• 음성과학
• /
• 제11권2호
• /
• pp.227-235
• /
• 2004

In this paper, we proposed a new practical implementation method of the lattice transversal joint (LTJ) adaptive filter using speech codec's information. And it was applied to the adaptive echo cancellation problem to verify the efficiency of the proposed method. Realtime implementation of the LTJ adaptive filter is very difficult due to high computational complexity for the filter coefficients compensation. However, in case of using speech codec, complexity can be reduced since linear predictive coding (LPC) coefficients are updated each frame or sub-frame instead of every sample. Furthermore, LPC coefficients can be acquired from speech decoder and transformed to the reflection coefficients. Therefore, the computational complexity for updates of the reflection coefficients can be reduced. The effectiveness of the proposed LTJ adaptive filter was verified by the experiments about convergence and tracking performance of the adaptive echo canceller.

• 대한후두음성언어의학회지
• /
• 제19권2호
• /
• pp.101-106
• /
• 2008

The speech signal has been put into a form suitable for storage and analysis by computer, several different operation can be performed. Filtering, sampling and quantization are the basic operation in digiting a speech signal. The waveform can be displayed, measured and even edited, and spectra can be computed using methods such as the Fast Fourier Transform (FFT), Linear predictive Coding (LPC), Cepstrum and filtering. The digitized signal also can be used to generate spectrograms. The spectrograph provide major advantages to the study of speech. So, author introduces the basic techniques for the acoustic recording, digital signal processing and the principles of spectrum and spectrogram.

• 대한전기학회:학술대회논문집
• /
• 대한전기학회 1988년도 전기.전자공학 학술대회 논문집
• /
• pp.33-36
• /
• 1988

In this paper, the starting point of affricate is detected from the first predictor coefficient of a 12-pole linear predictive coding (LPC) analysis and phonemic segmentation is done through measuring short time energy and zero crossing rate. By this segmentation method, the duration of an aspirate can be mearsured in order to detect an aspirate or not.

• 정보저장시스템학회논문집
• /
• 제6권1호
• /
• pp.18-24
• /
• 2010

The performance of the speech recognition mainly depends on uncertain factors such as speaker's conditions and environmental effects. The present study deals with the speech recognition of a number of multi-syllable isolated Korean words using soft computing techniques such as back-propagation neural network, fuzzy inference system, and fuzzy neural network. Feature patterns for the speech recognition are analyzed with 12th order thirty frames that are normalized by the linear predictive coding and Cepstrums. Using four models of speech recognizer, actual experiments for both single-speakers and multiple-speakers are conducted. Through this study, the recognizers of combined fuzzy logic and back-propagation neural network and fuzzy neural network show the better performance in identifying the speech recognition.

• 음성과학
• /
• 제11권4호
• /
• pp.17-28
• /
• 2004

In this paper, we propose a statistical approach to improve the performance of spectral quantization of speech coders. The proposed techniques compensate for the distortion in a decoded line spectrum pairs (LSP) vector based on a statistical mapping function between a decoded LSP vector and its corresponding original LSP vector. We first develop two codebook-based probabilistic matching (CBPM) methods based on linear mapping functions according to different assumption of distribution of LSP vectors. In addition, we propose an iterative procedure for the two CBPMs. We apply the proposed techniques to a predictive vector quantizer used for the IS-641 speech coder. The experimental results show that the proposed techniques reduce average spectral distortion by around 0.064dB.

• 자원환경지질
• /
• 제28권5호
• /
• pp.487-491
• /
• 1995

음성신호를 임펄스 반응으로 압축시키는데 사용되는 선형예측코드의 다중펄스 방법을 다중반사파를 제거시킬수 있도록 개선시켰다. 다중반사파는 층사이에서 연속 반사에 의해 발생하는 것으로서 탄성파 해석을 어렵게 한다. 본 논문에서는 개선된 다중펄스방법을 이용하여 음원 파형요소를 스파이크로 압축시키고 다중반사파를 제거하도록 하였으며, 지하 정보를 갖고 있는 반사계수 함수의 크기와 위치를 연속 계산방식에 의해 이끌어 냈었다. 개선된 다중펄스 방법의 탄성파 자료에의 적용은 좋은 결과를 보여주고 있다.

