• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.103-106
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• 1994

Generally we are used to apply waveform coding method obtaining the high quality synthesized speech. But we have to solve the problems, memory capacity and pitch alteration, for applying the waveform coding method to speech synthesis by rule. The former problem is conquered by improving the integrated semiconductor technology, but the latter problem remains. In this paper, we compare the methods that have proposed for pitch alteration in our laboratory until now. These methods are not change properties of vocal tract formants and only altered the pitch halving method, 1.14% for cepstrum analysis method, and 2.36% for hamonics compensated with the phase method.

• MALSORI
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• no.65
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• pp.81-91
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• 2008

In this paper, we propose a new variable bit-rate speech coder based on the waveform interpolation concept. After the coder extracted all parameters, the amounts of the distortions between the current and the predicted parameters which are estimated by extrapolation using past two parameters are measured for all parameters. A parameter would not be transmitted unless the distortion exceeds the preset threshold. At the decoder side, the non-transmitted parameter is reconstructed by extrapolation with past two parameters used to synthesize signals. In this way, we can reduce 26% of the total bit rate while retaining the speech quality degradation below 0.1 PESQ score.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.89-95
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• 1995

The waveform coding technique has concerned with simply preserving the waveform shape of speech signal through a redundancy reduction process. In the case of speech synthesis, the waveform coding with high sound quality is mainly used to the synthesis by analysis. However, since the parameters of this coding are not classified into either excitation or vocal tract parameters, it is difficult to applying the waveform coding to the synthesis by rule. In order to apply the waveform coding to the synthesis by rule, the pitch alteration technique is required in prosody control. In this paper, we propose a new pitch alteration method that can change the pitch period in waveform coding by scaling the time-axis and compensating the spectrum. This is relevant to the time-frequency domain method were the phase components of the waveform is preserved with a little spectrum distortion of 2.5 % and less for 50% pitch change.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.7-10
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• 1991

In a text-to-speech system, sound units (phonemes, words, or phrases, etc.) can be concatenated together to produce required utterance. The quality of the resulting speech is dependent on factors including the phonological/prosodic contour, the quality of basic concatenation units, and how well the units join together. Thus although the quality of each basic sound unit is high, if occur the discontinuity in the concatenation part then the quality of synthesis speech is decrease. To solve this problem, a smoothing operation should be carried out in concatenation parts. But a major problem is that, as yet, no method of parameter smoothing is available for joining the segment together. Thus in this paper, we proposed a new aigorithm that smoothing the unnatural discountinuous parts which can be occured in speech waveform editing. This algorithm used the exponential smoothing method.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.9 no.2
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• pp.142-146
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• 1998

Background : It has been established that the fundamental frequency(Fo) of the vowels varies systemically as a function of vowel height. Specifically, high vowels have a higher Fo than low vowels. Two major explanations or hypotheses dominate contemporary accounts of fired to explain the mechanisms underlying intrinsic variation in vowel Fo, source-tract coupling hypothesis and tongue-pull hypothesis. Objectives : Total laryngectomy surgery necessiates removal of all structures between the hyoid bone and the tracheal rings. Therefore, the assumption that no direct interconnection exists between the tongue and pharyngoesophageal segment that would mediate systematic variation in vowel Fo appears quite reasonable. If tongue-pull hypothesis is correct, systemic differences in Fo between high versus low vowels produced by esophageal speakers would not Or expected. We analyzed the Fo in the vowels of esophageal voice. Materials and method : The subjects were 11 cases of laryngectomee patients with fluent esophageal voice. The five essential vowels were recorded and analyzed with computer speech analysis system(Computerized Speech Lab). The Fo was measured using acoustic waveform, automatically and manually, and narrow band spectral analysis. Results : The results of this study reveal that intrinsic variation in vowel Fo is clearly evident in esophageal speech. By analysis using acoustic waveform automatically, the signals were too irregular to measure the Fo precisely. So the data from automatic analysis of acoustic waveform is not logical. But the Fo by measuring with manually calculated acoustic waveform or narrowband spectral analysis resulted in acceptable results. These results were interpreted to support neither the source-tract coupling nor the tongue-pull hypotheses and led us to offer an alternative explanation to account for intrinsic variation of Fo.

