• Journal of Communications and Networks
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• v.9 no.4
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• pp.473-481
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• 2007

Energy efficiency is one of the most critical concerns for wireless sensor networks. By allowing sensor nodes in close proximity to cooperate in transmission to form a virtual multiple-input multiple-output(MIMO) system, recent progress in wireless MIMO communications can be exploited to boost the system throughput, or equivalently reduce the energy consumption for the same throughput and BER target. However, these cooperative transmission strategies may incur additional energy cost and system overhead. In this paper, assuming that data collectors are equipped with antenna arrays and superior processing capability, energy efficiency of relevant traditional and cooperative transmission strategies: Single-input-multiple-output(SIMO), space-time block coding(STBC), and spatial multiplexing(SM) are studied. Analysis in the wideband regime reveals that, while receive diversity introduces significant improvement in both energy efficiency and spectral efficiency, further improvement due to the transmit diversity of STBC is limited, as opposed to the superiority of the SM scheme especially for non-trivial spectral efficiency. These observations are further confirmed in our analysis of more realistic systems with limited bandwidth, finite constellation sizes, and a target error rate. Based on this analysis, general guidelines are presented for optimal transmission strategy selection in system level and link level, aiming at minimum energy consumption while meeting different requirements. The proposed selection rules, especially those based on system-level metrics, are easy to implement for sensor applications. The framework provided here may also be readily extended to other scenarios or applications.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.9
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• pp.1293-1298
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• 2013

This paper proposes a noise reduction algorithm using Wiener filter to remove the noise components from the noisy speech in order to improve the speech signal. The proposed algorithm first removes the noise spectrums of white noise from the noisy signal based on the noise reshaping and reduction method at each frame. And this algorithm enhances the speech signal using Wiener filter based on linear predictive coding analysis. In this experiment, experimental results of the proposed algorithm demonstrate using the speech and noise data by Japanese male speaker. Based on measuring the spectral distortion (SD) measure, experiments confirm that the proposed algorithm is effective for the speech by contaminated white noise. From the experiments, the maximum improvement in the output SD values was 4.94 dB better for white noise compared with former Wiener filter.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.16 no.5
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• pp.239-246
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• 2023

In this paper, we propose an open-loop algorithm to classify speech and music signals using the spectral flux parameters and Mel Frequency Cepstral Coefficients(MFCC) parameters for the audio coder. To increase responsiveness, the MFCC was used as a short-term feature parameter and spectral fluxes were used as a long-term feature parameters to improve accuracy. The overall voice/music signal classification decision is made by combining the short-term classification method and the long-term classification method. The Gaussian Mixed Model (GMM) was used for pattern recognition and the optimal GMM parameters were extracted using the Expectation Maximization (EM) algorithm. The proposed long-term and short-term combined speech/music signal classification method showed an average classification error rate of 1.5% on various audio sound sources, and improved the classification error rate by 0.9% compared to the short-term single classification method and 0.6% compared to the long-term single classification method. The proposed speech/music signal classification method was able to improve the classification error rate performance by 9.1% in percussion music signals with attacks and 5.8% in voice signals compared to the Unified Speech Audio Coding (USAC) audio classification method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.8
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• pp.1675-1686
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• 2011

We propose a spectrally efficient relaying scheme with multiple antennas for cellular networks which consist of base station (BS), relay station (RSs), and mobile stations (MSs). In general, a BS has more antennas than an RS or an MS. By using multiple antennas, a BS can support another MS with the same frequency resource, where an RS transmits data to a specific MS. In this case, the MS receiving data form an RS also receives the interference form the BS because the BS also uses the same frequency resource at the same time. In this paper, we propose ing and pre-whitening techniques as a pre-coding scheme at the BS for reducing the interference at the MS receiving data from the RS.

• Journal of Communications and Networks
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• v.7 no.2
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• pp.157-170
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• 2005

The combination of multiple antennas and multi-carrier code division multiple-access (MC-CDMA) is a strong candidate for the downlink of the next generation mobile communications. The study of such systems in scenarios that model real-life trans-missions is an additional step towards an optimized achievement. We consider a realistic MIMO channel with two or four transmit antennas and up to two receive antennas, and channel state information (CSI) mismatches. Depending on the mobile terminal (MT) class, its number of antennas or complexity allowed, different data-rates are proposed with turbo-coding and asymptotic spectral efficiencies from 1 to 4.5 bit/s/Hz, using three algorithms developed within the European IST-MATRICE project. These algorithms can be classified according to the degree of CSI at base-station (BS): i) Transmit space-frequency prefiltering based on constrained zero-forcing algorithm with complete CSI at BS; ii) transmit beamforming based on spatial correlation matrix estimation from partial CSI at BS; iii) orthogonal space-time block coding based on Alamouti scheme without CSI at BS. All presented schemes require a reasonable complexity at MT, and are compatible with a single-antenna receiver. A choice between these algorithms is proposed in order to significantly improve the performance of MC-CDMA and to cover the different environments considered for the next generation cellular systems. For beyond-3G, we propose prefiltering for indoor and pedestrian microcell environments, beamforming for suburban macrocells including high-speed train, and space-time coding for urban conditions with moderate to high speeds.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.8 no.4
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• pp.342-353
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• 1997

