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REFERENCE LINKING PLATFORM OF KOREA S&T JOURNALS
> Journal Vol & Issue
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 21, Issue 8 - Nov 2002
Volume 21, Issue 7 - Oct 2002
Volume 21, Issue 6 - Aug 2002
Volume 21, Issue 5 - Jul 2002
Volume 21, Issue 4 - May 2002
Volume 21, Issue 3 - Apr 2002
Volume 21, Issue 2 - Feb 2002
Volume 21, Issue 1 - Jan 2002
Volume 21, Issue 1E - 00 2002
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Design and Fabrication of a Convex Array Ultrasonic Transducer with Finite Element Analysis
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 592~599
In this study, an ultrasonic transducer was designed with a commercial finite element analysis (FEA) code, PZFlex, and fabricated based on the design. The transducer has the dimension and shape suitable for abdomen diagnosis working at 5 ㎒ and consists of 128 piezoelectric elements disposed in a convex linear array form. The transducer is composed of two impedance matching layers, one backing layer, and kerfs placed between the piezoelectric elements. Validity of the design with the FEA was illustrated through experimental characterization of a sample transducer. Comparison with the design results by equivalent circuit analysis method was also made to check the superiority of the FEA design.
Characteristics Control of a Thickness Mode Piezoelectric Vibrator Using a Negative Impedance Converter Circuit
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 600~605
In this paper, a Negative Impedance Converter (NIC) circuit was employed for the electro-mechanical characteristic control of a thickness mode piezoelectric vibrator. Two circular plane piezoelectric vibrators were bonded together and the NIC circuit was connected to one of the vibrators. The theoretical and experimental analysis of the characteristics shown that the quality factor and the electro-acoustic efficiency of the vibrator with the NIC circuit could be improved by 20 times and 2.5 times, respectively.
Performance Enhancement of Underwater Acoustic Communication System Using Hydrophone Transmit Array
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 606~613
In this paper we applied a transmit beamforming technique to the underwater acoustic communication system for high rate data transmission. A prototype transmit system was designed and implemented with the general purpose DSP processor and multiple digital-to-analog converters. The performances of the implemented system were evaluated by the experiment in water tank. In order to simplify the procedure the channel coding and equalizer were omitted. And the simplest OOK (On-Off Keying) technique in digital communication methods was applied. The experimental result shows that the transmission data rate is higher about 3 times in the case of 5 hydrophone transmitting may than 1 hydrophone transmitter at bit error rate 10/sup -2/. We verified that the maximum data rate was 400 bps for speech signal transmission in water tank.
Detection of Partial Discharge Acoustic Signal Using the Optical Fiber Interferometric Sensor
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 614~623
In this paper, it was manufactured an interferometric optical fiber sensor and measured partial discharge acoustic signal caused by defect of power facilities such as power cables, transformers and gas insulation. Acrylic and aluminium mandrels wound with fiber-optic were chosen as optical fiber sensor, Sagnac and Mach-Zehnder interferometers were chosen to detect discharge acoustic signals. The two fiber optic interferometers were identified by using the PZT. Discharge experimentation set in the discharge imitation cell in oil tank and the discharge phenomena was generated. Based on the experimental result, to detect the discharge acoustic signal, Sagnac interferometer can detect stably the acoustic signal than the Mach-Zehnder interferometer. It is shown that Sagnac optical fiber sensor can detect the discharge acoustic signals effectively.
A Study on the Insertion Loss of Noise Barrier with the Variation of Top Shape
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 624~631
The insertion loss of the noise barriers with several top shape is measured in an anechoic room by using a reduced scale model test. The insertion loss differences between a straight vertical barrier having 0.3 m height and several barriers with simple top shaped are compared. The results show that the latter is more effective than the former and absorptive barrier is more effective than the reflective one. Among the barrier types of 'T', 'Y', and '(equation omitted)', type 'Y' is the best one and the rest have similar effect. This result is well agree with Alfredson (PIOC. Inter-Noise 95, p. 381, 1995)'s but contradict to May (J. Sound Vb. 71, p. 73, 1980)'s. Therefore, it is difficult to determine which type is the best. In order to find out this discrepancy, boundary element method is adopted and the result shows one can have different result because each supposed different experimental conditions like height of noise barrier, positions of sound source and receiver, etc.
