• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 제13회 신호처리 합동 학술대회 논문집
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• pp.171-174
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• 2000

In this paper, we improve statistical-model-based speech detector using the probability that a speech occupies a frequency bin. While the previous method assumes speech energy occupies all the frequency components and use them with equal weights in the likelihood ratio test for speech detection, the proposed method assumes speech energy occupies just some frequency componets and use them with different weights in accordance with the probabilities of spectral occupancy in the test. The probability is iteratively up-dated for speech frames to contribute to the likelihood ratio test. The proposed method well reflects the characteristic distribution of speech spectrum, and yields better detection performance.

• 음성과학
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• 제7권3호
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• pp.153-165
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• 2000

During the phonation of voiced sounds, instants exist where the glottis is opened or closed, due to the periodic vibration of the vocal cord. When closed, this is called the glottal closure instant(GCI) or epoch.. The correct detection of the GCI is one of the important problems in speech processing for pitch detection, pitch synchronous analysis, and so on. Recently, it has been shown that the local maxima points of the wavelet transformed speech signal correspond to the GCIs of speech signal. In this paper, we investigate the accuracy of Gels estimated from this wavelet transformed speech signal. For this purpose we compare them with the negative peak points of the differentiated EGG signal that represents the actual GCIs of speech signal.

• 말소리와 음성과학
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• 제15권2호
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• pp.53-59
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• 2023

Detection of children with autism spectrum disorder (ASD) based on speech has relied on predefined feature sets due to their ease of use and the capabilities of speech analysis. However, clinical impressions may not be adequately captured due to the broad range and the large number of features included. This paper demonstrates that the knowledge-driven speech features (KDSFs) specifically tailored to the speech traits of ASD are more effective and efficient for detecting speech of ASD children from that of children with typical development (TD) than a predefined feature set, extended Geneva Minimalistic Acoustic Standard Parameter Set (eGeMAPS). The KDSFs encompass various speech characteristics related to frequency, voice quality, speech rate, and spectral features, that have been identified as corresponding to certain of their distinctive attributes of them. The speech dataset used for the experiments consists of 63 ASD children and 9 TD children. To alleviate the imbalance in the number of training utterances, a data augmentation technique was applied to TD children's utterances. The support vector machine (SVM) classifier trained with the KDSFs achieved an accuracy of 91.25%, surpassing the 88.08% obtained using the predefined set. This result underscores the importance of incorporating domain knowledge in the development of speech technologies for individuals with disorders.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2000년도 제15차 학술회의논문집
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• pp.338-338
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• 2000

This paper proposes a modified TD-PSOLA algorithm for Korean speech synthesis. A WSS (Weighted score search) algorithm is proposed for pitch detection and speech synthesis engine is designed using 46 phones database.

• 한국전자통신학회논문지
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• 제8권5호
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• pp.665-670
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• 2013

이 논문에서 우리는 음성 검출 및 음소 분할에 대한 새로운 방법을 제안한다. 배경 잡음으로부터 신호를 구분하기 위해 에너지를 활용하게 되는데, 그 이전에 프리엠퍼시스 FIR 필터링을 적용하는 효과에 대해 조사한다. 이 방법에 의해, 에너지 프로필에서 진폭과 주파수의 곱이 동시에 작은 부분이 두드러지게 나타나게 된다. 이 처방에 의해, 묵음/음성 경계가 종전의 방법에 비해 더 선명해짐을 실험적으로 확인하였다. 또한 이 방법을 적용함으로써, 음소 분할 또한 더 수월해짐을 밝혔다.

