• 정보처리학회 논문지
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• 제13권4호
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• pp.174-180
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• 2024

본 논문에서는 wav2vec 2.0과 KcELECTRA 모델을 활용하여 멀티모달 학습을 통한 감정 분류 방법을 탐색한다. 음성 데이터와 텍스트 데이터를 함께 활용하는 멀티모달 학습이 음성만을 활용하는 방법에 비해 감정 분류 성능을 유의미하게 향상시킬 수 있음이 알려져 있다. 본 연구는 자연어 처리 분야에서 우수한 성능을 보인 BERT 및 BERT 파생 모델들을 비교 분석하여 텍스트 데이터의 효과적인 특징 추출을 위한 최적의 모델을 선정하여 텍스트 처리 모델로 활용한다. 그 결과 KcELECTRA 모델이 감정 분류 작업에서 뛰어난 성능이 보임을 확인하였다. 또한, AI-Hub에 공개되어 있는 데이터 세트를 활용한 실험을 통해 텍스트 데이터를 함께 활용하면 음성 데이터만 사용할 때보다 더 적은 양의 데이터로도 더 우수한 성능을 달성할 수 있음을 발견하였다. 실험을 통해 KcELECTRA 모델을 활용한 경우가 정확도 96.57%로 가장 우수한 성능을 보였다. 이는 멀티모달 학습이 감정 분류와 같은 복잡한 자연어 처리 작업에서 의미 있는 성능 개선을 제공할 수 있음을 보여준다.

• 음성과학
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• 제12권2호
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• pp.183-191
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• 2005

This paper describes the automated call routing for call center system based on speech recognition. We focus on the task of automatically routing telephone calls based on a users fluently spoken response instead of touch tone menus in an interactive voice response system. Vector based call routing algorithm is investigated and normalization method suggested. Call center database which was collected by KT is used for call routing experiment. Experimental results evaluating call-classification from transcribed speech are reported for that database. In case of small training data, an average call routing error reduction rate of 9% is observed when normalization method is used.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1998년도 제15회 음성통신 및 신호처리 워크샵(KSCSP 98 15권1호)
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• pp.87-92
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• 1998

In general, the synthesis unit database was constructed by recording isolated word. In that case, each boundary of word has typical prosodic pattern like a falling intonation or preboundary lengthening. To get natural synthetic speech using these kinds of database, we must artificially distort original speech. However, that artificial process rather resulted in unnatural, unintelligible synthetic speech due to the excessive prosodic modification on speech signal. To overcome these problems, we gathered thousands of sentences for synthesis database. To make a phone level synthesis unit, we trained speech recognizer with the recorded speech, and then segmented phone boundaries automatically. In addition, we used laryngo graph for the epoch detection. From the automatically generated synthesis database, we chose the best phone and directly concatenated it without any prosody processing. To select the best phone among multiple phone candidates, we used prosodic information such as break strength of word boundaries, phonetic contexts, cepstrum, pitch, energy, and phone duration. From the pilot test, we obtained some positive results.

• 한국산학기술학회논문지
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• 제14권8호
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• pp.3992-3998
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• 2013

본 논문은 고음질의 대용량 코퍼스 기반 음성 합성기에 감정 음성 코퍼스를 추가하여 보다 다양한 합성음을 생성할 수 있는 방법에 관한 것이다. 파형 접합형 합성기에서 사용할 수 있는 형태로 감정 음성 코퍼스를 구축하여 기존의 일반 음성 코퍼스와 동일한 합성단위 선택과정을 통해 합성음을 생성할 수 있도록 구현하였다. 감정 음성 합성을 위해 태그를 사용하여 텍스트를 입력하고, 억양구 단위로 일치하는 데이터가 존재하는 경우 감정 음성으로 합성하고, 그렇지 않은 경우 일반 음성으로 합성하도록 하였다. 그리고 음성에서 운율을 구성하는 요소로 휴지기(break)가 있는데, 감정 음성의 휴지기는 일반 음성보다 불규칙한 특성이 있다. 따라서 합성기에서 생성되는 휴지기 정보를 감정 음성 합성에 그대로 사용하는 것이 어려워진다. 이 문제를 해결하기 위해 가변 휴지기(Variable break)[3] 모델링을 적용하였다. 실험은 일본어 합성기를 사용하였고, 그 결과 일반 음성의 휴지기 예측 모듈을 그대로 사용하면서 자연스러운 감정 합성음을 얻을 수 있었다.

