• Title/Summary/Keyword: Cepstrum

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A Study on the Thickness Measurement of Thin Film and the Flaw Detection of the Interface by Digital Signal Processing (디지털 신호처리에 의한 박판두께측정 및 접합경계면의 결함검출에 관한 연구)

  • Kim, Jae-Yeol;Yiu, Shin;Kim, Byung-Hyun
    • Proceedings of the Korean Society of Precision Engineering Conference
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    • 1997.04a
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    • pp.123-127
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    • 1997
  • Recently, it is gradually raised necessity that interface is measured accurately and managed in industrial circles and medical world, An Ultrasonic wave transmitted from a focused beam transducer is being expected as a powerful tool for NDE of micro-defect. The ultrasonic NDE of the defect is based on the form of the wave reflected form the interface In this study, regarding to the thickness of film which is in opaque object and thickness measurement was done by MEM-cepstrum analysis of received ultrasonic wave. In measument results, film thickness which is beyond distance resolution capacity was measured accurately. Also, automatically repeated discrimination analysis method can be decided in the category of all kinds of defects on semiconductor package.

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Spectrum Representation Based on LPC Cepstral VQ for Low Bit Rate CELP Coder (LPC Cepstral 벡터 양자화에 의한 저 전송율 CELP 음성부호기의 스펙트럼 표기)

  • 정재호
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.19 no.4
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    • pp.761-771
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    • 1994
  • This paper focuses on how spectrum information can be represented efficiently in a very low bit rate CELP speech coder. To achieve the goal, an LPC cepstral coefficients VQ scheme representing the spectrum information in a CELP coder is proposed. To represent the spectrum information using LPC cepstrums, three different cepstral distance measures having different spectral meanings in the frequency domain are considered, and their performances are compared and analyzed. The experimental results show that spectrum information in low bit rate CELP coders can be represented very efficiently using the proposed LPC cepstral vector quantization scheme.

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Organ Recognition in Ultrasound images Using Log Power Spectrum (로그 전력 스펙트럼을 이용한 초음파 영상에서의 장기인식)

  • 박수진;손재곤;김남철
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.9C
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    • pp.876-883
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    • 2003
  • In this paper, we propose an algorithm for organ recognition in ultrasound images using log power spectrum. The main procedure of the algorithm consists of feature extraction and feature classification. In the feature extraction, as a translation invariant feature, log power spectrum is used for extracting the information on echo of the organs tissue from a preprocessed input image. In the feature classification, Mahalanobis distance is used as a measure of the similarity between the feature of an input image and the representative feature of each class. Experimental results for real ultrasound images show that the proposed algorithm yields the improvement of maximum 30% recognition rate than the recognition algorithm using power spectrum and Euclidean distance, and results in better recognition rate of 10-40% than the recognition algorithm using weighted quefrency complex cepstrum.

Noise Source Identification Scheme for Multi-Source Signal using the Cepstrum Technique (캡스트럼을 이용한 다중 응답신호의 소음원 해석기법)

  • Kim, Kyung-Yong;Lee, Jang-Myung
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.37 no.6
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    • pp.76-82
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    • 2000
  • To reduce the radiated noise of ships, the noises which are generated from onboard machinery, propulsion system and transfer characteristics of structure must be identified. While the ship is operating, however, we can not directly measure each signal of inputs and characteristics of transfer passage, because measured signals are superimposed by multi source and multi transfer passage. In this paper, the signal processing method for separating noise sources and transfer functions from the measured response signal by the cepstrum technique is proposed. The proposed method is verified by application of simulated signal and impact test and shows usefulness by application of real ship test.

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On a Pitch Change of the Waveform Coding by the Cepstrum Analysis of Speech Waveforms (켑스트럼 분석에 의한 파형부호화의 피치변경에 관한 연구)

  • Bae, Myung-Jin;Lee, Mi-Suk
    • The Journal of the Acoustical Society of Korea
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    • v.11 no.4
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    • pp.14-21
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    • 1992
  • The waveform coding is concerned with simply preserving the wave shape of speech signal through a redundancy reduction process. In area of the speech synthesis, the waveform codings with high quality are mainly used to the synthesis by analysis. However, because the parameters of this coding are not classified as either excitation parameters and vocal tract parameters, it is difficult to applying the waveform coding to the synthesis by rule. In this paper, we proposed a new pitch alternation method that can change the pitch periods in the waveform coding by using the cepstrum analysis. Thus, it is possible that the waveform coding is carried out the synthesis by rule in speech processing.

