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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 14, Issue 6 - Dec 1995
Volume 14, Issue 5 - Oct 1995
Volume 14, Issue 4 - Aug 1995
Volume 14, Issue 3 - Jun 1995
Volume 14, Issue 2 - Apr 1995
Volume 14, Issue 1 - Feb 1995
Volume 14, Issue 2E - 00 1995
Volume 14, Issue 1E - 00 1995
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Propagation Loss Variability due to Hourly Variations of Underwater Sound Speed profiles in the Korea Strait
Na, Youn-Nam ; Shim, Tae-Bo ; Kim , Seong-Il ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 5~5
In order to estimate the variability of the wave propagation loss (PL) du e to hourly variations of the sound speed profiles (SSPs), we conducted oceanographic measurements every hour for 39 hours in October 1993 in the Korea Strait. Currents and meteorological data were measured simultaneously to examine the causes of the temporal variations of temperatures. During the experiment, the temporal variations of temperatures in the surface layer highly depend on the water mass transport from adjacent seas. The PL for low frequency (75-300 Hz) is calculated using the parabolic equation scheme and averaged over the whole water depth. The hourly variation of the SSP may cause a PL difference of up to 10 dB over a 30-50 km range. The variability of PL, represented by standard deviation for the 39 SSPs, is as large as 3 dB over a 50 km range.
Parallel Sorting Algorithm by Median-Median
Min, Yong-Sik ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 14~14
This paper presents a parallel sorting algorithm suitable for the SIMD multiprocessor. The algorithm finds pivots for partitioning the data into ordered subsets. The data can be evenly distributed to be sorted since it uses the probability theory. For n data elements to be sorted on p processors, when
, the algorithm is shown to be asymptotically optimal. In practice, sorting 8 million data items on 64 processors achieved a 48.43-fold speedup, while the PSRS required a 44.4-fold speedup. On a variety of shared and distributed memory machines, the algorithm achieved better than half-linear speedups.
On a Reduction of Pitch Searching Time by Separating the Speech Components in the CELP Vocoder
Hyeon, Jin-Il ; Byeon, Gyeong-Jin ; Han, Gi-Cheon ; Kim, Jong-Jae ; Yu, Ha-Yeong ; Kim, Jae-Seok ; Kim, Dae-Sik ; Bae, Myeong-Jin ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 22~22
Code excited Linear Prediction(CELP) vocoder exhibits good performance at data rates below 4.8 kbps. The major drawback of CELP type coders is their large amount of computation. In this paper, we propose a new pitch searching method that preseves the quality of the CELP vodocer reducing computational complexity. The basic idea is that pregrasps preliminary pitches about signal and performs pitch search only about the preliminary pitches. Applying the proposed method to the CELP vocoder, we can reduce complexity about 90% in th pitch search.
A Comparative Performance Study of Speech Coders for Three-Way Conferencing in Digital Mobile Communication Networks
Lee, Mi-Suk ; Lee, Yun-Geun ; Kim, Gi-Cheol ; Lee, Hwang-Su ; Jo, Wi-Deok ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 30~30
In this paper, we evaluated the performance of vocoders for three-way conferencing using signal summation technique in digital mobile communication network. The signal summation technique yields natural mode of three-way conferencing, in shich the mixed voice signal from two speakers are transmitted to a third person, though there has been no useful speech coding technique for the mixed voice signal yet. We established Qualcomm code term prediction (RPE-LTP) vocoders to provide three-way conferencing using signal summation techinique. In addition, as the conventional speech quality measures are not applicable to the vocoders for mixed voice signals, we proposed two kinds of subjective quality measures. These are the sentence discrimination (SD) test and the modified degraded mean opinion score (MDMOS) test. The experimental results show that the output speech quality of the VSELP vocoder is superior to other two.
Noisy Speech Recognition Based on Spectral Mapping Techniques
Lee, Ki-Young ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 39~39
This paper presents noisy speech recognition method based on spectral mapping techniques of speaker adaptation method. In the presented method, the spectral mapping training makes the spectral distortion of noisy speech reduced, and for the more correctively spectral mapping, let the adjustment window;s slope be adaptive to several word lengths. As a result of recognition experiment, the recognition rate is higher than that of the conventional method using VQ and DTW without noise processing. Even when SNR level is 0 dB, the recognition rate is 10 times more than that using the conventional method. It is confirmed that the speacker adaptation technique using the spectral mapping training has an ability to improve the recognition performance for noisy speech.
