• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.192-192
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• 2000

In this paper, a platoon merging is considered as a remote-controlled system with the state represented by a stochastic process. In this system, it becomes to encounter situations where a single decision maker controls a large number of subsystems, and observation and control signals are sent over a communication channel with finite capacity and significant transmission delays. Unlike classical estimation problem in which the observation is a continuous process corrupted by additive noise, there is a constraint that the observation must be coded and transmitted over a digital communication channel with finite capaci쇼. A recursive coder-estimator sequence is a state estimation scheme based on observations transmitted with finite communication capacity constraint. Using the coder-estimator sequence, the remote control station designs a feedback controller. In this paper, we introduce a stochastic model for the lead vehicle in a platoon of vehicles considering the angle between a road surface and a horizontal plane as a stochastic process. The simulation results show that the inter-vehicle distance and the deviation from the desired inter-vehicle distance are well regulated.

• The Journal of the Acoustical Society of Korea
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• v.20 no.8
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• pp.19-23
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• 2001

DSP-based implementation is pervasive in wireless communication parts for systems and handsets according to developing high-speed and low-power programmable Digital Signal Processor (DSP). In this paper, we present a real-time implementation of multi-channel Adaptive Multi-rate (AMR) speech coder. The real-time implementation of an AMR algorithm is achieved using 32-bit fixed-point TMS320C6202 DSP chip that operates at 250 MHz. We performed cross compile, linear assembly optimization and TMS320C62xx assembly optimization for real-time implementation. Furthermore, speech data input/output function and communication function with external CPU is included in an AMR speech coder. The AMR Speech coder developed using DSP EVM board was evaluated in ETRI IMT-2000 Test-bed system.

• Journal of Korea Multimedia Society
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• v.10 no.12
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• pp.1551-1558
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• 2007

The standard of evaluating function of speech coder for mobile telecommunication can be shown in channel capacity, noise immunity, encryption, complexity and encoding delay largely. This study is an algorithm to reduce complexity applying to CDMA(Code Division Multiple Access) mobile telecommunication system, which has a benefit of keeping the existing advantage of telecommunication quality and low transmission rate. This paper has an objective to reduce the computing complexity by controlling the frequency band nonuniform during the changing process of LSP(Line Spectrum Pairs) parameters from LPC(Line Predictive Coding) coefficients used for EVRC(Enhanced Variable-Rate Coder, IS-127) speech coders. Its experimental result showed that when comparing the speech coder applied by the proposed algorithm with the existing EVRC speech coder, it's decreased by 45% at average. Also, the values of LSP parameters, Synthetic speech signal and Spectrogram test result were obtained same as the existing method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.1 no.1
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• pp.77-83
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• 1997

In this paper, we present a comparison between QCELP(Qualcomm Code Excited Linear Predictor) speech coder that is operating in digital mobile communication system and CS-ACELP(Conjugate Structure Algebraic Code Excited Linear Prediction) speech coder that is scheduled to use for IMT-2000 (International Mobile Telecommunications 2000) system. The performance comparison might give help to design of the speech coding algorithms so that the robustness of the algorithms to channel errors engaged by mobile communication system be optimized.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.37 no.1
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• pp.66-75
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• 2000

In this paper, we present the efficient image transmission system using designed source-channel combined coder in W-CDMA mobile communication system. In proposed schemes, we decompose the wavelet transformed hierarchical band-images into some types of different size blocks which have different properties in error sensitivity. The RS(Reed-Solomon) coder with different coding rate is used for each decomposed source blocks which has different importance. In addition, we combine retransmitted error frames in Truncated Hybrid Type I ARQ. The proposed algorithm shows efficient image transmission methods because it is not much degraded in PSNR compared with the existing not combined source-channel coder in erroneous wireless channel.

