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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 22, Issue 8 - Nov 2003
Volume 22, Issue 7 - Oct 2003
Volume 22, Issue 6 - Aug 2003
Volume 22, Issue 5 - Jul 2003
Volume 22, Issue 4 - May 2003
Volume 22, Issue 3 - Apr 2003
Volume 22, Issue 2 - Feb 2003
Volume 22, Issue 1 - Jan 2003
Volume 22, Issue 1E - 00 2003
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Nonlinear Sound Amplification and Directivity Due to Underwater Bubbles
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 250~260
Since a bubble in water is a highly nonlinear acoustic scatterer, the acoustic scattered waves from underwater bubbles show highly nonlinear acoustic properties. These acoustic scattered waves can be observed at the second or higher harmonics as well as at the fundamental primary frequency of incident acoustic wave. When two primary acoustic waves of different frequencies are incident on a bubble, the acoustic scattered waves can be also observed at the sum and the difference frequencies of the primary waves. In this study, when the two primary acoustic waves were incident on a bubble screen in water, we observed that the amplitude of difference frequency wave was amplified by the bubble nonlinearity and its directivity was oriented in the propagation directions of primary waves. The directivity of scattered difference frequency wave was analyzed as a coherent scattering for virtual source by using the directivity of the primary acoustic wave.
Echo Canceller with Improved Performance in Noisy Environments
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 261~268
Conventional acoustic echo cancellers using ES algorithm have simple structure and fast convergence speed compared with those using NLMS algorithm, but they are very weak to external noise because ES algorithm updates the adaptive filter taps based on average energy reduction rate of room impulse response in specific acoustical condition. To solve this problem, in this paper, a new update algorithm for acoustic echo canceller with stepsize matrix generator is proposed. A set of stepsizes is determined based on residual error energy which is estimated by two moving average operators, and applied to the echo canceller in matrix from, resulting in improved convergence speed. Simulations in various noise condition show that the proposed algorithm improves the robustness of acoustic echo canceller to external noise.
Audio Watermarking Using Specific Frequency Coefficients
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 269~275
In this paper, we proposed the robust watermark diminishing distortion of the original data inserting the watermark in specific coefficients of the frequency domain. In case the alpha is more than 0.5. we found that proposed watermark is detected by experiment of MP3, FFT, Cropping and Echo attack. Our proposed method improved the Cox's method in the SNR aspect.
Real-time Implementation of a Tone Sender/Receiver on a High Performance DSP
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 276~285
In this paper, we present real-time implementation of a R2MFC/DTMF (R2 Multi Frequency Combinations/Dual Tone Multiple Frequency) tone receiver/sender using a high performance DSP (Digital Signal Processor) and apply it to a carrier class VoIP (Voice over Internet Protocol) gateway system. The Receiver utilizes the Goertzel filter and the sender adopts the harmonic resonant filter. We describe, in detail, the techniques of multi-channel real-time implementation on a Texas Instruments TMS320C62x DSP such as effective PCM (Pulse Code Modulation) in/out by means of DMA (Direct Memory Access) and McBSP (Multi Channel Buffered Serial Port) and message communication via HPI (Host Port Interface), etc. From experimental results, we confirmed that the optimized code provided 780 channel capacity at 250㎒ C6202, and the our R2MFC/DTMF receiver/sender met ITU-T (International Telecommunication Union-Telecommunication) specifications.
Extraction of Unvoiced Consonant Regions from Fluent Korean Speech in Noisy Environments
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 286~292
Voice activity detection (VAD) is a process that separates the noise region from silence or noise region of input speech signal. Since unvoiced consonant signals have very similar characteristics to those of noise signals, it may result in serious distortion of unvoiced consonants, or in erroneous noise estimation to can out VAD without paying special attention on unvoiced consonants. In this paper, we propose a method to extract in an explicit way the boundaries between unvoiced consonant and noise in fluent speech so that more exact VAD could be performed. The proposed method is based on histogram in frequency domain which was successfully used by Hirsch for noise estimation, and a1so on similarity measure of frequency components between adjacent frames, To evaluate the performance of the proposed method, experiments on unvoiced consonant boundary extraction was performed on seven kinds of noisy speech signals of 10 ㏈ and 15 ㏈ SNR respectively.
