• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.4 no.2
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• pp.37-43
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• 2005

A simple scheduler satisfying the proportional fairness (PF) was introduced in wireless access networks and revealed that it can achieve a good compromise between total throughput and user fairness. Though it has received much attention for some time, its application was mainly restricted to the single channel systems. In this paper, we study how to implement the PF scheduler in the multi-channel environments such as OFDMA/TDMA. Besides the traditional PF-SC scheme, we propose a new PF-OPT scheme that is the genuine PF scheduler in a sense of maximizing the total log-utility of users. The simulation results show that PF-OPT gives large throughput under the heterogeneous subchannel statistics.

• The Journal of the Acoustical Society of Korea
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• v.17 no.4E
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• pp.17-21
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• 1998

This paper presents an implementation of a multi-channel speech surveillance system over telephone lines using TMS320C31 DSP chips. The incoming speech into each telephone line are first compressed simultaneously in real-time by the popular vector-sum excited linear predictive (VSELP) speech coding algorithm at the rate of 8 Kbps. The compressed steech bit streams are then multiplexed with those of other users. The multiplexed speech bit streams are transferred to the system storage equipments with some other required information so that a system operator can later monitor the stored speech data whenever it is necessary. The host program runs under Microsoft Windows95 for an efficient man-machine interface and a future upgrade-ability. We have confirmed that the overall 64-channel system operates satisfactorily in realtime. We also have checked approximately up to 2,880 total hours of recording capability of the system on a playback module and two removable backup drives.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.8
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• pp.2033-2042
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• 1996

In this papre, we describe a recently implemented echo canceller for digital cellular communication of Code Division Multiple Access(CDMA) that features time sharing of digital signal processor(DSP) over four channels in one DSP to reduce per channel costs. In the Public Land Mobile Network(PLMN), it is important to cancel the echo reflected from the Public Switched Telephone Network(PSTN) side. In case of digital mobile system, the round-trip delay of the echo is in excess of about 180 milliseconds due to frame-by-frame voice coding. It is necessary to cancel the echo in PLMN. We have developed a multi-channel echo canceller tht operates with Time Switch Module in a Mobile Switching Center(MSC). The general echo canceller needs PCM trunk interface circuits and the tone detection and disabling circuits, but the multi-channel echo canceller linked with Time Switch Module does not need them. Therefore we could develop the effective and economical echo canceller.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1211-1214
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• 1998

Real-time implementation of Conjugate-Structure Algebraic CELP(CS-ACELP) is presented. ITU-T Study Group(SG) 15 has standardized the CS-ACELP speech coding algorithm as G.729. A real-time implementation of the CS-ACELP is achieved using 16 bit fixed point DSP16210 Digital Signal Processor (DSP) of Lucent Technologies. The speech coder has been implemented in the bit-exact manner using the fixed point CS-ACELP C source which is the part of the G.729 standard. To provide a multi-channel vocoder solution to digital communication system, we try to minimize the complexity(e.g., MIPS, ROM, RAM) of CS-ACELP. Our speech coder shows 15.5 MIPS in performance which enables 4 channel CS-ACELP to be processed with one DSP16210.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.92-97
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• 2000

An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.3
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• pp.737-746
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• 1996

In this paper, we present simulation result of quantization effects about a new Angular From Carrier Recovery Loop(AF CR loop) for PSK modulation technique. AF CR loop includes detected angle symbol and Multi Level hardimiter. In general, detected angle is used in dtermining symbol. Because detected angle is used to make an error signal of phase detector output, hardware implementation of AF CR loop is simpler than that of other loops. Before hardware implementation of AF CR loop, the result due to quantization effect should be investigated. In order to confirm quntization effect of AF CR loop, we evaluate performance of this loop by Monte-Carlosimulation method. Under both in the AWGN and Jake's fading noise channel environments, we confirmed the characteristics of AF CR loop in terms of RMS jitter due to quntization effect. Differential APSK modulation schemeis used in this paper. Especially, Jake's fading channel is used as a channel model and also AGC(Automatic Gain Control) is used in the overall process of performance evaluation. We obtained the resonable result of quantization effect about AF CR loop. With the result of performanceevaluation based on quantization effects, we can expect to operate AF CRloop under the fading channel environments reasonably well.

• ETRI Journal
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• v.41 no.6
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• pp.863-872
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• 2019

Elliptic curve cryptography is a relatively lightweight public-key cryptography method for key generation and digital signature verification. Some lightweight curves (eg, Curve25519 and Curve Ed448) have been adopted by upcoming Transport Layer Security 1.3 (TLS 1.3) to replace the standardized NIST curves. However, the efficient implementation of Curve Ed448 on Internet of Things (IoT) devices remains underexplored. This study is focused on the optimization of the Curve Ed448 implementation on low-end IoT processors (ie, 8-bit AVR and 16-bit MSP processors). In particular, the three-level and two-level subtractive Karatsuba algorithms are adopted for multi-precision multiplication on AVR and MSP processors, respectively, and two-level Karatsuba routines are employed for multi-precision squaring. For modular reduction and finite field inversion, fast reduction and Fermat-based inversion operations are used to mitigate side-channel vulnerabilities. The scalar multiplication operation using the Montgomery ladder algorithm requires only 103 and 73 M clock cycles on AVR and MSP processors.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.3
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• pp.32-38
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• 2001

Multi-carrier data transmission system performance is mostly limited by Inter- symbol-interference that is caused by a dispersive characteristic of the transmission channel. In order to enhance the system performance to meet the service requirements of local access, the channel impulse response shortening method incorporated with a channel frequency response compensation method is proposed. For a fast and efficient implementation of the equalizer proposed, Kalman and LMS algorithms are successively used. To verify the channel equalization performance, a set of computer simulation is performed on a filter bank based multitone system operating in a typical high-speed local area data transmission environment. The results showed us a comparable signal-to-interference improvement over the conventional multitone equalization scheme.

• Journal of Communications and Networks
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• v.14 no.6
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• pp.710-718
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• 2012

The implementation of a common control channel is one of the most challenging issues in cognitive radio networks, since a fully reliable control channel cannot be created without reserving bandwidth specifically for this purpose. In this paper, we investigate a promising solution that exploits the ultra wide band (UWB) technology to let cognitive radio nodes discover each other and exchange control information for establishing a communication link. The contribution of this paper is threefold: (i) We define the communication protocol needed to let cognitive radio nodes discover each other and exchange control information for link set up, (ii) we overcome the gap in coverage, which typically exists between UWB and long-medium range technologies, by using multi-hop communication, (iii) we evaluate the performance of our approach by adopting an accurate channel model and show its benefits with respect to an in-band signaling solution.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.378-381
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• 2003

The Channel Coding Unit (CCU) is an integral component of Payload Data Transmission System (PDTS) for the Multi-Spectral Camera (MSC) data. The main function of the CCU is channel coding and encryption. CCU has two channels (I ＆ Q) for data processing. The input of CCU is the output of DCSU (Data Compression ＆ Storage Unit). The output of CCU is the input of QTX which modulate data for RF communication. In this paper, there are the overview, short H/W description and operation concept of CCU.