• Phonetics and Speech Sciences
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• v.10 no.4
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• pp.121-125
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• 2018

This paper describes procedures of changing and measuring voice quality in terms of jitter. Jitter synthesis method was applied to the TD-PSOLA analysis system of the Praat software. The jitter component is synthesized based on a Gaussian random noise model. The TD-PSOLA re-synthesize process is used to synthesize the modified voice with artificial jitter. Various vocal jitter parameters are used to measure the change in quality caused by artificial systematic jitter change. Synthetic vowels, natural vowels and short sentences are used to check the change in voice quality through the synthesizer model. The results shows that the suggested method is useful for voice quality control in a limited way and can be used to alter the jitter component of voice.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.1
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• pp.122-131
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• 2016

In the smart societies, such as the current unlike in the past, the voice that listeners will feel favor is changing through the development of ICT technologies and infrastructure. In other words, in the past, loud, intensive and fast voice is a favorite but now a new social and cultural situation that is changing them with ICT technologies. Now, this becomes one of the important things that we clarify 'Is it a voice that feels a favor?'. For this, in this paper, we identified what voice that listeners feel favor by applying ICT technologies. Studies were carried out to proceed largely divided into two categories. Firstly, as the quantified data, we extracted the impact on favorable feeling of listeners which related with emotional speech by empirical analysis work. To do this, we performed the experiment for the public. Secondly, we identified what kind of voice which listeners feel a good impression. For this, we identified voice characteristics that there are people who are influential in the real society. Also, we extracted both the voice characteristics of each influential people and common voice characteristics. In addition, we want to overcome the problems of qualitative methods that have originally limitations in objective respects which is significant to the voice analysis. For this, we performed the experiments of the voice analysis by numerical and visual approaches.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.201-203
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• 2005

The best method to study language is to talking with a native speaker. A voice recognition technology can be used to develope a language study machine. SD(Speaker dependant) and SI(speaker independant) voice recognition method is used for the language study machine. MP3 Player. FM Radio. Alarm clock functions are added to enhance the value of the product. The machine is designed with a DSP(Digital Signal Processing) chip for voice recognition. MP3 encoder/decoder chip. FM tumer and SD flash memory card. This paper deals with the application of SD ad SD voice recognition. flash memory file system. PC download function using USB ports, English conversation text function by the use of SD flash memory. LCD display control. MP3 encoding and decoding, etc. The study contents are saved in SD flash memory. This machine can be helpful from child to adult by changing the SD flash memory.

• Phonetics and Speech Sciences
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• v.9 no.1
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• pp.41-47
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• 2017

The main advantage of the statistical parametric speech synthesis is its flexibility in changing voice characteristics. A personalized text-to-speech(TTS) system can be implemented by combining a speech synthesis system and a voice transformation system, and it is widely used in many application areas. It is known that the fundamental frequency and the spectral envelope of speech signal can be independently modified to convert the voice characteristics. Also it is important to maintain naturalness of the transformed speech. In this paper, a speech synthesis system based on Hidden Markov Model(HMM-based speech synthesis, HTS) using the STRAIGHT vocoder is constructed and voice transformation is conducted by modifying the fundamental frequency and spectral envelope. The fundamental frequency is transformed in a scaling method, and the spectral envelope is transformed through frequency warping method to control the speaker's vocal tract length. In particular, this study proposes a voice transformation method using the correlation between fundamental frequency and vocal tract length. Subjective evaluations were conducted to assess preference and mean opinion scores(MOS) for naturalness of synthetic speech. Experimental results showed that the proposed voice transformation method achieved higher preference than baseline systems while maintaining the naturalness of the speech quality.

• Phonetics and Speech Sciences
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• v.8 no.4
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• pp.131-138
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• 2016

This study analyzes the characteristics of good impression using speech manipulation scripts and investigates the characteristics of preferred speech voice. Fourty male and female college students participated in this study. They have been exposed to the Gyeongsang dialect spoken by their friends and family for more than 15 years. Two sample voices(1 male and 1 female), considered as giving good impression, were subject to voice analysis. Two students were asked to read the sample paragraph of 'Walking' and their voice samples were analyzed through Praat. The collected speech data were manipulated into 4 different sets by changing pitch level, degree of loudness and speech rate. First, both men and women received good impression more from pitch-lowered sound than from the original one. Second, men tended to receive good impression more from slightly louder voice than from the natural-pitched one. Third, it was shown that men often felt more drowned to a voice at slightly faster speech rate than at the original speech rate. Overall, both male and female listeners favored lower pitch over the original pitch. Men tended to prefer louder voice sound while women preferred less loud one. Men received better impression at a lower speech rate but women at a faster speech rate.