Improvement of Domain-specific Keyword Spotting Performance Using Hybrid Confidence Measure
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 632~640
In this paper, we proposed ACM (Anti-filler confidence measure) to compensate shortcoming of conventional RLJ-CM (RLJ-CM) and NCM (normalized CM), and integrated proposed ACM and conventional NCM using HCM (hybrid CM). Proposed ACM analyzes that FA (false acceptance) happens by the construction method of anti-phone model, and presumed phoneme sequence in actuality using phoneme recognizer to compensate this. We defined this as anti-phone model and used in confidence measure calculation. Analyzing feature of two confidences measure, conventional NCM shows good performance to FR (false rejection) and proposed ACM shows good performance in FA. This shows that feature of each other are complementary. Use these feature, we integrated two confidence measures using weighting vector α And defined this as HCM. In MDR (missed detection rate) 10% neighborhood, HCM is 0.219 FA/KW/HR (false alarm/keyword/hour). This is that Performance improves 22% than used conventional NCM individually.
A Study on Trend Sharing in Segmental-feature HMM
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 641~647
In this paper, we propose the reduction method of the number of parameters in the segmental-feature HMM using trend quantization method. The proposed method shares the trend information of the polynomial trajectories by quantization. The trajectory is obtained by the sequence of feature vectors of speech signals and can be divided by trend and location information. The trend indicates the variation of consequent frame features, while the location points to the positional difference of the trajectories. Since the trend occupies the large portion of SFHMM, if the trend is shared, the number of parameters maybe decreases. To exploit the proposed system the experiments are performed on TIMIT corpus. The experimental results show that the performance of the proposed system is roughly similar to that of previous system. Therefore, the proposed system can be considered one of parameter reduction method.
Implementation of a G,723.1 Annex A Using a High Performance DSP
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 648~655
This paper describes implementation of a multi-channel G.723.1 Annex A (G.723.1A) focused on code optimization using a high performance general purpose Digital Signal Processor (DSP), To implement a multi-channel G.723.1A functional complexities of the ITU-T G.723.1A fixed-point C-code are measures an analyzed. Then we sort and optimize C functions in complexity order. In parallel with optimization, we verify the bit-exactness of the optimized code using the ITU-T test vectors. Using only internal memory, the optimized code can perform full-duplex 17 channel processing. In addition, we further increase the number of available channels per DSP into 22 using fast codebook search algorithms, referred to as bit -compatible optimization.
Statistical Analysis of Korean Phonological Variations Using a Grapheme-to-phoneme System
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 656~664
We present a statistical analysis of Korean phonological variations using a Grapheme-to-Phoneme (GPT) system. The GTP system used for experiments generates pronunciation variants by applying rules modeling obligatory and optional phonemic changes and allophonic changes. These rules are derived form morphophonological analysis and government standard pronunciation rules. The GTP system is optimized for continuous speech recognition by generating phonetic transcriptions for training and constructing a pronunciation dictionary for recognition. In this paper, we describe Korean phonological variations by analyzing the statistics of phonemic change rule applications for the 60,000 sentences in the Samsung PBS Speech DB. Our results show that the most frequently happening obligatory phonemic variations are in the order of liaison, tensification, aspirationalization, and nasalization of obstruent, and that the most frequently happening optional phonemic variations are in the order of initial consonant h-deletion, insertion of final consonant with the same place of articulation as the next consonants, and deletion of final consonant with the same place of articulation as the next consonant's, These statistics can be used for improving the performance of speech recognition systems.
A Study on Extraction of Vocal Tract Characteristic After Canceling the Vocal Cord Property Using the Line Spectrum Pairs
The Journal of the Acoustical Society of Korea, volume 21, issue 7, 2002, Pages 665~670
The most common form of pre-emphasis is y(n)=s(n)-As(n-1), where A typically lies between 0.9 and 1.0 in voiced signal. Also, this value reflects the degree of pre-emphasis and equals R(1)/R(0) in conventional method. This paper proposes a new flattening method to compensate the weaked high frequency components that occur by vocal cord characteristic. We used interval information of LSP to estimate formant frequency, After obtaining the value of slope and inverse slope using linear interpolation among formant frequency, flattening process is followed. Experimental results show that the proposed method flattened the weaked high frequency components effectively. That is, we could improve the flattening characteristics by using interval information of LSP as flattening factor at the process that compensates weaked high frequency components.