• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1999년도 하계종합학술대회 논문집
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• pp.1107-1110
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• 1999

In speech signal processing, the accurate decision of the voiced/unvoiced sound is important for robust word recognition and analysis and a high coding efficiency. In this paper, we propose the mehod of the voiced/unvoiced decision using the LSP parameter which represents the spectrum characteristics of the speech signal. The voiced sound has many more LSP parameters in low frequency region. To the contrary, the unvoiced sound has many more LSP parameters in high frequency region. That is, the LSP parameter distribution of the voiced sound is different to that of the unvoiced sound. Also, the voiced sound has the minimun value of sequantial intervals of the LSP parameters in low frequency region. The unvoiced sound has it in high frequency region. we decide the voiced/unvoiced sound by using this charateristics. We used the proposed method to some continuous speech and then achieved good performance.

• 음성과학
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• 제4권2호
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• pp.41-54
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• 1998

Speech signals are, depending on the characteristics of waveform, classified as voiced sound, unvoiced sound, and silence. Voiced sound, produced by an air flow generated by the vibration of the vocal cords, is quasi-periodic, while unvoiced sound, produced by a turbulent air flow passed through some constriction in the vocal tract, is noise-like. Silence represents the ambient noise signal during the absence of speech. The need for deciding whether a given segment of a speech waveform should be classified as voiced, unvoiced, or silence has arisen in many speech analysis systems. In this paper, a voiced/unvoiced/silence classification algorithm using spectral change in the wavelet transformed signal is proposed and then, experimental results are demonstrated with our discussions.

• 음성과학
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• 제10권2호
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• pp.7-25
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• 2003

Predicting the right prosodic elements is a key factor in improving the quality of synthesized speech. Prosodic elements include break, pitch, duration and loudness. Pitch, which is realized by Fundamental Frequency (F0), is the most important element relating to the quality of the synthesized speech. However, the previous method for predicting the F0 appears to reveal some problems. If voiced and unvoiced sounds are not correctly classified, it results in wrong prediction of pitch, wrong unit of triphone in synthesizing the voiced and unvoiced sounds, and the sound of click or vibration. This kind of feature is usual in the case of the transformation from the voiced sound to the unvoiced sound or from the unvoiced sound to the voiced sound. Such problem is not resolved by the method of grammar, and it much influences the synthesized sound. Therefore, to steadily acquire the correct value of pitch, in this paper we propose a new model for predicting and classifying the voiced and unvoiced sounds using the CART tool.

• 인터넷정보학회논문지
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• 제19권6호
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• pp.1-7
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• 2018

본 논문에서는 합성 음성파형의 일그러짐을 제어하기 위하여 V/UV/S(Voiced / Unvoiced / Silence)의 스위칭을 사용하고, 피치구간마다 멀티펄스를 보정하며, 무성자음(Unvoiced)의 근사합성에 특정주파수를 이용하는 ACFBD-MPC(Amplitude Compensation Frequency Band Division - Multi Pulse Coding)와 LMS-MPC(Least Mean Square - Multi Pulse Coding)를 통합한 8kbps ACLMS-MPC(Amplitude Compensation and Least Mean Square - Multi Pulse Coding) 부호화 방식을 제안하고자 한다. 여러 방식을 통합하는데 있어서, 음성파형의 일그러짐을 줄이면서 유성음과 무성음의 비트율을 8kbps로 조정하는 것이 중요하다. 유성음과 무성음의 비트율을 8kbps로 조정함에 있어서, 개별피치를 이용하여 대표구간의 멀티펄스를 피치구간마다 복원함으로서 음성파형을 효율적으로 합성할 수 있다. 8kbps의 부호화 조건에서 ACLMS-MPC 방식을 구현하고 SNR를 평가한 결과, ACLMS-MPC의 SNR는 남자음성에서 15.0dB, 여자음성에서 14.3dB 임을 확인할 수 있었다. 따라서 ACLMS-MPC가 기존의 MPC, ACFBD-MPC, LMS-MPC에 비하여 남자음성에서 0.3dB~1.8dB, 여자음성에서 0.3dB~1.6dB 정도 개선된 것을 알 수 있었다. 이러한 방법들은 셀룰러폰이나 인터넷폰과 같이 낮은 비트율의 음원을 사용하여 음성신호를 부호화하는 방식에 활용할 수 있을 것으로 기대된다. 향후 멀티펄스 음원의 진폭과 위치를 동시에 보정하는 6.9kbps 음성부호화 방식의 음질평가를 수행하고자 한다.

