• Asia-pacific Journal of Multimedia Services Convergent with Art, Humanities, and Sociology
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• v.8 no.6
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• pp.927-937
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• 2018

In this paper the comparative study of the consonants of Pyongyang and Seoul dialects of Korean is performed from the perspective of the signal processing which can be regarded as the basis of engineering applications. Until today, the most of speech signal studies were primarily focused on the vowels which are playing important role in the language evolution. In any language, however, the number of consonants is greater than the number of vowels. Therefore, the research of consonants is also important. In this paper, with the vowel study of the Pyongyang dialect, which was conducted by phonological research and experimental phonetic methods, the consonant studies are processed based on an engineering operation. The alveolar consonant, which has demonstrated many differences in the phonetic value between Pyongyang and Seoul dialects, was used as the experimental data. The major parameters of the speech signal analysis - formant frequency, pitch, spectrogram - are measured. The phonetic values between the two dialects were compared with respect to /시/ and /씨/ of Korean language. This study can be used as the basis for the voice recognition and the voice synthesis in the future.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2005.11a
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• pp.250-255
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• 2005

In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and reduces the computational complexity. The proposed filter is applied to wavelet based adaptive algorithm. Actually we compared the performance of the proposed algorithm with other algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, the frequency domain algorithm is prefer than the existent time domain. we analyzed the Wavelet algorithm, short-length fast running FIR algorithm, fast-short-length fast running FIR algorithm and proposed algorithm.

• Journal of Institute of Control, Robotics and Systems
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• v.11 no.12
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• pp.1020-1026
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• 2005

In the era of ubiquitous computing, human-friendly man-machine interface is getting more attention due to its possibility to offer convenient services. For this, in this paper, we introduce a 'Half-Mirror Interface System (HMIS)' as a novel type of human-friendly man-machine interfaces. Basically, HMIS consists of half-mirror, USB-Webcam, microphone, 2ch-speaker, and high-speed processing unit. In our HMIS, two principal operation modes are selected by the existence of the user in front of it. The first one, 'mirror-mode', is activated when the user's face is detected via USB-Webcam. In this mode, HMIS provides three basic functions such as 1) make-up assistance by magnifying an interested facial component and TTS (Text-To-Speech) guide for appropriate make-up, 2) Daily weather information provider via WWW service, 3) Health monitoring/diagnosis service using Chinese medicine knowledge. The second one, 'display-mode' is designed to show decorative pictures, family photos, art paintings and so on. This mode is activated when the user's face is not detected for a time being. In display-mode, we also added a 'healing-window' function and 'healing-music player' function for user's psychological comfort and/or relaxation. All these functions are accessible by commercially available voice synthesis/recognition package.

• The Journal of the Acoustical Society of Korea
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• v.18 no.4
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• pp.60-64
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• 1999

In this paper, we apply CART(Classification And Regression tree) and OC1(Oblique Classifier1) which methods are widely used for continuous speech recognition and synthesis. We prediet prosodic boundary index by applying CART and OC1, which combine right depth of tree-structured method and To_Right of link grammar method with tri_gram model. We assigned four prosodic boundary index level from 0 to 3. Experimental results show that OC1 method is superior to CART method. In other words, in spite of OC1's having fewer nodes than CART, it can make more improved prediction than CART.

• Transactions on Control, Automation and Systems Engineering
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• v.3 no.4
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• pp.223-228
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• 2001

We have been developed an intelligent mobile robot with an artificial olfactory function to recognize odours and to track odour source location. This mobile robot also has ben installed an engine for speech recognition and synthesis and is controlled by wireless communication. An artificial olfactory system based on array of 7 gas sensors has been installed in the mobile robot for odour recognition, and 11 gas sensors also are located in the obttom of robot to track odour sources. 3 optical sensors are also in cluded in the intelligent mobile robot, which is driven by 2 D. C. motors, for clash avoidance in a way of direction toward an odour source. Throughout the experimental trails, it is confirmed that the intelligent mobile robot is capable of not only the odour recognition using artificial neural network algorithm, but also the tracking odour source using the step-by-step approach method. The preliminary results are promising that intelligent mobile robot, which has been developed, is applicable to service robot system for environmental monitoring, localization of odour source, odour tracking of hazardous areas etc.

• Proceedings of the KSPS conference
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• 1996.10a
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• pp.53-69
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• 1996

Much research has been done on the rues differentiating the three Korean stops in word initial position. This paper focuses on a more neglected area: the acoustic cues differentiating the medial tense and lax unaspirated stops. Eight adult Korean native speakers, four males and four females, pronounced sixteen minimal pairs containing the two series of medial stops with different preceding vowel qualities. The average duration of vowels before lax stops is 31 msec longer than before their tense counterparts (70 msec for lax vs 39 msec for tense). In addition, the average duration of the stop closure of tense stops is 135 msec longer than that of lax stops (69 msec for lax vs 204msec for tense). THESE DURATIONAL DIFFERENCES ARE 50 LARGE THAT THEY MAY BE PHONOLOGICALLY DETERMINED, NOT PHONETICALLY. Moreover, vowel duration varies with the speaker's sex. Female speakers have 5 msec shorter vowel duration before both stops. The quality of voicing, tense or lax, is also a cue to these two stop types, as it is in initial position, but the relative duration of the stops appears to be much more important cues. The duration of stops changes the stop perception while that of preceding vowel does not. The consequences of these results for the phonological description of Korean as well as the synthesis and automatic recognition of Korean will be discussed.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.10a
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• pp.150-155
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• 2014

