• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2008.05a
• /
• pp.421-424
• /
• 2008

This paper proposes an adaptive noise suppression system based on human auditory model to enhance speech signal that is degraded by various background noises. The proposed system detects voiced and unvoiced sections for each frame and implements the adaptive auditory process, then reduces the noise speech signal using neural network including amplitude component and phase component. Base on measuring signal-to-noise ratios, experiments confirm that the proposed system is effective for speech signal that is degraded by various noises.

• Journal of the Korean Institute of Telematics and Electronics B
• /
• v.31B no.7
• /
• pp.33-41
• /
• 1994

A single channel adaptive noise canceling (ANC) technique with a recursive time delay estimator (RTDE) is presented for removing effects of additive noise on the speech signal. While the conventional method makes a reference signal for the adaptive filter using the pitch estimated on a frame basis from the input speech, the proposed method makes the reference signal using the delay estimated recursively on a sample-by-sample basis. As the RTDEs, the recursion formulae of autocorrelation function (ACF) and average magnitude difference function (AMDF) are derived. The normalized least mean square (NLMS) and recursive least square (RLS) algorithms are applied for adaptation of filter coefficients. Experimental results with noisy speech demonstrate that the proposed method improves the perceived speech quality as well as the signal-to-noise ratio and cepstral distance when compared with the conventional method.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.13 no.4
• /
• pp.652-657
• /
• 2009

This paper proposes an adaptive noise suppression system based on human auditory model to enhance speech signal that is degraded by various background noises. The proposed system detects voiced, unvoiced and silence sections for each frame and implements an adaptive auditory process, then reduces the noise speech signal using a neural network including amplitude component and phase component. Based on measuring signal-to-noise ratios, experiments confirm that the proposed system is effective for speech signal that is degraded by various noises.

• 제어로봇시스템학회:학술대회논문집
• /
• 1990.10a
• /
• pp.575-579
• /
• 1990

This paper describes an algorithm that can efficiently analyze the multichannel ECG signal using the frame. The input is a set of significant features (points) which have been extracted from an original sampled signal by using the split-and-merge algorithm. A signal from each channel can be hierarchical ADN/OR graph on the basis of the priori knowledge for ECG signal. The search mechanisms with some heuristics and the mixed paradigms of data-driven hypothesis formation are used as the major control mechanisms. The mutual relations among features are also considered by evaluating a score based on the relational spectrum. For recognition of morphologies corresponding to OR nodes, an hypothesis modification strategy is used. Other techniques such as instance, priority update of prototypes, and template matching facility are also used. This algorithm exactly recognized the primary points and supporting points from the multichannel ECG signals.

• Proceedings of the IEEK Conference
• /
• 2001.06d
• /
• pp.281-284
• /
• 2001

Speech is classified into voiced signal and unvoiced signal. Since the amplitude of voiced fall off at about -20dB/decade, dynamic range is often compressed prior to spectral analysis so that details at weak, high frequencies may be visible[5][6] There is a distinct difference in spectrum slope between voiced signal and unvoiced signal. In this paper, we got the slope of each frame by using autocorrelation method, and determined voiced /unvoiced region. Also, we used energy to decide region of silence. To show experimental results, we allot to 1 value in voiced region, -1 value in unvoiced region and 0 value in silence region.

• Journal of Power Electronics
• /
• v.16 no.4
• /
• pp.1513-1525
• /
• 2016

Rapid and accurate phase synchronization is critical for the reliable control of grid-tied inverters. However, the commonly used software phase-locked loop methods do not always satisfy the need for high-speed and accurate phase synchronization under severe grid imbalance conditions. To address this problem, this study develops a novel open-loop phase locking scheme based on a synchronous reference frame. The proposed scheme is characterized by remarkable response speed, high accuracy, and easy implementation. It comprises three functional cascaded blocks: fast orthogonal signal generation block, fast fundamental-frequency positive sequence component construction block, and fast phase calculation block. The developed virtual orthogonal signal generation method in the first block, which is characterized by noise immunity and high accuracy, can effectively avoid approximation errors and noise amplification in a wide range of sampling frequencies. In the second block, which is the foundation for achieving fast phase synchronization within 3 ms, the fundamental-frequency positive sequence components of unsymmetrical grid voltages can be achieved with the developed orthogonal signal construction strategy and the symmetrical component method. The real-time grid phase can be consequently obtained in the third block, which is free from self-tuning closed-loop control and thus improves the dynamic performance of the proposed scheme. The proposed scheme is adaptive to severe unsymmetrical grid voltages with sudden changes in magnitude, phase, and/or frequency. Moreover, this scheme is able to eliminate phase errors induced by harmonics and random noise. The validity and utility of the proposed scheme are verified by the experimental results.

