• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.302-305
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• 1998

음소의 확률적 분포를 이용하는 음소 HMM 모델을 결정하기 위한 여러 가지 거리 측정방법에 대한 연구이다. 음소 HMM 모델 결정을 위해서 LPC 계수를 이용하고, 거리 측정자를 LPC 계수, LPC 스첵트럼, LPC 켑스트럼 등의 파라미터를 이용하고, 또한 양자화 과정은 k-means 와 LBG 알고리즘을 혼합한 하이브리드 알고리듬을 사용하였다. LPC 코드북을 구성하기 위해 세 가지 파라미터를 유클리디안 거리로 거리측정에 이용하였다. 이렇게 양자화한 파라미터의 평균과 분산을 구하고, 양자화한 파라미터 코드북의 확률갑승ㄹ 비교해 한국어 음소 HMM 모델 결정을 위한 거리 측정 파라미터를 비교하였으며, 그 결과 LPC 계수를 주파수 영역으로 변환하여 유클리디안 거리를 이용한 코드북의 분산이 작으므로 상대적으로 높은 확률을 가짐을 보았다.

• Journal of IKEEE
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• v.2 no.1 s.2
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• pp.61-67
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• 1998

In this thesis, experiment have performed with the speaker recognition using multilayer feedforward neural network(MFNN) model using Korean and Japanese digits . The 5 adult males and 5 adult females pronounciate form 0 to 9 digits of Korean, Japanese 7 times. And then, they are extracted characteristics coefficient through Pitch deletion algorithm, LPC analysis, and LPC Cepstral analysis to generate input pattern of MFNN. 5 times among them are used to train a neural network, and 2 times is used to measure the performance of neural network. Both Korean and Japanese, Pitch coefficients is about 4%t more enhanced than LPC or LPC Cepstral coefficients.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.7
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• pp.1370-1376
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• 2012

This paper proposes a gender recognition algorithm which classifies a male or female speaker. In this paper, characteristic vectors for the male and female speaker are analyzed, and recognition experiments for the proposed gender recognition by a neural network are performed using these characteristic vectors for the male and female. Input characteristic vectors of the proposed neural network are 10 LPC (Linear Predictive Coding) cepstrum coefficients, 12 LPC cepstrum coefficients, 12 FFT (Fast Fourier Transform) cepstrum coefficients and 1 RMS (Root Mean Square), and 12 LPC cepstrum coefficients and 8 FFT spectrum. The proposed neural network trained by 20-20-2 network are especially used in this experiment, using 12 LPC cepstrum coefficients and 8 FFT spectrum. From the experiment results, the average recognition rates obtained by the gender recognition algorithm is 99.8% for the male speaker and 96.5% for the female speaker.

• Journal of the Korea Computer Industry Society
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• v.2 no.4
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• pp.471-480
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• 2001

In this thesis, we perform the experiment of speaker recognition by identifying vowels in the pronunciation of each speaker using Adaptive Resource Theory 2(ART2) model. The 5 adult males and 5 adult females pronounce from 0 to 9 digits. We extract the vowels from the pronunciation of each speaker first, we are extracted characteristic coefficient through a pitch detection algorithm, a LPC analysis, and a LPC cepstral analysis to generate an input pattern of ART2. The experimental results showed that pitch coefficients are somewhat more enhanced than LPC or LPC cepstral coefficient.

• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.107-110
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• 2003

본 논문은 LTJ 적응필터의 실용적 구현에 관한 연구이다. 음성코덱(codec)을 사용하는 응용분야에서는 코덱 복호화단의 LPC 계수정보를 얻을 수 있으므로 이를 반사계수로 변환하여 사용하므로서 반사계수 적응에 소용되는 계산량을 감소시킬 수 있으며, 코덱에서는 프레임 또는 서브프레임 단위로 LPC 계수를 적응시키므로 시변 변환 영역 적응필터에 해당하는 LTJ 적응필터의 필터 계수 보상에 필요한 계산량을 감소시킬 수 있다. 실제 음성신호를 사용하여 제안된 실용적 구현 방법의 타당성을 검증하였다.

• Proceedings of the KIEE Conference
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• 2001.07d
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• pp.2516-2518
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• 2001

The quantization of the LPC parameters is a very important aspect of the speech compression algorithm. This paper analyzes the quantization effect of the LPC coefficients and presents the implementation of a fixed-point CELP coder based on the LPC analysis.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.7
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• pp.146-156
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• 1989

This paper discusses how to choose speech parameters and decision logics to improve the performance of speaker recognition systems. It also considers the influence of the reference patterns on the speaker recognition. It is observed from the performance analysis based on LPSs, PARCOR coefficients and LPC-cepstrum coefficients that LPC-cepstrum coefficients are superior to the others in speaker recognition without regard to the reference patterns. In order to improve the recognition performance, a new decision logic is proposed based on a generalized-distance concept. It differs from the existing methods in that it considers the statistics of customer and impostors at the same time. It turns out from a speaker verification test that the proposed decision logic ferforms better than the existing ones.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.15 no.5
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• pp.61-64
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• 1996

To improve the performance of a recognition system, namely the recognition rate, we propose a hew implementation of perceptual distance using LPC cepstrum(perceptual cepstrum, PLC). The PLC is caculated by convolution of a usual LPC cepstrum and a perceptual lifter(PL). To caculate PL, we define a new weighting function in the linear frequency domain considering the frequency scale(Bark-scale) characteristics. The PL is the inverse Fourier transform of the exponents of the weighting function. We verified our method through the speech recognition experiments. The performance of PLC was compared with that of the rasied sine liftering method.