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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 18, Issue 8 - Nov 1999
Volume 18, Issue 7 - Oct 1999
Volume 18, Issue 6 - Aug 1999
Volume 18, Issue 5 - Jul 1999
Volume 18, Issue 4 - May 1999
Volume 18, Issue 3 - Apr 1999
Volume 18, Issue 2 - Feb 1999
Volume 18, Issue 1 - Jan 1999
Volume 18, Issue 4E - 00 1999
Volume 18, Issue 3E - 00 1999
Volume 18, Issue 2E - 00 1999
Volume 18, Issue 1E - 00 1999
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An Active Noise Canceller with Blind Source Separation
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 3~8
In this paper, we propose a new active noise control system that cancels the only noise signal from the mixture selectively. A blind source separation realized by a dynamic recurrent neural network is used as a preprocessor of the active noise control system and separates the desired signal and the noise signal. The active noise control system adaptively generates an anti-noise signal to remove the only noise signal separated by the blind source separation. Computer simulation results show that the proposed scheme is effective to construct a selective attention system.
Target Motion Analysis for a Passive Sonar System with Observability Enhancing
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 9~16
As a part of target motion analysis(TMA) with highly noisy bearings-only measurements from a passive sonar system, a nonlinear batch estimator is proposed to provide the initial estimates to a sequential estimator called the modified gain extended Kalman filter(MGEKF). Based on the system observability analysis of passive target tracking, a practical and effective method is suggested to determine the observer maneuvers for improved TMA performance through system observability enhancing. Also suggested is a method to determine observer location for enhanced system observability at the initial phase of TMA from various engagement boundaries which represent the relationship between observer-target relative geometrical data and system observability. The proposed TMA methods are tested by a series of computer simulation runs.
Pseudo Stereophonic Acoustic Echo Canceller using Hyper-plane Projection Algorithm
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 17~30
This paper proposes a new stereophonic acoustic echo canceller to prevent impairments on the voice quality and to remove acoustic echo effectively appearing in stereo environment at the instant of abrupt change of the transmission room environment in teleconferencing system. In stereophonic acoustic echo canceller, the major defective problems are the large computational complexity of estimating echo path systems due to the long impulse response of the true echo paths and the performance degradation of echo canceller due to large correlation between dual stereo signals. Moreover, the change of the suboptimal solution for the echo canceller was considered as a critical deficient factor on to the performance of stereophonic echo canceller. To overcome these problems, this paper proposes pseudo stereophonic acoustic echo canceller using Hyper-plane projection algorithm, which shows the robustness to the environment change of the transmission room and the efficiency of computational complexity.
Far-Field Sound Field Estimation from Near-Field Sound Field Data Using Boundary Collocation Method ; Decision of Optimum Points of Measurement
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 31~37
This paper describes the far-field estimation using the near-field measurement data. Measurement in far-field region gives us the acoustical characteristics of the source but in general measurement is made in near-field such as acoustic water tank or anechoic chamber, so far-field acoustical characteristics of the source should be predicted from near-field data. In this case, the number of measurement points in the near field which relates to the accuracy of the predicted field and the amount of data processing, should be optimized. Existing papers say that measurement points is proportional to kL and depends on geometry and directivity of the source. But they do not give us any definite criterion for the required number of measurement points. Boundary Collocation Method which is one of the far-field prediction methods, is analyzed based on Helmholtz integral equation and Green function and it has been found that the number of measurement points is optimized as 0.54kL which is about one half of the existing results.
EVRC Speech Quality Enhancement Using Pitch Prediction and Gradual Increase of the Decoded Speech
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 38~43
The EVRC vocoder is a toll quality coder, but it shows significant degradation or the quality in weak RF environment. In this paper, the speech quality degradation phenomenon of the EVRC is analyzed, and two methods are proposed as the solution - the pitch prediction and the gradual increase. The preference tests for various Rf environment are performed for speech quality assessments and both the methods show better performance.
An End Point Detection Technique Using the LSP Distance in EVRC Packets
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 44~48
This paper presents a simple and fast method for end point detection under low-level noisy environment. The proposed algorithm uses a threshold logic with LSP distances and takes vocoded packets as input to the recognition system. The results from the proposed method are compared with those manually checked in decoded speeches. From the result it exhibits acceptable accuracy.
