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An efficient algorithm for bandpass sampling of multiple RF signals
Bae, Junghwa,Park, Jinwoo IEEE Signal Processing Society 2006 IEEE signal processing letters Vol.13 No.4
This letter proposes, based on a bandpass sampling theory, a novel method to find available sampling ranges with a low computational complexity and high accuracy for multiple bandpass radio frequency signals. Guard-bands between downconverted signal spectrums are also taken into consideration in determining a minimum sampling frequency. We verify its validity through simulations in terms of the sampling ranges, the minimum sampling frequency, and computational efficiency.
Yu Gwang Jin,Jong Won Shin,Nam Soo Kim IEEE Signal Processing Society 2014 IEEE signal processing letters Vol. No.
<P>In this letter, we propose a spectro-temporal filtering algorithm for multichannel speech enhancement in the short-time Fourier transform (STFT) domain. Compared with the traditional multiplicative filtering technique, the proposed method takes account of interdependencies between components in adjacent frames and frequency bins. For spectro-temporal filtering, speech and noise power spectral density (PSD) matrices are estimated based on an extended formulation utilizing temporal and spectral correlations, and the parametric noise reduction filter based on these PSD matrices is applied to the input microphone array signal. Moreover, multichannel speech presence probabilities are also estimated within a unified framework. A number of experimental results show that the proposed spectro-temporal filtering method improves the performance of multichannel speech enhancement.</P>
Feature Compensation Incorporating Modeling Error Statistics
Woohyung Lim,Nam Soo Kim IEEE Signal Processing Society 2007 IEEE signal processing letters Vol.14 No.7
<P>In this letter, we propose a novel approach to feature compensation for robust speech recognition in noisy environments. We analyze the error distribution of speech corruption model in the log spectral domain and represent the statistics as functions with respect to the signal-to-noise ratio. The proposed algorithm incorporates modeling error statistics into the interacting multiple model technique and shows a performance improvement over the AURORA2 speech recognition task.</P>
Jae-Mo Kang,Hyung-Myung Kim IEEE Signal Processing Society 2015 IEEE signal processing letters Vol. No.
<P>This letter proposes channel training designs for two-hop multi-relay networks, where a source, a destination and multiple amplify-and-forward (AF) relays are all equipped with multi-antenna, based on mean square error (MSE) and signal-to-noise ratio (SNR) criteria with taking into account spatial fading correlation between multiple-input multiple-output (MIMO) channel elements. The minimum MSE and maximum SNR channel estimators are initially derived and, then, optimal structures on source training signal and relay matrices are determined. An iterative and a closed-form power allocation solutions are proposed for both channel estimators. Simulation results show that the proposed schemes outperform the conventional schemes.</P>
IEEE Signal Processing Society 2017 IEEE signal processing letters Vol.24 No.7
<P>Presents corrections to the paper, “Indistinguishability of compressed encryption with circulant matrices for wireless security,” (Yu, N.Y.), IEEE Signal Process. Lett., vol. 24, no. 2, pp. 181–185, Feb. 2017.</P>
Amplitude Clipping and Iterative Reconstruction of STBC/SFBC-OFDM Signals
Kwon, Ui-Kun,Kim, D.,Kim, Kiho,Im, Gi-Hong IEEE Signal Processing Society 2007 IEEE signal processing letters Vol.14 No.11
<P>As with orthogonal frequency-division multiplexing (OFDM), one main disadvantage of multi-input multi-output (MIMO) OFDM is the prohibitively large peak-to-average power ratio (PAPR) of transmitted signals on different antennas, which can be reduced by a deliberate amplitude clipping. However, clipping causes distortion that degrades the system performance. In this letter, we propose clipping noise mitigation techniques for space-time and space-frequency block coded (STBC/SFBC)-OFDMs. A new SFBC transmitter for clipped OFDM is proposed, and the optimum equalizer is derived for reconstruction of the clipped signals. Simulation results show that the proposed receivers effectively recover contaminated STBC/SFBC-OFDM signals with a moderate computational complexity.</P>
Weight-Space Viterbi Decoding Based Spectral Subtraction for Reverberant Speech Recognition
Ban, Sung Min,Kim, Hyung Soon IEEE Signal Processing Society 2015 IEEE signal processing letters Vol. No.
<P>A single-channel blind dereverberation algorithm is proposed in this letter for distant-talking speech recognition. The proposed method is based on spectral subtraction (SS) method, in which the spectrum of a late reverberant signal is estimated using a delayed and attenuated version of the reverberant signal. Through some assumptions, the conventional SS method regards the attenuation weight as a constant that is a function of reverberation time. However, these assumptions are not valid in real situations, and the ideal weight varies with the frame. Therefore, in the proposed method, the variable weight sequence is estimated using Viterbi decoding scheme based on the reverberation model. This weight sequence is then substituted for the fixed weight in the conventional SS method without explicitly estimating the reverberation time. The proposed method performs better than the conventional SS method in both isolated word recognition and connected digit recognition experiments in reverberant environments.</P>
A TR-MISI Serial Prefilter for Robustness to ISI and Noise in Indoor Wireless Communication System
Misun Yoon,Chungyong Lee IEEE Signal Processing Society 2014 IEEE signal processing letters Vol. No.
<P>We propose a time reversal and minimum inter-symbol interference (TR-MISI) serial prefilter for indoor wireless communication system. Proposed TR-MISI serial prefilter minimizes inter-symbol interference (ISI) and makes received peak power be robust to noise. Since the TR prefilter does not minimize the ISI and the joint zero-forcing and TR prefilter does not guarantee the performance in low signal-to-noise ratio (SNR) environment, we consider these two facts in designing an optimization problem for proposed TR-MISI serial prefilter. We use the semidefinite relaxation to solve the optimization problem and get a near-optimal solution. The simulation results show that proposed TR-MISI serial prefilter improves the bit error rate performance at overall SNR.</P>
Radar Signals With ZACZ Based on Pairs of D-Code Sequences and Their Compression Algorithm
Ipanov, Roman N.,Baskakov, Alexandr I.,Olyunin, Nikolay,Ka, Min-Ho IEEE Signal Processing Society 2018 IEEE signal processing letters Vol.25 No.10
<P>Radar signals are synthesized having a region of zero autocorrelation sidelobes with a relative width of <TEX>$1/3\leq \varepsilon < 1$</TEX> in the vicinity of the central peak of an aperiodic autocorrelation function. These signals represent a train of two binary phase-coded pulses encoded by pair of complementary sequences of length <TEX>$N=2^k$</TEX>, with an integer <TEX>$k\geq 2$</TEX> . The correlation characteristics of the synthesized signal are analyzed. A compression algorithm for these signals, multichannel with respect to the Doppler frequency, is considered. It includes a combined algorithm for fast Walsh and Fourier transforms.</P>
Perceptual Reinforcement of Speech Signal Based on Partial Specific Loudness
Shin, Jong Won,Kim, Nam Soo IEEE Signal Processing Society 2007 IEEE signal processing letters Vol.14 No.11
<P>In the presence of background noise, the perceptual loudness of speech signal significantly decreases, resulting in the deterioration of intelligibility and clarity. In this letter, we propose a novel approach to enhance the perceived quality of the speech signal when the additive noise cannot be directly controlled. Instead of controlling the background noise, we propose to reinforce the speech signal so that it can be heard more clearly in noisy environments. To find a suitable reinforcement rule, the loudness perception model proposed by Moore et al. is adopted. Experimental results show that the loudness of the reinforced signal can be maintained at the level almost the same as that of the original noise-free speech, and the proposed algorithm can enhance the perceived speech quality under various noise environments.</P>