조가 필요한 연산을DSP의 내부 데이터 메모리 영역에서 수행하여 소요시간을 단축하였으며, 음성신호 및 비트스트림의 입출력에는 background DMA(direct memory access) 방식을 이용하였다. 이와 같은 최적화결과 2kbps 및 4kbps의 비트율에서 압축 및 복원을 실시간으로 수행할 수 있다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2018.05a
• /
• pp.495-496
• /
• 2018

As more and more companies introduce artificial intelligent(AI) speakers, the price of the speakers has become a burden to someone. Based on some knowledge and dexterity, it is not difficult to make an AI speaker that acquires sensor information and environmental information of the house in accordance with your own taste. In this paper, we implement an AI speaker using Raspberry Pie, Google Cloud Speech (GCS) and Naver's Clova Speech Synthesis (CSS) API.

• Proceedings of the Acoustical Society of Korea Conference
• /
• autumn
• /
• pp.13-16
• /
• 2001

본 논문에서는 3GPP 규격에 따른 IMT-2000 시스템용 UE AMR 코덱의 소프트웨어 및 하드웨어 개발에 관하여 논한다 UE AMR 코덱은 ASIC 개발을 고려하여 Teak-Lite DSP 칩을 사용하여 개발하였다 AMR 코덱을 구현하기 위한 효율적인 소프트웨어 개발 기법을 설명하고 하드웨어 디자인도 논한다 개발된 UE AMR 코덱에는 음성 데이터 입출력 기능은 물론 리부 호스트 프로세서와의 통신 기능도 포함된다. Teak-Lite EVM보드를 사용하여 실시간으로 동작하는 AMR 코덱 소프트웨어를 개발하였다. 또한 동시에 UE AMR 코덱용 하드웨어도 개발하였다. ETRI에서 개발 및 시험 중인 IMT-2000 시스템 상에서 개발한 UE AMR 코덱의 동작 및 기능을 검증하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2013.10a
• /
• pp.922-923
• /
• 2013

최근 고령화가 증가하면서 헬스케어기술도 함께 발달해 가고 있다. 또한 스마트폰이 발전해 가면서 헬스케어기술과 함께 융합한 기술이 많이 연구되어 가고 있다. 기존의 대형화 되었던 기술들이 스마트폰을 통해 제작되면서 생체센서만 부착되면 U-헬스케어 기술이 구현될 수 있는 세상이 실현된 것이다. 본 논문에서는 스마트폰에서 공용으로 부착되어 있는 오디오(AUX) 단자를 사용해 생체신호를 입력받았다. 일반적으로 스마트폰의 오디오단자는 음성의 입출력을 할 수 있도록 설계되었으나 오디오의 마이크 단자를 활용할 경우 생체신호를 입력 받을 수 있다. 본 연구에서는 PPG회로를 구현하고 오디오 단자를 통해 입력받은 생체신호를 애플리케이션을 통해 모니터링 하는 프로그램을 제작하였다.