• Title/Summary/Keyword: Linear Predictive Coding (LPC)

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On Speech Digitization and Bandwidth Compression Techniques[II]-Vocoding (음성신호의 디지탈화와 대역폭축소의 방법에 관하여 [II]-Vocoding)

  • 은종관
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.15 no.5
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    • pp.1-6
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    • 1978
  • This paper is a sequel of the previous paperl) on speech digitization and bandwidth compression techniques. Several recently developed vocoding techniques, that is, linear predictive coding (LPC), formant vocoding, residual excited linear prediction (RELP) vocoding, and adaptive predictive coding(APC) are discussed. Throughout the leaper emphasis is placed on the LPC approach that is presently the most promising technique in speech compression. In addition, current problems and possible solutions are discussed.

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Fault Diagnosis System of Rotating Machines Using LPC Residual Signal Energy (LPC 잔여신호의 에너지를 이용한 회전기기의 고장진단 시스템)

  • Lee, Sung-Sang;Cho, Sang-Jin;Chong, Ui-Pil
    • Journal of the Institute of Convergence Signal Processing
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    • v.6 no.3
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    • pp.143-147
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    • 2005
  • Monitoring and diagnosis of the operating machines are very important for safety operation and maintenance in the industrial fields. These machines are most rotating machines and the diagnosis of the machines has been researched for long time. We can easily see the faulted signal of the rotating machines from the changes of the signals in frequency. The Linear Predictive Coding(LPC) is introduced for signal analysis in frequency domain. In this paper, we propose fault detection and diagnosis method using the Linear Predictive Coding(LPC) and residual signal energy. We applied our method to the induction motors depending on various status of faulted condition and could obtain good results.

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Quantization of LPC Coefficients Using a Multi-frame AR-model (Multi-frame AR model을 이용한 LPC 계수 양자화)

  • Jung, Won-Jin;Kim, Moo-Young
    • The Journal of the Acoustical Society of Korea
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    • v.31 no.2
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    • pp.93-99
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    • 2012
  • For speech coding, a vocal tract is modeled using Linear Predictive Coding (LPC) coefficients. The LPC coefficients are typically transformed to Line Spectral Frequency (LSF) parameters which are advantageous for linear interpolation and quantization. If multidimensional LSF data are quantized directly using Vector-Quantization (VQ), high rate-distortion performance can be obtained by fully utilizing intra-frame correlation. In practice, since this direct VQ system cannot be used due to high computational complexity and memory requirement, Split VQ (SVQ) is used where a multidimensional vector is split into multilple sub-vectors for quantization. The LSF parameters also have high inter-frame correlation, and thus Predictive SVQ (PSVQ) is utilized. PSVQ provides better rate-distortion performance than SVQ. In this paper, to implement the optimal predictors in PSVQ for voice storage devices, we propose Multi-Frame AR-model based SVQ (MF-AR-SVQ) that considers the inter-frame correlations with multiple previous frames. Compared with conventional PSVQ, the proposed MF-AR-SVQ provides 1 bit gain in terms of spectral distortion without significant increase in complexity and memory requirement.

Speaker Recognition using LPC cepstrum Coefficients and Neural Network (LPC 켑스트럼 계수와 신경회로망을 사용한 화자인식)

  • Choi, Jae-Seung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.15 no.12
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    • pp.2521-2526
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    • 2011
  • This paper proposes a speaker recognition algorithm using a perceptron neural network and LPC (Linear Predictive Coding) cepstrum coefficients. The proposed algorithm first detects the voiced sections at each frame. Then, the LPC cepstrum coefficients which have speaker characteristics are obtained by the linear predictive analysis for the detected voiced sections. To classify the obtained LPC cepstrum coefficients, a neural network is trained using the LPC cepstrum coefficients. In this experiment, the performance of the proposed algorithm was evaluated using the speech recognition rates based on the LPC cepstrum coefficients and the neural network.

Neural-network-based Driver Drowsiness Detection System Using Linear Predictive Coding Coefficients and Electroencephalographic Changes (선형예측계수와 뇌파의 변화를 이용한 신경회로망 기반 운전자의 졸음 감지 시스템)

  • Chong, Ui-Pil;Han, Hyung-Seob
    • Journal of the Institute of Convergence Signal Processing
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    • v.13 no.3
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    • pp.136-141
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    • 2012
  • One of the main reasons for serious road accidents is driving while drowsy. For this reason, drowsiness detection and warning system for drivers has recently become a very important issue. Monitoring physiological signals provides the possibility of detecting features of drowsiness and fatigue of drivers. One of the effective signals is to measure electroencephalogram (EEG) signals and electrooculogram (EOG) signals. The aim of this study is to extract drowsiness-related features from a set of EEG signals and to classify the features into three states: alertness, drowsiness, sleepiness. This paper proposes a neural-network-based drowsiness detection system using Linear Predictive Coding (LPC) coefficients as feature vectors and Multi-Layer Perceptron (MLP) as a classifier. Samples of EEG data from each predefined state were used to train the MLP program by using the proposed feature extraction algorithms. The trained MLP program was tested on unclassified EEG data and subsequently reviewed according to manual classification. The classification rate of the proposed system is over 96.5% for only very small number of samples (250ms, 64 samples). Therefore, it can be applied to real driving incident situation that can occur for a split second.

