• Journal of Global Scholars of Marketing Science
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• v.16 no.1
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• pp.93-116
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• 2006

The main purpose of this study is to examine whether power-of-speech have an impact on persuasion in radio CM and to determine what is stronger variable either power-of-speech or sex of source. Previous research showed confused result about this issue. Some research supposed the effect of power-of-speech depend on sex of source/Carli, 1990) but the others supposed it is not(Holtgraves & Lasky, 1999; Erickson, Lind, Johnson, & O'Barr, 1978). So present research is designed to show empirical evidence about this issue examining the interaction between effect of power-of-speech and sex of source. Result revealed that power-of-speech affects strongly on persuasion and that effect have no interaction with sex of source. This result means that the effect of power-of-speech is stronger than the effect of sex of source. But in attitude toward product exceptionally, female source using powerful speech induce more favorable attitude toward product than female source using powerless speech whereas male source using powerful and powerless speech induce same attitude toward product.

• Proceedings of the Korea Information Processing Society Conference
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• 2009.11a
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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).

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.5
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• pp.807-812
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• 2020

This paper proposes a blind speech source separation algorithm using a microphone to separate only the target speech source signal in an environment in which various speech source signals are mixed. The proposed algorithm is a model of frequency domain representation based on independent component analysis method. Accordingly, for the purpose of verifying the validity of independent component analysis in the frequency domain for two speech sources, the proposed algorithm is executed by changing the type of speech sources to perform speech sources separation to verify the improvement effect. It was clarified from the experimental results by the waveform of this experiment that the two-channel speech source signals can be clearly separated compared to the original waveform. In addition, in this experiments, the proposed algorithm improves the speech source separation performance compared to the existing algorithms, from the experimental results using the target signal to interference energy ratio.

• MALSORI
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• no.48
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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.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.24 no.2
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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.

• Proceedings of the KSPS conference
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• 2007.05a
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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.

• Speech Sciences
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• v.3
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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.

• Phonetics and Speech Sciences
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• v.3 no.2
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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.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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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.

• Phonetics and Speech Sciences
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• v.4 no.2
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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.