• Journal of the Institute of Electronics Engineers of Korea TE
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• v.37 no.5
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• pp.36-47
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• 2000

The CDMA digital cellular systems provide better voice Quality than analog systems, however there exists inherent delays due to speech coding and transmission processing, which brings echoes returned by the BSC and PSTN interface. In this paper, we show the performance improvement of a proposed echo canceller by real time implementation, where Block Update NLMS algorithm is applied into the TMS320C54X DSP. By applying the proposed method into the practical mobile phone, we verify that various types of echoes (LE, ESE, AE) may be removed more precisely. We also cope with echo path change resulting from change of delay length after taking VAD to find echo path delay.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.2
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• pp.118-123
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• 1989

In this paper, as we have integrated SCF for voice signal processing using CMOS circuit with the low power dissipation and the easy circuit design, it has been presented the simplified CMOS OP AMP design method with ${\pm}$5V pwoer source in order to use together with digital part. After an example about SCF CMOS OP AMP design, it has been performed layout appling channel width and length obtained by design method, and then its characteristics were simulated by SPICE 2G program. Therefoe, this design method will be applied the general CMOS OP AMP design in the electronic circuit.

• Electronics and Telecommunications Trends
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• v.33 no.5
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• pp.121-134
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• 2018

The Von Neumann based architecture of the modern computer has dominated the computing industry for the past 50 years, sparking the digital revolution and propelling us into today's information age. Recent research focus and market trends have shown significant effort toward the advancement and application of artificial intelligence technologies. Although artificial intelligence has been studied for decades since the Turing machine was first introduced, the field has recently emerged into the spotlight thanks to remarkable milestones such as AlexNet-CNN and Alpha-Go, whose neural-network based deep learning methods have achieved a ground-breaking performance superior to existing recognition, classification, and decision algorithms. Unprecedented results in a wide variety of applications (drones, autonomous driving, robots, stock markets, computer vision, voice, and so on) have signaled the beginning of a golden age for artificial intelligence after 40 years of relative dormancy. Algorithmic research continues to progress at a breath-taking pace as evidenced by the rate of new neural networks being announced. However, traditional Von Neumann based architectures have proven to be inadequate in terms of computation power, and inherently inefficient in their processing of vastly parallel computations, which is a characteristic of deep neural networks. Consequently, global conglomerates such as Intel, Huawei, and Google, as well as large domestic corporations and fabless companies are developing dedicated semiconductor chips customized for artificial intelligence computations. The AI Processor Research Laboratory at ETRI is focusing on the research and development of super low-power AI processor chips. In this article, we present the current trends in computation platform, parallel processing, AI processor, and super-threaded AI processor research being conducted at ETRI.

• Journal of Institute of Control, Robotics and Systems
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• v.11 no.5
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• pp.471-475
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• 2005

Recently, development of digital technology is occupying a large part of multimedia information like character, voice, image, video, etc. Research about video indexing and retrieval progresses especially in research relative to video. This paper proposes the novel notation in order to retrieve MPEG video in the international standards of moving picture encoding For realizing the retrieval-system, we detect DCT DC coefficient, and then we obtain shot to apply MVC(Mean Value Comparative) notation to image constructed DC coefficient. We choose the key frame for start-frame of a shot, and we have the codebook index generating it using feature of DC image and applying PCA(principal Component Analysis) to the key frame. Also, we realize the retrieval-system through similarity after indexing. We could reduce error detection due to distinguish shot from conventional shot detection algorithm. In the mean time, speed of indexing is faster by PCA due to perform it in the compressed domain, and it has an advantage which is to generate codebook due to use statistical features. Finally, we could realize efficient retrieval-system using MVC and PCA to shot detection and indexing which is important step of retrieval-system, and we using retrieval-system over the internet.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7A
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• pp.659-667
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• 2002

In this dissertation, in the Turbo Code used for error correction coding of the recent digital communication systems, we propose a new S-R interleaver that has the better performance than the existing block interleaver, and the Turbo Decoder that has the parallel concatenated New structure using the MAP algorithm. For real-time voice and video services over the third generation mobile communications, the performance of two proposed methods is analyzed by the reduced decoding delay using the variable decoding method by computer simulation over multipath channels of DS-CDMA system. Also, a Modified NCMA based on conventional NCMA is proposed to improve the channel efficiency in the mobile communication system, and is investigated over the multi-user environment of DS-CDMA system through computer simulation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.3 no.1
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• pp.97-104
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• 1999

In this study we show a telemedicine system using a DSP, which gives fast and exact medical data such as the ECG signal of the external emergency patient and enables the patient to get temporary treatments under the direction of a doctor in the hospital. This transceiver, which is able to treat the real time transmission of dynamic medical data captured by the measuring instrument and bidirectional communication of voice signal, is implemented using DSB-SC as modulation and demodulation technique and digital filter of each terminal are implemented as FIR filter. The system designed with DSP in this study is very small and compact and it can k, furthermore, to support additional biomedical signals just by renewing the software.

