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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 19, Issue 8 - Nov 2000
Volume 19, Issue 7 - Oct 2000
Volume 19, Issue 6 - Aug 2000
Volume 19, Issue 5 - Jul 2000
Volume 19, Issue 4 - May 2000
Volume 19, Issue 3 - Apr 2000
Volume 19, Issue 2 - Feb 2000
Volume 19, Issue 1 - Jan 2000
Volume 19, Issue 3E - 00 2000
Volume 19, Issue 2E - 00 2000
Volume 19, Issue 1E - 00 2000
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Performance Analysis of a Dynamic Priority Control Scheme for Delay-Sensitive Traffic
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 3~11
This paper considers the performance of a dynamic priority control function (DPCF) of a threshold-based Bernoulli priority jump (TBPJ) scheme. Loss-sensitive and delay-sensitive traffics are applied to a system with a TBPJ scheme that is a general state-dependent Bernoulli scheduling scheme. Loss-sensitive and delay-sensitive traffics represent sound and data, respectively. Under the TBPJ scheme, the first packet of the loss-sensitive traffic buffer goes into the delay-sensitive traffic buffer with Bernoulli probability p according to system states which represent the buffer thresholds and the number of packets waiting for scheduling. Performance analysis shows that TBPJ scheme obtains large performance build-up for the delay-sensitive traffic without performance degradation for the loss-sensitive traffic. TBPJ scheme shows also better performance than that of HOL scheme.
A Novel Multi-Channel Hearing Aid Algorithm with SMR(signal-to-masking ratio) Improvement
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 12~21
In this paper, we propose a novel hearing aid algorithm for sensorinural hearing loss restoration with multi-channel(band) dynamic range compression and psychoacoustics. In this way, we can present a normal perception condition to the impaired listener. The proposed algorithm make loudness scaling function achieve proper loudness level, and analysis masking property for the signal will be perceived to impaired listener, and then, restore normal spectral contrast using SMR(signal-to-masking ratio) defined by distance between the level of each frequency and masking threshold.
On a Sound Analysis of Samulnori Instruments
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 22~25
In this paper, we examine the characteristic of samulnori by the sound analysis techniques. Four instruments of samulnori are constituted with the dual principle of the negative and positive (the male and female) which mean the harmony of each instruments high and low sounds. These sounds are covered the audible whole sounds. Since the fundamental frequencies of jing and janggu sounds are similar to the fundamental frequencies of male and female speech respectively, we feel the affection when we hear the sounds of jing and janggu. Additionally, since human feels the sound not only by ears, but also by the vibration to the skin, the put jing and janggu have the remarkable characteristic of the vibration.
Impulse Noise Cancellation Using Adaptive Threshold Algorithm
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 26~34
This paper presents a new adaptive impulse noise cancelling technique based on the adaptive nonlinear suppressing function. The proposed "adaptive threshold algorithm (ATA)" is controlled by the normalized power prior input data term, and this adaptive threshold makes the cancelling system highly robust against additive impulse noise. For the performance evaluation, we have tested the proposed algorithm with the observed signals simulated in various impulsive noise environments and real EMG signals. As a result the proposed algorithm shows superior performance of 51.7% to the available techniques in the points of SNR and MSE.
Subband Acoustic Echo Canceller with Double-Talk Detector Using Weighted Overlap-add Method and Dedicated filter
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 35~46
In this paper, we propose a subband acoustic echo canceller using the weighted Overlap-add adaptive filter bank to prevent the decrease of convergence speed in full-band US processing, and make it possible to realize the adaptive filter in block-parallel processing, this paper introduces the weighted overlap-add technique for subband echo canceller. Moreover, we propose a new double-talk detector which employs dedicated filter in addition to the energy comparison method simultaneously. The computer simulation results show that the performance of the proposed subband adaptive echo canceller double-talk detection
Maximum Likelihood Classifier Using Detection of Amplitude Modulation Frequency due to Propulsion of Underwater Vehicle
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 47~53
In order to classify the underwater vehicles due to propeller propulsion, maximum likelihood classifier was developed. Propeller produces the cavitation and noise during its work. Cavitation-bubble makes the nonlinear medium in the water. The nonlinearity of cavitation leads to the generation of a complete spectrum of combination harmonics of the tonals of noise, and modulation of cavitation noise with propeller shaft-rates and blade-rates. The optimal estimator was derived mathematically and its capabilities were proven by simulation and real test.