• Journal of Haehwa Medicine
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• v.14 no.1
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• pp.113-128
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• 2005

1. Acute laryngitis is a hoarse voice or the complete loss of the voice because of irritation to the vocal folds. 2. Acute laryngitis belongs with the GeupHuEum, HuBi, HuPung in oriental medicine. 3. GeupHuEum is caused by wind and cold, weak of lung and kidney, evil energy of liver, sore throat, etc. It is treated with the methods of cooling lung and wetting, removing heat and changing phlegm, etc. 4. Hubi is caused by fire and wind, dampness, large lung. It is treated with the methods of removing heat and antidote, reinforcing and descending fire, bleeding by acupuncture, vomiting. 5. Hupung is caused by phlegm and heat of lung and stomach, wind and heat. It is treated with the methods of dispersing wind and removing heat and changing phlegm by medicine, acupuncture, moxibustion, vomiting, fumigation.

• The KIPS Transactions:PartB
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• v.18B no.6
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• pp.397-404
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• 2011

In this paper, we were carried out experiments to apply parameter of voice analysis to measure changing characteristic articulator according to inhale the helium gas. The helium gas was used to overcome air embolism nitrogen gas to deal a fatal blow in body nitrogen gas by diver. However, the helium gas has been much trouble interpretation about abnormal voice of diver to cause squeaky voice of low articulation. Therefor, we was carried out experiments about pitch and spectrogram measurement, analysis based on to influence in acoustic organs before and after of inhaled helium gas.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.2362-2367
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• 2005

This paper reports a technique for discriminating double talk and echo path change using the stochastic characteristics of power change for an adaptive noise canceller. The causes of rapid error increasing are double talk and echo path change. When the echo path is changed, the system corrects the impulse response in order to reduce the error. However, in the case of double talk, the system has to suspend the updating impulse response in order to maintain the quality of the voice signal. In the conventional system, it was difficult to discriminate between the two situations. In this research, the stochastic characteristics of the voice power change in the double talk period were experimentally verified to be different from the power change during echo path changing. Based on the results, a new double talk detection method is proposed.

• Journal of Digital Contents Society
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• v.11 no.2
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• pp.235-242
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• 2010

Differently from instruments which synthesized sounds and tones in the past, voice synthesis engine for music production has reached to the level of creating music as if actual artists were singing. It uses the samples of human voices naturally connected to the different levels of phoneme within the frequency range. Voice synthesis engine is not simply limited to the music production but it is changing cultural paradigm through the second creations of new music type including character music concerts, media productions, albums, and mobile services. Currently, voice synthesis engine technology makes it possible that users input pitch, lyrics, and musical expression parameters through the score editor and they mix and connect voice samples brought from the database to sing. New music types derived from such a development of computer music has sparked a big impact culturally. Accordingly, this paper attempts to examine the specific case studies and the synthesis technologies for users to understand the voice synthesis engine more easily, and it will contribute to their variety of music production.

• Journal of Internet Computing and Services
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• v.6 no.1
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• pp.39-50
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• 2005

Speech DB is basically required factor when we are study for phonetics, speech recognition and speech synthesis and so on. The quantity and quality of speech DB decide the efficiency of system that we develop. therefore. speech DB has an extremely important factor, Recently, development of the various telephone service technique such as voice portal. it is actual condition where the necessity of collection of telephone speech DB. The existing IVR application telephone speech DB collection system used C/C++ language or the exclusive development tool. Thus it is the actual condition where the recycle of each application service for resources is difficult and have a problem of many labors and time necessity. But. VoiceXML is a language having tag form ipredicated in XML. which has easy and simple grammar system. Therefore, if we make a few efforts we could draw up easily. it has a merit reducing labors and time, Also, VoiceXML has many advantages of various telephone speech DB gathering because of changing contents of DB. In this paper, we introduce telephone speech DB gathering system which is the mast important factor for development of speech information processing technique.