• 마케팅과학연구
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• 제16권1호
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• pp.93-116
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• 2006

본 연구의 목적은 라디오 광고에서 정보원이 사용하는 언어 힘이 설득에 효과가 있는지를 관찰하고, 그 효과를 정보원의 성 그 자체의 효과와 비교해 봤을 때 과연 어떤 것이 설득을 결정하는 더 큰 요소인지를 알아보는 데 있다. 이 문제와 관련하여 선행 연구들은 일관성 없는 결괄르 보여주었다. 어떤 연구는 오디오 커뮤니케이션에서 언어 힘의 설득 효과는 학자의 성에 달려있다고 주장했고(Carli, 1990), 또 어ㅏ떤 연구는 오디오 커뮤니케이션에서 언어 힘의 설득 효과는 학자의 성에 관계없이 나타나는 굳건한 효과임을 주장했다(Holtgraves & Lasky, 1999; Erickson, Lind, Johnson, & O'Barr, 1978). 이에 본 연구는 라디오 광고에서 언어 힘과 정보원 성의 상호작용을 관찰함으로써 언어 힘의 설득 효과에 정보원의 성이 어떤 영향을 미치는지에 대한 실험적 증거를 제시하고자 했다. 실험 결과, 라디오 광고에서 정보원이 사용하는 언어 힘은 설득에 강한 효과가 있으며, 그 효과는 정보원 성의 영향을 받지 않는다는 것이 밝혀졌다. 이는 언어 힘의 지위 효과가 성의 지위 효과보다 더 강력하다는 것을 함축한다.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2009년도 추계학술발표대회
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• pp.239-240
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• 2009

We live in a flat world in which globalization fosters communication, travel, and trade among more than 150 countries and thousands of languages. To surmount the barriers among these languages, translation is required; Speech-to-Speech translation will automate the process. Thanks to recent advances in Automatic Speech Recognition (ASR), Machine Translation (MT), and Text-to-Speech (TTS), one can now utilize a system to translate a speech of source language to a speech of target language and vice versa in affordable manner. The three phase process establishes that the source speech be transcribed into a (set of) text of the source language (ASR) before the source text is translated into the target text (MT). Finally, the target speech is synthesized from the target text (TTS).

• 한국전자통신학회논문지
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• 제15권5호
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• pp.807-812
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• 2020

본 논문은 여러 음원신호가 혼합된 환경에서 목적으로 하는 음원신호만을 분리하기 위하여 마이크로폰을 사용한 블라인드 음원분리 알고리즘을 제안한다. 제안하는 알고리즘은 독립성분분석 방법을 기반으로 한 주파수영역 표현모델이다. 따라서 2 음원에 대한 주파수영역 독립성분분석의 실제 환경에서의 유효성 검증을 목적으로, 음원의 종류를 변경하여 주파수영역 독립성분분석을 실행하여 음원분리를 실시하여 그 향상효과를 검증한다. 파형에 의한 실험결과로부터 원래의 파형과 비교하여 2채널의 음원신호를 깨끗하게 분리할 수 있음을 명확히 하였다. 또한 목표 신호 대 간섭 에너지비율을 사용하여 비교한 실험 결과로부터 본 논문에서 제안한 알고리즘의 음원분리 성능이 기존의 알고리즘에 비하여 성능이 향상되었다는 것을 알 수 있었다.

• 대한음성학회지:말소리
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• 제48호
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• pp.117-126
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• 2003

source of speech signal is actually composed of combination of periodic and aperiodic components, although it is often modeled to either one of those. In the paper an experiment which can separate periodic and aperiodic components from speech source. Linear predictive residual signal was used as a approximated vocal source the original speech to obtain the estimated aperiodic part. Iterative extrapolation method was used to compute the aperiodic part.

