• Proceedings of the KSPS conference
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• 대한음성학회 2006년도 춘계 학술대회 발표논문집
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• pp.127-130
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• 2006

This paper suggests an algorithm that can estimate the direction of the sound source with three microphones arranged on a circle. The algorithm is robust to microphones' gains because it uses only the time differences between microphones. To make this possible, a cost function which normalizes the microphone's gains is utilized and a procedure to detect the rough position of the sound source is also proposed. Through our experiments, we obtained significant performance improvement compared with the energy-based localizer.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제4권2호
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• pp.254-258
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• 2004

In this paper, we propose a robust speech recognition system under noisy environments. Since the presence of noise severely degrades the performance of speech recognition system, it is important to design the robust speech recognition method against noise. The proposed method adopts a new distance measure technique based on stochastic probability instead of conventional method using minimum error. For evaluating the performance of the proposed method, we compared it with conventional distance measure for the 10-isolated Korean digits with car noise. Here, the proposed method showed better recognition rate than conventional distance measure for the various car noisy environments.

• Speech Sciences
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• 제9권4호
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• pp.253-263
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• 2002

In this paper, we propose new algorithm of speaker recognition which identifies the speaker using the information obtained by the intensive speech feature analysis such as pitch, intensity, duration, and formant, which are crucial parameters of individual voice, for candidates of high percentage of wrong recognition in the existing speaker recognition algorithm. For testing the power of discrimination of individual parameter, DTW (Dynamic Time Warping) is used. We newly set the range of threshold which affects the power of discrimination in speech verification such that the candidates in the new range of threshold are finally discriminated in the next stage of sound parameter analysis. In the speaker verification test by using voice DB which consists of secret words of 25 males and 25 females of 8 kHz 16 bit, the algorithm we propose shows about 1% of performance improvement to the existing algorithm.

• Phonetics and Speech Sciences
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• 제4권4호
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• pp.79-86
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• 2012

In the two-band excitation (TBE) model, maximum voiced frequency (MVF) is the most important feature of the excitation parameter because the synthetic speech quality depends on MVF. Thus, this paper proposes an enhanced MVF estimation scheme based on the peak picking method. In the proposed scheme, the local peak and the peak lobe are picked from the spectrum of a linear predictive residual signal. The normalized distance between neighboring peak lobes is calculated and utilized as a feature to estimate MVF. Experimental results of both objective and subjective tests show that the proposed scheme improves synthetic speech quality compared with that of the conventional one.

• MALSORI
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• 제47호
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• pp.131-140
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• 2003

We propose a new method of generating client models for HMM based text-dependent speaker verification system with only a small amount of training data. To make a client model, statistical methods such as segmental K-means algorithm are widely used, but they do not guarantee the quality or reliability of a model when only limited data are avaliable. In this paper, we propose a blind speech segmentation based on level building DTW algorithm as an alternative method to make a client model with limited data. In addition, considering the fact that voiced sounds have much more speaker-specific information than unvoiced sounds and energy of the former is higher than that of the latter, we also propose a new score evaluation method using the observation probability raised to the power of weighting factor estimated from the normalized log energy. Our experiment shows that the proposed methods are superior to conventional HMM based speaker verification system.

• Journal of rehabilitation welfare engineering & assistive technology
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• 제11권2호
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• pp.143-146
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• 2017

In this paper, a novel approach is proposed to improve the performance of speech/music classification using the recurrent neural network (RNN) in the enhanced voice services (EVS) of 3GPP for hearing aids. Feature vectors applied to the RNN are selected from the relevant parameters of the EVS for efficient speech/music classification. The performance of the proposed algorithm is evaluated under various conditions and large speech/music data. The proposed algorithm yields better results compared with the conventional scheme implemented in the EVS.

• Speech Sciences
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• 제13권3호
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• pp.169-183
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• 2006

The purpose of this study was to find an effect of Metaphon Intervention on the improvement of speech intelligibility of preschool children with articulation-phonological disorders. Subjects were 4 preschool children with articulation-phonological disorders. A multiple baseline design across subjects was used to examine the effect of the program. The program consisted of 2 steps. The first step was composed of concept level, sound level, phoneme level, and word level. The second step was on sentence level. Results were as follows: First, metaphon ability of all subjects was improved after the Metaphon Intervention. Second, speech intelligibility of all subjects was improved after Metaphon Intervention. From the results above, Metaphon Intervention can be effective to improve not only phonological awareness and metaphon but also overall speech intelligibility of preschool children with articulation-phonological disorders.

• Speech Sciences
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• 제9권2호
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• pp.157-165
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• 2002

The purpose of this study was to determine the effects of associative drawing task in a Korean bilingual aphasic. The subject is a 41-year old male and lived and was educated in the United States for over 25 years(from the age of 14 through 39). His former occupation was a psychiatrist. He has had a massive lesion in the occipital lobe. This study focused on improving his spontaneous language performances by associative drawing task. The associative drawing task along with spontaneous speech is to help the subject's cognition. The ten target words in this treatment were familiar words and could be drawn easily. The results were that the associative drawing task was effective on improving the patient's drawing ability-writing ability in English only-and naming performance both in English and Korean. However, the patient's writing ability in Korean did not show any improvement.

• Phonetics and Speech Sciences
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• 제9권2호
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• pp.103-110
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• 2017

The pole filtering concept has been successfully applied to cepstral feature normalization techniques for noise-robust speech recognition. In this paper, it is proposed to apply the pole filtering selectively only to the speech intervals, in order to further improve the recognition performance for short utterances in noisy environments. Experimental results on AURORA 2 task with clean-condition training show that the proposed selectively pole-filtered cepstral mean normalization (SPFCMN) and selectively pole-filtered cepstral mean and variance normalization (SPFCMVN) yield error rate reduction of 38.6% and 45.8%, respectively, compared to the baseline system.

• Journal of the Korea Institute of Information and Communication Engineering
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• 제23권5호
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• pp.555-567
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• 2019

Speech recognition rate is lowered even in a noisy environment, and it is difficult for a person with a speech disability or an inconvenient language to use it in a social life. In addition to improving the inconvenience of using the language, 280 words were selected using the word selection method which was improved when the word was selected considering the pronunciation characteristics of the language impaired. The MEMS development device used in the experiment was made considering material, lead wire type, length and direction. We improved the speech recognition rate by using the developed word selection method and the MEMS device developed to improve the speech recognition rate due to incorrect pronunciation and severe noise. The new method of selecting words and the mems device were improved and the results were included.