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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 20, Issue 8 - Nov 2001
Volume 20, Issue 7 - Oct 2001
Volume 20, Issue 6 - Aug 2001
Volume 20, Issue 5 - Jul 2001
Volume 20, Issue 4 - May 2001
Volume 20, Issue 3 - Apr 2001
Volume 20, Issue 2 - Feb 2001
Volume 20, Issue 1 - Jan 2001
Volume 20, Issue 4E - 00 2001
Volume 20, Issue 3E - 00 2001
Volume 20, Issue 2E - 00 2001
Volume 20, Issue 1E - 00 2001
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Physical Modeling of Plucked String Based on Fixed Spatial Sampling Interval
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 3~12
In physical modeling of plucked string instruments, the vibration of a string is typically simulated by the linear system. Currently the Digital Waveguides of J.O.Smith are widely used to get a high quality sound of the plucked string instrument. He used the wave equation to derive the Digital Waveguides and emphasized the time variable. In this thesis, new model of plucked string is proposed to improve the sound quality emphasizing the spatial variable of the wave equation. In our model, we used the fixed sampling interval which is not dependent on the speed of the wave. So we could get more detailed description of wave movement by the time variable. As a result, the new model could produce a higher quality sound of plucked string instrument.
Nonuniform Delayless Subband Filter Structure with Tree-Structured Filter Bank
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 13~20
Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.
Enhancement of Bearing Estimation Performance at Endfire Using Cardioid Inverse Beamforming
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 21~29
In order to detect the precise port/starboard direction of arrival of target signal in real noisy ocean environments, Inverse beamforming (IBF) algorithm is surveyed theoretically and the detection performances of IBF are analyzed with simulations. Cardioid Inverse beamforming algorithm was proposed for port/starboard discrimination and the performance was studied with simulations. It is shown that IBF has a 3dB array gain advantage over Conventional beamforming (CBF) under ideal conditions. This 3 dB advantage is proven theoretically and illustrated with simulations. The fact that the IBF beamwidth is narrower than the CBF beamwidth by a factor of 0.68 proves the performance of defection and spatial resolution improvement. Comparing the simulation results of Cardioid Inverse beamforming and Conventional Cardioid beamforming, it is shown that Cardioid Inverse beamformer has enhanced performance in minimum detection level, detection accuracy and resolution. Due to the results of moving target bearing detection test in endfire, it is shown that Cardioid Inverse beamformer has better performance, comparing the Conventional Cardioid beamformer.
A Correlation between Emile Sound and Other Waves
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 30~35
The most important characteristic of Emile Bell's sound is a beating. It is modulation phenomenon which appears as a result of interference multiplication in time domain. This modulation phenomenon can be modeled as DSB-SC which suppress carrier and signals distributed both sides. The beatiog wave is observed in Laman distribution signal for polyvinyl speech signal, water vein wave, tide wave. The beating wave is caused by asymmetry Property of the bell.
A Study on the Robust Sound Localization System Using Subband Filter Bank
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 36~42
This paper propose new sound localization algorithm that detects the sound source bearing in a closed office environment using two microphone array. The proposed Subband CPSP (Cross Power Spectrum Phase) algorithm is a development of previously Down CPSP method using subband approach. It first split the received microphone signals into subbands and then calculates subband CPSP which result in possible source bearings. This type of algorithm, Subband CPSP, can provide more robust and reliable sound localization system because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, a real time simulation was conducted and it was compared with previous CPSP method. From the simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than 5％ average accuracy for sound source detection.
Fast Implementation Algorithms for EVRC
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 43~49
EVRC (Enhanced Variable Rate Codec) has been adopted as a standard coder for the CDMA digital cellular system in North America and Korea, and known to provide good call quality at 8kbps. In this paper, fast implementation algorithms for EVRC encoder are proposed. The proposed algorithms are based on both efficient pitch detection scheme and fast fixed codebook search algorithm. In the codebook search, computational complexity is reduced down to 70％ of the original EVRC by limiting the number of pulse position combination and by using a truncated impulse response. The proposed algorithms enable us to implement the EVRC with much smaller computational works. Also, informal subjective tests confirmed that the difference in the speech quality between the original EVRC and the proposed method was indistinguishable.
