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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
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The Journal of the Acoustical Society of Korea
Journal Basic Information
Journal DOI :
The Acoustical Society of Korea
Editor in Chief :
Volume & Issues
Volume 13, Issue 6 - Dec 1994
Volume 13, Issue 5 - Oct 1994
Volume 13, Issue 4 - Aug 1994
Volume 13, Issue 3 - Jun 1994
Volume 13, Issue 2 - Apr 1994
Volume 13, Issue 1 - Feb 1994
Volume 13, Issue 2E - 00 1994
Volume 13, Issue 1E - 00 1994
Selecting the target year
Effects of the Finite Ground Impedance on the Excess Attenuation of Noise
Kim, Dong-Il ; Kang, Byoung-Yong ; Chang, Ho-Gyeong ; Kim, Ye-Hyun ;
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 5~14
In this study, the ground impedance is measured using the standing wave method in a free field on the grass, the soil, and the ground covered with asphalt and cement. And the excess attenuation of sound is investigated. Results are obtained in the frequency range between 300Hz and 1000Hz. There are very good agreements between the results of the measured ground impedance and the prediction of Delanyand Bazley. The ground impedance is increased in order the grass, the soil, the asphalt and the cement road, decreased with frequency for each the ground. The excess attenuation of sound is mainly determined by the ground impedance. The experimental results of the excess attenuation over the different types and the microphone heights are compared with the theoretical values.
Development and Experiment of a Linear Array Acoustic Lens with 31 Microphones
Hyun, Seok-Bong ; Min, Dong-Hyun ; Kim, Su-Young ;
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 15~23
We developed an electronic lens for acoustic imaging systems, which is linear array with 31 microphones equally spaced with distance 34mm. Resonant frequency fo receiver circuit coupled to microphone is 20 kHz. We arranged 16 microphones horizontally and 15 microphones vertically, so that the array allows us to obtain a 2 dimensional angle of source, and to track the motion of source in real time. Due to the problem of aliasing in discrete Fourier Transfrom, the maximum observable angle of the lens is limited to 15
. We also employed quadrature phase detection scheme to adjust the focus. We have tested the acoustic lens with a personal computer in an anechoic room and obtained the results agreed with the acoustic imaging theory.
A Study on Word Juncture Modeling for Continuous Speech Recognition of Korean Language
Choi, In-Jeong ; Un, Chong-Kwan ;
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 24~31
In this paper, we study continuous speech recognition of Korean language using acoustic models of word juncture coarticulation. To alleviate the performance degradation due to coarticulation problems, we use context-dependent units that model inter-word transitions in addition to intra-word transitions. In all cases the initial phone of each word has to be specified for each possible final phone of the previous word similarly for the final phone of each word. To improve the robustness of the HMM parameters, the covariance matrix is smoothed. We also use position-dependent units to improve the discriminative power between units. Simulation results show that when the improved models of word juncture coarticulation are used. the recognition performance is considerably improved compared to the baseline system using only intra-word units.
A Study on the Speech Transmission Index Method for Estimating Articulation of Loudspeaking Telephony
Jang, Dae-Young ; Kang, Seong-Hoon ; Sim, Dong-Yeon ; Kim, Chun-Duck ;
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 32~39
The speech transmission quality in telephone is quantified in terms of loudness rating, but this method has been validated only for the handset telephony. The transmission quality of loudspeaking telephony in any room must be evaluated not only with speech transmission but also with background noise, echo and reverberation since the effect of room acoustics is much stroger for loudspeaking telephoy. Therefore, it requires a better approach to specify the quality of loudspeaking telephony. By calcuating the speech transmission index (STI), a physical method for measuring the quality of speech transmission was proposed by Steeneken. In this paper, the application of a STI method for estimating articulation of loudspeaking telephony was discussed. And the STI measurement system with high speed calculation was also three rooms, having different reverberation times. The result show that the STI decreases as the reverberation time of rooms increases. It suggests that speech transmission index method can be useful evaluating articulation of a loudspeaking telephony including the sound field characteristics.
