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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 23, Issue 8 - Nov 2004
Volume 23, Issue 7 - Oct 2004
Volume 23, Issue 6 - Aug 2004
Volume 23, Issue 5 - Jul 2004
Volume 23, Issue 4 - May 2004
Volume 23, Issue 3 - Apr 2004
Volume 23, Issue 2 - Feb 2004
Volume 23, Issue 1 - Jan 2004
Volume 23, Issue 4E - 00 2004
Volume 23, Issue 3E - 00 2004
Volume 23, Issue 2E - 00 2004
Volume 23, Issue 1_E - 00 2004
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An FPGA Implementation of the Synthesis Filter for MPEG-1 Audio Layer III by a Distributed Arithmetic Lookup Table
Koh Sung-Shik ; Choi Hyun-Yong ; Kim Jong-Bin ; Ku Dae-Sung ;
The Journal of the Acoustical Society of Korea, volume 23, issue 8, 2004, Pages 554~561
As the technologies of semiconductor and multimedia communication have been improved. the high-quality video and the multi-channel audio have been highlighted. MPEG Audio Layer 3 decoder has been implemented as a Processor using a standard. Since the synthesis filter of MPEG-1 Audio Layer 3 decoder requires the most outstanding operation in the entire decoder. the synthesis filter that can reduce the amount of operation is needed for the design of the high-speed processor. Therefore, in this paper, the synthesis filter. the most important part of MPEG Audio, is materialized in FPGA using the method of DAULT (distributed arithemetic look-up table). For the design of high-speed synthesis filter, the DAULT method is used instead of a multiplier and a Pipeline structure is used. The Performance improvement by 30% is obtained by additionally making the result of multiplication of data with cosine function into the table. All hardware design of this Paper are described using VHDL (VHIC Hardware Description Language) Active-HDL 6.1 of ALDEC is used for VHDL simulation and Synplify Pro 7.2V is used for Model-sim and synthesis. The corresponding library is materialized by XC4013E and XC4020EX. XC4052XL of XILINX and XACT M1.4 is used for P&R tool. The materialized processor operates from 20MHz to 70MHz.
Distribution of Seagrass (Zostera marina) Beds and High Frequency Backscattering Characteristics by Photosynthesis
Yoon Kwan-Seob ; La Hyoung Sul ; Na Jungyul ; Lee Jae-Hyuk ;
The Journal of the Acoustical Society of Korea, volume 23, issue 8, 2004, Pages 562~569
An experiment for observation of the distribution of the seagrass (zostera marina) beds and characteristics of high-frequency backscattering by the photosynthesis was conducted off the coast. Acoustic data were taken as a function of the grazing angles and the relative azimuth angles on the seagrass beds of which bottom type was sandy-mud. The transmitted source signal was a 120 kHz CW waveform. Mapping of the seagrass beds distribution was drawn up using the seagrass backscattering strength with azimuth and grazing angles. The result of the comparison backscattering strength distribution of the seagrass beds was shown to be the similar to the photograph of real seagrass beds. The seagrass backscattering strength was also compared between day and night to verify the effects of the acoustical scattering by the bubbles of Photosynthetic oxygen formed on the seagrass. In these results. it is clear that observation of the seagrass beds between day and night showed the different characteristics because the bubbles of Photosynthetic oxygen affect the acoustical scattering.
An Array Beampattern Synthesis Using Adaptive Array Method and Partial Constrained Adaptation
Lim Jun-Seok ; Choi Nakjin ; Sung Koeng-Mo ; Kim Hyun-Seok ;
The Journal of the Acoustical Society of Korea, volume 23, issue 8, 2004, Pages 570~575
In the underwater acoustic systems. we can receive signals and retrieve information about a target by using a beamforming method. The most important thing in the beamforming is finding the way to optimize the mainlobe beamwidth and the sidelobe level to the desired value. One of the prominent results of beamforming method. which has been studied. is Philip's weighting function method(1) . Philip's method adaptively adjusts its weights of array to meet the desired mainlobe beamwidth and sidelobe level. It is very similar to the design method in adaptive filter. However. this method cannot easily bring us to the desired sidelobe level due to complementary relation between mainlobe beamwidth and sidelobe level. In this paper, we propose a new algorithm using partial constrained adaptation. This method makes us circumvent the above problem and meet the specification of design easily. The proposed algorithm presents a Pattern synthesis that designer can easily control the mainlobe beamwidth and the sidelobe level to the desired value while calculation time to converge is decreasing.