• 한국소음진동공학회논문집
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• 제24권2호
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• pp.146-153
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• 2014

This study investigates the effects of seat-design elements such as seating arrangement, shape, and height on speech privacy in high-speed trains. For the evaluation of speech privacy, acoustic simulation software was used to reproduce room acoustical conditions in passenger cars on the basis of in-situ measurement data. The influences of speech source directivity and source height on privacy distance ($r_P$) were investigated, and it was found that $r_P$ determined using an omni-directional source was relatively shorter than that determined using a directional source. It was also found that $r_P$ decreased when the source height was lower than the height of the seat-back because the seat-back blocked the propagation of speech from the sound source. The effect of seating arrangement was not significant when comparing the vis-a-vis seating and one-side seating arrangements. In addition, among the alternative seat-designs, the seats that block the space between the seats and cover the space near the ear were found to show significantly enhanced speech privacy in high-speed train passenger cars.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2007년도 한국음성과학회 공동학술대회 발표논문집
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• pp.315-318
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• 2007

This paper proposes the methods to enhance the speech quality of source controlled variable bit-rate coder based on the waveform interpolation. The methods are to estimate and generate the parameters that are not transmitted from encoder to decoder by the repetition and extrapolation schemes. For the performance evaluation, the PESQ(Perceptual Evaluation of Speech Quality) scores are measured. The experimental results shows that our proposed method outperforms the conventional source controlled variable bit-rate coder. Especially, the performance of the extrapolation method is better than that of the repetition method.

• 음성과학
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• 제3권
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• pp.132-147
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• 1998

Previously developed methods for pitch modification have not been based on the voice source model. Therefore, the synthesized speech often sounds unnatural although it may be highly intelligible. The purpose of this paper is to analyze the alteration of a voice source signal with pitch period and to establish the pitch-modification rule based on the result of this analysis. We examine the alteration of the interval of closing phase, closed phase and open phase using the excitation waveform as the pitch increases. In comparison to the previous methods which performed directly on the speech signal, the pitch modification method based on a voice source model shows high intelligibility and naturalness. This study might benefit the application to the speaker identification and the voice color conversion. Therefore the proposed method will provide high quality synthetic speech.

• 말소리와 음성과학
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• 제3권2호
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• pp.89-96
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• 2011

In this paper, details of the design and implementation of a voice source simulator using Simulink and Matlab are discussed. This simulator is an implementation by model-based design concept. Voice sources can be analyzed and manipulated through various factors by choosing options from GUI input and selecting pre-defined blocks or user created ones. This kind of simulation tool can simplify the procedure of analyzing speech signals for various purposes such as voice quality analysis, pathological voice analysis, and speech coding. Also, basic analysis functions are supported to compare the original signal and the manipulated ones.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 FIFTH WESTERN PACIFIC REGIONAL ACOUSTICS CONFERENCE SEOUL KOREA
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• pp.893-898
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• 1994

In this paper, the influence of voice source estimation and modeling on speech synthesis and coding is examined and then their new estimation and modeling techniques are proposed and verified by computer simulation. It is known that the existing speech synthesizer produced the speech which is dull and inanimated. These problems are arised from the fact that existing estimation and modeling techniques can not give more accurate voice parameters. Therefore, in this paper we propose a new voice source estimation algorithm and modeling techniques which can not give more accurate voice parameters. Therefore, in this paper we propose a new voice source estimation algorithm and modeling techniques which can represent a variety of source characteristics. First, we divide speech samples in one pitch region into four parts having different characteristics. Second, the vocal-tract parameters and voice source waveforms are estimated in each regions differently using sequential SVD. Third, we propose composite source model as a new voice source model which is represented by weighted sum of pre-defined basis functions. And finally, the weights and time-shift parameters of the proposed composite source model are estimeted uning EM(estimate maximize) algorithm. Experimental results indicate that the proposed estimation and modeling methods can estimate more accurate voice source waveforms and represent various source characteristics.

• 말소리와 음성과학
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• 제4권2호
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• pp.73-78
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• 2012

Emotion recognition is an important technology in the filed of human-machine interface. To apply speech technology to emotion recognition, this study aims to establish a relationship between emotional groups and their corresponding voice characteristics by investigating various speech features. The speech features related to speech source and vocal tract filter are included. Experimental results show that statistically significant speech parameters for classifying the emotional groups are mainly related to speech sources such as jitter, shimmer, F0 (F0_min, F0_max, F0_mean, F0_std), harmonic parameters (H1, H2, HNR05, HNR15, HNR25, HNR35), and SPI.