• Korean Journal of Microbiology
• /
• v.39 no.2
• /
• pp.83-88
• /
• 2003

We cloned a gene encoding acetyl xylan esterase(axeA) of Streptomyces coelicolor A3(2) and studied its expression pattern in Escherichia coli. The full sequence of axeA was amplified by PCR. Sequence analysis of the PCR product revealed an open reading frame of 1,008 nucleotides encoding a protein consisted of 335 amino acid residues, with a calculated molecular mass of about 38 kDa. The base sequence showed 98% homology to the same gene of Streptomyces lividans. Two different kinds of acetyl xylan esterases were produced in Escherichia coli(pLacI) by IPTG induction; their molecular weights were 38 kDa and 34 kDa, respectively. Of these, 38 kDa protein seemed to be a total protein holding N-terminal signal peptide region, whereas 34 kDa protein seemed to be a matured protein without signal peptide which was produced by peptide bond cleavage between two amino acid residues of alanine 41 and alanine 42.

• Journal of Aquaculture
• /
• v.16 no.2
• /
• pp.110-117
• /
• 2003

We have cloned and sequenced the cDNA encoding growth hormone (GH) from pituitary poly(A)$^{+}$ RNA of red-spotted grouper (Epinephelus akaara). The cDNA of red-spotted grouper GH is 883 base pairs (bp) consisting of 21 bp of 5'untranslated region (UTR), 615 Up of an open reading frame (ORF) and 247 Up of 3'UTR. The polyadenylation signal, AATAAA, was 20 bp upsteam of polyadenylation site. Based on the nucleotide sequences, the deduced putative polypeptide contains 204 amino acids (aa), representing 17 aa of a signal and 187 aa of a mature polypeptide. The putative GH cDNA encodes a polypeptide with four cysteine residues and only one N-gly- cosylation site. Comparative sequence alignment shows that red-spotted grouper GH exhibits high similarity with its corresponding other Perciformes species GH cDNAs.

• The Journal of the Acoustical Society of Korea
• /
• v.35 no.5
• /
• pp.375-382
• /
• 2016

This paper presents the results for the performance improvement of underwater communication in a passive time reversal mirror (PTRM) using ray-based blind deconvolution (RBD). In conventional PTRM, the signal to be recovered is found from matched-filtering the received probe signal. However, the communication performance is degraded because the time-varying impulse response for each data frame is not reflected in the received probe signal. In this study, the time-variant transfer function is estimated from each received data frame using RBD, and the estimated time-variant transfer function is then used to recover the data signal using PTRM. The results from the experimental data show that the suggested method improves the communication performance when comparing with the conventional PTRM.

• Journal of the Institute of Convergence Signal Processing
• /
• v.9 no.3
• /
• pp.195-200
• /
• 2008

To compress the moving picture data effectively, it is needed to reduce spatial and temporal redundancy of input image data. While motion estimation! compensation methods is effectively able to reduce temporal redundancy but it is increased computation complexity because of the prediction between frames. So, the study of algorithm for computation reduction and real time processing is needed. This paper is presenting quantizer effectively able to quantize DCT coefficient considering the human visual sensitivity. As quantizer that proposed DCT-based H.263 could make transmit more frame than TMN5 at a same transfer speed, and it could decrease the frame drop effect. And the luminance signal appeared the difference of $-0.3{\sim}+0.65dB$ in the average PSNR for the estimation of objective image quality and the chrominance signal appeared the improvement in about 1.73dB in comparision with TMN5. The proposed method reduces $30{\sim}31%$ compared with NTSS and $20{\sim}21%$ compared to 4SS in comparition of calculation quantity.