A Study on the Affinity Between Pairs of Korean Vowels Using the Dynamic Paremeters of Vocal Tract (성도의 다이내믹 피라미터에 의한 한글 모음간의 근사도에 관한 연구)

  • 김중규;안수길
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.19 no.1
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    • pp.1-8
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    • 1982
  • Many researches on the parametric representation of speech ,signals using the adaptive linear prediction method have been studied for the past few years. In this paper, we used the LPC(Linear Predictive Coding)method to analyae the parameters of Korean vowels and by using those parameters we studied the affinity between every pair of Korean vowels. As a result of our study, it is found that each pair of Korean vowels that has a greater phonetic affinity also has a greater affinity of vocal tract parameters than other pairs.

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Fault Detection and Diagnosis of Induction Motors using LPC and DTW Methods (LPC와 DTW 기법을 이용한 유도전동기의 고장검출 및 진단)

  • Hwang, Chul-Hee;Kim, Yong-Min;Kim, Cheol-Hong;Kim, Jong-Myon
    • Journal of the Korea Society of Computer and Information
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    • v.16 no.3
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    • pp.141-147
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    • 2011
  • This paper proposes an efficient two-stage fault prediction algorithm for fault detection and diagnosis of induction motors. In the first phase, we use a linear predictive coding (LPC) method to extract fault patterns. In the second phase, we use a dynamic time warping (DTW) method to match fault patterns. Experiment results using eight vibration data, which were collected from an induction motor of normal fault states with sampling frequency of 8 kHz and sampling time of 2.2 second, showed that our proposed fault prediction algorithm provides about 45% better accuracy than a conventional fault diagnosis algorithm. In addition, we implemented and tested the proposed fault prediction algorithm on a testbed system including TI's TMS320F2812 DSP that we developed.

VHDL Implementation of an LPC Analysis Algorithm (LPC 분석 알고리즘의 VHDL 구현)

  • 선우명훈;조위덕
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.32B no.1
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    • pp.96-102
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    • 1995
  • This paper presents the VHSIC Hardware Description Language(VHDL) implementation of the Fixed Point Covariance Lattice(FLAT) algorithm for an Linear Predictive Coding(LPC) analysis and its related algorithms, such as the forth order high pass Infinite Impulse Response(IIR) filter, covariance matrix calculation, and Spectral Smoothing Technique(SST) in the Vector Sum Exited Linear Predictive(VSELP) speech coder that has been Selected as the standard speech coder for the North America and Japanese digital cellular. Existing Digital Signal Processor(DSP) chips used in digital cellular phones are derived from general purpose DSP chips, and thus, these DSP chips may not be optimal and effective architectures are to be designed for the above mentioned algorithms. Then we implemented the VHDL code based on the C code, Finally, we verified that VHDL results are the same as C code results for real speech data. The implemented VHDL code can be used for performing logic synthesis and for designing an LPC Application Specific Integrated Circuit(ASOC) chip and DsP chips. We first developed the C language code to investigate the correctness of algorithms and to compare C code results with VHDL code results block by block.

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Electroencephalogram-based Driver Drowsiness Detection System Using AR Coefficients and SVM (AR계수와 SVM을 이용한 뇌파 기반 운전자의 졸음 감지 시스템)

  • Han, Hyungseob;Chong, Uipil
    • Journal of the Korean Institute of Intelligent Systems
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    • v.22 no.6
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    • pp.768-773
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    • 2012
  • One of the main reasons for serious road accidents is driving while drowsy. For this reason, drowsiness detection and warning system for drivers has recently become a very important issue. Monitoring physiological signals provides the possibility of detecting features of drowsiness and fatigue of drivers. One of the effective signals is to measure electroencephalogram (EEG) signals and electrooculogram (EOG) signals. The aim of this study is to extract drowsiness-related features from a set of EEG signals and to classify the features into three states: alertness, drowsiness, sleepiness. This paper proposes a drowsiness detection system using Linear Predictive Coding (LPC) coefficients and Support Vector Machine (SVM). Samples of EEG data from each predefined state were used to train the SVM program by using the proposed feature extraction algorithms. The trained SVM program was tested on unclassified EEG data and subsequently reviewed according to manual classification. The classification rate of the proposed system is over 96.5% for only very small number of samples (250ms, 64 samples). Therefore, it can be applied to real driving incident situation that can occur for a split second.

Performance Improvement of Double Talk Detection before Convergence of the Echo Canceller by Using Linear Predictive Coding Filter Gain of the Primary Input Signal (주입력신호의 LPC 필터 이득을 이용한 반향제거기의 수렴전 동시통화검출 성능 개선)

  • Yoo, Jae-Ha
    • Journal of the Korean Institute of Intelligent Systems
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    • v.24 no.6
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    • pp.628-633
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    • 2014
  • This paper proposes a performance improvement method of the conventional double talk detection method which can operate before convergence of the echo canceller. The proposed method estimates the coefficients of the linear predictive coding(LPC) filter by using the primary input signal. The time-varying threshold for double talk detection is determined based on the LPC filter gain of the primary input signal level. The proposed method can reduce not only false detection rate which means wrong detection of single talk as double talk but also double talk detection delay. Computer simulation was performed using a long-term real speech signals. It is shown that the proposed method improves the conventional method in terms of lowering the false detection rate and shortening the detection delay.