• 대한음성학회:학술대회논문집
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• 대한음성학회 1996년도 10월 학술대회지
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• pp.542-547
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• 1996

Prosodic characteristics of natural speech, especially intonation, in many cases represent specific feelings of the speaker at the time of the utterance, with relatively vast variations of speaking styles over the same text. We analyzed a collected speech corpus, recorded with ten Slovene speakers. Interpretation of observed intonation contours was done for the purpose of modelling the intonation contour in synthesis process. We devised a scheme for modeling the intonation contour for different types of intonation units based on the results of analyzing intonation contours. The intonation scheme uses a superpositional approach, which defines the intonation contour as the sum of global (intonation unit) and local (accented syllables or syntactic boundaries) components. Near-to-natural intonation contour was obtained by rules, using only the text of the utterance as input.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2003년도 5월 학술대회지
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• pp.87-90
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• 2003

In order to produce high quality synthesized speech, it is very important to get an accurate grapheme-to-phoneme conversion and prosody model from texts using natural language processing. Robust preprocessing for non-Korean characters should also be required. In this paper, we analyzed Korean texts using a morphological analyzer, part-of-speech tagger and syntactic chunker. We present a new grapheme-to-phoneme conversion method, i.e. a dictionary-based and rule-based hybrid method, for unlimited vocabulary Korean TTS. We constructed a prosody model using a probabilistic method and decision tree-based method.

• The Journal of the Acoustical Society of Korea
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• 제24권2E호
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• pp.54-60
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• 2005

A new multichannel blind deconvolution algorithm is proposed for speech mixtures. It employs unidirectional filters and normalization of gradient terms in the frequency domain. The proposed algorithm is shown to be approximately nonholonomic. Thus it provides improved convergence and separation performances without whitening effect for nonstationary sources such as speech and audio signals. Simulations using real world recordings confirm superior performances over existing algorithms and its usefulness for real applications.

• 한국음향학회지
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• 제1권1호
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• pp.54-59
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• 1982

We present an improved 4800 bps LPC vocoder system that virtually eleminates the buzzy effect from synthetic speech. Excitation signal in the new system is formed by adding high-pass filtered pitch pulses or random noise to a baseband residual signal that has been coded by pitch predictive PCM. Since the baseband residual is used as a part of excitation, the system is also robust to V/UV and pitch errors. According to our informal listening tests, the synthetic speech of the new system does not have the buzzy effect. As a result the vocoder speech quality is more natural than that of a conventioinal LPC vocoder.

• 음성과학
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• 제12권4호
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• pp.53-70
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• 2005

A Vowel of speech signals are multicomponent signals composed of AM-FM components whose instantaneous frequency and instantaneous amplitude are time-varying. The changes of emotion states cause the variation of the instantaneous frequencies and the instantaneous amplitudes of AM-FM components. Therefore, it is important to estimate exactly the instantaneous frequencies and the instantaneous amplitudes of AM-FM components for the extraction of key information representing emotion states and changes in speech signals. In tills paper, firstly a method decomposing speech signals into AM - FM components is addressed. Secondly, the fundamental frequency of vowel sound is estimated by the simple method based on the spectrogram. The estimate of the fundamental frequency is used for decomposing speech signals into AM-FM components. Thirdly, an estimation method is suggested for separation of the instantaneous frequencies and the instantaneous amplitudes of the decomposed AM - FM components, based on Hilbert transform and the demodulation property of the extended Fourier transform. The estimates of the instantaneous frequencies and the instantaneous amplitudes can be used for modification of the spectral distribution and smooth connection of two words in the speech synthesis systems based on a corpus.

• 대한음성학회지:말소리
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• 제50호
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• pp.41-50
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• 2004

Recently, there has been considerable progress in the natural language processing and digital signal processing components and this progress has led to the improved synthetic speech qualify of many commercial TTS systems. But there still remain many obstacles to overcome for the practical application of TTS. To resolve the problems, the cooperative research among the related areas is highly required and a common Korean TTS platform is essential to promote these activities. This platform offers a general framework for building Korean speech synthesis systems and a full C/C++ source for modules supports to implement and test his own algorithm. In this paper we described the aspect of a Korean TTS platform to be developed and a developing plan.