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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
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The Journal of the Acoustical Society of Korea
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Journal DOI :
The Acoustical Society of Korea
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Volume & Issues
Volume 15, Issue 6 - Dec 1996
Volume 15, Issue 5 - Oct 1996
Volume 15, Issue 4 - Aug 1996
Volume 15, Issue 2 - Apr 1996
Volume 15, Issue 1 - Feb 1996
Volume 15, Issue 4E - 00 1996
Volume 15, Issue 3E - 00 1996
Volume 15, Issue 2E - 00 1996
Volume 15, Issue 1E - 00 1996
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Speech Recognition Based on VQ/NN using Fuzzy
Ann, Tae-Ock ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 5~11
This paper is the study for recognizing single vowels of speaker-independent, and we suppose a method of speech recognition using VQ(Vector Quantization)/NN(Neural Network). This method makes a VQ codebook, which is used for obtaining the observation sequence, and then claculates the probability value by comparing each codeword with the data, finally uses these probability values for the input value of the neural network. Korean signle vowels are selected for our recognition experiment, and ten male speakers pronounced eight single vowels ten times. We compare the performance of our method with those of fuzzy VQ/HMM and conventional VQ/NN According to the experiment result, the recognition rate by VQ/NN is 92.3%, by VQ/HMM using fuzzy is 93.8% and by VQ/NN using fuzzy is 95.7%. Therefore, it is shown that recognition rate of speech recognition by fuzzy VQ/NN is better than those of fuzzy VQ/HMM and conventional VQ/HMM because of its excellent learning ability.
A Beamformer for Antenna Arrays with Faulty Elements
Kim, Gi-Man ; Cha, Il-Hwan ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 12~15
An array often has faulty elements in real operation. The faulty elements, producing no output or highly reduced gain than other normal elements, cause an elevated sidelobe level and fail to reject the interference signals in an adaptive beamformer. In this paper we have presented the beamforming algorithm for arrays with faulty elements. In the ideal case, an autocorrelation matrix computed from array output data is the toeplitz. However, the inverse of the autocorrelation matrix computed from array with faulty elements can not be obtained due to deficient values of matrix. To overcome this problem, an adaptive beamforming algorithm using the average values of the diagonal terms of matrix is proposed. The computer simulations have been performed to study the performance of the presented method. We have been able to solve the degrees-of-freedom problem that is the drawback of the previous subaperture processing technique.
A study on Chip Design for Hageul Type Classification using Content Addressable Memory
Park, Noh-Kyung ; Koo, Chang-Mo ; Jeong, Chang-Won ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 16~25
In this paper, we designed the chip which can classify the Korean characters using CAM(Content Addressable Memory). A high-speed OCR has been implemented by software to recognize the characters. However, it is difficult to process in real-time. The pipelined hardware implementation is one of the solution to recognize the characters in real-time by using the parallel processing techniques. We used the CAM which has the function of high-speed parallel-match to implement easily and twenty reference patterns are used for comparison. The chip has been evaluated result using DLAB of DAZIX. The simulation results have shown that the process speed was
per character. Also, we programed using C-language and compared the results.
The Rule of Korean Pitch Variation for a Natural Synthetic Female Voice
Kim, Chung-Won ; Park, Dae-Duck ; Kim, Boh-Hyun ; Kwon, Cheol-Hong ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 26~32
In this paper we make a rule of pitch variation for a natural synthetic female voice. Intonation phrase, which is the basic unit the rule is applied to, mostly consists of a syllable or syllables. The pitch values of the first, second, and final syllables make up the pitch contour of the intonation phrase. Those of the first and second syllable are determined by the initial consonants of the respective syllables, and that of the final syllable by the type of the function word. There are two kinds of boundaries between intonation phrases. One is a boundary with pause, and the other is a boundary without pause. The pitch contour of the intonation phrase with the boundary phenomena determines the pitch pattern of a sentence.