• 대한전자공학회논문지
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• 제26권9호
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• pp.1436-1443
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• 1989

In this paper, we propose a maximum-likelihood estimation(MLE) method to obtain the location and the amplitude of the pulses in MPE( multi-pulse excitation)-LPC speech synthesis using multi-pulses as excitation source. This MLE method computes the value maximizing the likelihood function with respect to unknown parameters(amplitude and position of the pulses) for the observed data sequence. Thus in the case of overlapped pulses, the method is equivalent to Ozawa's crosscorrelation method, resulting in equal amount of computation and sound quality with the cross-correlation method. We show by computer simulation: the multi-pulses obtained by MLE method are(1) pseudo-periodic in pitch in the case of voicde sound, (2) the pulses are random for unvoiced sound, (3) the pulses change from random to periodic in the interval where the original speech signal changes from unvoiced to voiced. Short time power specta of original speech and syunthesized speech obtained by using multi-pulses as excitation source are quite similar to each other at the formants.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1998년도 하계종합학술대회논문집
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• pp.599-602
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• 1998

In this paper, a speech enhancement method based on phonemic properties and masking effect is propsoed. It is a modified type of spectral subtraction wherein the spectral sharpening process is exploited in unvoiced state considering the phonemic properties. The masking threshold is used to remove the residual noise. The proposed spectral subtraction shows similar performance as that of the classical spectral subtraction method in view of the SNR. But by the prposed scheme, the unvoiced sound region is shown to exhibit relatively less signal distortion in the enhanced speech.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 제11회 음성통신 및 신호처리 워크샵 논문집 (SCAS 11권 1호)
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• pp.91-94
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• 1994

This paper is a study on the design and implementation of the Korean Tet-to-Speech system. In this paper, parameter symthesis method is chosen for speech symthesis method and PARCOR coeffient, one of the LPC analysis, is used as acoustic parameter, We use a diphone as synthesis unit, it include a basic naturalness of human speech. Diphone DB is consisted of 1228 PCM files. LPC synthesis method has defect that decline clearness of synthesis speech, during synthesizing unvoiced sound In this paper, we improve clearness of synthesized speech, using residual signal as ecitation signal of unvoiced sound. Besides, to improve a naturalness, we control the prosody of synthesized speech through controlling the energy and pitch pattern. Synthesis system is implemented at PC/486 and use a 70Hz-4.5KHz band pass filter for speech imput/output, amplifier and TMS320c30 DSP board.

• 한국통신학회논문지
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• 제33권3C호
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• pp.234-239
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• 2008

For the efficient coding and transmission, the speech signal can be classified into three distinctive classes: voiced, unvoiced, and transition classes. At low bit rate coding below 4 kbit/s, conventional sinusoidal transform coders synthesize speech of high quality for the purely voiced and unvoiced classes, whereas not for the transition class. The transition class including plosive sound and abrupt voiced-onset has the lack of periodicity, thus it is often classified and synthesized as the unvoiced class. In this paper, the efficient algorithm for the transition class detection is proposed, which demonstrates superior detection performance not only for clean speech but for noisy speech. For the detected transition frame, phase information is transmitted instead of magnitude information for speech synthesis. From the listening test, it was shown that the proposed algorithm produces better speech quality than the conventional one.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1999년도 하계학술대회 논문집 G
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• pp.2905-2907
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• 1999

The meaningful speech sound block classification provides very important information in the speech recognition. The following technique of the classification is based on the DWT (discrete wavelet transform), which will provide a more fast algorithm and a useful, compact solution for the pre-processing of speech recognition. The algorithm is implemented to the unvoiced/voiced classification and the denoising.

• 인터넷정보학회논문지
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• 제18권5호
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• pp.17-22
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• 2017

유성음원과 무성음원을 사용하는 멀티펄스 음성부호화 방식에 있어서, 대표구간의 멀티펄스 음원을 사용하는 경우에 유성음의 합성음성파형에서 일그러짐이 나타난다. 이러한 원인은 대표구간의 멀티펄스를 피치구간마다 복원하는 과정에서 재생 음성파형이 정규화 되는 것이 원인으로 작용한다. 본 논문에서는 합성 음성파형의 일그러짐을 제어하기 위하여 V/UV/S(Voiced / Unvoiced / Silence)의 스위칭을 사용하고, 피치구간 마다 멀티펄스의 위치를 보정하며, 무성자음(Unvoiced)의 근사합성에 특정주파수를 이용하는 PCFBD-MPC(Position Compensation Frequency Band Division-Multi Pulse Coding)를 제안하였다. 또한 8kbps의 부호화 조건에서 PCFBD-MPC 시스템을 구현하고, PCFBD-MPC의 SNRseg를 평가하였다. 그 결과 PCFBD-MPC의 남자음성에서 13.8dB, 여자음성에서 13.4dB 임을 확인할 수 있었다. 향후 멀티펄스 음원의 진폭과 위치를 동시에 보정하는 8kbps 음성부호화 방식의 음질을 평가하는 연구를 수행하고자 한다. 향후, 멀티펄스 음원의 진폭과 위치를 동시에 보정하는 8kbps 음성부호화 방식의 음질을 평가하는 연구를 하고자 한다. 이러한 방법들은 셀룰러폰이나 스마트폰과 같이 낮은 비트율의 음원을 사용하여 음성신호를 부호화하는 방식에 활용할 수 있을 것으로 기대된다.