This paper presents an analytical evaluation of a hybrid direct-sequence/slow frequencyhopped code division multiple-access (DS/SFH-CDMA) system employing noncoherent M-ary frequency shift keying(MFSK) modulation in a multiple Nakagami fading (m) environment. Multipath interference (MPI) and multi-access interference (MAI) is taken into account and the spectral efficiency is calculated for uncoded as well as channel coding systems. Predetection multipath maximal ratio combining (MRC) diversity in conjunction with interleaved channel coding (Hamming(7,4) code, BCH(15, 7) code and RS (7, 4), (15, 9)) code ) is employed for improving the bit error rate (BER) performance. The BER of noncoherent hybrid system is obtained using a Gaussian interference approximation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11C
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• pp.889-895
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• 2010

MIMO system generally requires more than one antenna at the communication device. However, many wireless devices are limited by size, cost or hardware complexity to one antenna. To overcome such restrictions, we used a new technique, called cooperative communication. We propose a new cooperative transmission strategy system using cross-layer coding method base on H-ARQ for optimal communication. Proposed cooperative H-ARQ system that can improve the above problems and can get the better performance. In proposed cooperative system with H-ARQ method, if the received signal from source node is satisfied by the destination preferentially, the destination transmit ACK message to both relay node and source node, and then recovers the received signal. In addition, if ARQ message indicates NACK message, relay node operates selective retransmission. Based on the simulation results in aspect to BER performance and throughput, the proposed method which combined cooperative system with H-ARQ based on cross-layer coding can improve spectral efficiency reliability of system compared with that of general one by one system.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4
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• pp.29-34
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• 2001

In this paper, we designed an LSF coefficient quantizer of the wideband speech codec that can produce high quality speech service. For the efficient LSF coefficient quantizer, the interframe correlation was used. Also we separately quantized the LSF coefficients with high and low interframe correlation. Predictive pyramid vector quantizer (PVQ) was used for quantizing the LSF coefficients with high interframe correlation, and PVQ was used for quantizing the LSF coefficients with low interframe correlation. Experiments show that the proposed UF quantizer can quantize LSF information in 40 bits/frame, with an average spectral distortion (SD) of 1 dB and less than 3.87% frames having SD greater than 2 dB.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.14 no.4
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• pp.247-256
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• 2021

In this paper, we propose a packet loss concealment (PLC) algorithm using pitch harmonic motion prediction and adaptive signal amplitude prediction and. The spectral motion prediction method divides the spectral motion of the previous usable frame into predetermined sub-bands to predict and restore the motion of the lost signal. In the proposed algorithm, the speech signal is classified into voiced and unvoiced sounds. In the case of voiced sounds, it is further divided into pitch harmonics using the pitch frequency to predict and restore the pitch harmonic motion of the lost frame, and for the unvoiced sound, the lost frame is restored using the spectral motion prediction method. When the continuous loss of speech frames occurs, a method of adjusting the gain using the least mean square (LMS) predictor is proposed. The performance of the proposed algorithm was evaluated through the objective evaluation method, PESQ (Perceptual Evaluation of Speech Quality) and was showed MOS 0.1 improvement over the conventional method.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.8 no.3
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• pp.221-231
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• 1997

This paper presents an analysis of a hybrid direct-sequence/slow frequency hopped code division multiple access(DS/SFH-CDMA) system employing noncoherent M-ary frequency shift keying(MFSK) modulation in a multiple m-distribution fading environment. Multipath interfer- ence(MPI) and multiuser interference(MUI) is taken into accout and the spectral efficiency is calculated for uncoded as well as simple channel coding systems. The predetection multipath CCI canceller in conjunction with convolution coding is employed for improving the bit error rate(BER) performance. The BER of noncoherent hybrid system is obtained using a Gaussian interference approximation. From the results, we know that the error performance more deteriorates as the depth of fading becomes deeper. The DS part of the modulation combats the multipath interference, whereas the FH part is a predetection against large multiuser interference. It is shown that, for the con- sidered types of a channel coding, the use of a predetection coding is still essential for obtained a satisfactory bit error performance. The results show that the capacity of the DS/SFG-CDMA MFSK communication system increases in proportion to the length of PN code sequence in the presence of AWGN and MUI. In m-distribution fading environment the capacity increases in proportion to the fading index. The capacity is increased and error performance is improved when the CCI Canceller and Convolution code technique are adopted, respectively. From the results, it is known that the error performance of $4\times10^{-2}$ by adopting Canceller technique. Also convolutional coding technique is the improvement of error performance attains about $10^{-5}$ in code rate 1/2.