Time Domain Acoustic Propagation Analysis Using 2-D Pseudo-spectral Modeling for Ocean Environment
Kim Keesan ; Lee Keunhwa ; Seong Woojae ;
The Journal of the Acoustical Society of Korea, volume 23, issue 8, 2004, Pages 576~582
A computer code that is based on the Pseudo-spectral finite difference algorithm using staggered grid is developed for the wave propagation modeling in the time domain. The advantage of a finite difference approximation is that any geometrically complicated media can be modeled. Staggered grids are advantageous as it provides much more accuracy than using a regular grid. Pseudo-spectral methods are those that evaluate spatial derivatives by multiplying a wavenumber by the Fourier transform of a pressure wave-field and performing the inverse Fourier transform. This method is very stable and reduces memory and the number of computations. The synthetic results by this algorithm agree with the analytic solution in the infinite and half space. The time domain modeling was implemented in various models. such as half-space. Pekeris waveguide, and range dependent environment. The snapshots showing the total wave-field reveals the Propagation characteristic or the acoustic waves through the complex ocean environment.
Vector Quantizer Based Speaker Normalization for Continuos Speech Recognition
Shin Ok-keun ;
The Journal of the Acoustical Society of Korea, volume 23, issue 8, 2004, Pages 583~589
Proposed is a speaker normalization method based on vector quantizer for continuous speech recognition (CSR) system in which no acoustic information is made use of. The proposed method, which is an improvement of the previously reported speaker normalization scheme for a simple digit recognizer, builds up a canonical codebook by iteratively training the codebook while the size of codebook is increased after each iteration from a relatively small initial size. Once the codebook established, the warp factors of speakers are estimated by comparing exhaustively the warped versions of each speaker's utterance with the codebook. Two sets of phones are used to estimate the warp factors: one, a set of vowels only. and the other, a set composed of all the Phonemes. A Piecewise linear warping function which corresponds to the estimated warp factor is adopted to warp the power spectrum of the utterance. Then the warped feature vectors are extracted to be used to train and to test the speech recognizer. The effectiveness of the proposed method is investigated by a set of recognition experiments using the TIMIT corpus and HTK speech recognition tool kit. The experimental results showed comparable recognition rate improvement with the formant based warping method.
Self-Adaptation Algorithm Based on Maximum A Posteriori Eigenvoice for Korean Connected Digit Recognition
Kim Dong Kook ; Jeon Hyung Bae ;
The Journal of the Acoustical Society of Korea, volume 23, issue 8, 2004, Pages 590~596
This paper Presents a new self-adaptation algorithm based on maximum a posteriori (MAP) eigenvoice for Korean connected digit recognition. The proposed MAP eigenvoice is developed by introducing a probability density model for the eigenvoice coefficients. The Proposed approach provides a unified framework that incorporates the Prior model into the conventional eigenvoice estimation. In self-adaptation system we use only one adaptation utterance that will be recognized, we use MAP eigenvoice that is most robust adaptation. In series of self-adaptation experiments on the Korean connected digit recognition task. we demonstrate that the performance of the proposed approach is better than that of the conventional eigenvoice algorithm for a small amount of adaptation data.
Distorted Speech Rejection For Automatic Speech Recognition under CDMA Wireless Communication
Kim Nam Soo ; Chang Joon-Hyuk ;
The Journal of the Acoustical Society of Korea, volume 23, issue 8, 2004, Pages 597~601
This paper introduces a pre-rejection technique for wireless channel distorted speech with application to automatic speech recognition (ASR) Based on analysis of distorted speech signals over a wireless communication channel. we propose a method to reject the channel distorted speech with a small computational load. From a number of simulation results. we can discover that tile pre-rejection algorithm enhances the robustness of speech recognition operation.