본 논문에서는, 원거리 음성인식을 위한 서브밴드 필터링 기반의 빠르고 효율적인 에코제거 시스템을 제안한다. 제안하는 에코제거 시스템은 우선 채널간 유사도 (correlation) 가 높을 경우 적응필터가 오작동하는 것을 방지하기 위해 spatial decorrelation 을 적용하게 된다. 그리고 tree 형태를 가지는 IIR filterbank 기반의 subband 구조를 채택함으로써, 적은 차수로도 효과적인 analysis, synthesis 필터링을 수행할 수 있도록 한다. 이 과정에서 불가피하게 발생하는 서브 밴드간 spectral aliasing은 notch filter를 적용해 해결할 수 있다. 또한 적응 필터로는 improved proportionate normalized least-mean-square (IP-NLMS) 알고리즘을 사용해 수렴속도 및 에코제거 성능에서 우수함을 확인하였다. 마지막으로 decision-directed estimation 기반의 residual echo suppressor를 적용해 잔여 에코를 제거하게 된다. 본 논문에서는 각 단계를 구성하게 된 이론적인 배경을 소개하고, 실제 에코가 존재하는 환경에서 ERLE, 원거리 음성 인식률, computational complexity를 통해 제안하는 에코제거 시스템의 효과를 입증하도록 한다.

• The Journal of the Acoustical Society of Korea
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• v.32 no.2
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• pp.174-180
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• 2013

This paper proposes a decision tree based context clustering algorithm for HMM-based speech synthesis systems using the cross likelihood ratio with a hierarchical prior (CLRHP). Conventional algorithms tie the context-dependent HMM states that have similar statistical characteristics, but they do not consider the statistical similarity of split child nodes, which does not guarantee the statistical difference between the final leaf nodes. The proposed CLRHP algorithm improves the reliability of model parameters by taking a criterion of minimizing the statistical similarity of split child nodes. Experimental results verify the superiority of the proposed approach to conventional ones.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.28 no.3
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• pp.133-141
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• 2005

The previous research has been mainly focused on individual technology of voice recognition, voice synthesis, translation, and bone transmission technical. Recently, commercial models have been produced using aforementioned technologies. In this research, a new automated translation support system concept has been proposed by combining established technology of bone transmission and wireless system. The proposed system has following three major components. First, the hybrid system consist of headset, bone transmission and other technologies will recognize user's voice. Second, computer recognized voice (using small server attached to the user) of the user will be converted into digital signal. Then it will be translated into other user's language by translation algorithm. Third, the translated language will be wirelessly transmitted to the other party. The transmitted signal will be converted into voice in the other party's computer using the hybrid system. This hybrid system will transmit the clear message regardless of the noise level in the environment or user's hearing ability. By using the network technology, communication between users can also be clearly transmitted despite the distance.

• Proceedings of the Korea Information Processing Society Conference
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• 2021.05a
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• pp.494-496
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• 2021

최근 경량 딥러닝 기반 키워드 음성인식은 가전, 완구, 키오스크 등 다양한 응용에 음성 인터페이스를 쉽게 적용할 수 있는 기술로서 주목받고 있다. 키워드 음성인식은 일부 키워드만 인식 가능한 음성인식 기술로서 저성능 디바이스에서 활용 가능한 장점이 있다. 그러나 응용에 따라 필요한 키워드에 대하여 다시 음성데이터를 수집해야하고 이를 학습하여 모델을 새로 준비해야하는 단점이 있다. 따라서 본 연구에서는 음성데이터 수집 없이 음성합성을 통해 생성한 음성으로만 키워드 음성인식 모델을 학습하는 음성합성 기반 자동 학습 기법을 제안하였다. 생성한 음성데이터를 활용하고자하는 시도가 활발히 이루어지고 있으나, 기존 연구에서는 정확도를 유지하기 위하여 수집한 실제 음성데이터가 필요한 한계가 있다. 제안한 자동 학습 기법은 생성한 음성데이터에 대해 복합 데이터 증대 기법을 적용하여 실제 음성데이터 없이 키워드 음성인식의 정확도를 높였다. 제안한 기법에 대하여 상용 음성합성 서비스를 기반으로 수집한 한국어 키워드 데이터세트를 활용하여 성능평가를 진행하였다. 20개 한국어 키워드에 대해 실험한 결과, 제안한 기법을 적용하여 학습시킨 키워드 음성인식 모델의 정확도는 86.44%임을 확인하였다.