• Proceedings of the KIEE Conference
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• 1996.07b
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• pp.1295-1297
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• 1996

CCITT G.721, G.723 standard ADPCM algorithm is implemented by using TI's fixed point DSP start kit (DSK). ADPCM can be implemented on a various rates, such as 16K, 24K, 32K and 40K. The ADPCM is sample based compression technique and its complexity is not so high as the other speech compression techniques such as CELP, VSELP and GSM, etc. ADPCM is widely applicable to most of the low cost speech compression application and they are tapeless answering machine, simultaneous voice and fax modem, digital phone, etc. TMS320C50 DSP is a low cost fixed point DSP chip and C50 DSK system has an AIC (analog interface chip) which operates as a single chip A/D and D/A converter with 14 bit resolution, C50 DSP chip with on-chip memory of 10K and RS232C interface module. ADPCM C code is compiled by TI C50 C-compiler and implemented on the DSK on-chip memory. Speech signal input is converted into 14 bit linear PCM data and encoded into ADPCM data and the data is sent to PC through RS232C. The ADPCM data on PC is received by the DSK through RS232C and then decoded to generate the 14 bit linear PCM data and converted into the speech signal. The DSK system has audio in/out jack and we can input and out the speech signal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12C
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• pp.273-279
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• 2001

This paper describes a design and implementation of ISDN system on a chip which provides ISDN service and used to develop a low-price multimedia communication terminal. This ISDN SOC is an ISDN system control chip which has 32bit RISC processor, and it includes ISDN S interface transceiver, G.711 voice CODEC, PC interface for data communication, ISDN protocol which includes Q.931 call control protocol and internet protocol. It provides good solution to develope ISDN terminal equipment and ISDN terminal adaptor which connected with basic rate interface, because it minimize external peripheral devices.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2004.05b
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• pp.78-81
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• 2004

As a society has Progressed rapidly toward a highly advanced digital information age, a multimedia communication service for acquisition, transmission and storage of image data as well as voice has being commercialized externally and internally. However, in the process of digitalization or transmission of data, noise is generated by several causes, and researches for eliminating those noises have been continued until now. The mean filter is useful method to remove AWGN (additive white gaussian noise) from degraded image and has excellent low-frequency properties. However, it brings about degradation of high-frequency properties in image. So in this paper we removed noise with mean filters added directional information and minimized degradation of high-frequency properties.

• Proceedings of the Korea Information Processing Society Conference
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• 2010.04a
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• pp.782-785
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• 2010

본 논문은 연속된 음성 신호를 전송로 상에 전송하기 위해 음성 신호를 G.711 표준 권고인 PCM으로 다중화한 후 고정 타임슬롯을 배정하여 전송하는 시스템에서 PCM 화된 디지털 음성 데이터를 실시간으로 암호화하여 전송하는 스트림 암호화 기법에 관한 것이다. 실시간으로 처리되는 음성 데이터의 암호화 시에는 하드웨어 방식이 적합한 데, 본 논문에서는 고정 타임슬롯을 배정받는 음성 데이터의 실시간 암호화 기법에 관한 것이다. 일반적으로 아날로그 음성 신호 코딩 시에 국내에서는 북미 방식인 ${\mu}-law$ 코딩 기법을 적용하는 데 이는 표본화한 음성 데이터를 양자화전에 압축하고 복호화 후 신장하는 비선형 양자화 기법을 적용하는 것으로 표본화된 값을 8 비트의 PCM 데이터로 변화하여 E1(2.048Mbps) 급 속도로 전송한다. 본 논문에서는 PCM 전송로 상에 전송되기 전의 직렬 입력 데이터를 암호화 장치를 거쳐 해당 타임슬롯에 해당하는 8 비트의 데이터를 실시간으로 암호화하여 전송로 상으로 전송하고 역으로 수신 단에서는 PCM 전송로를 거친 직렬 입력 데이터를 암호화된 타임슬롯을 판별하여 해당 타임슬롯의 데이터를 복호화하여 원래 데이터를 복원한다. 본 논문에서는 고정 타임슬롯을 배정받은 PCM 데이터를 암호화하여 전송한 후 수신 단에서 복호화 과정을 거친 후 타임슬롯 단위로 데이터 암호화/복호화가 가능함을 보여준다.