• MALSORI
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• no.54
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• pp.27-43
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• 2005

Existing standard speech coders can provide speech communication of high quality while they degrade the performance of speech recognition systems that use the reconstructed speech by the coders. The main cause of the degradation is that the spectral envelope parameters in speech coding are optimized to speech quality rather than to the performance of speech recognition. For example, mel-frequency cepstral coefficient (MFCC) is generally known to provide better speech recognition performance than linear prediction coefficient (LPC) that is a typical parameter set in speech coding. In this paper, we propose a speech coder using MFCC instead of LPC to improve the performance of a server-based speech recognition system in network environments. However, the main drawback of using MFCC is to develop the efficient MFCC quantization with a low-bit rate. First, we explore the interframe correlation of MFCCs, which results in the predictive quantization of MFCC. Second, a safety-net scheme is proposed to make the MFCC-based speech coder robust to channel error. As a result, we propose a 8.7 kbps MFCC-based CELP coder. It is shown from a PESQ test that the proposed speech coder has a comparable speech quality to 8 kbps G.729 while it is shown that the performance of speech recognition using the proposed speech coder is better than that using G.729.

• Journal of Korea Society of Digital Industry and Information Management
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• v.18 no.2
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• pp.27-37
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• 2022

In this paper, we presents a Multiple User Multiple Input Multiple Output (MU-MIMO) test-bed system for Time Division Duplex (TDD) Long Term Evolution-Advanced (LTE-A). Using two parameters, the condition number of the channel matrix and the path gain, the MU-MIMO system could switch pre-coder to maintain target Bit Error Rate (BER) level. This paper also introduces a calibration procedure for compensating error of Radio Frequency (RF) paths of the antennas and RF transceivers. From experimental measurements, dynamic pre-coding scheme could maintain target BER, set to 10^-3, with the pre-coder set configured with Zero Forcing (ZF), Tomlinson Harashima Pre-coding (THP), Lattice Reduction (LR). The simplest pre-coder ZF is adopted in stable channel, and when path gain become less than 0.25, LR is adopted. Lastly, when condition number of channel matrix become larger than 7, THP is adopted.

• Journal of Broadcast Engineering
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• v.8 no.2
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• pp.163-171
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• 2003

In this paper, we develop an approach toward Joint Source-Channel Coding for MPEG-4(Moving picture Experts Group) based video coding In fixed and mobile reception environment. We have considered channel environment of AWGN and mobile reception. The source coder used MPEG-4 video. the channel coder used RCPC(Rate Compatible Punctured Convolution) and the modulation method used QPSK(Quadrature Phase Shift Keying) modulation. This study determined optimum Trade-off point between source bit rate and channel coding rate In variable channel states. We compared Joint Source/channel Coding method and general constant bit rate transmission. In this results, Joint Source/channel Coding was appeared better performance than constant bit rate transmission.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.7 no.2
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• pp.47-54
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• 1982

In this paper, a improved per-channel PCM Coder with 1-bit interpolation is proposed. The coder converts a telephone signal to 15-segments u-law PCM signal of a large dynamic range. The A/D conversion technique of the proposed converter requires a feedback loop around a quantizer operates at high speed, and a accumulater for accumulating the quantized values to provide PCM outputs. To obtain both linear and compressed PCM signals a improved table look-up method is presented. The operations of the proposed converter are certified through the experiments to be good. The experimental circuit comprises TTL logic gates, a resistive D/Z converter and a simple differential amplifier. From the results of the experiments, it is known that the proposed converter has many advantage to be adopted economically for per-channel onverter used in rural area service.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.261-269
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• 2005

Throughout the history of digital communication, the digital speech coder is used as speech compression tool. Nowadays, the speech coder has been rapidly developed in the area of mobile communication system to overcome severe channel error and limitation of radio frequency resources. Due to the development of high performance communication system, high quality of speech coder is needed. This kind of speech coder can be used not only in communication services but also in digital multimedia services. In this paper, we describe the technologies of digital speech coder which are used in wire and wireless communication. We also present a summary of recent speech coding standards for narrowband and wideband applications. Finally we introduce the technical trends of next generation speech coder.