• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2415-2418
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• 2003

In this paper, we present a new filter to adjust formant information. Spectral envelope in speech analysis shows information about characteristics of speech and formant information determines speech timbre. So, if formant position is adjusted, we can verify adjusted speech timbre. A presented filter is to adjust this formant. This filter is composed of triangular filters. Using this filter we could locate the formant frequency at target position.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.25 no.8
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• pp.525-531
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• 2015

Driving sound is an effective element to promote the product identity of a vehicle by providing customers with attractive sound which reflects the concept of a vehicle. Recently, major automakers are focusing on the target sound setting so that the sound can represent the brand image as well as the unique concept of a vehicle. In this study, a new method of target setting for the driving sound will be introduced based on using formant filter and musical harmonics characteristics. In addition, a target sound suggested from this method will be realized and verified by using active noise control in vehicle.

• Journal of Biomedical Engineering Research
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• v.28 no.2
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• pp.271-279
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• 2007

We suggest a novel method for the evaluation of cochlear implant (CI) speech processing strategy based on neural spike train decoding. From formant trajectories of input speech and auditory nerve responses responding to the electrical pulse trains generated from a specific CI speech processing strategy, optimal linear decoding filter was obtained, and used to estimate formant trajectory of incoming speech. Performance of a specific strategy is evaluated by comparing true and estimated formant trajectories. We compared a newly-developed strategy rooted from a closer mimicking of auditory periphery using nonlinear time-varying filter, with a conventional linear-filter-based strategy. It was shown that the formant trajectories could be estimated more exactly in the case of the nonlinear time-varying strategy. The superiority was more prominent when background noise level is high, and the spectral characteristic of the background noise was close to that of speech signals. This confirms the superiority observed from other evaluation methods, such as acoustic simulation and spectral analysis.

• Speech Sciences
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• v.14 no.2
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• pp.65-71
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• 2007

The severe-profound hearing impaired children have various disorders in everday communication due to the lack of hearing feedback. Especially, their speech produced unstable voice, omission and distortion of articulation, pitch break, cul-de-sac voice, and so on so that they were difficult to accurately deliver an intended message. This study attempts to analyze the acoustic characteristics of 4 vowel sounds produced by 35 severe-profound hearing impaired children using CSL(Computerized Speech Lab, Model 4300b). The formant data were obtained from the spectrogram and analyzed data by 12 formant filter and auto-correlation among the formants. Results showed that the hearing impaired children's formant values came out very high. They produced the vowels at the mode of hypertension with unstable voice. In order to improve their speech, they would need some adequate auditory feedback.

• Journal of Korea Multimedia Society
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• v.19 no.12
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• pp.1919-1926
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• 2016

The people who complain of pain due to voice causes such as vocal cord nodules is increasing year by year. If the voice is changed, it is possible to give to colleagues discomfort or inconvenience during conversation. In this paper, we propose a way to reduce discomfort by improving the husky voice during the conversation. A VBF (voice boosting filter) is firstly designed to improve the husky voices. This filter may further emphasize the formant frequency components than the frequency components around the formant frequency, because the value is relatively greater than the other frequency. And a fixed-point type DSP chipset, TMS320F2812 is applied to the system, the operating frequency is 150MHz. The system was implemented as a compact for use as a portable, its size is $2.5cm{\times}10cm$. Through the test using three husky voices with some type of statement, it was satisfactory in processing speed and sound quality improvement.

• Proceedings of the KOSOMBE Conference
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• v.1992 no.05
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• pp.163-166
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• 1992

In this study, the extractions of formant and articulately motion trajectorles from Korean diphthongs are performed by using the RISL adaptive linear prediction filter. This enables us to extract spectrum transition of speech signal accurately. This study showes that the RISL algorithm is superior to the Levinson algorithm, specially in transition part of speech.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.221-224
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• 1995

In speech signal processing, it is necessary to detect exactly the pitch. The algorithms of pitch extraction with have been proposed until now are difficult to detect pitches over wide range speech signals. In this paper, thus, we proposed a new pitch detection algorithm that used a low pass filter with variable bandwidth. It is the method that preprosses to find the first formant of speech signals by the FFT at each frame and detects the pitches for signals LPFed with the cut off frequency according to the first formant. Applying the method, we obtained the pitch contours, improving the accuracy of pitch detection in some noise environments.

• Speech Sciences
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• v.12 no.4
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• pp.43-52
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• 2005

The common form of pre-emphasis filter is $H(z)\;=\;1\;- az^{-1}$, where a typically lies between 0.9 and 1.0 in voiced signal. Also, this value reflects the degree of filter and equals R(1)/R(0) in Auto-correlation method. This paper proposes a new flattening algorithm to compensate the weaked high frequency components that occur by vocal cord characteristic. We used interval information of LSP to estimate formant frequency. After obtaining the value of slope and inverse slope using linear interpolation among formant frequency, flattening process is followed. Experimental results show that the proposed algorithm flattened the weaked high frequency components effectively. That is, we could improve the flattened characteristics by using interval information of LSP as flattening factor at the process that compensates weaked high frequency components.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.5
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• pp.458-467
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• 1991

In this paper, the cause of linear prediction error is analysed and the theoretical basis for nomalizing the format bandwidth to 0is given and its validity is verified. The formant and bandwidth in relation to the position of the poles of AR filter are measured for an alaysis of the relation between the pole position and the formant bandwidth. By changing the glottis reflection coefficient to 1. the pole position and the formant bandwidth. By changing the glottis reflection coefficient to 1. the effect of the glottis is eliminated and as the result a new linear preiction coefficients are obtained by normalizing the formant bandwidth of the signal to 0. since these coefficients are symmetrical, the standard deviation is larger than the coefficients with fixed glottis reflection coefficient. The bit rate for speech coding can be reduced by a factor of 2 without any loss of information. Through computer simulation, recognition rate of 96.7% is botained by using the proposed algorithm in recognizing 5 Korean vowels in noisy environment.

• Agricultural and Biosystems Engineering
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• v.2 no.1
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• pp.15-23
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• 2001

On the assumption that the voices of the cows are produced by the linear prediction filter, we characterized the cows’voices. The order of this filter was determined by examining the voice characteristics both in time and frequency domains. The proposed order of the linear prediction filter is 15 for modeling voice production of the cow. The characteristics of the amplitude envelope of the voice signal was investigated by analyzing the sequence of the short time variance both in time and frequency domains, and the new parameters were defined. One of the coefficients o the linear prediction filter generating the voice signal, the fundamental frequency, the slope of the straight line regressed from the log-log spectra of the short time variance and the coefficients of the linear prediction filter generating the sequence of the short time variance of the voice signal can differentiate the two cows.