A Study on the Performance Enhancement of Blind Equalizer for CATV Receiver Using the Variable Step Size Algorithm
Lee, Hyeon-Cheol ; Jo, Il-Jun ; Jin, Hyeon-Su ; Kim, Seong-Hwan ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 33~40
In this paper, we resolved a trade-off problem of the blind equalizer based on the stop-and-go algorithm that is commonly used for QAM demodulation in CATV receiver. The stop-and-go algorithm has used the LMS(least mean square) algorithm in the updating operation of tap weights so that the structure of equalizer is simple, but there is a trade-off between convergence speed and steady state error as in the typical LMS algorithm. We used the variable step size algrithm to improve the convergence speed with the steady state error in the constant level. With respect to the same level of the steady state error, the variable step size stop-and-go algortihm improved convergence speed by about
as compared with that of the constant step size algortihm
Scrambler Design and Real Time Implementation for Secure Communication
Seok, Gwang-Won ; Yeo, Song-Pil ; Park, Jung-Ho ; Jeong, Jeong-Gyun ; Du, Hyeon-Ung ; Kim, Seong-Hwan ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 41~46
Conversations over telephone line are easily monitored and therefore acceptable level of security is not provided. In this paper, we propose the design of scrambler for secure communication and real time implementation of it. Especially, we con reduce the complexity of hardware and implement it easily by designing the scrambling system which doesn't require synchronization signal, and provide the scrambling filter with a sufficient level of security of adding new gain into the scrambling system.
A Study of Comparison with Free Wave Number Between a New Cylinderical Wave Equation and the Wave Equation by Junger and Feit
Jo, Heung-Kuk ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 47~51
The Cylindrical Shell Equation is one of the fundamental tools in the study of the noise analysis in the cylindrical shell. Therefore, lot of the acousticians induced many cylindrical shell motion equations. In the Reference, we introduced the newly induced cylindrical Shell Equation and Junger and Feit's shell equation, and computed the free wave number with the linear Equation with the supposed solution, in the case of the free motion of the shell. In this paper, we compared above cylindrical shell equations by using dispersion curve of free wave number and we describe the physical mean for the dispersion curve with ring-frequency and ring-extention-frequency. With this result, we proves the useful of a newly induced cylindrical shell equation and we can analyse the Structure-Borne Sound of the shell with this equation in the application.
Development and Performance Tests of the Waste Water Diffusers using Acoustic Resonance and Oscillatory Pulsation
Hong, Suk-Yoon ; Moon, Jong-Duck ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 52~58
Using the acoustic resonances and oscillatory pulsations considered as the branch of wave technologies, the concept of the acoustic resonance diffusers for waste water treatment which maximize the oxygen transfer efficiency in gas-liquid two phase medium have been proposed, and studies for the principles and performance tests were accomplished. Besides, the design concepts for the low pressure Helmholtz resonator, cylinder and annular type reflection resonator and combined type resonance system have been implemented. The acoustic resonance energy which can speed up the mass transfer process increase the oxygen transfer efficiency, and periodic pulsations generated from the instability of air jet from nozzle make very small air bubbles. Then, the annular type jet resonator(AJR) applying these two principles successfully was evalulated as the most promising device and also the efficiency showing
better than conventional diffusers has been verified experimentally.
A 4 kbps PSI-VSELP Speech Coding Algorithm
Choi, Yong-Soo ; Kang, Hong-Goo ; Park, Sang-Wook ; Youn, Dae-Hee ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 59~65
This paper proposes a 4 kbps PSI-VSELP(Pitch Synchronous Innovation-Vector Sum Excited Linear Prediction) speech coder which produces speech equivalent to that of the conventional 4.8 kbps VSELP. Since the 'half-rate' is differently defined from country to country, there may be a need to reduce the bit rate of conventional half-rate coder. To minimize the degradation of speech quality caused by bit-rate reduction, it is desirable to perform bit-allocation based on the carefull consideration of the effect of various transmission parameters. This paper adopts this analytical approach for bit-allocation at 4 kbps. To improve the quality of the VSELP coder at 4 kbps, basis vectors which play the most important role in the performance, are optimized by an iterative closed-loop training process and the PSI technique is employed in the VSELP performance, are optimized by an iterative closed-loop training process and the PSI technique is employed in the VSELP coder. To demonstrate the performance of the proposed speech coder, we peformed experiments under the noiseless and error free conditions. From experimental results, even though the proposed 4 kbps PSI-VSELP coder showed lower scores in the objective measure, higher scores in subjective measure was obtained compared with those of the conventional 4.8 kbps VSELp.
