• The Journal of the Acoustical Society of Korea
• /
• v.16 no.4
• /
• pp.5-10
• /
• 1997

In this paper, a new time delay estimation algorithm, WTD-LMSTDE was proposed. This method has great improvement in convergence rate relative to the time domain approach by decreasing the eigen value spread of input signal autocorrelation matrix. The performance of the algorithm was evaluated for the cases of time invariant time delay and time varying time delay. In the case of time invariant time delay, the estimation accuracy of WTD-LMSTDE was better than that of LMSTDE from 3.3% to 12.5% with respect to SNR. In the case of time varying time delay, the mean error power of WTD-LMSTDE in linear increased delay environment was decreased about 2.4dB compared to that of LMSTDE under noise-free condition. As a result, we showed that the performance of WTD-LMSTDE is better than of LMSTDE.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.8 no.2
• /
• pp.389-405
• /
• 2014

Outdated Channel State Information (CSI) was proved to have negative effect on performance in two-way relay networks. The diversity order of widely used opportunistic relay selection (ORS) was degraded to unity in networks with outdated CSI. This paper proposed a multiple relay selection scheme for amplify-and-forward (AF) based two-way relay networks (TWRN) with outdated CSI. In this scheme, two sources exchange information through more than one relays. We firstly select N best relays out of all candidate relays with respect to signal-noise ratio (SNR). Then, the ratios of the SNRs on the rest of the candidate relays to that of the Nth highest SNR are tested against a normalized threshold ${\mu}{\in}[0,1]$, and only those relays passing this test are selected in addition to the N best relays. Expressions of outage probability, average bit error rate (BER) and ergodic channel capacity were obtained in closed-form for the proposed scheme. Numerical results and Simulations verified our theoretical analyses, and showed that the proposed scheme had significant gains comparing with conventional ORS.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.3E
• /
• pp.18-24
• /
• 2001

Speaking rate represents how many phonemes in speech signal have in limited time. It is various and changeable depending on the speakers and the characters of each phoneme. The preprocessing to remove the effect of variety of speaking rate is necessary before recognizing the speech in the present speech recognition systems. So if it is possible to estimate the speaking rate in advance, the performance of speech recognition can be higher. However, the conventional speech vocoder decides the transmission rate for analyzing the fixed period no regardless of the variety rate of phoneme but if the speaking rate can be estimated in advance, it is very important information of speech to use in speech coding part as well. It increases the quality of sound in vocoder as well as applies the variable transmission rate. In this paper, we propose the method for presenting the speaking rate as parameter in speech vocoder. To estimate the speaking rate, the variety of phoneme is estimated and the Line Spectrum Pairs is used to estimate it. As a result of comparing the speaking rate performance with the proposed algorithm and passivity method worked by eye, error between two methods is 5.38% about fast utterance and 1.78% about slow utterance and the accuracy between two methods is 98% about slow utterance and 94% about fast utterances in 30 dB SNR and 10 dB SNR respectively.

• Phonetics and Speech Sciences
• /
• v.2 no.3
• /
• pp.141-148
• /
• 2010

In this paper, voice activity detection (VAD) for dual-channel noisy speech recognition is proposed in which spatial cues are employed. In the proposed method, a probability model for speech presence/absence is constructed using spatial cues obtained from dual-channel input signal, and a speech activity interval is detected through this probability model. In particular, spatial cues are composed of interaural time differences and interaural level differences of dual-channel speech signals, and the probability model for speech presence/absence is based on a Gaussian kernel density. In order to evaluate the performance of the proposed VAD method, speech recognition is performed for speech segments that only include speech intervals detected by the proposed VAD method. The performance of the proposed method is compared with those of several methods such as an SNR-based method, a direction of arrival (DOA) based method, and a phase vector based method. It is shown from the speech recognition experiments that the proposed method outperforms conventional methods by providing relative word error rates reductions of 11.68%, 41.92%, and 10.15% compared with SNR-based, DOA-based, and phase vector based method, respectively.

• Journal of the Korea Academia-Industrial cooperation Society
• /
• v.12 no.5
• /
• pp.2261-2266
• /
• 2011

Being the most widely used in ultrasound Doppler color flow mapping, the Kasai algorithm, also known as lag-1 autocorrelation method, is capable of estimating the Doppler mean frequency relatively accurately with a modest amount of computation. Particularly in the case of imaging deep lying areas, however, its performance suffers due to low signal-to-noise ratios. The purpose of this paper is to propose a dealiased lag-2 autocorrelation method which is superior to the Kasai algorithm even at low signal-to-noise ratios and to compare their performances through simulations. The proposed algorithm is found to be better by about 2 to 3 dB than the Kasai algorithm in terms of Doppler mean frequency estimation error in the presence of measurement noise.

