• Journal of the Korean Institute of Telematics and Electronics C
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• v.34C no.8
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• pp.71-78
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• 1997

An LPC cepstrum processor for speech recognition is implemented on CMOS array process. The designed processor contains a 24-bit floating-point MAC unit to perform the correlation quickly, which occupies the majority of operations used in the algorithm, and has 22 register files to store temporary variables. For the purpose of fast operations, the floating-point MAC consists of a 3-stage pipeline and the new post-normalization shceme is proposed and applied to it. Experimental result shows that it takes approximately 266.mu.s to process 200 samples/frame at 15 MHz clock rate. This processor runs at the maximum rate of 16.6 MHz and the number of gates are 27,760.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.63 no.11
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• pp.1559-1563
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• 2014

Recently, the prevention of crime using CCTV draws special in accordance with the higher crime incidence rate. Therefore security systems like a CCTV with audio capability are developing for giving an instant alarm. This paper proposes a scream detecting engine from various ambient noises in real environment for surveillance systems. The proposed engine detects scream signals among the various ambient noises using the features extracted in time/frequency domain. The experimental result shows the performance of our engine is very promising in comparison with the traditional engines using the model based features like LPC, LPCC and MFCC. The proposed method has a low computational complexity by using FFT and cross correlation coefficients instead of extracting complex features like LPC, LPCC and MFCC. Therefore the proposed engine can be efficient for audio-based surveillance systems with low SNRs in real field.

• Speech Sciences
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• v.3
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• pp.167-176
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• 1998

Conventional time-scale modification methods have the problem that as the modification rate gets higher the time-scale modified speech signal becomes less intelligible, because they ignore the effect of articulation rate on speech characteristics. Results of research on speech perception show that the timing information of transient portions of a speech signal plays an important role in discriminating among different speech sounds. Inspired by this fact, we propose a novel scheme for modifying the time-scale of speech. In the proposed scheme, the timing information of the transient portions of speech is preserved, while the steady portions of speech are compressed or expanded somewhat excessively for maintaining overall time-scale change. In order to identify the transient and steady portions of a speech signal, we employ a simple method using LPC cepstral distance between neighboring frames. The result of the subjective preference test indicates that the proposed method produces performance superior to that of the conventional SOLA method, especially for very fast playback case.

• The Korean Journal of Physiology and Pharmacology
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• v.22 no.4
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• pp.399-408
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• 2018

A lipidomic study on extensive plasma lipids in bacterial peritonitis (cecal ligation and puncture, CLP)-induced sepsis in mice was done at 24 h post-CLP. The effects of administration of lysophosphatidylcholine (LPC) and lysophosphatidic acid (LPA), compounds known to have beneficial effects in CLP, on the sepsis-induced plasma lipid changes were also examined. Among the 147 plasma lipid species from 13 lipid subgroups (fatty acid [FA], LPA, LPC, lysophosphatidylethanolamine [LPE], phosphatidic acid [PA], phosphatidylcholine [PC], phosphatidylethanolamine [PE], phosphatidylinositol [PI], monoacylglyceride [MG], diacylglyceride [DG], triacylglyceride [TG], sphingomyelin [SM], and ceramide [Cer]) analyzed in this study, 40 and 70 species were increased, and decreased, respectively, in the CLP mice. Treatments with LPC and LPA affected 14 species from 7 subgroups, and 25 species from 9 subgroups, respectively. These results could contribute to finding the much needed reliable biomarkers of sepsis.

• Speech Sciences
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• v.2
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• pp.57-66
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• 1997

This paper studies the effects of the method, CMS(Cepstral Mean Subtraction), (which compensates for some of the speech distortion. caused by telephone channels), on the performance of the text-dependent speaker identification system. This system is based on the VQ(Vector Quantization) and HMM(Hidden Markov Model) method and chooses the LPC-Cepstrum and Mel-Cepstrum as the feature vectors extracted from the speech data transmitted through telephone channels. Accordingly, we can compare the correct recognition rates of the speaker identification system between the use of LPC-Cepstrum and Mel-Cepstrum. Finally, from the experiment results table, it is found that the Mel-Cepstrum parameter is proven to be superior to the LPC-Cepstrum and that recognition performance improves by about 10% when compensating for telephone channel using the CMS.

• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.738-741
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• 1988

In this paper, we deal with the movements identification of EMG signals by LPC cepstrum coefficients. Movements were identified by extration of characteristics of similar patterns in Euclid distance measurement method for EMG signals generated by voluntary contractions of subject's musculature. As number of coefficients is larger, we obtain the better rate of movements identification. By exact extraction of signals and decision of optimal coefficient, it is expected that these results will apply to prosthesis control in real-time.

• KIEE International Transactions on Electrophysics and Applications
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• v.3C no.5
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• pp.177-181
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• 2003

Partial discharge pulse shapes from variable PD (partial discharge) sources sustain many characteristics such as types of PD. Ultra high frequency antennas have wide bandwidth from 30KHz to 2㎓. Therefore, signals taken from a UHF antenna have important attributes (rising time, falling time, shape factor, etc.) for electromagnetic sources, such as PD sources. We investigated PD pulse shapes from several PD sources using a UHF antenna and the results were used for classification of PD sources. Features for discrimination are extracted from frequency distribution and LPC (Linear Prediction Coefficient) of time signal. RBFN are used for investigating the possibility of classification of multi-PD sources.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.45-49
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• 1991

In area of the speech synthesis, the waveform coding with high quality are mainly used to the synthesis by analysis. However, it is difficult to applying the waveform coding to the synthesis by rule, because the parameters of this coding are not classified as either excitation parameters and vocal tract parameters. In this paper, we proposed a new pitch change method that can alter the pitch periods in the waveform coding. The proposed method expands the pitch period by the LPC synthesis method, and then the period is compressed by the waveform halving technique. Thus, it is possible that the waveform coding is carried out the synthesis by rule in speech processing.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.28B no.11
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• pp.866-875
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• 1991

In this paper, multi pulse excited-linear predictive coding (MPE-LPC), where the correlation eliminated residual signal is modeled by a few pules, is shown to be effective for the compression of electrocardiogram (ECG) data, and a more efficient scheme for a faithful reconstruction of ECG is proposed. The reconstruction charateristic of QRS's and P.T waves is improved using the adaptive pulse allocation (APA), and the compression ratio (CR) can be changed by controlling the mumber of modeling pulses. The performance of the proposed method was evaluated using 10 normal and 10 abnormal ECG data. The proposed method had a better performance than the variable threshold amplitude zone time epoch coding (AZTEC) algorithm and the scan-along polygonal approximation (SAPA) algorithm with the same CR. With the CR in kthe range of 8:1 to 14:1, we could compress ECG data efficiently.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1799-1802
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• 2002

This paper proposes an improved method for formant frequencies estimation based on the complex autocorrelation function of the speech signal. Instead of using the incoming signal as an input fur the LPC analysis, the analytic signal of the autocorrelation function of the speech signal is computed and itself used as an input for the LPC analysis. Due to the properties of the analytic signal, which occupies half of the bandwidth of the original signal, the required model order for the LPC analysis is halved. The accuracy of the proposed method in noisy environments is examined on five natural vowels. The effectiveness of the proposed method is shown by the estimated spectral shapes and the estimation errors of the formant frequencies.