• 대한음성학회:학술대회논문집
• /
• 대한음성학회 2005년도 추계 학술대회 발표논문집
• /
• pp.51-54
• /
• 2005

This paper analyzes the vocal tract area estimation algorithm used as a part of a speech analysis program to help deaf children correct their pronunciations by comparing their vocal tract shape with normal children's. Assuming that a vocal tract is a concatenation of cylinder tubes with a different cross section, we compute the relative vocal tract area of each tube using the reflection coefficients obtained from linear predictive coding. Then, obtain the absolute vocal tract area by computing the height of lip opening with a formula modified for children's speech. Using the speech data for five Korean vowels (/a/, /e/, /i/, /o/, and /u/), we investigate the effects of the sampling frequency, frame size, and model order. We compare vocal tract shapes obtained from deaf and normal children's speech.

• ETRI Journal
• /
• 제43권1호
• /
• pp.82-94
• /
• 2021

In this work, six voiced/unvoiced speech classifiers based on the autocorrelation function (ACF), average magnitude difference function (AMDF), cepstrum, weighted ACF (WACF), zero crossing rate and energy of the signal (ZCR-E), and neural networks (NNs) have been simulated and implemented in real time using the TMS320C6713 DSP starter kit. These speech classifiers have been integrated into a linear-predictive-coding-based speech analysis-synthesis system and their performance has been compared in terms of the percentage of the voiced/unvoiced classification accuracy, speech quality, and computation time. The results of the percentage of the voiced/unvoiced classification accuracy and speech quality show that the NN-based speech classifier performs better than the ACF-, AMDF-, cepstrum-, WACF- and ZCR-E-based speech classifiers for both clean and noisy environments. The computation time results show that the AMDF-based speech classifier is computationally simple, and thus its computation time is less than that of other speech classifiers, while that of the NN-based speech classifier is greater compared with other classifiers.

• 한국음향학회지
• /
• 제10권1호
• /
• pp.20-29
• /
• 1991

기존의 선형 예측법에 의한 음성 분석의 기본적인 가정은 전극점 성도 필터의 입력은 백색 신호라는 것이다. 그러나, 주기성 입력 신호의 경우 피치 바이오스 오차가 기존 선형 예측 계수에 개입된다. 만일 여기 신호의 추정값을 이용할 수 있다면 멀티 펄스에 의한 선형 예측 분석으로 이러한 바이어스를 제거할 수 있다. 기존의 선형 예측 분석에서의 예측 오차는 멀티 펄스 여기 신호열과 불규칙 잡음 신호열의합으로 나타내어질 수 있으므로 선형 예측 오차로부터 멀티 펄스 신호열을 찾아내는 것은 고전적인 검출 및 추정의 문제로 생각될 수 있다. 본 논문에서는 먼저 LRT 를 이용하여 예측오차로부터 멀티 펄스 신호의 위치와 크기를 찾아낸 다음 이 신호열로부터 피치 바이어스가 제거된 선형 예측 계수를 구하는 알고리즘을 제안한다. 매번 적응된 임계값을 적용하여 반복 수행을 함으로써 성능향상을 입증하였다.

• 한국음향학회지
• /
• 제21권3호
• /
• pp.304-309
• /
• 2002

LSP (Line Spectrum Pairs) 파라미터는 음성코덱 (codec)이나 인식기에서 음성신호를 분석하여 전송형이나 저장형 파라미터로 변환되어, 주로 저전송률 음성부호화기에 사용된다. 그러나 LPC (Linear Predictive Coding) 계수를 LSP로 변환하는 방법이 복잡하여 계산시간이 많이 소요된다는 단점이 있다. 기존의 LSP변환 방법 중 음성 부호화기에서 주로 사용하는 실근 (real root)방법은 근을 구하기 위해 주파수 영역을 순차적으로 검색하기 때문에 계산시간이 많이 소요되는 단점을 갖는다. 본 논문에서 기존의 실근 방법과 비교 평가한 알고리즘은 첫 번째 검색 대역에 멜 스케일 (met scale)을 사용하였고, 두 번째는 LSP 파라미터의 분포 특성을 조사하여 이를 토대로 검색구간의 순서와 검색간격을 달리 하였다. 실험결과, 기존의 실근 방식에 비하여 두 가지 방식 모두가 변환시간의 47% 이상이 감소되는데 반하여 동일한 근을 찾음을 알 수가 있었다.