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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
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Volume & Issues
Volume 16, Issue 6 - Dec 1997
Volume 16, Issue 5 - Oct 1997
Volume 16, Issue 4 - Aug 1997
Volume 16, Issue 3 - Jun 1997
Volume 16, Issue 2 - Apr 1997
Volume 16, Issue 1 - Feb 1997
Volume 16, Issue 3E - 00 1997
Volume 16, Issue 2E - 00 1997
Volume 16, Issue 1E - 00 1997
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Determination of the Optimum-Bandwidth of Chirp-Signal for Pulse Compression Technique
Ko, Dae-Sik ; Moon, Gun ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 5~9
In this paper, when we use the chirp signal as input signal of ultrasonic signal system the technique for determining the bandwidth of the chirp signal that maximizes the amplitude of the compressed ultrasonic echo signal has been studied. In ultrasonic signal processing systems, the signal-to-noise ratio of the echo signal can be too low due to damping and scattering of the ultrasonic wave during transmission. Method of pulse compression using chirp signal is a means to increase the signal-to-noise ratio in ultrasonic pulse-echo systems. Simulation and experimental results showed that the output signal of ultrasonic system was increased by pulse width of chirp signal and the optimum-bandwidth of the chirp signal was 1.15 times larger than the bandwidth of the ultrasonic system.
Design and Evaluations of Underwater Hydrophone with Self Noise Suppressing Structures -Part Ⅰ. Noise Transfer Characteristics & Effects of Structure Modifications -
Im, Jong-In ; Roh, Young-Rae ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 10~15
The hydrophones is mounted in many applications on a vibrating surface and functions as an underwater acoustic signal receiver without sensing the vibrations from the mounting surface. However, their performance is usually degraded by the interference of exterior noises such as acoustic cavitation in water stream, host structural vibration in the hull, and propeller motions. This paper describes the design and evaluation of a self noise suppressing hydrophones which shows very poor sensitivity to the external noises, first, effects of the external noise on the its receiver performance is simulated with finite element method(FEM). Second, the geometrical variations are implemented on the original structure that include additional air pockets and acoustic walls which work as acoustic shied or scatter of the noises. The results show that the effect of the external noise is the most significant when it is applied near to the bottom of the side wall of the hydrophones. The transverse noise induced by the outside water flow is isolated most effectively when a thin compliant (damping) layer combined with two air pockets is inserted to the circumference of the nose. Noise level is reduced about fifty nine percent of that of the original structure.
Modeling of the Head-Related Transfer Functions with Optimum Reflection Wave Transfer Characteristics in Free-Field Listening over Headphones
Yim, Jeong-Bin ; Kim, Chun-Duck ; Kang, Seong-Hoon ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 16~25
A new method to model the HRTF's(Head-Related Transfer Function), which could give improvement of the sound localization accuracy using the spatial effects by the reflected sound wave transfer characteristics, is proposed. When using the HRTF model having reflected sound wave transfer characteristics, the accuracy of sound localization was quite improved up to about 23%, compared with using the direct wave transfer characteristics only. Furthermore, it is verified that the spatial impression could be a factor to enhance the ability of sound localization.
Thickness Measurement of Adhesive Layer of Multilayer Using Power Cepstrum Technique
Shin, Jin-Seob ; Jun, Kye-Suk ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 26~30
In this paper, the thickness measurement method of adhesive layers of multilayers using power cepstrum signal processing technique has been proposed. The peak values for reflected signal from each layer have been separated by power cepstrum technique. Therefore, thickness of adhesive layers have been measured by the intervals of peak signal. In the experiment, the adhesive layers of 0.5mm-0.75mm thickness using epoxy(2-Ton and Plastic Steel Putty(A)) between the aluminum and the brass were formed. The adhesive layer thickness which is calculated with data of reflected signal by ultrasonic pulse-echo method was within error 1.34% of the measured values.
Variable Vocabulary Word Recognizer using Phonetic Knowledge-based Allophone Model
Kim, Hoi-Rin ; Lee, Hang-Seop ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 31~35
In this paper, we propose a variable vocabulary word recognizer that is able to recognize new words not exist in training data. For the variable vocabulary word recognizer, we must have an on-line lexicon generator to transform new candidate words to the corresponding pronunciation sequences of phones without any large lexicon table. And, we also must make outputs. In order to model the phones and allophones reliably, we define Korean allophones by triphone clustering based on phonetic knowledge of preceding and succeeding phones of each phone. Using the clustering method, we generated 1,548 allophones with POW (Phonetically Optimized Words) 3,848 word DB. We evaluated the proposed word recognizer with POW 3,848 DB, PBW (Phonetically Balanced Words) 445 DB, and 244 word DB in hotel reservation task. Experimental results showed word recognition accuracy of 79.6% for the POW DB corresponding to vocabulary-dependent case, 79.4% in case of 445 word lexicon and 88.9% in case of 100 word lexicon for the PBW DB, and 71.4% for the hotel reservation DB corresponding to vocabulary-independent case.