• The Journal of the Acoustical Society of Korea
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• v.15 no.6
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• pp.99-103
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• 1996

Most waveform coding techniques attempt to reduce redundancy of speech signal while preserving the shape of the waveform. In speech synthesis, wavefrom coding methods are used to the synthesis by rule for high quality speech. However, it is difficult to apply the waveform coding to the synthesis by rule because the parameters of the wavefrom coding cannot be classified as either the excitation or the vocal tract parameters. The proposed method shows little spectrum distortion of 2.7% or less for 50% pitch changes. It also achieves smooth connection of wavefrom magnitudes among the frames by compensating the phase in time domain.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.3
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• pp.465-470
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• 2021

A robust speech recognition technology is required that does not degrade the performance of speech recognition and the quality of the speech when speech recognition is performed in an actual environment of the speech mixed with noise. With the development of such speech recognition technology, it is necessary to develop an application that achieves stable and high speech recognition rate even in a noisy environment similar to the human speech spectrum. Therefore, this paper proposes a speech enhancement algorithm that processes a noise suppression based on the MMSA-STSA estimation algorithm, which is a short-time spectral amplitude method based on the error of the least mean square. This algorithm is an effective nonlinear speech enhancement algorithm based on a single channel input and has high noise suppression performance. Moreover this algorithm is a technique that reduces the amount of distortion of the speech based on the statistical model of the speech. In this experiment, in order to verify the effectiveness of the MMSA-STSA estimation algorithm, the effectiveness of the proposed algorithm is verified by comparing the input speech waveform and the output speech waveform.

• Phonetics and Speech Sciences
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• v.7 no.4
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• pp.35-40
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• 2015

We propose a speech waveform forgery detection algorithm based on the flatness of frequency distribution. We devise a new measure of flatness which emphasizes the local change of the frequency distribution. Our measure calculates the sum of the differences between the energies of neighboring frequency bands. We compare the proposed measure with conventional flatness measures using a set of a large amount of test sounds. We also compare- the proposed method with conventional detection algorithms based on spectral distances. The results show that the proposed method gives lower equal error rate for the test set compared to the conventional methods.

• The Journal of the Acoustical Society of Korea
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• v.18 no.1E
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• pp.32-36
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• 1999

In the area of the speech synthesis techniques, the waveform coding methods maintain the intelligibility and naturalness of synthetic speech. In order to apply the waveform coding techniques to synthesis by rule, we must be able to alter the pitches of synthetic speech. In this paper, we propose a new pitch altering method that compensates phase distortion of the cepstral pitch alteration method with time scaling method in the time domain. This method can remove some spectrum distortion which is occurred in conjunction point between the waveforms. For performance test the spectrum distortion rate was used as objective criterion and the MOS(Mean Opinion Score) was used as subjective criterion. As a result, the spectrum distortion and MOS are obtained by 0.66% and 3.9, respectively.

• The Journal of the Acoustical Society of Korea
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• v.20 no.2
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• pp.3-9
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• 2001

In this paper, a random number generator for simulation of speech-waveform coders was designed. A random number generator having a desired probability density function and a desired power spectral density is discussed and experimental results are presented. The technique is based on Sondhi algorithm which consists of a linear filter and a memoryless nonlinearity. Several methods of obtaining memoryless nonlinearities for some typical continuous distributions are discussed. Sondhi algorithm is analyzed in the time domain using the diagonal expansion of the bivariate Gaussian probability density function. It is shown that the Sondhi algorithm gives satisfactory results when the memoryless nonlinearity is given in an antisymmetric form as in uniform, Cauchy, binary and gamma distribution. It is shown that the Sondhi algorithm does not perform well when the corresponding memoryless nonlinearity cannot be obtained analytically as in Student-t and F distributions, and when the memoryless nonlinearity can not be expressed in an antisymmetric form as in chi-squared and lognormal distributions.