Improved Decision Tree-Based State Tying In Continuous Speech Recognition System
;Xintian Wu;Chaojun Liu;;;
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 49~56
In many continuous speech recognition systems based on HMMs, decision tree-based state tying has been used for not only improving the robustness and accuracy of context dependent acoustic modeling but also synthesizing unseen models. To construct the phonetic decision tree, standard method performs one-level pruning using just single Gaussian triphone models. In this paper, two novel approaches, two-level decision tree and multi-mixture decision tree, are proposed to get better performance through more accurate acoustic modeling. Two-level decision tree performs two level pruning for the state tying and the mixture weight tying. Using the second level, the tied states can have different mixture weights based on the similarities in their phonetic contexts. In the second approach, phonetic decision tree continues to be updated with training sequence, mixture splitting and re-estimation. Multi-mixture Gaussian as well as single Gaussian models are used to construct the multi-mixture decision tree. Continuous speech recognition experiment using these approaches on BN-96 and WSJ5k data showed a reduction in word error rate comparing to the standard decision tree based system given similar number of tied states.
Endpoint Detection of Speech Signal Using Wavelet Transform
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 57~64
In this paper, we investigated the robust endpoint detection algorithm in noisy environment. A new feature parameter based on a discrete wavelet transform is proposed for word boundary detection of isolated utterances. The sum of standard deviation of wavelet coefficients in the third coarse and weighted first detailed scale is defined as a new feature parameter for endpoint detection. We then developed a new and robust endpoint detection algorithm using the feature found in the wavelet domain. For the performance evaluation, we evaluated the detection accuracy and the average recognition error rate due to endpoint detection in an HMM-based recognition system across several signal-to-noise ratios and noise conditions.
Flow Control Algorithm for ABR Service in VS/VD Switch
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 65~70
In ATM network there exist several traffics according to QoS, such as CBR, rt-VBR, nrt-VBR, UBR, and ABR. Many studies have done at the traffic management of ABR which uses the unused network bandwidth. Many flow control mechanisms have proposed to use efficiently the unused bandwidth. In TMWG(Traffic Management Working Group) of ATM Forum, there exist rate-based, credit-based, and mixture of them to manage flow control of ABR traffic. Among these, rate-based mechanisms adopted as standard because it is flexible and also makes it possible to implement ATM switch with low price and high capacity. In this paper, we study the switch that uses EFCI, ER and VS/VD(Virtual Source/Virtual Destination) with rate-based mechanism. Instead of using queue threshold, we propose a new algorithm which uses bandwidth threshold of input stage of switch, and manages efficiently ABR traffic with EPRCA algorithm.
Voice Packet Conversion from 13kbps QCELP to 8kbps QCELP Speech Codecs
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 71~76
In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.
The Design of Adaptive Quantizer to Improve Image Quality of the H.263
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 77~83
H.263 is an international standard of ITU-T that can makes the service such as video phone, video conference in the transmission line less than 64Kbps. This recommendation draft has used motion estimation/compensation, transform coding and quantizing methods. TMN5 used for the performance estimation of H.263 has fundamentally used DCT in transform coding method and presented quantizer for quantizing the DCT transform coefficient. This paper is presenting adaptive quantizer effectively able to quantize DCT coefficient considering the human visual sensitivity while the structure of TMN5 is maintaining. As quantizer that proposed DCT-based H.263 could make transmit more frame than TMN5 in a same transfer speed, it could lower the frame drop effect. And the luminance signal appeared the difference of -0.3 ～ +0.7dB in the average PSNR for the estimation of objective image quality and the chrominance signal appeared the improvement in about 1.5dB in comparision with TMN5. As a result it can attain the better image quality compared to TMN5 in the estimation of subjective image quality.
A Neural Network Based Korean Segmental Duration Modeling Using Tonal Information of Phonemes
The Journal of the Acoustical Society of Korea, volume 18, issue 6, 1999, Pages 84~88
The accurate estimation of segmental duration is crucial for natural-sounding text-to-speech synthesis. For predicting Korean segmental durations, conventional methods utilized phonemic context, part-of-speech context and locational information in prosodic phrase. In this paper, the tonal information of phonemes is employed for more accurate prediction. After defining two non-boundary tones and six boundary tones, we annotated the tonal label on each syllable of 400 sentences. To predict segmental duration using tonal information, we constructed neural networks with a real-valued output node predicting phonemic duration and trained them by backpropagation algorithm. Experimental results showed that the proposed features are effective for predicting Korean segmental durations, and we got 0.863 correlation coefficient of the observed durations and predicted ones.