On a Pitch Extraction of Speech Signal using Residual Signal of the Uniform Quantizer
Bae, Myung-Jin ; Han, Ki-Cheon ; Cha, Jin-Jong ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 36~40
In speech signal processing, it is necessary and important to detect exactly the pitch. The algorithms of pitch extraction which have been proposed until now are difficult exactly pitches over wide range speech signals. In this paper, thus, we proposed a new pitch detection algorithm that finds the fundamental period of speech signal in the residual signal quantized by the uniform quantizer as PCM. The proposed method shows little gross error of average 0.25% for clean speech and average 3.39% for SNR of 0dB. It also achieves results of the pitch contours, improving the accuracy of pitch detection in transient phonemes and noise environments.
The Development of a Speech Recognition Method Robust to Channel Distortions and Noisy Environments for an Audio Response System(ARS)
Ahn, Jung-Mo ; Yim, Kye-Jong ; Kay, Young-Chul ; Koo, Myoung-Wan ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 41~48
This paper proposes the methods for improving the recognition rate of theARS, especially equipped with the speech recognition capability. Telephone speech, which is the input to the ARS, is usually affected by the announcements from the system, channel noise, and channel distortion, thus directly applying the recognition algorithm developed for clean speech to the noisy telephone speech will bring the significant performance degradation. To cope with this problem, this paper proposes three methods: 1)the accurate detection of the inputting instant of the speech in order to immediately turn off the announcements from the system at that instant, 2)the effective end-point detection of the noisy telephone speech on the basis of Teager energy, and 3)the SDCN-based compensation of the channel distortion. Experiments on speaker-independent, noisy telephone speech reveal that the combination of the above three proposed methods provides great improvements on the recognition rate over the conventional method, showing about 77% in contrast to only 23%.
Stochastic Pronunciation Lexicon Modeling for Large Vocabulary Continous Speech Recognition
Yun, Seong-Jin ; Choi, Hwan-Jin ; Oh, Yung-Hwan ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 49~57
In this paper, we propose the stochastic pronunciation lexicon model for large vocabulary continuous speech recognition system. We can regard stochastic lexicon as HMM. This HMM is a stochastic finite state automata consisting of a Markov chain of subword states and each subword state in the baseform has a probability distribution of subword units. In this method, an acoustic representation of a word can be derived automatically from sample sentence utterances and subword unit models. Additionally, the stochastic lexicon is further optimized to the subword model and recognizer. From the experimental result on 3000 word continuous speech recognition, the proposed method reduces word error rate by 23.6% and sentence error rate by 10% compare to methods based on standard phonetic representations of words.
Automatic Classification Method for Time-Series Image Data using Reference Map
Hong, Sun-Pyo ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 58~65
A new automatic classification method with high and stable accuracy for time-series image data is presented in this paper. This method is based on prior condition that a classified map of the target area already exists, or at least one of the time-series image data had been classified. The classified map is used as a reference map to specify training areas of classification categories. The new automatic classification method consists of five steps, i.e., extraction of training data using reference map, detection of changed pixels based upon the homogeneity of training data, clustering of changed pixels, reconstruction of training data, and classification as like maximum likelihood classifier. In order to evaluate the performance of this method qualitatively, four time-series Landsat TM image data were classified by using this method and a conventional method which needs a skilled operator. As a results, we could get classified maps with high reliability and fast throughput, without a skilled operator.
Acquisition of natural Emotional Voice Through Autobiographical Recall Method
Jo, Eun-Kyung ; Jo, Cheol-Woo ; Min, Kyung-Hwan ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 66~70
In order to obtain natural emotional voice in laboratory, an autobiographical recall method was used and happy, angry, sad and afraid feelings were induced in 16 college students. Three independent judges rated the subject's facial expressions and vocal characteristics. The mood induction results were compared with those from the actor-initiated method. Data analysis showed that recall-induced voices successfully conveyed subtle emotional cues, while actor-induced voices signaled more extreme emotioms. Implications of the autobiographical recall method in emotional voice research and potential problems are discussed.
Codeword-Dependent Distance Normalization and Smoothing of Output Probalities Based on the Instar-formed Fuzzy Contribution in the FVQ-DHMM
Choi, Hwan-Jin ; Kim, Yeon-Jun ; Oh, Yung-Hwan ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 71~79
In this paper, a codeword-dependent distance normalization(CDDN) and an instar-formed fuzzy smoothing of output distribution are proposed for robust estimation of output probabilities in the FVQ(fuzzy vector quantization)-DHMM(discrete hidden Markov model). The FVQ-DHMM is a variant of DHMM in which the state output probability is estimated by the sum oft he product of the output probability and its weighting factor for each codeword on an input vector. As the performance of the FVQ-DHMM is influenced by weighting factor and output distribution from a state, it is required to get a method to get robust estimation of weighting factors and output distribution for each state. From experimental results, the proposed CDDN method has reduced 24% of error rate over the conventional FVQ-DHMM, and also reduced 79% of error rate when the smoothing of output distribution is also applied to the computation of an output probability. These results indicate that the use of CDDN and the fuzzy smoothing of output distribution to the FVQ-DHMM lead to improved recognition, and therefore it may be used as an alternative to the robust estimation of output probabilities for HMMs.