A Stduy on Acoustics Estimation of PANSORI hall by Scale Model
Shin, Young-Moo ; Chung, Sa-Hee ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 66~72
In order to the sound effects and acoustics estimation of PANSORI hall, we are researched into the impulse response measuring and convolution integral of dry music(PANSORI) by using 1/10 scale model. Results are as follwo. First, impulse responses are measured by spark sound of electrodes and it is absolutely necessary many times of synchronous calculating for the obtain to enough S/N ratio. Second, a simulation technique of scale model is confirmed one of an effectual method of indoor acoustics estimation. Further, using the these new techniques and hearing test, its are recognized that reverberation time of PANSORI hall is about
A Human Face Recognition System : Incorporation of Complementary Utilization of Front and Profile Human Images
Choi, Dong-Sun ; Lee, Ju-Shin ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 73~80
Success of a face recognition system depends on which parameters are used. Generally the parameters are affected by environment of facial images such as illumination. To reduce the influence of the evcironment, since side images are insensitive to variance of brightness, it might be an appropriate approach to make the defect of front face images complete with the features extracted from side images. This paper proposes a method which collects and completes the information of front and side images. It is intended to prove the usefulness of the method that it is compared with other methods.
A Study on Suppression of UT Grain Noise Using SSP MPO Algorithms
Koo, Kil-Mo ; Jun, Kye-Suk ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 81~89
It is very important for ultrasonic test method to evaluate the integrity of the class I components in nuclear power plants. However, as the rltrasonic test is affected by internal structures and configurations of test materials, backscattering, that is, time invariant noise is generated in large grain size materials. Due to the above reason, the received signal results in low signal to noise(S/N) ratio. Split spectrum processing(SSP) technique is effective to suppress the grain noise. The conventional SSP technique. however, has been applied to unique algorithm. This paper shows that MPO(minimization and polarity threshold) algorithm which two algorithms are applied simulatancously, was utilized, the signal processing time was shorten by using the new constant-Q SSP with the FIR filter which frequency to bandwidth ratio is constant and the optimum parameters were analysed for the signal processing to longitudinal wave and shear wave with the same requirements of inspection on nuclear power plant site. Moreover, the new ultrasonic test instrument, the reference block of the same product form and material specification, stainless stell test specimens and copper test specimens block of the same fabricated for the application of new SSP technique. As the result of experimental test with new ultrasonic test instrument and test specimens, the signal to noise ratio was improved by appying the new SSP technique.
Sign Language Transformation System based on a Morpheme Analysis
Lee, Yong-Dong ; Kim, Hyoung-Geun ; Jeong, Woon-Dal ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 90~98
In this paper we have proposed the sign language transformation system for deaf based on a morpheme analysis. The proposed system extracts phoneme components and connection informations of the input character sequence by using a morpheme analysis. And then the sign image obtained by component analysis is correctly and automatically generated through the sign image database. For the effective sign language transformation, the language description dictionary which consists of a morpheme analysis part for analysis of input character sequence and sign language description part for reference of sign language pattern is costructed. To avoid the duplicating sign language pattern, the pattern is classified a basic, a compound and a similar sign word. The computer simulation shows the usefulness of the proposed system.
On a Pitch Alteration Technique in the V/UV Spectrum for High Quality Speech Synthesis Technique
Jo, Wang-Rae ; Bae, Myung-Jin ; Kim, Dong-Sung ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 99~103
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 extraction method of unstable frequency line generated by underwater target using extended Kalman filter
Lee, Sung-Eun ; Hwang, Soo-Bok ; Nam, Ki-Gon ; Kim, Jae-Chang ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 104~109
In passive sonar system, frequency lines generated by underwater target are very important for detection, tracking and classification. In this paper, the extraction method of unstable frequency line from the time samples of the radiated noise of underwater target is studied. As unstable frequency line is time varying, an extended Kalman filter algorithm which is desirable for nonlinear system is applied to extract unstable frequency line. The proposed method shows good extraction of unstable frequency line by application of simulated signal and real target.
Improvement of DTMF Tone Detection in ARS System
Kim, Hee-Dong ; Kim, Je-Woo ; Hong, Young-Jin ;
The Journal of the Acoustical Society of Korea, volume 15, issue 6, 1996, Pages 110~116
In this paper a novel method improving the accuracy of DTMF tone reception in ARS system is proposed. ARS system should allow users to generate DTMF signals while it is sending voice guidance. It is not unocmmon, in this case, that a portion of transmitting voice signals cross-talks to the receiving channel and it often results in interfering with the receiving DTMF signals. Serious degradations including DTMF tone missing, false alarm and so forth have been introduced for the above reason. To overcome this phenomena, we have proposed a way eliminating the frequency spectra representing DTMF signals bands from the transmitting voice signal by using notch filters. This method also employs bandpass filters of which the frequency responses are reciprocal to those of the notch filters incorporated with the DTMF receiver. It is shown that a drastic improvement has been achieved with respect to the DTMF tone detection with little deterioration of voice guidance quality.