• 융합신호처리학회논문지
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• 제9권2호
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• pp.121-128
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• 2008

정확한 음성 구간 검출은 음성 인식 및 음성 코딩 그리고 음성 통신 시스템 등과 같은 음성 어플리케이션의 성능에 큰 영향을 미친다. 본 논문에서는 실제 운전하고 있는 상태에서 다양한 차량 노이즈 환경의 음성 구간 검출 방법을 제안한다. 기존의 음성 구간 검출은 시간 에너지, 주파수 에너지, 영 교차율, spectral entropy 등 다양한 방법을 사용하였으며 잡음 환경에서 급격하게 성능이 저하되는 단점이 있었다. 본 논문에서는 기존의 spectral entropy를 기반으로 하여 MFB(Mel-frequency Filter Banks) spectral entropy, 기울기 FFT(Fast Fourier Transform) spectral entropy, 기울기 MFB spectral entropy를 이용한 음성 구간 검출 방법을 제안한다. MFB는 멜 스케일과 FFT를 곱한 것으로 멜 스케일은 인간이 소리를 인지할 때 주파수에 대해 비선형적인 스케일이며 음성의 특징을 잘 반영한다. 제안한 MFB spectral entropy 방법은 다양한 차량 잡음 환경에서 음성 및 비음성 분별 능력을 향상시킬 수 있으며 실험 결과 93.21%의 음성 구간 검출율을 나타내었다. 이는 기존의 spectral entropy 방법과 비교할 때 MFB를 이용한 음성 구간 검출 방법이 3.2%의 검출율이 향상되었다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1984년도 추계학술발표회 논문집
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• pp.66-69
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• 1984

This paper is a study on the Endpoint Detection for Korean Speech Recognition. In speech signal process, analysis parameter was classification from Zero Crossing Rate(Z.C.R), Log Energy(L.E), Energy in the predictive error(Ep) and fundamental Korean Speech digits, /영/-/구/ are selected as date for the Recognition of Speech. The main goal of this paper is to develop techniques and system for Speech input ot machine. In order to detect the Endpoint, this paper makes choice of Log Energy(L.E) from various parameters analysis, and the Log Energy is very effective parameter in classifying speech and nonspeech segments. The error rate of 1.43% result from the analysis.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제15권4호
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• pp.300-304
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• 2015

A low bit rate speech coder based on the non-uniform sampling technique is proposed. The non-uniform sampling technique is based on the detection of inflection points (IP). A speech block is processed by the IP detector, and the detected IP pattern is compared with entries of the IP database. The address of the closest member of the database is transmitted with the energy of the speech block. In the receiver, the decoder reconstructs the speech block using the received address and the energy information of the block. As results, the coder shows fixed data rate contrary to the existing speech coders based on the non-uniform sampling. Through computer simulation, the usefulness of the proposed technique is shown. The SNR performance of the proposed method is approximately 5.27 dB with the data rate of 1.5 kbps.

• 말소리와 음성과학
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• 제5권1호
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• pp.11-16
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• 2013

This paper proposes robust feature extraction for accurate voice activity detection (VAD). VAD is one of the principal modules for speech signal processing such as speech codec, speech enhancement, and speech recognition. Noisy environments contain nonstationary noises causing the accuracy of the VAD to drastically decline because the fluctuation of features in the noise intervals results in increased false alarm rates. In this paper, in order to improve the VAD performance, harmonic-weighted energy is proposed. This feature extraction method focuses on voiced speech intervals and weighted harmonic-to-noise ratios to determine the amount of the harmonicity to frame energy. For performance evaluation, the receiver operating characteristic curves and equal error rate are measured.

• ETRI Journal
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• 제38권3호
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• pp.425-434
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• 2016

A novel average magnitude difference function (AMDF)-based pitch detection scheme (PDS) is proposed to achieve better performance in speech quality. A performance evaluation of the proposed PDS is carried out through both a simulation and a real-time implementation of a speech analysis-synthesis system. The parameters used to compare the performance of the proposed PDS with that of PDSs that are based on either a cepstrum, an autocorrelation function (ACF), an AMDF, or circular AMDF (CAMDF) methods are as follows: percentage gross pitch error (%GPE); a subjective listening test; an objective speech quality assessment; a speech intelligibility test; a synthesized speech waveform; computation time; and memory consumption. The proposed PDS results in lower %GPE and better synthesized speech quality and intelligibility for different speech signals as compared to the cepstrum-, ACF-, AMDF-, and CAMDF-based PDSs. The computational time of the proposed PDS is also less than that for the cepstrum-, ACF-, and CAMDF-based PDSs. Moreover, the total memory consumed by the proposed PDS is less than that for the ACF- and cepstrum-based PDSs.