A Study of Real-Time Implementation of Audio/Data Processor for Digital/Analog Dual mode Mobile Phone
Byun, Kyung-Jin ; Kim, Jong-Jae ; Han, Ki-Chun ; Yoo, Hah-Young ; Cha, Jin-Jong ; Kim, Kyung-Su ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 80~88
In this paper, the implementation of audio/data processor using ETRI DSP to support analog mode in digital/analog dual mode mobile phone is presented. Audio/data processor performs the wideband data processing, audio signal processing, demodulation function, and data rate conversion when it is operated in analog mode. These functions are programmed in assembly language, and then loaded to ETRI DSP together with vocoder program for the digital mode operation. This is a very efficient implementation of the dual mode cellular phone ASIC since the vocoder for the digital mode and audio/data processor for the analog mode are programmed together in the same hardware.
The Study on development of a SAW SO
Lee, Young-Jin ; Kim, Hak-Bong ; Roh, Yong-Rae ; Cho, Hyun-Min ; Baik, Sung ; , ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 89~94
A new type SO
gas sensor with a particular inorganic thin film on SAW devices was developed. The sensor consisted of twin SAW oscillators of the center frequency of 54 MHz fabricated on the LiTaO
piezoelectric single crystal. One delay line of the sensor was coated with a CdS thin film that selectively adsorbed and desorbed SO
, while the other was uncoated for use as a stable reference. Deposition of the CdS thin film was carried out by the spray pyrolysis method using an ultrasonic nozzle. The sensor could measure the concentration in air less than 0.25 parts per million of SO
. Stability of the sensor turned out to be as good as less than 20ppm, recovery time after each measurement was as short as 5 minutes. Repeatability of the measurement was confirmed through so many reiterated experiments. Hence, the SAW sensor developed through this work showed promising performance as a microsensing tool of SO
. Further work required to improve the performance of the sensor includes enhancement of the reactivity of the CdS thin film with SO
through appropriate dopant addition, an increase of the center frequency of the SAW device.
A Fast VQ Encoding Algorithm
Baek, Seong-Joon ; Lee, Dae-Ryong ; Jeon, Bum-Ki ; Sung, Koeng-Mo ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 95~100
In this paper, we present a new fast VQ encoding algorithm. The proposed algorithm facilitates two characteristics of a vector, i.e., mean and variance to reject many unlikely codewords and save a lot of computation time. Since the proposed algorithm, which is based upon geometric considerations, rejects those codewords that are impossible to be the closest codeword, it provides the same results as a conventional exhaustive(or full) search algorithm. The simulation results confirm the effectiveness of the proposed algorithm.
Time-varying Estimation of Vocal Track Parameters During the Speech Transition Regions
Choi, Hong-Sub ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 101~106
In this paper, sample selective RLS(SSRLS) method is proposed, which aims to eliminate the influence of pitch bias. Its basic concepts are as follows. First it extracts the open glottis interval by using the residual signals, then estimates the formant values from the selected speech samples excluding above open glottis interval. This method has some analogy with the SSLPS, the simulation is conducted upon the synthetic and real speech. From these results, we find more usefulness of the proposed method than the conventional ones.
Noisy Environmental Adaptation for Word Recognition System Using Maximum a Posteriori Estimation
Lee, Jung-Hoon ; Lee, Shi-Wook ; Chung, Hyun-Yeol ;
The Journal of the Acoustical Society of Korea, volume 16, issue 2, 1997, Pages 107~113
To achive a robust Korean word recognition system for both channel distortion and additive noise, maximum a posteriori estimation(MAP) adaptation is proposed and the effectiveness of environmental adaptation for improving recognition performance is investigated in this paper. To do this, recognition experiments using MAP adaptation are carried out for the three different speech ; 1) channel distortion is introduced, 2) environmental noise is added, 3) both channel distortion and additive noise are presented. Theeffectiveness of additive feature parameters, such as regressive coefficients and durations, for environmental adaptation are also investigated. From the speaker independent 100 words recognition tests, we had 9.0% of recognition improvement for the case 1), more than 75% for the case 2), and 11%~61.4% for the case 3) respectively, resulting that a MAP environmental adaptation is effective for both channel distorted and noise added speech recognition. But it turned out that duration information used as additive feature parameter did not played an important role in the tests.