This paper evaluate the performance of a PN (pseudonoise) code acquisition for a direct-sequence/code-division-multiple-access (DS/CDMA) system in a mobile satellite environment. The acquisition scheme considered consists of a parallel matched-filter and a FFT processor. The uplink of mobile satellite channel is modeled as shadowed Rayleigh fading channel. The effects of power control error and shadowing are considered in the analysis of acquisition performance. It is shown that the power control error causes acquisition to be slower than the case of perfect power control, and for high SNR/chip, the effect of power control error becomes less significant. It is also shown that the case with heavy shadowing takes longer time to achieve acquisition than that with light shadowing. For the subinterval-based PN code search, the parallel MF scheme is thought to be more appropriate than the serial MF scheme at the cost of complexity. The analysis in the paper can be applied to the uplink of a DS/CDMA system for packet-type services in a mobile satellite channel.

Current research trends in computer vision have tended towards achieving the goal of recognizing human action, due to the potential utility of such recognition in various applications. Among many potential approaches, an approach involving Gaussian Mixture Model (GMM) supervectors with a Support Vector Machine (SVM) and a nonlinear GMM KL kernel has been proven to yield improved performance for recognizing human activities. In this study, based on tensor analysis, we develop and exploit an extended class of action features that we refer to as gradient-flow tensor divergence. The proposed method has shown a best recognition rate of 96.3% for a KTH dataset, and reduced processing time.

Digital hearing aid users often complain of difficulty in understanding speech in the presence of background noise. To improve speech perception in a noisy environment, various speech enhancement algorithms have been applied in digital hearing aids. In this study, a speech enhancement algorithm using modified spectral subtraction and companding is proposed for digital hearing aids. We adjusted the biases of the estimated noise spectrum, based on a subtraction factor, to decrease the residual noise. Companding was applied to the channel of the formant frequency based on the speech presence indicator to enhance the formant. Noise suppression was achieved while retaining weak speech components and avoiding the residual noise phenomena. Objective and subjective evaluation under various environmental conditions confirmed the improvement due to the proposed algorithm. We tested segmental SNR and Log Likelihood Ratio (LLR), which have higher correlation with subjective measures. Segmental SNR has the highest and LLR the lowest correlation of the methods tested. In addition, we confirmed by spectrogram that the proposed method significantly reduced the residual noise and enhanced the formants. A mean opinion score that represented the global perception score was tested; this produced the highest quality speech using the proposed method. The results show that the proposed speech enhancement algorithm is beneficial for hearing aid users in noisy environments.

A new matched-filter (MF) acquisition scheme with adaptive threshold is proposed for a frequency-hopped/spread-spectrum multiple-access (FH/SSMA) system. Detection and false alarm probabilities are derived for combined interference environments. The combined interference consists of partialband noise jamming or tone jamming, multiple access interference (MAI), multipath interference, and thermal noise. We use Gaussian approximation for modeling the MAI and multipath interference. Equal power assumption of the users is employed which is typically used in the SSMA system analysis. In the proposed scheme, MF output is compared to an adaptive threshold determined by the number of jammed frequency slots. It is shown that the proposed adaptive-threshold acquisition scheme achieves higher detection probability and lower false alarm probability than a conventional fixed-threshold scheme for each jammed fractional bandwidth, JSR, the number of multipaths, and the number of users. It is also shown that adaptive threshold achieves faster acquisition and higher packet throughput than fixed threshold in application to FH/SSMA packet radio system.

In this paper, a novel visual signal reliability (VSR) measure is proposed to consider video degradation at the signal level in audio-visual speaker identification (AVSI). The VSR estimation is formulated using a~ Gaussian fuzzy membership function (GFMF) to measure lighting variations. The variance parameters of GFMF are optimized in order to maximize the performance of the overall AVSI. The experimental results show that the proposed method outperforms the score-based reliability measuring technique.

This paper proposes a novel wavelet de-noising scheme regarding the existing burst noises that consist of background and impulsive noises in power-line communications. The proposed de-noising scheme employs multi-level threshold functions to efficiently and adaptively reduce the given burst noises. The experiment results show that the proposed de-noising scheme significantly outperformed the conventional schemes.