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Speaker-dependent Speech Recognition Algorithm for Male and Female Classification (남녀성별 분류를 위한 화자종속 음성인식 알고리즘)

  • Choi, Jae-Seung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.17 no.4
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    • pp.775-780
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    • 2013
  • This paper proposes a speaker-dependent speech recognition algorithm which can classify the gender for male and female speakers in white noise and car noise, using a neural network. The proposed speech recognition algorithm is trained by the neural network to recognize the gender for male and female speakers, using LPC (Linear Predictive Coding) cepstrum coefficients. In the experiment results, the maximal improvement of total speech recognition rate is 96% for white noise and 88% for car noise, respectively, after trained a total of six neural networks. Finally, the proposed speech recognition algorithm is compared with the results of a conventional speech recognition algorithm in the background noisy environment.

A Study on Human Training System for Prosthetic Arm Control (의수제어를 위한 인체학습시스템에 관한 연구)

  • 장영건;홍승홍
    • Journal of Biomedical Engineering Research
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    • v.15 no.4
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    • pp.465-474
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    • 1994
  • This study is concerned with a method which helps human to generate EMG signals accurately and consistently to make reliable design samples of function discriminator for prosthetic arm control. We intend to ensure a signal accuracy and consistency by training human as a signal generation source. For the purposes, we construct a human training system using a digital computer, which generates visual graphes to compare real target motion trajectory with the desired one, to observe EMG signals and their features. To evaluate the effect which affects a feature variance and a feature separability between motion classes by the human training system, we select 4 features such as integral absolute value, zero crossing counts, AR coefficients and LPC cepstrum coefficients. We perform a experiment four times during 2 months. The experimental results show that the hu- man training system is effective for accurate and consistent EMG signal generation and reduction of a feature variance, but is not correlated for a feature separability, The cepstrum coefficient is the most preferable among the used features for reduction of variance, class separability and robustness to a time varing property of EMG signals.

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Performance Improvement of Speech Recognition Based on Independent Component Analysis (독립성분분석법을 이용한 음성인식기의 성능향상)

  • 김창근;한학용;허강인
    • Proceedings of the Korea Institute of Convergence Signal Processing
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    • 2001.06a
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    • pp.285-288
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    • 2001
  • In this paper, we proposed new method of speech feature extraction using ICA(Independent Component Analysis) which minimized the dependency and correlation among speech signals on purpose to separate each component in the speech signal. ICA removes the repeating of data after finding the axis direction which has the greatest variance in input dimension. We verified improvement of speech recognition ability with training and recognition experiments when ICA compared with conventional mel-cepstrum features using HMM. Also, we can see that ICA dealt with the situation of recognition ability decline that is caused by environmental noise.

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Character-Based Video Summarization Using Speaker Identification (화자 인식을 통한 등장인물 기반의 비디오 요약)

  • Lee Soon-Tak;Kim Jong-Sung;Kang Chan-Mi;Baek Joong-Hwan
    • Journal of the Institute of Convergence Signal Processing
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    • v.6 no.4
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    • pp.163-168
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    • 2005
  • In this paper, we propose a character-based summarization algorithm using speaker identification method from the dialog in video. First, we extract the dialog of shots containing characters' face and then, classify the scene according to actor/actress by performing speaker identification. The classifier is based on the GMM(Gaussian Mixture Model) using the 24 values of MFCC(Mel Frequency Cepstrum Coefficient). GMM is trained to recognize one actor/actress among four who are all trained by GMM. Our experiment result shows that GMM classifier obtains the error rate of 0.138 from our video data.

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Speaker Verification Model Using Short-Time Fourier Transform and Recurrent Neural Network (STFT와 RNN을 활용한 화자 인증 모델)

  • Kim, Min-seo;Moon, Jong-sub
    • Journal of the Korea Institute of Information Security & Cryptology
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    • v.29 no.6
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    • pp.1393-1401
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    • 2019
  • Recently as voice authentication function is installed in the system, it is becoming more important to accurately authenticate speakers. Accordingly, a model for verifying speakers in various ways has been suggested. In this paper, we propose a new method for verifying speaker verification using a Short-time Fourier Transform(STFT). Unlike the existing Mel-Frequency Cepstrum Coefficients(MFCC) extraction method, we used window function with overlap parameter of around 66.1%. In this case, the speech characteristics of the speaker with the temporal characteristics are studied using a deep running model called RNN (Recurrent Neural Network) with LSTM cell. The accuracy of proposed model is around 92.8% and approximately 5.5% higher than that of the existing speaker certification model.