• International Journal of Contents
• /
• v.11 no.1
• /
• pp.7-14
• /
• 2015

In this paper, we derive the outage probability for M-ary phase shifting keying (MPSK) and M-ary quadrature amplitude modulation (MQAM) burst transmission (BT) of adaptive decode-and-forward (ADF) cooperative relay systems over quasi-static Rayleigh fading channels. Within a burst, there are pilot symbols and data symbols. Pilot symbols are used for channel estimation schemes and each relay node's transmission mode selection schemes. At first, we focus on ADF relay systems in which the probability density function (PDF) is derived on the basis of error events at relay nodes corresponding to channel estimation errors. Next, the average outage probability is derived as an approximate expression for an arbitrary link signal-to-noise ratio (SNR) for different modulation orders. Its accuracy is demonstrated by comparison with simulation results. Further, it is confirmed that BT-ADF relay systems with pilot symbol based channel estimation schemes enables to select correctly decoded relay nodes without additional signaling between relay nodes and the destination node, and it is verified that the ideal performance is achieved with small SNR loss.

• Journal of Advanced Navigation Technology
• /
• v.13 no.1
• /
• pp.97-103
• /
• 2009

In this paper, the structure and detailed specifications of the HDR-WPAN physical layer have been analyzed and the block module of transmitter and receiver have been also designed, and analyzed the performance as well. In the process of transmitter design, it concentrated on all possibility of modulation of QPSK, DQPSK and 16/32/64QAM-TCM, which could be available for mode selection due to the transmission rate. In addition to the receiver module, DQPSK and TCM decoding algorithm is mainly concerned. After designing the transceiver MODEM using VHDL, we have programmed on the platform board and verified the functions of the MODEM. Some experimental results showed that it can be considered a possibility of data communication without error over SNR 22dB.

• 제어로봇시스템학회:학술대회논문집
• /
• 2005.06a
• /
• pp.1222-1227
• /
• 2005

In this paper, an efficient scheme of the adaptive notch filter is presented for rejecting the narrow bandwidth interferences(NBI) in GPS receiver. Designed is the lattice IIR double notch filter for more efficient suppression of the NBI with less computational complexity. The algorithm is of recursive prediction error form and uses a special constrained model of IIR with a minimal number of parameters. This paper chooses seven different jamming scenarios including one without jamming for evaluating the proposed filter algorithm. The simulation results to the jamming scenarios show that the proposed algorithm adjusts the double notch filter effectively for the given JSR, and provides better SNR than the conventional algorithms. Finally, it is shown that the advantages of the proposed filter algorithm can range as high as JSR 79dB in time domain processing. Also, the ADNF(adaptive double notch filter) guarantees that more than SNR 10dB of GPS receiver can be always maintained. In conclusion, there is enough evidence to believe that the proposed algorithm will perform quite well for removing interference signals.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.18 no.3
• /
• pp.348-354
• /
• 1993

In this research, a multi-tree coder at 9.6Kbits/sec using a novel scheme for adaptation of the short-term coefficients is developed. The overall delay of the tree coder is maintained at 2.5 msec(16 samples at the 6.4KHz sampling frequency). This coder produces good quality speech over ideal channels, and it is very robust to channel errors up to a bit error rate (BER) of $10^{-3}$. This robustness is achieved by using a parallel adaptation scheme in combination with the use of a smoothed version of the received excitation sequence for adaptation of the short-term prediction coefficients. For the multi-tree coder, reconstructed output speech is evaluated using signal-to-quantization noise ratios (SNR), segmental SNRs, and informal listening tests.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.1 s.343
• /
• pp.19-24
• /
• 2006

Wireless sensor networks(WSN) technology has various applications such as surveillance and information gathering in the uncontrollable area of human. One of major issues in WSN is the research for reducing the energy consumption and reliability of data. A system with forward error correction(FEC) can provide an objective reliability while using less transmission power than a system without FEC. In this paper, we propose to use LDPC codes of various code rate(0.53, 0.81, 0.91) for FEC for WSN. Also, we considered node-to-node interference in addition to AWGN channel. The proposed system has not only high reliable data transmission at low SNR, but also reduced transmission power usage.