A Study on the Sensitivity Compensation of Three-dimensional Acoustic Intensity Probe in the Higher Frequency Range
Kim, Suk-Jae ; Hideo, Suzuki ; Kim, Chun-Duck ;
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 40~50
In this paper, the sensitivity compensation method for three-dimensional acoustic intensity probe in the higher frequency range has been studied. The measurement error in the higher frequency range is generated from the phase mismatch between microphone's signals of the probe. If the wavelength of sound signal measured is less than those of the distance between microphones of the probe, that is, the higher frequency of the sound signal, the bigger measurement error is generated. In this study, we proposed the compensation methods for one-dimensional acoustic intensity probe with two-microphones, and the efficiency of those methods were investigated by numerical calculation of computer. It was most effective method to compensate the phase mismatch between microphone for the acoustic intensity probe was investigated for the sound estimated. and the efficiency of this method in a three-dimensional probe was investigated for the sound wave travelling in the arbitrary direction by numerical calculation of computer. In this result, the efficiency was proved that, for the measurement error of 1dB or less with the three-dimensional probe of 60mm space, the frequency should be less than 1.2kHz without the error compensation method, but the frequency increased up to 2.8kHz with the error compensation method.
A Study on the Generation of Multi-syllable Nonsense Wordset for the Assessment of Synthetic Speech
Jo, Cheol-Woo ; Kim, Kyung-Tae ; Lee, Yong-Ju ;
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 51~58
These times many kinds of man-machine Interfaces using speech signal, speech recognizers or speech synthesizers, are proposed and utilized in practice. Especially speech synthesis system is widely used in our life. But its assessment method is still in its first stage. In this paper we propose a method to generate multi-syllable nonsense wordset for the purpose of synthetic speech assessment and applies the wordset to one commercial text-to-speech system. Some results about the experiment is suggested and it is verified that the method to generate a nonsense wordset can be used to assess the intelligibility of the synthesizer in phoneme level or in phonemic environmental level.
Performance Assessment of Speech Recogniger using Lombard Speech
Jung, Sung-Yun ; Chung, Hyun-Yeol ; Kim, Kyung-Tae ;
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 59~68
This paper describes the performance assessment test and analysis of test results on a Korean speech recognizer which recognizes Lombard effect received speech in noisy environment, as a basic performance assessment research. In the assessement test, standard speech data were first manipulated close to speech uttered in a noisy environment, and then performance assessment tests were carried out along with the assessment items (the type of noise, SNR) in two ways-one with Lombard effect received speech(LES), the other with not received(NLES). As a result, when 90% of recognition rate is set to be a recognition limit, it was achieved at 10dB SNR point with LES, while at 30dB with NLES. This 20dB of SNR difference indicates Lombard effect should be considered in real world assessment test. The type of noises didn't affect performance of recognizers in out tests. ANOVA analysis, in evaluating several kinds of recognizers, showed every assessment item affecting the recognition performance could be quantified.
자음의 단어내 음운환경별로 본 음가변화
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 69~76
Acoustic cues of some consonantal phonology were tested in Korean words. All Korean consonants were recorded and acoustically analyzed in controlled phonological environments :ⅰ) word-initial, ⅱ) inter-vocalic, and ⅲ) word-final positions. The observed acoustic regulations are : ⅰ) The lengths of obstruents are longer word-initially than word-finally, ⅱ) The lengths of sonorants are longer word-finally than in word-initial or inter-vocalic positions, ⅲ) The formants of the lateral sound /l/ are higher word-finally than intervocalically. The phonological explanations of these acoustic regulations can be found in the rules of ⅰ) inter-vocalic voicing of plain stops, ⅱ) syllable-final unreleasing of obstruents, ⅲ) word-initial aspiration of stops, and ⅳ) liquid alternation between [r] and [l]. Numerical data of all these acoustic regulations are reported in order to facilitate their application toward improving naturalness for speech synthesis and accurateness for speech recognition
Pitch Detection Using Variable Bandwidth LPF
Keum, Hong ; Baek, Guem-Ran ; Bae, Myung-Jin ; Jang, Ho-Sung ;
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 77~82
In speech signal processing, it is very important to detect the pitch exactly. Although various methods for detecting the pitch of speech signals have been developed, it is difficult to exactly extract the pitch for wide range of speakers and various utterances. Thus we propose a new pitch detection algorithm which takes advantage of the G-peak extraction. It is a method to detect the pitch period of the voiced signals by finding MZCI (maximum zero-crossing interval) of the G-peak which is defined as cut-off bandwidth rate of LPF (low pass filter). This algorithm performs robustly with a gross error rate of 3.63% even in 0 dB SNR environement. The gross error rate for clean speech is only 0.18%. Also it is able to process all courses with high speed.
The Journal of the Acoustical Society of Korea, volume 13, issue 5, 1994, Pages 83~110