Fast Algorithm for Recognition of Korean Isolated Words
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 50~55
This paper presents a korean isolated words recognition algorithm which used new endpoint detection method, auditory model, 2D-DCT and new distance measure. Advantages of the proposed algorithm are simple hardware construction and fast recognition time than conventional algorithms. For comparison with conventional algorithm, we used DTW method. At result, we got similar recognition rate for speaker dependent korean isolated words and better it for speaker independent korean isolated words. And recognition time of proposed algorithm was 200 times faster than DTW algorithm. Proposed algorithm had a good result in noise environments too.
Implementation of Adaptive Multi Rate (AMR) Vocoder for the Asynchronous IMT-2000 Mobile ASIC
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 56~61
This paper presents the real-time implementation of an AMR (Adaptive Multi Rate) vocoder which is included in the asynchronous International Mobile Telecommunication (IMT)-2000 mobile ASIC. The implemented AMR vocoder is a multi-rate coder with 8 modes operating at bit rates from 12.2kbps down to 4.75kbps. Not only the encoder and the decoder as basic functions of the vocoder are implemented, but VAD (Voice Activity Detection), SCR (Source Controlled Rate) operation and frame structuring blocks for the system interface are also implemented in this vocoder. The DSP for AMR vocoder implementation is a 16bit fixed-point DSP which is based on the TeakLite core and consists of memory block, serial interface block, register files for the parallel interface with CPU, and interrupt control logic. Through the implementation, we reduce the maximum operating complexity to 24MIPS by efficiently managing the memory structure. The AMR vocoder is verified throughout all the test vectors provided by 3GPP, and stable operation in the real-time testing board is also proved.
LSAW Velocity Measurement by Using a PVDF Line-Focus Ultrasonic Transducer
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 62~67
Velocities of leaky surface acoustic waves (LSAW/sub s/) were measured by using a line-focus polyvinylidene fluoride (PVDF) transducer and compared with theoretically calculated ones. Isotropic materials of Cu, Al, fused quartz, and anisotropic one of Z-cut α-quartz crystal were used as specimens. The velocities were obtained by the separation time between wave components reflected directly from the surface of specimen and LSAW components according to the defocusing distance. The measured velocities well agree with the theoretical results within 1％ error, and it was shown that the leaky pseudo-surface acoustic wave (LPSAW) as well as the LSAW propagates with the typical 6-fold anisotropy on the (0,0,1) surface of α-quartz.
Finite Element Analysis for Acoustic Characteristics of Piezoelectric Underwater Acoustic Sensors
The Journal of the Acoustical Society of Korea, volume 20, issue 1, 2001, Pages 68~76
Sonar is the system that detects objects and finds their location in water by using the echo ranging technique. In order to have excellent performance in variable environment, acoustic characteristics of this system must be analyzed accurately. In this paper, based on the finite element analysis, modeling and analysis of acoustic characteristics of underwater acoustic sensors are preformed. Couplings between piezoelectric and elastic materials, and fluid and structure systems associated with the modeling of piezoelectric underwater acoustic sensors are formulated. In the finite element modeling of unbounded acoustic fluid, IWEE (Infinite Eave Envelop Element) is adopted to take into account the infinite domain. When an incidence wave excites the surface of Tonpilz underwater acoustic sensor, the scattered wave on the sensor is founded by satisfying the radiation condition at the artificial boundary approximately. Based on this scattering analysis, the electrical response of the underwater acoustic sensor under incidence, so called RVS (Receiving Voltage Signal) is founded accurately. This will devote to design Sonar systems accurately.