In this paper, multiuser detector (MUD) based on radial basis function (RBF) is proposed and simulated for a multicode DS/CDMA system in an AWGN and a multipath fading channels. The performance of RBF-based MUD is compared with that of many suboptimal multiuser detectors in terms of bit error probability. To obtain simulation results, importance sampling technique is employed. From the simulation results, it is confirmed that the RBF-based MUD outperforms decorrelating detector, and achieves near-optimum performance under various environments. The results in this paper can be applied to design of MUD for a multicode DS/CDMA system.

With the increasing demand of Internet-of-Things applicability in various devices and location-based services (LBSs) with positioning capabilities, we proposed simple and effective post-processing techniques to reduce positioning error and provide more precise navigation to users in a pedestrian environment in this letter. The proposed positioning error reduction techniques (Technique 1-minimum range securement and bounce elimination, Technique 2-direction vector-based error correction) were studied considering low complexity and wide applicability to various types of positioning systems, e.g., global positioning system (GPS). Through the real field tests in urban areas, we have verified that an average positioning error of the proposed techniques is significantly decreased compared to that of a GPS-only environment.

In this letter, we study a scenario based on decoupled RF energy harvesting networks (DRF-EHNs) that separate energy sources from information sources to overcome the doubly near-far problem and improve harvesting efficiency. We propose an algorithm to maximize energy efficiency (EE) while satisfying constraints on the maximum transmit power of the hybrid access point (H-AP) and power beacon (PB), while further satisfying constraints on the minimum quality of service and minimum amount of harvested power in multi-user Rayleigh fading channel. Using nonlinear fractional programming and Lagrangian dual decomposition, we optimize EE with four optimization arguments: the transmit power from the H-AP and PB, time-splitting ratio, and power-splitting ratio. Numerical results show that the proposed algorithm is more energy-efficient compared to baseline schemes.

In this paper, outage performance of a turbo-coded CDMA system is analyzed and simulated in a multiple-beam satellite channel. From the simulation results, it is confirmed that turbo coding provides considerable coding gain over an uncoded system. And, it is demonstrated that Max-Log-MAP decoding algorithm is most promising in terms of performance and complexity.

Recently many researches concerning heart sound analysis are being processed with development of digital signal processing and electronic components. But there are few researches about recognition of heart sound, especially full cardiac cycled heart sound. In this paper, three new recognition methods about full cardiac cycled heart sound were proposed. The first method recognizes the characteristics of heart sound by integrating important peaks and analyzing statistical variables in time domain. The second method builds a database by principal components analysis on training heart sound set in time domain. This database is used to recognize new input of heart sound. The third method builds the same sort of the database not in time domain but in time-frequency domain. We classify the heart sounds into seven classes such as normal (NO) class, pre-systolic murmur (PS) class, early systolic murmur (ES) class, late systolic murmur (LS) class, early diastolic murmur (ED) class, late diastolic murmur (LD) class and continuous murmur (CM) class. As a result, we could verify that the third method is better efficient to recognize the characteristics of heart sound than the others and also than any precedent research. The recognition rates of the third method are 100% for NO, 80% for PS and ES, 67% for LS, 93 for ED, 80% for LD and 30% for CM.

Howling is very annoying problem to the hearing-aid users and it limits the maximum usable gain of hearing aids. We propose a new feedback cancellation system by inserting a time-varying decorrelation filter in the forward path. We use a second-order all-pass filter with control parameters whose time variation is implemented using a low-frequency modulator. A noticeable reduction of weight-vector misalignment is achievable using our proposed method.

The aim of this letter is to guarantee the ability of low probability of intercept (LPI) and anti-jamming (AJ) by maximizing the energy efficiency (EE) to improve wireless communication survivability and sustain wireless communication in jamming environments. We studied a scenario based on one transceiver pair with a partial-band noise jammer in a Rician fading channel and proposed an EE optimization algorithm to solve the optimization problem. With the proposed EE optimization algorithm, the LPI and AJ can be simultaneously guaranteed while satisfying the constraint of the maximum signal-to-jamming-and-noise ratio and combinatorial subchannel allocation condition, respectively. The results of the simulation indicate that the proposed algorithm is more energy-efficient than those of the baseline schemes and guarantees the LPI and AJ performance in a jamming environment.