Enhanced Wavelet Transform-based CELP Coder with Band Selection and Selective VQ
Chang, Dong-Il ; Cho, Young-Kwon ; Ann, Sou-Guil ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 46~46
In this paper, we present a new wavelet transform-based CELP coder, called band selection wavelet transform CELP (BS-WTCELP) operated at 4.8 kbps. The proposed algorithm uses a band selection scheme of frequency bands of wavelet transform and selective vector quantization (VQ). The band selection and selective VQ structure is implemented by using a classified VQ structure. The proposed algorithm has about 0.5-1.0 dB improvement in segmental SNR compared with the conventional CELP that uses the random codebook search, while is has significantly reduced computational and storage complexity. Many experimental results have shown that the proposed algorithm is more suitable for most real-applications than the conventional CELP and wavelet transform CELP.
Convergence Analysis of the Least Mean Fourth Adaptive Algorithm
Cho, Sung-Ho ; Kim, Hyung-Jung ; Lee, Jong-Won ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 56~56
The least mean fourth (LMF) adaptive algorithm is a stochastic gradient method that minimizes the error in the mean fourth sense. Despite its potential advantages, the algorithm is much less popular than the conventional least mean square (LMS) algorithm in practice. This seems partly because the analysis of the LMF algorithm is much more difficult than that of the LMS algorithm, and thus not much still has been known about the algorithm. In this paper, we explore the statistical convergence behavior of the LMF algorithm when the input to the adaptive filter is zero-mean, wide-sense stationary, and Gaussian. Under a system idenrification mode, a set of nonlinear evolution equations that characterizes the mean and mean-squared behavior of the algorithm is derived. A condition for the conbergence is then found, and it turns out that the conbergence of the LMF algorithm strongly depends on the choice of initial conditions. Performances of the LMF algorithm are compared with those of the LMS algorithm. It is observed that the mean convergence of the LMF algorithm is much faster than that of the LMS algorithm when the two algorithms are designed to achieve the same steady-state mean-squared estimation error.
A Novel Approach to Improving the Performance of Randomly Perturbed Sensor Arrays
Chang, Byong-Kun ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 65~65
The effects of random errors in array weight and sensor positions on the performance of a Linearly constrained linear sensor array is analyzed in a weight vector space. It is observed that a nonorthogonality exists between an optimum weight vector and the steering vector of an interference direction du e to random errors. A novel approach to improving the nulling performance by compensating for the nonorthogonality is proposed. Computer simulation results are presented.
Merging Algorithm for Relaxed Min-Max Heaps Relaxed min-max 힙에 대한 병합 알고리즙
Min, Yong-Sik ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 73~73
This paper presents a data structure that implements a mergeable double-ended priority queue ; namely, an improved relaxed min-max-pair heap. It suggests a sequential algorithm to merge priority queues organized in two relaxed min-max heaps : kheap and nheap of sizes k and n, respecrively. This new data sturuture eliminates the blossomed tree and the lazying method used to merge the relaxed min-max heaps in . As a result, the suggested method in this paper requires the time complexity of O(log(log(n/k))*log(k)) and the space complexity of O(n+), assuming that
are in two heaps of different sizes.
QCELP Implementation on TMS320C30 DSP Board TMS320C30 DSP를 이용한 QCELP Codec의 실현
Han, Kyong-Ho ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 83~83
The implementation of the voice dodec is imjplemented by using TMS320C30, which is the floating point DSP chip from Texas Instrument. QCELP (Qualcomm Code Excited Linear Prediction) is used to encode and decode the voice. The QCELP code is implemented by the TMS320C30 C-dode. The DSP board is controlled by the PC. The PC program tranfors the voice file from and to the DSP board, which is also implemented by C-code. The voice is encoded by the DSP board and the encoded data is transferred to PC to be stored as a file. To hear the voice. the voice data file is sent to DSP board and decoded to synthesize audible voice. Two flags are used by both programs to notify the status of the operation. By checking the flags, DSP and PC decides when the voice data is transferred between them.
Performance Analysis of Highly Effective Proposed Direction Finding Method
Rhee, Ill-Keun ;
The Journal of the Acoustical Society of Korea, volume 14, issue 1E, 1995, Pages 88~88
The main purpose of this paper is to show the realizaability of the proposed highly effective direction finiding method which performs extremely well under the circumstances like low signal-to-noise ratio (S/N), very closely located signal sources, and so on. In order to achieve the purpose, the degree to which the proposed method is superior to the MUSIC(multiple signal classification) with respect to the S/N is discussed, and the result is analyzed in terms of the S/N and the number of sample data.