A Study on the Fast Search Algorithm for Vector Quantization
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 293~298
In this paper. we propose a fast search algorithm for nearest neighbor vector quantization (NNVQ). The proposed algorithm rejects those codewords which can not be the nearest codeword and reduces the search range of codebook. Hence it reduces computational time and complexity in encoding process, while it provides the same SD performance as the conventional full search algorithm. We apply the proposed algorithm to the adaptive multi-rate (AMR) speech coder and a general vector quantizer designed by LBG. algorithm. Simulation results show effectiveness of the proposed algorithm.
Quantization Based Speaker Normalization for DHMM Speech Recognition System
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 299~307
There have been many studies on speaker normalization which aims to minimize the effects of speaker's vocal tract length on the recognition performance of the speaker independent speech recognition system. In this paper, we propose a simple vector quantizer based linear warping speaker normalization method based on the observation that the vector quantizer can be successfully used for speaker verification. For this purpose, we firstly generate an optimal codebook which will be used as the basis of the speaker normalization, and then the warping factor of the unknown speaker will be extracted by comparing the feature vectors and the codebook. Finally, the extracted warping factor is used to linearly warp the Mel scale filter bank adopted in the course of MFCC calculation. To test the performance of the proposed method, a series of recognition experiments are conducted on discrete HMM with thirteen mono-syllabic Korean number utterances. The results showed that about 29% of word error rate can be reduced, and that the proposed warping factor extraction method is useful due to its simplicity compared to other line search warping methods.
Efficient Codebook Search Method for AMR Wideband Speech Codec
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 308~314
Wideband speech communications with 7㎑ bandwidth can provide high-quality speech services that are almost impossible with current narrow-band speech communications with 3.4 ㎑ bandwidth, and AMR wideband codec was recently developed for these services. The performance of AMR wideband codec is excellent due to its wideband information and partially to ACELP structure, but it requires high computational complexity especially in codebook search. In this paper, to solve this problem, an efficient codebook search method for AMR wideband codec is proposed. The proposed method first determines the coarse initial codevector, then improves the performance of codevector by replacing a poor pulse in codevector with better one iteratively. Simulations show that AMR wideband codec with proposed codebook search method has higher performance with much less computational cost than conventional AMR wideband codec.
Channel-attentive MFCC for Improved Recognition of Partially Corrupted Speech
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 315~322
We propose a channel-attentive Mel frequency cepstral coefficient (CAMFCC) extraction method to improve the recognition performance of speech that is partially corrupted in the frequency domain. This method introduces weighting terms both at the filter bank analysis step and at the output probability calculation of decoding step. The weights are obtained for each frequency channel of filter bank such that the more reliable channel is emphasized by a higher weight value. Experimental results on TIDIGITS database corrupted by various frequency-selective noises indicated that the proposed CAMFCC method utilizes the uncorrupted speech information well, improving the recognition performance by 11.2% on average in comparison to a multi-band speech recognition system.
Generating Korean Energy Contours Using Vector-regression Tree
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 323~328
This study describes an energy contour generation method for Korean n systems. We propose a vector-regression tree, which is a vector version of a scalar regression tree. A vector-regression tree predicts a response vector for an unknown feature vector. In our study, the tree yields a vector containing ten sampled energy values for each phone. After collecting 500 sentences and its corresponding speech corpus, we trained trees on 300 sentences and tested them on 200 sentences. We construct a bagged tree and a born again one to improve the performance of contour prediction. In the experiment, we got a 0.803 correlation coefficient for the observed and predicted energy values.
A Study on an Ultrasonic Circular Array Transducer for Intra-vascular Ultra-sound Diagnosis
The Journal of the Acoustical Society of Korea, volume 22, issue 4, 2003, Pages 329~336
Intra-Vascular Ultra-Sound (IVUS) transducers were developed for the application to diagnose coronary diseases. The transducer consists of 32 piezoelectric elements with a front insulation layer and a polymeric acoustic backing layer on a hollow alumina tube. The optimal geometrical structure of the transducer was designed through theoretical analysis of radiation patterns of the transducer. Samples of the IVUS transducers of the diameter of 3㎜ were fabricated to illustrate the design scheme. For the piezoelectric elements, 2-2 mode piezocomposite materials were employed. Experimental performance of the transducers showed good agreement with the design results, which verified feasibility of the transducer for IVUS applications.