Acoustic Event Classification (AEC) poses difficult technical challenges as a result of the complexity in capturing and processing sound data. Of the various applicable approaches, Support Vector Machine (SVM) with Gaussian Mixture Model (GMM) supervectors has been proven to obtain better solutions for such problems. In this paper, based on the multiple kernel selection model, we introduce two non-linear kernels, which are derived from the linear kernels of GMM Kullback-Leibler divergence (GMM KL) and GMM-UBM mean interval (GUMI). The proposed method improved the AEC model's accuracy from 85.58% to 90.94% within the domain of home AEC.

In this paper, we propose a non-cooperative line-of-sight (LOS)/non-LOS channel identification algorithm with single node channel measurements based on time-of-arrival statistics. In order to improve the accuracy of channel identification, we adopt a recalibration interval in terms of measured distance to the proposed algorithm. Experimental results are presented in terms of identification probability and recalibration interval. The proposed algorithm involves a trade-off between the channel identification quality and the recalibration rate. However, depending on the recalibration interval, it is possible to greatly improve the sensitivity of the channel identification system.

In this paper, performance of a PN code acquisition scheme is analyzed and simulated for a DS/CDMA overlay system where a CDMA user and a narrowband user coexist in the same frequency band. A narrowband user is modelled as a narrowband interference (NBI) located at the fraction of a CDMA user's bandwidth. To suppress the NBI, an interference suppression filter is employed at the receiver frontend. Acquisition performance is evaluated in terms of mean acquisition time using state transition diagram for acquisition process. To apply for a DS/CDMA mobile cellular environments, multiple access interference and imperfection of power control are taken into account in the analysis of acquisition performance. The imperfect power control is considered by modelling the power of each user to be lognormally distributed about nominal received power. From the simulation results, it is shown that for the cases of perfect and imperfect power control, the interference suppression filter is very effective for supprssion of the NBI and rapid PN code acquisition in a DS/CDMA overlay environment. It is also shown that the one-sided tap number of 5 for interference suppression filter is sufficient to suppress the NBI. And, capacity estimates are compared based on acquisition and BER performance. The analysis in this paper can be applied to the practical situations for a DS/CDMA overlay environment.

In this letter, we study an interference scenario under a smart interferer which observes color channels and interferes with a visible light communication (VLC) network. We formulate the smart interference problem based on a Stackelberg game and propose an optimal response algorithm to overcome the interference by optimizing transmit power and sub-color channel allocation. The proposed optimization algorithm is composed with Lagrangian dual decomposition and non-linear fractional programming to have stability to get optimum points. Numerical results show that the utility by the proposed algorithm is increased over that of the algorithm based on the Nash game and the baseline schemes.

In this paper, we propose a single-channel speech enhancement method for a push-to-talk enabled wireless communication device. The proposed method is based on adaptive weighted β-order spectral amplitude estimation under speech presence uncertainty and enhanced instantaneous phase estimation in order to achieve flexible and effective noise reduction while limiting the speech distortion due to different noise conditions. Experimental results confirm that the proposed method delivers higher voice quality and intelligibility than the reference methods in various noise environments.

Detection loss due to phase error in a carrier tracking loop is analyzed and simulated for a code division multiple access system with BPSK and QPSK modulations in a Rayleigh fading channel. For a specific BER, the detection loss due to phase error is defined as an increase of required SNR to maintain the same BER without phase error. A nonlinear Fokker-Planck method is employed to analyze first-order PLL (phase locked loop) performance. From the simulation results, it is confirmed that the phase noise induces significant detection loss, which eventually leads to degradation of the BER performance.

The performance of a noncoherent parallel matched-filter (MF) acquisition scheme with a reference filter (RF) is evaluated for a direct-sequence/spread-spectrum multiple access (DS/SSMA) packet radio system in a mobile cellular environment. This acquisition scheme employs a RF to estimate the variance of interference at the output of detecting MF. Acquisition-based packet throughput of the parallel NM-RF scheme is derived for an AWGN and a Rayleigh fading channels. Packet throughput of a parallel MF-RF acquisition scheme is compared with those of a serial MF scheme, a serial MF-RF scheme, and a parallel MF. From the numerical results, it is shown that the packet throughput decreases with the number of users in the system, and increases with the preamble length. Imperfect power control causes packet throughput to decrease especially when the power control error is large. The considerations in this paper can be applied to the reverse link (mobile-to-base station) design of a DS/SSMA system for packet-type services.