• 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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• 1993.06a
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• pp.249-252
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• 1993

In speech signal processing, The accurate detection of the voiced/unvoiced is important for robust word recognition and analysis. This algorithm is based on the MD in the frame of speech signals that does not require statistical information about either signal or background-noise to decide a voiced/unvoiced. This paper presents a method of estimation the Characteristic of Magnitude Distribution from noisy speech and also of estimation the optimal threshold based on the MD of the voiced/unvoiced decision. The performances of this detectors is evaluated and compared to that obtained from classifying other paper.

• 제어로봇시스템학회:학술대회논문집
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• 1987.10b
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• pp.680-683
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• 1987

The speech signal processing in the auditory system can be analysized based on two representations : Average discharge rate and Temporal discharge pattern. But the average discharge rate representation is restricted by the narrow dynamic range because of the rate saturation and the two tone suppression phenomena, and the temporal discharge pattern representation needs a sophisticate frequency analysis and synchrony measure. In this paper, a simple representation is proposed : using a model considering the interaction of Cochlear fluid-BM movement and a haircell model, the feature of speech signals (formant frequency and pitch of vowels) is easily estimated in the Average Synchronized Rate.

• The Journal of the Acoustical Society of Korea
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• v.26 no.1
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• pp.8-15
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• 2007

Speech intelligibility of the eavesdropping sound is investigated on a acoustic cavity - glass window coupled system. Using MLS (Maximum Length Sequency) signal as a sound source, acceleration and velocity responses of the glass window are measured by accelerometer and laser doppler vibrometer. MTF (Modulation Transfer Function) is used to identify tile speech transmission characteristics of the cavity and window system. STI (Speech Transmission Index) based upon MTF is calculated and speech intelligibility of the vibration sound of the glass window is estimated. Speech intelligibilities by the acceleration signal and the velocity signal are compared. Finally, intelligibility of the conversation sound is confirmed by the subjective test.

• ETRI Journal
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• v.43 no.1
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• pp.82-94
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• 2021

In this work, six voiced/unvoiced speech classifiers based on the autocorrelation function (ACF), average magnitude difference function (AMDF), cepstrum, weighted ACF (WACF), zero crossing rate and energy of the signal (ZCR-E), and neural networks (NNs) have been simulated and implemented in real time using the TMS320C6713 DSP starter kit. These speech classifiers have been integrated into a linear-predictive-coding-based speech analysis-synthesis system and their performance has been compared in terms of the percentage of the voiced/unvoiced classification accuracy, speech quality, and computation time. The results of the percentage of the voiced/unvoiced classification accuracy and speech quality show that the NN-based speech classifier performs better than the ACF-, AMDF-, cepstrum-, WACF- and ZCR-E-based speech classifiers for both clean and noisy environments. The computation time results show that the AMDF-based speech classifier is computationally simple, and thus its computation time is less than that of other speech classifiers, while that of the NN-based speech classifier is greater compared with other classifiers.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.7 no.1
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• pp.11-19
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• 1996

Reinke's edema is used for describing varying degrees of chronic swelling of the vocal folds. The acoustic analysis of Reinke's edema has not been reported so far in this country. The purpose of this study is to clarify acoustic and aerodynamic characteristics of the Reinke's edema. Several acoustic evaluations ＆ aerodynamic studies were done in 20 Reinke's edema patients and the data was compared with those of 20 normal controls. Videolaryngoscopy also was done to classify the severity in grading. We used C-Speech, Doctor speech science, and Phonatory function analyser. In C-Speech, we compared jitter, shimmer, and SNR(signal to noise ratio) of normal and Rrinke's edema patient. In Doctor speech science, we compared NNE(Glottal noise energy), speech fundamental frequency, voice quality between two groups. And in phonatory function analyser for aerodynamic function test, we compared speech intensity, airflow rate, and expiratory pressure between two groups. In conclusion, Reinke's edema patients showed lower voice pitches than normal, additionally jitter, shimmer, SNR(signal to noise ratio), NNE(Glottal noise energy), airflow rate, and expiratory pressure may be meaningful parameters for diagnosis and prognosis for treatment.

• Proceedings of the IEEK Conference
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• 1999.06a
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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.

• Proceedings of the IEEK Conference
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• 2000.07b
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• pp.685-688
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• 2000

In this paper, we used image information for the enhancement of Korean numeral speech recognition. First, a noisy environment was made by Gaussian generator at each 10 dB level and the generated signal was added to original Korean numeral speech. And then, the speech was analyzed to recognize Korean numeral speech. Speech through microphone was pre-emphasized with 0.95, Hamming window, autocorrelation and LPC analysis was used. Second, the image obtained by camera, was converted to gray level, autocorrelated, and analyzed using LPC algorithm, to which was applied in speech analysis, Finally, the Korean numerial speech recognition with image information was more ehnanced than speech-only, especially in ‘3’, ‘5’and ‘9’. As the same LPC algorithm and simple image management was used, additional computation a1gorithm like a filtering was not used, a total speech recognition algorithm was made simple.

• The Journal of the Acoustical Society of Korea
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• v.35 no.2
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• pp.134-142
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• 2016

In this paper, a method synthesizing speech signal using the 40 kHz ultrasonic signals reflected from the articulatory muscles was introduced and performance was evaluated. When the ultrasound signals are radiated to articulating face, the Doppler effects caused by movements of lips, jaw, and chin observed. The signals that have different frequencies from that of the transmitted signals are found in the received signals. These ADS (Acoustic-Doppler Signals) were used for estimating of the speech parameters in this study. Prior to synthesizing speech signal, a quantitative correlation analysis between ADS and speech signals was carried out on each frequency bin. According to the results, the feasibility of the ADS-based speech synthesis was validated. ADS-to-speech transformation was achieved by the joint Gaussian mixture model-based conversion rules. The experimental results from the 5 subjects showed that filter bank energy and LPC (Linear Predictive Coefficient) cepstrum coefficients are the optimal features for ADS, and speech, respectively. In the subjective evaluation where synthesized speech signals were obtained using the excitation sources extracted from original speech signals, it was confirmed that the ADS-to-speech conversion method yielded 72.2 % average recognition rates.

• The KIPS Transactions:PartB
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• v.15B no.1
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• pp.17-24
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• 2008

This paper proposes Sasang constitution classification through speech signals analysis values and comparison. For this, this paper wishes to propose Taeumin classification method of output values signals that comes out speech signal analysis to connect with process classification of Soeumin through skin diagnosis by first step in the whole system configuration to provide for objective index of Sasang constitution. First of all, these characteristic of voices wish to extract phonetic elements that each Sasang constitution groups' clear features. Also, we wish to classify Taeumin through constitution groups' difference and similarity on the basis of results value. Finally, the effectiveness of this method is verified through the experiments.