Transform Domain Adaptive Filtering with a Chirp Discrete Cosine Transform LMS
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 54~62
Adaptive filtering method is one of signal processing area which is frequently used in the case of statistical characteristic change in time-varing situation. The performance of adaptive filter is usually evaluated with complexity of its structure, convergence speed and misadjustment. The structure of adaptive filter must be simple and its speed of adaptation must be fast for real-time implementation. In this paper, we propose chirp discrete cosine transform (CDCT), which has the characteristics of CZT (chrip z-transform) and DCT (discrete cosine transform), and then CDCTLMS (chirp discrete cosine transform LMS) using the above mentioned algorithm for the improvement of its speed of adaptation. Using loaming curve, we prove that the proposed method is superior to the conventional US (normalized LMS) algorithm and DCTLMS (discrete cosine transform LMS) algorithm. Also, we show the real application for the ultrasonic signal processing.
Performance Improvement of Speaker Recognition System Using Genetic Algorithm
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 63~67
This paper deals with text-prompt speaker recognition based on dynamic time warping (DTW). The Genetic Algorithm was applied to the creation of reference patterns for suitable reflection of the speaker characteristics, one of the most important determinants in the fields of speaker recognition. In order to overcome the weakness of text-dependent and text-independent speaker recognition, the text-prompt type was suggested. Performed speaker identification and verification in close and open set respectively, hence the Genetic algorithm-based reference patterns had been proven to have better performance in both recognition rate and speed than that of conventional reference patterns.
Perceptual and Adaptive Quantization of Line Spectral Frequency Parameters
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 68~77
Line special frequency (LSF) parameters have been widely used in low bit-rate speech coding due to their efficiency for representing the short-time speech spectrum. In this paper, a new distance measure based on the masking properties of human ear is proposed for quantizing LSF parameters whereas most conventional quantization methods are based on the weighted Euclidean distance measure. The proposed method derives the perceptual distance measure from the definition of noise-to-mask ratio (NMR) which has high correspondence with the actual distortion received in the human ear and uses it for quantizing LSF parameters. In addition, we propose an adaptive bit allocation scheme, which allocates minimal bits to LSF parameters maintaining the perceptual transparency of given speech frame for reducing the average bit-rates. For the performance evaluation, we has shown the ratio of perceptually transparent frames and the corresponding average bit-rates for the conventional and proposed methods. By jointly combining the proposed distance measure and adaptive bit allocation scheme, the proposed system requires only 770 bps for obtaining 95.5% perceptually transparent frames, while the conventional systems produce 89.9% at even 1800 bps.
A Study on the Automatic Detection and Extraction of Narrowband Multiple Frequency Lines
The Journal of the Acoustical Society of Korea, volume 19, issue 8, 2000, Pages 78~83
Passive sonar system is designed to classify the underwater targets by analyzing and comparing the various acoustic characteristics such as signal strength, bandwidth, number of tonals and relationship of tonals from the extracted tonals and frequency lines. First of all the precise detection and extraction of signal frequency lines is of particular importance for enhancing the reliability of target classification. But, the narrowband frequency lines which are the line formed in spectrogram by a tonal of constant frequency in each frame can be detected weakly or discontinuously because of the variation of signal strength and transmission loss in the sea. Also, it is very difficult to detect and extract precisely the signal frequency lines by the complexity of impulsive ambient noise and signal components. In this paper, the automatic detection and extraction method that can detect and extract the signal components of frequency tines precisely are proposed. The proposed method can be applied under the bad conditions with weak signal strength and high ambient noise. It is confirmed by the simulation using real underwater target data.